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[alsa-devel] [PATCH] ALSA: ak4117: remove redundant check on err being < 0
by Colin King 12 Jul '16

12 Jul '16

From: Colin Ian King <colin.king(a)canonical.com> snd_ak4117_create checks if the error return err is less than zero or not. This is a redundant check, err can only be < 0 to get to the __fail label, in which case just return err and remove the redundant check (since we never return -EIO). Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/i2c/other/ak4117.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index 48848909..0702f05 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -110,7 +110,7 @@ int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t __fail: snd_ak4117_free(chip); - return err < 0 ? err : -EIO; + return err; } void snd_ak4117_reg_write(struct ak4117 *chip, unsigned char reg, unsigned char mask, unsigned char val) -- 2.8.1

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[alsa-devel] [PATCH] ALSA: hda: add AMD Stoney PCI ID with proper driver caps
by Awais Belal 12 Jul '16

12 Jul '16

This allows the device to correctly show up as ATI HDMI rather than a generic one and allows the driver to use the available caps. Signed-off-by: Awais Belal <awais_belal(a)mentor.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 09920ba..4b80b16 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2106,6 +2106,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x157a), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0x15b3), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x7919), -- 1.9.1

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[alsa-devel] Problem getting USB DAC to work
by John Westwood 12 Jul '16

12 Jul '16

Dear ALSA Developers I have a problem getting my obscure USB DAC (Starting Point Systems DAC3) to work in Linux. I believe the problem may be similar to this one, but am not sure: https://groups.google.com/forum/#!topic/linux.debian.kernel/HwwkRkbL5aY If I try to use aplay to play a wav I receive the following error: aplay -D default test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo aplay: pcm_write:1939: write error: No such device syslog shows the following: Jul 9 15:38:31 atlas kernel: [ 4805.148059] usb 7-2: USB disconnect, device number 19 Jul 9 15:38:32 atlas kernel: [ 4805.532022] usb 7-2: new full-speed USB device number 20 using uhci_hcd Jul 9 15:38:32 atlas kernel: [ 4805.725024] usb 7-2: New USB device found, idVendor=2504, idProduct=0201 Jul 9 15:38:32 atlas kernel: [ 4805.725029] usb 7-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 Jul 9 15:38:32 atlas kernel: [ 4805.725032] usb 7-2: Product: TDA 1543 DAC 3 v1.4 Jul 9 15:38:32 atlas kernel: [ 4805.725034] usb 7-2: Manufacturer: STARTING POINT SYSTEMS Jul 9 15:38:32 atlas kernel: [ 4805.725035] usb 7-2: SerialNumber: Jul 9 15:38:32 atlas kernel: [ 4805.745018] usb 7-2: Warning! Unlikely big volume range (=65535), cval->res is probably wrong. Jul 9 15:38:32 atlas kernel: [ 4805.745023] usb 7-2: [2] FU [PCM Playback Volume] ch = 1, val = -32768/32767/1 Jul 9 15:38:32 atlas mtp-probe: checking bus 7, device 20: "/sys/devices/pci0000:00/0000:00:1d.1/usb7/7-2" Jul 9 15:38:32 atlas mtp-probe: bus: 7, device: 20 was not an MTP device I have another USB DAC that works fine, in syslog the working DAC has the following lines which are not present with the non-working DAC: Jul 11 19:56:35 atlas kernel: [83940.049124] input: Lakewest Audio Audiolab M-DAC as /devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.0/0003:0451:ADAC.0002/input/input12 Jul 11 19:56:35 atlas kernel: [83940.104367] hid-generic 0003:0451:ADAC.0002:input,hiddev0,hidraw1: USB HID v1.11 Device [Lakewest Audio Audiolab M-DAC] on usb-0000:00:1d.1-2/input0 With the non-working DAC I get these lines instead: Jul 11 19:56:25 atlas kernel: [83930.079063] usb 7-2: Warning! Unlikely big volume range (=65535), cval->res is probably wrong. Jul 11 19:56:25 atlas kernel: [83930.079067] usb 7-2: [2] FU [PCM Playback Volume] ch = 1, val = -32768/32767/1 I am using Linux Mint 17.3 64bit, kernel 3.19.0-32. I have also tried kernel 4.4.0-28 but to no avail. The ALSA version is 1.0.27.2 . Any help or suggestions would be greatly appreciated. Thanks very much. Best regards John Westwood

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[alsa-devel] Fwd: Audio capture on OMAP3530/DM3730 (Yocto Poky Linux 3.18+) occasionally swaps channels
by Akram Hameed 12 Jul '16

12 Jul '16

Hello there, (Apologies for cross-posting from user list if you are seeing this twice). I have a fairly serious problem using ALSA for capture on an Gumstix Overo based system. The audio device in question is a TWL4030 device. It is connected to a TI DM3730 (or in some cases, an OMAP3530). Previously I was using a 3.0 kernel and my audio capture worked just fine with no extra tweaks to ALSA. Now for other reasons I have to move to a more recent kernel, and have tried with 3.18 and 4.4 stable releases. Using these kernels, at random times, the audio capture channels will swap. They will remain swapped for an unpredictable period of time, and then they will swap back. This will continue forever, again, with the swapping occurring at random times. Our system totally relies on predictable audio capture - so for this random behaviour to occur means we must seek a different platform unless we can either: a) Find a solution for 3.18 or newer kernel (the problem is still occurring with ALSA 1.1 on kernel 4.4) or, b) Make some code change to solve the issue. I found this alsa-devel thread regarding the 2.6.32 kernel that seems to be almost exactly what I am experiencing: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-February/036895.h… I also found this one which is probably more closely related: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-March/060557.html My current thought is that it is indeed a DMA issue, but I am not familiar with that area of the kernel source and therefore hoping someone might have some suggestions about what to try next. Best regards, Akram

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[alsa-devel] [PATCH v2] sound: oss: Remove useless initialisation
by Amitoj Kaur Chawla 12 Jul '16

12 Jul '16

Remove useless initialisation of variable whose value is reinitialised later. The Coccinelle semantic patch used to make this change is as follows: @@ type T; identifier x; constant C; expression e; @@ T x - = C ; x = e; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- Changes in v2: -Modify commit message sound/oss/ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 10c8de1..6368e5c 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -254,7 +254,7 @@ static void ad_write(ad1848_info * devc, int reg, int data) static void wait_for_calibration(ad1848_info * devc) { - int timeout = 0; + int timeout; /* * Wait until the auto calibration process has finished. -- 1.9.1

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[alsa-devel] [PATCH] ASoC: intel: Fix sst-dsp dependency on dw stuff
by Takashi Iwai 12 Jul '16

12 Jul '16

The recent commit [a92ea59b74e2: ASoC: Intel: sst: only select sst-firmware when DW DMAC is built-in] introduced more strict kconfig dependency (depends on DW_DMAC_CORE=y) for avoiding the build failures due to dependency messes in intel-sst. This makes, however, it impossible to use this driver with the modularized systems, i.e. typically on Linux distros. The problem addressed in the commit above is that sst_dsp_new() and sst_dsp_free() includes the firmware init / finish that call dw_*() functions. Thus building it as built-in with DW_DMAC_CORE module results in the missing symbols. However, these sst_dsp functions are basically called only from the drivers that depend on DW_DMAC_CORE already. That is, once when these functions are split out, the rest can be independent from dw stuff. This patch attempts to solve the issue by the following: - Split sst-dsp stuff into two modules: snd-soc-sst-dsp and snd-soc-sst-firmware. - Move sst_dsp_new() and sst_dsp_free() to the latter module so that the former module can be independent from DW_DMAC_CORE. - Add a new kconfig SND_SOC_INTEL_SST_FIRMWARE to select the latter module by machine drivers. One only remaining pitfall is that each machine driver has to select SND_SOC_INTEL_SST_FIRMWARE carefully depending on DW_DMAC_CORE. This can't be done cleanly due to the restriction of the current kbuild. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=988117 Fixes: a92ea59b74e2 ('ASoC: Intel: sst: only select sst-firmware when DW DMAC is built-in') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/Kconfig | 18 +++++++--- sound/soc/intel/common/Makefile | 4 +-- sound/soc/intel/common/sst-dsp-priv.h | 4 --- sound/soc/intel/common/sst-dsp.c | 67 ---------------------------------- sound/soc/intel/common/sst-dsp.h | 2 +- sound/soc/intel/common/sst-firmware.c | 68 +++++++++++++++++++++++++++++++++++ 6 files changed, 85 insertions(+), 78 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 9c86459d0fc3..a20c3dfbcb5d 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -32,6 +32,12 @@ config SND_SOC_INTEL_SST select SND_SOC_INTEL_SST_MATCH if ACPI depends on (X86 || COMPILE_TEST) +# firmware stuff depends DW_DMAC_CORE; since there is no depends-on from +# the reverse selection, each machine driver needs to select +# SND_SOC_INTEL_SST_FIRMWARE carefully depending on DW_DMAC_CORE +config SND_SOC_INTEL_SST_FIRMWARE + tristate + config SND_SOC_INTEL_SST_ACPI tristate @@ -47,8 +53,9 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE=y + depends on DW_DMAC_CORE select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -91,8 +98,9 @@ config SND_SOC_INTEL_BXT_RT298_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n) + depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n) select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help @@ -103,8 +111,9 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n) + depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n) select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help @@ -115,8 +124,9 @@ config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" depends on X86_INTEL_LPSS && I2C && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE=y + depends on DW_DMAC_CORE select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index fbbb25c2ceed..1a35149bcad7 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -2,9 +2,9 @@ snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o snd-soc-sst-match-objs := sst-match-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o - -snd-soc-sst-dsp-$(CONFIG_DW_DMAC_CORE) += sst-firmware.o +snd-soc-sst-firmware-objs := sst-firmware.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o obj-$(CONFIG_SND_SOC_INTEL_SST_MATCH) += snd-soc-sst-match.o +obj-$(CONFIG_SND_SOC_INTEL_SST_FIRMWARE) += snd-soc-sst-firmware.o diff --git a/sound/soc/intel/common/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h index 97dc1ae05e69..d13c84364c3c 100644 --- a/sound/soc/intel/common/sst-dsp-priv.h +++ b/sound/soc/intel/common/sst-dsp-priv.h @@ -383,10 +383,6 @@ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, u32 index, void *private); void sst_mem_block_unregister_all(struct sst_dsp *dsp); -/* Create/Free DMA resources */ -int sst_dma_new(struct sst_dsp *sst); -void sst_dma_free(struct sst_dma *dma); - u32 sst_dsp_get_offset(struct sst_dsp *dsp, u32 offset, enum sst_mem_type type); #endif diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index ff2196ef359f..c00ede4ea4d7 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -420,73 +420,6 @@ void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) } EXPORT_SYMBOL_GPL(sst_dsp_inbox_read); -#ifdef CONFIG_DW_DMAC_CORE -struct sst_dsp *sst_dsp_new(struct device *dev, - struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) -{ - struct sst_dsp *sst; - int err; - - dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id); - - sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); - if (sst == NULL) - return NULL; - - spin_lock_init(&sst->spinlock); - mutex_init(&sst->mutex); - sst->dev = dev; - sst->dma_dev = pdata->dma_dev; - sst->thread_context = sst_dev->thread_context; - sst->sst_dev = sst_dev; - sst->id = pdata->id; - sst->irq = pdata->irq; - sst->ops = sst_dev->ops; - sst->pdata = pdata; - INIT_LIST_HEAD(&sst->used_block_list); - INIT_LIST_HEAD(&sst->free_block_list); - INIT_LIST_HEAD(&sst->module_list); - INIT_LIST_HEAD(&sst->fw_list); - INIT_LIST_HEAD(&sst->scratch_block_list); - - /* Initialise SST Audio DSP */ - if (sst->ops->init) { - err = sst->ops->init(sst, pdata); - if (err < 0) - return NULL; - } - - /* Register the ISR */ - err = request_threaded_irq(sst->irq, sst->ops->irq_handler, - sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); - if (err) - goto irq_err; - - err = sst_dma_new(sst); - if (err) - dev_warn(dev, "sst_dma_new failed %d\n", err); - - return sst; - -irq_err: - if (sst->ops->free) - sst->ops->free(sst); - - return NULL; -} -EXPORT_SYMBOL_GPL(sst_dsp_new); - -void sst_dsp_free(struct sst_dsp *sst) -{ - free_irq(sst->irq, sst); - if (sst->ops->free) - sst->ops->free(sst); - - sst_dma_free(sst->dma); -} -EXPORT_SYMBOL_GPL(sst_dsp_free); -#endif - /* Module information */ MODULE_AUTHOR("Liam Girdwood"); MODULE_DESCRIPTION("Intel SST Core"); diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h index 0b84c719ec48..859f0de00339 100644 --- a/sound/soc/intel/common/sst-dsp.h +++ b/sound/soc/intel/common/sst-dsp.h @@ -216,7 +216,7 @@ struct sst_pdata { void *dsp; }; -#ifdef CONFIG_DW_DMAC_CORE +#if IS_ENABLED(CONFIG_DW_DMAC_CORE) /* Initialization */ struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 25993527370b..a086c35f91bb 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -1211,3 +1211,71 @@ u32 sst_dsp_get_offset(struct sst_dsp *dsp, u32 offset, } } EXPORT_SYMBOL_GPL(sst_dsp_get_offset); + +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) +{ + struct sst_dsp *sst; + int err; + + dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id); + + sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); + if (sst == NULL) + return NULL; + + spin_lock_init(&sst->spinlock); + mutex_init(&sst->mutex); + sst->dev = dev; + sst->dma_dev = pdata->dma_dev; + sst->thread_context = sst_dev->thread_context; + sst->sst_dev = sst_dev; + sst->id = pdata->id; + sst->irq = pdata->irq; + sst->ops = sst_dev->ops; + sst->pdata = pdata; + INIT_LIST_HEAD(&sst->used_block_list); + INIT_LIST_HEAD(&sst->free_block_list); + INIT_LIST_HEAD(&sst->module_list); + INIT_LIST_HEAD(&sst->fw_list); + INIT_LIST_HEAD(&sst->scratch_block_list); + + /* Initialise SST Audio DSP */ + if (sst->ops->init) { + err = sst->ops->init(sst, pdata); + if (err < 0) + return NULL; + } + + /* Register the ISR */ + err = request_threaded_irq(sst->irq, sst->ops->irq_handler, + sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); + if (err) + goto irq_err; + + err = sst_dma_new(sst); + if (err) + dev_warn(dev, "sst_dma_new failed %d\n", err); + + return sst; + +irq_err: + if (sst->ops->free) + sst->ops->free(sst); + + return NULL; +} +EXPORT_SYMBOL_GPL(sst_dsp_new); + +void sst_dsp_free(struct sst_dsp *sst) +{ + free_irq(sst->irq, sst); + if (sst->ops->free) + sst->ops->free(sst); + + sst_dma_free(sst->dma); +} +EXPORT_SYMBOL_GPL(sst_dsp_free); + +MODULE_DESCRIPTION("Intel SST Firmware Loader"); +MODULE_LICENSE("GPL v2"); -- 2.9.0

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[alsa-devel] [PATCH] sound: oss: Remove useless initialisation
by Amitoj Kaur Chawla 12 Jul '16

12 Jul '16

@@ type T; identifier x; constant C; expression e; @@ T x - = C ; x = e; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/oss/ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 10c8de1..6368e5c 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -254,7 +254,7 @@ static void ad_write(ad1848_info * devc, int reg, int data) static void wait_for_calibration(ad1848_info * devc) { - int timeout = 0; + int timeout; /* * Wait until the auto calibration process has finished. -- 1.9.1

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[alsa-devel] Verification of settings
by Rob Nertney 12 Jul '16

12 Jul '16

Hi all, Could I please get a sanity check on my hw_params? I have a DMA which is providing between 1-16 channels of 4 bytes/ch worth of data as a frame. I get an interrupt to my driver every frame's worth of data (64 Bytes). The data is S32_LE, 16000Hz. My DMA has 2 buffers, PING and PONG. Each receives an IRQ on a frame length, and these local buffers are the size of a frame length (64Bytes). #define MAX_BUFFER (64 * 2) static struct snd_pcm_hardware my_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S32, .rates = SNDRV_PCM_RATE_16000, .rate_min = 16000, .rate_max = 16000, .channels_min = 1, .channels_max = NUM_CHANNELS, .buffer_bytes_max = MAX_BUFFER, .period_bytes_min = 4, .period_bytes_max = 64, .periods_min = 2, .periods_max = 2, }; My understanding is that the MAX_BUFFER needs to be at least twice the size of a period so I don't underrun. .periods_max means the maximum number of periods in a the alsa dma_area buffer, right? So when my DMA fires its ISR, I copy from its local PING buffer to the dma_area at offset 0, increment the buf_pos by the frame_length (64Bytes), and call snd_pcm_period_elapsed. My DMA fires its ISR for its local PONG buffer, copies to the dma_area+buf_pos, increments buf_pos (now back to 0, since buffer only holds 2 frames/periods), and I call snd_pcm_period_elapsed again, correct? Thanks, Rob

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[alsa-devel] [PATCH 0/4] ASoC: Intel: Add support for KBL
by Vinod Koul 11 Jul '16

11 Jul '16

Kabylake is next generation Intel SoC which also sports a DSP. This controller is quite similar to Skylake. This patch series add KBL IDs to SKL driver, hdmi driver, sets combo jack for rt286 codec and finally adds entry to skl machine driver Shreyas NC (1): ASoC: hdac_hdmi: Add device id for Kabylake Vinod Koul (3): ASoC: Intel: add kablake device IDs ASoC: rt286: set combo jack for Kabylake ASoC: Intel: board: add kabylake machine id sound/soc/codecs/hdac_hdmi.c | 1 + sound/soc/codecs/rt286.c | 7 +++++++ sound/soc/intel/boards/skl_rt286.c | 9 +++++++++ sound/soc/intel/skylake/skl-messages.c | 6 ++++++ sound/soc/intel/skylake/skl.c | 8 ++++++++ 5 files changed, 31 insertions(+) -- 1.9.1

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Re: [alsa-devel] Problem getting USB DAC to work
by Alan Stern 11 Jul '16

11 Jul '16

On Mon, 11 Jul 2016, John Westwood wrote: > I think the MTP stuff is a red herring. I checked syslog when I use a DAC that > works (Audiolab MDAC) and that has the MTP lines too, but it doesn't have the > warning line about Unlikely big volume range. > > In addition, the working DAC has the following lines which are not present the > non-working DAC: > > Jul 11 19:56:35 atlas kernel: [83940.049124] input: Lakewest Audio Audiolab > M-DAC as /devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.0/0003:0451: > ADAC.0002/input/input12 > Jul 11 19:56:35 atlas kernel: [83940.104367] hid-generic 0003:0451:ADAC.0002: > input,hiddev0,hidraw1: USB HID v1.11 Device [Lakewest Audio Audiolab M-DAC] on > usb-0000:00:1d.1-2/input0 > > With the non-working DAC I get these lines instead: > > Jul 11 19:56:25 atlas kernel: [83930.079063] usb 7-2: Warning! Unlikely big > volume range (=65535), cval->res is probably wrong. > Jul 11 19:56:25 atlas kernel: [83930.079067] usb 7-2: [2] FU [PCM Playback > Volume] ch = 1, val = -32768/32767/1 > > > I think maybe the problem is similar to this one: > > https://groups.google.com/forum/#!topic/linux.debian.kernel/HwwkRkbL5aY > > Is my problem a bug in snd-usb-audio? Or am I doing something wrong? I would > really appreciate some thoughts on this, I feel like I'm talking to myself :-( Considering that you're not really doing anything, you're probably not doing something wrong. :-) On the other hand, you're likely to get more of a response if you direct your questions to the person who solved the problem mentioned in that Debian bug report, and if you CC: the sound developers' mailing list rather than the general USB list. Alan Stern

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Re: [alsa-devel] [PATCH] snd/hda: fix use-after-free after module unload
by Takashi Iwai 11 Jul '16

11 Jul '16

On Sat, 09 Jul 2016 16:38:57 +0200, Peter Wu wrote: > > register_vga_switcheroo() sets the PM ops from the hda structure which > is freed later in azx_free. Make sure that these ops are cleared. > > Caught by KASAN. > > Fixes: 246efa4a072f ("snd/hda: add runtime suspend/resume on optimus support (v4)") > Signed-off-by: Peter Wu <peter(a)lekensteyn.nl> > --- > 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 94089fc..a339066 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -1219,6 +1219,7 @@ static int azx_free(struct azx *chip) > snd_hda_unlock_devices(&chip->bus); > if (hda->vga_switcheroo_registered) > vga_switcheroo_unregister_client(chip->pci); > + vga_switcheroo_fini_domain_pm_ops(chip->card->dev); The domain pm ops is set only when hda->vga_switcheroo_registered flag is set. So the call should be in the previous if block. Also, the indentation looks wrong. Please use the correct indentation. Could you resubmit with these fixes? thanks, Takashi

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: Fix potential NULL dereference
by Vinod Koul 11 Jul '16

11 Jul '16

Static checker warns: Pointer 'hlink' returned from call to function 'snd_hdac_ext_bus_get_link' at line may be NULL and will be dereferenced" So we should always check the return of snd_hdac_ext_bus_get_link() before referencing the link pointer Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 13002f33384e..cc7bb77a28f1 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1388,6 +1388,11 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec) * exit, we call pm_runtime_suspend() so that will do for us */ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + if (!hlink) { + dev_err(&edev->hdac.dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(edev->ebus, hlink); ret = create_fill_widget_route_map(dapm); @@ -1494,6 +1499,11 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) /* hold the ref while we probe */ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + if (!hlink) { + dev_err(&edev->hdac.dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(edev->ebus, hlink); hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL); @@ -1600,6 +1610,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) } hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + if (!hlink) { + dev_err(dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_put(ebus, hlink); return 0; @@ -1621,6 +1636,11 @@ static int hdac_hdmi_runtime_resume(struct device *dev) return 0; hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + if (!hlink) { + dev_err(dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(ebus, hlink); err = snd_hdac_display_power(bus, true); -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Initialize module list for Broxton
by Vinod Koul 11 Jul '16

11 Jul '16

The module list was not initialized for Broxton DSP code, so initialize it. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 622da5d3e3b3..9d36440c802a 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -325,6 +325,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ), SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ); + INIT_LIST_HEAD(&sst->module_list); ret = skl_ipc_init(dev, skl); if (ret) return ret; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: atom: fix missing breaks that would cause the wrong operation to execute
by Vinod Koul 11 Jul '16

11 Jul '16

From: Alan Cox <alan(a)linux.intel.com> Now we correctly error an attempt to execute an unsupported operation. Signed-off-by: Alan Cox <alan(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/atom/sst-mfld-platform-compress.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-compress.c b/sound/soc/intel/atom/sst-mfld-platform-compress.c index 395168986462..1bead81bb510 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-compress.c +++ b/sound/soc/intel/atom/sst-mfld-platform-compress.c @@ -182,24 +182,29 @@ static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) case SNDRV_PCM_TRIGGER_START: if (stream->compr_ops->stream_start) return stream->compr_ops->stream_start(sst->dev, stream->id); + break; case SNDRV_PCM_TRIGGER_STOP: if (stream->compr_ops->stream_drop) return stream->compr_ops->stream_drop(sst->dev, stream->id); + break; case SND_COMPR_TRIGGER_DRAIN: if (stream->compr_ops->stream_drain) return stream->compr_ops->stream_drain(sst->dev, stream->id); + break; case SND_COMPR_TRIGGER_PARTIAL_DRAIN: if (stream->compr_ops->stream_partial_drain) return stream->compr_ops->stream_partial_drain(sst->dev, stream->id); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (stream->compr_ops->stream_pause) return stream->compr_ops->stream_pause(sst->dev, stream->id); + break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (stream->compr_ops->stream_pause_release) return stream->compr_ops->stream_pause_release(sst->dev, stream->id); - default: - return -EINVAL; + break; } + return -EINVAL; } static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, -- 1.9.1

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[alsa-devel] [PATCH alsa-lib v2 0/3] PCM thread-safety patchset
by Takashi Iwai 11 Jul '16

11 Jul '16

Hi, here is a v2 patchset for supporting multi thread safety to alsa-lib PCM API. Now it's out from RFC and should be reviewed as an official patchset. Basically ALSA PCM functions are thread-unsafe, and applications are supposed to do the proper protection against racy accesses. The reality is, however, that application developers don't care such, as alsa-lib works in most cases. Of course, users still get occasionally mysterious crashes. As a workaround, this patchset adds the pthread mutex protection to most of exported PCM functions. This is slightly an overkill, but the biggest merit is that it's easy to implement; I just wrapped the functions and replaced the internal ones with unlocked versions. To be noted, there is an optimization for the direct hw PCM access. Performance-sensitive applications like JACK should work like before without any overhead. Another bonus by this addition is that we can finally get rid of home brew (and deadly smelling) atomic macros from the tree, since it's now more widely protected. I lightly tested on my local machines (dmix, PA, jack and plugins) and all seem working well, so far. Takashi === v1->v2: - Drop locking for some API functions like hw_params - Lock some functions even for hw plugin to keep PCM internals consistent - Fix deadlock with jack ioplug - More documentations and comments - Add debugging option via $LIBASOUND_THREAD_SAFE=0 - Reordered patches Takashi Iwai (3): pcm: Add thread-safety to PCM API pcm: Remove home brew atomic operations pcm: Add LIBASOUND_THREAD_SAFE env variable check INSTALL | 9 ++ configure.ac | 15 ++ include/Makefile.am | 2 +- include/iatomic.h | 170 -------------------- include/pcm_ioplug.h | 10 +- src/pcm/Makefile.am | 2 +- src/pcm/atomic.c | 43 ----- src/pcm/pcm.c | 433 +++++++++++++++++++++++++++++++++++++++++--------- src/pcm/pcm_direct.c | 4 +- src/pcm/pcm_dmix.c | 13 +- src/pcm/pcm_dshare.c | 13 +- src/pcm/pcm_dsnoop.c | 15 +- src/pcm/pcm_file.c | 21 ++- src/pcm/pcm_generic.c | 10 +- src/pcm/pcm_hw.c | 3 + src/pcm/pcm_ioplug.c | 70 +++++--- src/pcm/pcm_local.h | 131 +++++++++++---- src/pcm/pcm_mmap.c | 18 ++- src/pcm/pcm_plugin.c | 72 ++------- src/pcm/pcm_plugin.h | 2 - src/pcm/pcm_rate.c | 41 ++--- src/pcm/pcm_route.c | 2 +- 22 files changed, 648 insertions(+), 451 deletions(-) delete mode 100644 include/iatomic.h delete mode 100644 src/pcm/atomic.c -- 2.9.0

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[alsa-devel] [PATCH 0/3] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 11 Jul '16

11 Jul '16

Skylake onwards we support GTS capability and thus can report audio link time. This series adds support for parsing the capability and then reports link time. While at it, fix the timestamping Documentation. Guneshwor Singh (2): ALSA - hda: Add support for GTS capability ALSA - hda: Add support for link audio time reporting Vinod Koul (1): ALSA - hda: Fix timestamping documentation Documentation/sound/alsa/timestamping.txt | 12 +- include/sound/hda_register.h | 23 ++++ sound/pci/hda/hda_controller.c | 192 +++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.h | 27 +++++ sound/pci/hda/hda_intel.c | 1 + 5 files changed, 248 insertions(+), 7 deletions(-) -- 1.9.1

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[alsa-devel] [alsa-lib][PATCH 0/7] fix compiler warnings
by Takashi Sakamoto 11 Jul '16

11 Jul '16

Hi, This patchset is just for fixes of compiler warnings. Nothing others. Takashi Sakamoto (7): topology: remove unused variable in tplg_parse_fe_dai() topology: remove unused variable in get_tokens() topology: apply the same type to compared two variables topology: initialize variable for error output pcm: set error code at failure path mixer: remove assignment to unused variable pcm: add 'default' case for switch statement modules/mixer/simple/sbase.c | 2 -- src/pcm/pcm_direct.c | 2 ++ src/pcm/pcm_hw.c | 1 + src/topology/data.c | 10 +++++----- src/topology/pcm.c | 1 - 5 files changed, 8 insertions(+), 8 deletions(-) -- 2.7.4

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Re: [alsa-devel] ALSA sound core device deinit crash
by Takashi Iwai 11 Jul '16

11 Jul '16

(Please don't drop Cc to ML) On Fri, 08 Jul 2016 07:35:37 +0200, b_lkasam(a)codeaurora.org wrote: > > On 2016-07-08 10:58, Takashi Iwai wrote: > > On Fri, 08 Jul 2016 07:19:05 +0200, > > b_lkasam(a)codeaurora.org wrote: > >> > >> Hi Alsa team, > >> There is kernel crash observed when soundcard register failure case as > >> below -> > >> > >> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c old mode > >> 100644 new mode 100755 index 0495890..60a1eb0 > >> --- a/sound/soc/soc-core.c > >> +++ b/sound/soc/soc-core.c > >> @@ -1382,6 +1382,7 @@ static int soc_probe_link_dais(struct > >> snd_soc_card *card, int num, int order) > >> /* do machine specific initialization */ if (dai_link->init) { ret = > >> dai_link->init(rtd); > >> + ret = -ENODEV; // -> we can force error here to reproduce crash > >> easily. > >> if (ret < 0) { > >> dev_err(card->dev, "ASoC: failed to init %s: %d\n", dai_link->name, > >> ret); > >> > >> If sound card fails at initialize at above, it is crashing in > >> pcm_chmap_ctl_private_free(). > >> > >> <1>[ 40.646642] [01-01-2016 00:07:16 CPU:0x3] Unable to handle > >> kernel > >> paging request at virtual address ffffffc0da644b68 > >> <1>[ 40.646664] [01-01-2016 00:07:16 CPU:0x3] pgd = ffffffc001d28000 > >> <1>[ 40.646673] [01-01-2016 00:07:16 CPU:0x3] [ffffffc0da644b68] > >> *pgd=00000000857fc003, *pud=00000000857fc003, *pmd=000000017ddd8003, > >> *pte=00c000015a644793 > >> <0>[ 40.646697] [01-01-2016 00:07:16 CPU:0x3] Internal error: Oops: > >> 9600004f [#1] PREEMPT SMP > >> <6>[ 40.646708] [01-01-2016 00:07:16 CPU:0x3] Modules linked in: > >> brcm_bt_drv fm_drv brcm_hci_ldisc texfat(PO) > >> <6>[ 40.646735] [01-01-2016 00:07:16 CPU:0x3] CPU: 3 PID: 299 Comm: > >> kworker/u8:8 Tainted: P O 3.18.24-g41b2dda2-00002-gbe25a74 > >> #1 > >> <6>[ 40.646744] [01-01-2016 00:07:16 CPU:0x3] Hardware name: > >> Qualcomm > >> Technologies, Inc. MSM 8996 v3.x + PMI8996 MTP (DT) > >> <6>[ 40.646763] [01-01-2016 00:07:16 CPU:0x3] Workqueue: deferwq > >> deferred_probe_work_func > >> <6>[ 40.646773] [01-01-2016 00:07:16 CPU:0x3] task: ffffffc0e951c880 > >> ti: ffffffc03368c000 task.ti: ffffffc03368c000 > >> <6>[ 40.646787] [01-01-2016 00:07:16 CPU:0x3] PC is at > >> pcm_chmap_ctl_private_free+0x1c/0x2c > >> <6>[ 40.646798] [01-01-2016 00:07:16 CPU:0x3] LR is at > >> snd_ctl_free_one+0x20/0x34 > >> > >> > >> FIX: > >> > >> Can you look at the change below and share your comments on this? > >> diff --git a/sound/core/device.c b/sound/core/device.c old mode 100644 > >> new mode 100755 index 41bec30..eaffde1 > >> --- a/sound/core/device.c > >> +++ b/sound/core/device.c > >> @@ -219,6 +219,7 @@ void snd_device_free_all(struct snd_card *card) > >> > >> if (snd_BUG_ON(!card)) > >> return; > >> - list_for_each_entry_safe_reverse(dev, next, &card->devices, > >> list) > >> + list_for_each_entry_safe(dev, next, &card->devices, > >> list) > >> __snd_device_free(dev); } > >> > >> > >> Since control sound device has the lowest type value > >> (SNDRV_DEV_CONTROL), it will be the first entry linked in the > >> card->devices linked list head and will be the last one to be freed. > >> > >> This issue seems to be resolved by modifying the sequence the sound > >> devices in the card->devices list are freed as shown below (from > >> “prev” > >> direction to “next” direction) but I’m not sure if this is a right > >> approach from ALSA perspective. > > > > This doesn't look correct. The strange thing is that this error > > shouldn't happen no matter which free loop direction is. The chmap > > ctl should have been already removed by the disconnection before > > freeing. It means that either the disconnection isn't done properly > > or something else is missing. > > > > Could you give the full stack trace? It's important to know which > > code path triggers it. > > > > > > thanks, > > > > Takashi > > > Hi Takashi, > > Here is full stack trace --> > > > [Callstack] > : (struct snd_pcm_chmap *)info = 0xFFFFFFC0DA6A5200 > : (struct snd_pcm *)pcm = 0xFFFFFFC0DA644A80 //part of buddy page, > read-only > : info->stream = 0 > : snd_card_free() was called > > -012|pcm_chmap_ctl_private_free() > //info->pcm->streams[info->stream].chmap_kctl = NULL > -013|snd_ctl_free_one() > -014|snd_ctl_remove() > -015|snd_ctl_dev_free() > -016|__snd_device_free() > -017|snd_device_free_all() > -018|snd_card_do_free(inline) > -018|release_card_device() > -019|device_release() > -020|kobject_cleanup(inline) > -020|kobject_release() > -021|kobject_put() > -022|put_device() > -023|snd_card_free_when_closed() > -024|snd_card_free() > -025|snd_soc_instantiate_card(inline) -> //Here instantiate card > failed for some reason, then triggers snd_card_free > -025|snd_soc_register_card() > -026|msm8996_asoc_machine_probe() > -027|platform_drv_probe() > -028|really_probe(inline) > -028|driver_probe_device() > -029|__device_attach() > -030|bus_for_each_drv() > -031|device_attach() > -032|bus_probe_device() > -033|deferred_probe_work_func() > -034|static_key_count(inline) > -034|static_key_false(inline) > -034|trace_workqueue_execute_end(inline) > -034|process_one_work() > -035|worker_thread() > -036|kthread() > -037|ret_from_fork(asm) > ---|end of frame > > > ----------------------- > trigger point in soc-core.c > in API snd_soc_instantiate_card(), > > card_probe_error: > if (card->remove) > card->remove(card); > > snd_card_free(card->snd_card); -> this is > snd_card_free which internally leads to above function call. > ------------------------------------------ OK, thanks. So this is the case where it frees before registering, and indeed there is a bug in the PCM chmap code. It's freed at disconnection but the disconnect is called only when it was registered. Below is a quick fix. Give it a try. thanks, Takashi --- diff --git a/sound/core/control.c b/sound/core/control.c index 9ff081cd03f4..fb096cb20a80 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -160,6 +160,8 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, if (snd_BUG_ON(!card || !id)) return; + if (card->shutdown) + return; read_lock(&card->ctl_files_rwlock); #if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 308c9ecf73db..8e980aa678d0 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -849,6 +849,14 @@ int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, } EXPORT_SYMBOL(snd_pcm_new_internal); +static void free_chmap(struct snd_pcm_str *pstr) +{ + if (pstr->chmap_kctl) { + snd_ctl_remove(pstr->pcm->card, pstr->chmap_kctl); + pstr->chmap_kctl = NULL; + } +} + static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; @@ -871,6 +879,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) kfree(setup); } #endif + free_chmap(pstr); if (pstr->substream_count) put_device(&pstr->dev); } @@ -1135,10 +1144,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) for (cidx = 0; cidx < 2; cidx++) { if (!pcm->internal) snd_unregister_device(&pcm->streams[cidx].dev); - if (pcm->streams[cidx].chmap_kctl) { - snd_ctl_remove(pcm->card, pcm->streams[cidx].chmap_kctl); - pcm->streams[cidx].chmap_kctl = NULL; - } + free_chmap(&pcm->streams[cidx]); } mutex_unlock(&pcm->open_mutex); mutex_unlock(&register_mutex);

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[alsa-devel] [alsa-lib][PATCH] hctl: remove assertion from element event handler because removal event overwrites all of queued events
by Takashi Sakamoto 11 Jul '16

11 Jul '16

In ALSA control core, when several events occurs for an element, they're represented bit mask (struct snd_ctl_event.data.elem.mask). Thus userspace applications can handles these events separately in one I/O operation. There's an exception; removal event. This is represented by all of bits in the mask. Therefore, when a removal event occurs for an element, all of queued events for the element are overwritten. This works worse for a combination of applications which add/remove control elements and applications which keep a cache of elements. For example, let's assume a case that: 1.some elements are added by ioctl(SNDRV_CTL_IOCTL_ELEM_ADD). 2.ALSA control core handles the request and queues 'add' event. 3.applications don't read the event yet. 4.the elements are removed by ioctl(SNDRV_CTL_IOCTL_ELEM_REMOVE) 5.ALSA control core handles the request and queues 'removal' event by overwriting queued events. 6.applications read the event at last. 7.the applications cannot find removed elements in their cache data. Especially, when applications use hctl interface of ALSA userspace library, this situation occurs because the interface is designed to keep element cache implicitly inner the library. PulseAudio is such an application. When executing a test program of ALSA library (test/user-ctl-element-set.c), pulseaudio process aborts due to assertion at calls of snd_hctl_handle_events(). The rule to represent events is in ALSA kernel/userspace interfaces (UAPI header) and it's difficult to improve the rule. Therefore, developers should pay enough attention to the fact that userspace applications can receive removal events for elements which are not notified with 'add' event. For this reason, this commit removes assertion from event handler of ALSA hctl API. CC: pulseaudio-discuss(a)lists.freedesktop.org Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/hcontrol.c | 1 - 1 file changed, 1 deletion(-) diff --git a/src/control/hcontrol.c b/src/control/hcontrol.c index 7645c57..c40a807 100644 --- a/src/control/hcontrol.c +++ b/src/control/hcontrol.c @@ -742,7 +742,6 @@ static int snd_hctl_handle_event(snd_hctl_t *hctl, snd_ctl_event_t *event) if (event->data.elem.mask == SNDRV_CTL_EVENT_MASK_REMOVE) { int dir; res = _snd_hctl_find_elem(hctl, &event->data.elem.id, &dir); - assert(res >= 0 && dir == 0); if (res < 0 || dir != 0) return -ENOENT; snd_hctl_elem_remove(hctl, (unsigned int) res); -- 2.7.4

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[alsa-devel] [alsa-lib][PATCH] ctl: improve documentation of APIs to add an element set
by Takashi Sakamoto 11 Jul '16

11 Jul '16

Currently, documentation of APIs to add an element set describes causes of errors partly. But it gives an opinion to describe them thoroughly. This is not good for users. This commit changes the documentation so that it partly describes the causes. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 85485aa..9bb8021 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -358,7 +358,8 @@ static bool validate_element_member_dimension(snd_ctl_elem_info_t *info) * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 127. + * <dd>Some arguments include invalid value; i.e. ID field in \a info has no + * name, or the number of members is not between 1 to 127. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO @@ -446,7 +447,8 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 127. + * <dd>Some arguments include invalid value; i.e. ID has no name, or the number + * of members is not between 1 to 127. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO @@ -530,7 +532,8 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 127. + * <dd>Some parameters include invalid value; i.e. ID has no name, or the number + * of members is not between 1 to 127. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO @@ -591,9 +594,10 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A control element with ID \a id already exists. * <dt>-EINVAL - * <dd>\a element_count is not between 1 to 127, or \a items is not at least - * one, or a string in \a labels is empty, or longer than 63 bytes, or total - * length of the labels requires more than 64 KiB storage. + * <dd>Some arguments include invalid value; i.e. \a element_count is not + * between 1 to 127, or \a items is not at least one, or a string in \a + * labels is empty, or longer than 63 bytes, or total length of the labels + * requires more than 64 KiB storage. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user control elements. * <dt>-ENXIO @@ -677,7 +681,8 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 511. + * <dd>Some arguments include invalid value; i.e. ID has no name, or the number + * of members is not between 1 to 511. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO -- 2.7.4

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[alsa-devel] [PATCH v6] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 11 Jul '16

11 Jul '16

Because we need to support the multiple codecs (MAX98357A/RT5514/DA7219) on the RK3399 GRU boards, this patch can help us to support these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v6: - remove enable/disable dapm pins, and to switch codecs via user space - fix "Speaker" dapm route for max98357a - add SND_JACK_LINEOUT flag for da7219 jack detection Changes in v5: - fix the build warning for missing I2C/CLKDEV_LOOKUP dependency Changes in v4: - the compatible name needs to be based on a single reference design, and the name of the board is called "GRU" - use one i2sX card "rockchip,cpu = <&i2s0>" instead of array that is based on GRU - rename DOC to rockchip,rk3399-gru-sound.txt - rename compatible to rockchip,rk3399-gru-sound - rename source code to rk3399_gru_sound.c Changes in v3: - rename DOC to rockchip,rk3399-max98357a-rt5514-da7219.txt - rename compatible to rockchip,rk3399-max98357a-rt5514-da7219 - rename source code to rk3399_max98357a_rt5514_da7219.c Changes in v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../bindings/sound/rockchip,rk3399-gru-sound.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 322 ++++++++++++++++++++ 4 files changed, 350 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt create mode 100644 sound/soc/rockchip/rk3399_gru_sound.c diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 0000000..f19b6c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..6d39032 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_RK3399_GRU_SOUND + tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support multiple codecs for SoC + audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..84e5c7c 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_RK3399_GRU_SOUND) += snd-soc-rk3399-gru-sound.o diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c new file mode 100644 index 0000000..43a537c --- /dev/null +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -0,0 +1,322 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rk3399-gru-sound" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speaker"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int mclk; + int ret; + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int mclk; + int ret; + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int mclk, ret; + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rk3399-gru-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); + if (!cpu_node) { + dev_err(&pdev->dev, "Property 'rockchip,cpu' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rk3399-gru-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

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[alsa-devel] [GIT PULL] skylake audio firmware update
by Vinod Koul 09 Jul '16

09 Jul '16

Hello, Please PULL to receive the updated audio firmware for SKL which contains update for supporting multiple languages. The following changes since commit 3ef7857d511ce6a91c5ce609da76c4702651cfa5: amdgpu: update polaris ucode (2016-06-28 14:31:11 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to d4274de33ed9448e91b970225ce9afdb74d5f13e: linux-firmware: intel: Update Skylake audio firmware (2016-07-01 13:10:07 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v951.bin | Bin 0 -> 221184 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_release_v951.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: cht: fix uninit variable warning
by Vinod Koul 08 Jul '16

08 Jul '16

Kbuild bot reports that we might use dai_index uninitialized. sound/soc/intel/boards/cht_bsw_rt5645.c:391:37: warning: 'dai_index' may be used uninitialized in this function [-Wmaybe-uninitialized] Since it is theoretically possible, set it while initializing. Signed-off-by: Vinod Koul <vinod.koul(a)intel.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 f26c7b8545ae..56056ed7fcfd 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -357,7 +357,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) char codec_name[16]; struct sst_acpi_mach *mach; const char *i2c_name = NULL; - int dai_index; + int dai_index = 0; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); if (!drv) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix to use the actual size for TLV control
by Vinod Koul 08 Jul '16

08 Jul '16

From: Dharageswari R <dharageswari.r(a)intel.com> DSP expects the actual length of parameters that is set through TLV to be passed in large config set, so pass the actual size received in tlv_control_set() instead of max size. Signed-off-by: Dharageswari R <dharageswari.r(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 13 +++++++++---- sound/soc/intel/skylake/skl-topology.h | 1 + 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index b284b3cf5f94..cc0150fc2601 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -411,7 +411,7 @@ static int skl_tplg_set_module_params(struct snd_soc_dapm_widget *w, if (bc->set_params == SKL_PARAM_SET) { ret = skl_set_module_params(ctx, - (u32 *)bc->params, bc->max, + (u32 *)bc->params, bc->size, bc->param_id, mconfig); if (ret < 0) return ret; @@ -446,7 +446,7 @@ static int skl_tplg_set_module_init_data(struct snd_soc_dapm_widget *w) continue; mconfig->formats_config.caps = (u32 *)&bc->params; - mconfig->formats_config.caps_size = bc->max; + mconfig->formats_config.caps_size = bc->size; break; } @@ -1054,7 +1054,7 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, if (w->power) skl_get_module_params(skl->skl_sst, (u32 *)bc->params, - bc->max, bc->param_id, mconfig); + bc->size, bc->param_id, mconfig); /* decrement size for TLV header */ size -= 2 * sizeof(u32); @@ -1088,6 +1088,10 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, struct skl *skl = get_skl_ctx(w->dapm->dev); if (ac->params) { + if (size > ac->max) + return -EINVAL; + + ac->size = size; /* * if the param_is is of type Vendor, firmware expects actual * parameter id and size from the control. @@ -1103,7 +1107,7 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, if (w->power) return skl_set_module_params(skl->skl_sst, - (u32 *)ac->params, ac->max, + (u32 *)ac->params, ac->size, ac->param_id, mconfig); } @@ -1719,6 +1723,7 @@ static int skl_init_algo_data(struct device *dev, struct soc_bytes_ext *be, ac->max = dfw_ac->max; ac->param_id = dfw_ac->param_id; ac->set_params = dfw_ac->set_params; + ac->size = dfw_ac->max; if (ac->max) { ac->params = (char *) devm_kzalloc(dev, ac->max, GFP_KERNEL); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 170d68b4e12d..22d3ef83817d 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -321,6 +321,7 @@ struct skl_algo_data { u32 param_id; u32 set_params; u32 max; + u32 size; char *params; }; -- 1.9.1

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[alsa-devel] [PATCH 0/3] Add audio support for Microsoft Surface 3 tablet
by Stephen Just 08 Jul '16

08 Jul '16

This series adds audio support for the Surface 3 tablet. This tablet identifes an rt5645 audio codec using the ACPI identifer 10EC5640, which is already associated with the rt5640, thus a quirk is required to have the correct drivers loaded. Additionally, in order for jack-detection and the tablet's internal microphone to work, the Surface 3 must be configured in the audio codec driver similarly to other devices. Resolves: https://bugzilla.kernel.org/show_bug.cgi?id=98001 UCM configuration to be added to alsa-lib is included in patch 3, not included in the diffstat below. Stephen Just (2): ASoC: rt5645: Add support for Surface 3 tablet ASoC: Intel: Atom: add quirk for CHT w/ RT5645 sound/soc/codecs/rt5645.c | 7 +++++++ sound/soc/intel/atom/sst/sst_acpi.c | 28 +++++++++++++++++++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 1 + 3 files changed, 35 insertions(+), 1 deletion(-) -- 2.7.4

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[alsa-devel] [RFC 00/11] ASoC: hdac: Add hdac generic driver
by Subhransu S. Prusty 08 Jul '16

08 Jul '16

HDA devices generically can be modelled with DAPM in ASoC. This series adds a framework in ASoC to model HDA devices. HDA widgets are enumerated and one or multiple DAPM widget(s) are created. Connection list is queried for each widget to identify the connection between two endpoints and modelled using DAPM graph. Set of event handlers are defined for each widget type. Based on DAPM events required verbs are sent to program codec. Finally a function is exported to query for the device endpoint configuration to create machine controls. Hardik T Shah (1): ASoC: Add dai_ops to set the stream tag. Subhransu S. Prusty (10): ALSA: hdac: Add codec helper library ALSA: hda - Add macro to test pin widget's input capability ASoC: hdac: Add a generic hdac driver framework ASoC: hdac: Create DAPM model for HDA widgets ASoC: dapm: Create API to add a single route element ASoC: hdac: Build DAPM graph by querying through widget connection list ASoC: hdac: Register widget event handlers ALSA: hda - macro to get default config device of pin widgets ASoC: dapm: Export snd_soc_dapm_new_control ASoC: hdac: Export API to create machine controls include/sound/hdaudio.h | 1 + include/sound/soc-dai.h | 12 + include/sound/soc-dapm.h | 5 + sound/hda/ext/Makefile | 3 +- sound/hda/ext/hdac_codec.c | 188 +++++ sound/hda/ext/hdac_codec.h | 52 ++ sound/hda/local.h | 13 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_generic.c | 1561 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_generic.h | 31 + sound/soc/soc-core.c | 35 + sound/soc/soc-dapm.c | 22 + 13 files changed, 1929 insertions(+), 1 deletion(-) create mode 100644 sound/hda/ext/hdac_codec.c create mode 100644 sound/hda/ext/hdac_codec.h create mode 100644 sound/soc/codecs/hdac_generic.c create mode 100644 sound/soc/codecs/hdac_generic.h -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda/realtek - add new pin definition in alc225 pin quirk table
by Hui Wang 08 Jul '16

08 Jul '16

We have some Dell laptops which can't detect headset mic, the machines use the codec ALC225, they have some new pin configuration values, after adding them in the alc225 pin quirk table, they work well. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- comparing with the previous patch (I said the patch should be skipped), this patch adds one more group of pin definition in the pin quirk table. sound/pci/hda/patch_realtek.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 45d4dc1..ddd29b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5753,7 +5753,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {} }; #define ALC225_STANDARD_PINS \ - {0x12, 0xb7a60130}, \ {0x21, 0x04211020} #define ALC256_STANDARD_PINS \ @@ -5778,10 +5777,24 @@ static const struct hda_model_fixup alc269_fixup_models[] = { static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, {0x14, 0x901701a0}), SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, {0x14, 0x901701b0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60150}, + {0x14, 0x901701a0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60150}, + {0x14, 0x901701b0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, + {0x1b, 0x90170110}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, {0x14, 0x90170110}, {0x21, 0x02211020}), -- 1.9.1

2 1

[alsa-devel] ALSA sound core device deinit crash
by b_lkasam＠codeaurora.org 08 Jul '16

08 Jul '16

Hi Alsa team, There is kernel crash observed when soundcard register failure case as below -> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c old mode 100644 new mode 100755 index 0495890..60a1eb0 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1382,6 +1382,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) /* do machine specific initialization */ if (dai_link->init) { ret = dai_link->init(rtd); + ret = -ENODEV; // -> we can force error here to reproduce crash easily. if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", dai_link->name, ret); If sound card fails at initialize at above, it is crashing in pcm_chmap_ctl_private_free(). <1>[ 40.646642] [01-01-2016 00:07:16 CPU:0x3] Unable to handle kernel paging request at virtual address ffffffc0da644b68 <1>[ 40.646664] [01-01-2016 00:07:16 CPU:0x3] pgd = ffffffc001d28000 <1>[ 40.646673] [01-01-2016 00:07:16 CPU:0x3] [ffffffc0da644b68] *pgd=00000000857fc003, *pud=00000000857fc003, *pmd=000000017ddd8003, *pte=00c000015a644793 <0>[ 40.646697] [01-01-2016 00:07:16 CPU:0x3] Internal error: Oops: 9600004f [#1] PREEMPT SMP <6>[ 40.646708] [01-01-2016 00:07:16 CPU:0x3] Modules linked in: brcm_bt_drv fm_drv brcm_hci_ldisc texfat(PO) <6>[ 40.646735] [01-01-2016 00:07:16 CPU:0x3] CPU: 3 PID: 299 Comm: kworker/u8:8 Tainted: P O 3.18.24-g41b2dda2-00002-gbe25a74 #1 <6>[ 40.646744] [01-01-2016 00:07:16 CPU:0x3] Hardware name: Qualcomm Technologies, Inc. MSM 8996 v3.x + PMI8996 MTP (DT) <6>[ 40.646763] [01-01-2016 00:07:16 CPU:0x3] Workqueue: deferwq deferred_probe_work_func <6>[ 40.646773] [01-01-2016 00:07:16 CPU:0x3] task: ffffffc0e951c880 ti: ffffffc03368c000 task.ti: ffffffc03368c000 <6>[ 40.646787] [01-01-2016 00:07:16 CPU:0x3] PC is at pcm_chmap_ctl_private_free+0x1c/0x2c <6>[ 40.646798] [01-01-2016 00:07:16 CPU:0x3] LR is at snd_ctl_free_one+0x20/0x34 FIX: Can you look at the change below and share your comments on this? diff --git a/sound/core/device.c b/sound/core/device.c old mode 100644 new mode 100755 index 41bec30..eaffde1 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -219,6 +219,7 @@ void snd_device_free_all(struct snd_card *card) if (snd_BUG_ON(!card)) return; - list_for_each_entry_safe_reverse(dev, next, &card->devices, list) + list_for_each_entry_safe(dev, next, &card->devices, list) __snd_device_free(dev); } Since control sound device has the lowest type value (SNDRV_DEV_CONTROL), it will be the first entry linked in the card->devices linked list head and will be the last one to be freed. This issue seems to be resolved by modifying the sequence the sound devices in the card->devices list are freed as shown below (from “prev” direction to “next” direction) but I’m not sure if this is a right approach from ALSA perspective. Thanks Laxminath Kasam

2 1

[alsa-devel] [alsa-lib][PATCH] test: add linker flag with pthread for pcm-multi-thread stress test
by Takashi Sakamoto 07 Jul '16

07 Jul '16

This test program is a pthread application, while Makefile has no flags for linker about pthread library. This can cause below error. $ make pcm-multi-thread CCLD pcm-multi-thread /usr/bin/ld: pcm-multi-thread.o: undefined reference to symbol 'pthread_cancel@@GLIBC_2.2.5' //lib/x86_64-linux-gnu/libpthread.so.0: error adding symbols: DSO missing from command line collect2: error: ld returned 1 exit status Makefile:523: recipe for target 'pcm-multi-thread' failed make: *** [pcm-multi-thread] Error 1 This commit fixes the bug by adding a corresponding flag. Fixes: 147093ac5b8d('test: Add pcm-multi-thread program') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/Makefile.am | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/test/Makefile.am b/test/Makefile.am index 970595a..5f35159 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -24,7 +24,7 @@ code_CFLAGS=-Wall -pipe -g -O2 chmap_LDADD=../src/libasound.la audio_time_LDADD=../src/libasound.la pcm_multi_thread_LDADD=../src/libasound.la - +pcm_multi_thread_LDFLAGS=-lpthread user_ctl_element_set_LDADD=../src/libasound.la user_ctl_element_set_CFLAGS=-Wall -g -- 2.7.4

2 1

[alsa-devel] [alsa-lib][PATCH v2] ctl: add dimension validator
by Takashi Sakamoto 07 Jul '16

07 Jul '16

Linux 4.7 or former have no validator of dimension information. This can causes an issue related to user-defined element set. For example, When calculated total members in multi-dimensional matrix is larger than actual capacity of snd_ctl_elem_value_t, processes to handle the element can cause buffer-over-run. For backward portability of this userspace library, this commit adds a validator of dimension information. When userspace applications give invalid dimension information to APIs to add element set, they receive -EINVAL. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/src/control/control.c b/src/control/control.c index 2ad3e0d..85485aa 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -91,6 +91,7 @@ I/O operations. #include <fcntl.h> #include <signal.h> #include <sys/poll.h> +#include <stdbool.h> #include "control_local.h" /** @@ -302,6 +303,32 @@ int snd_ctl_elem_info(snd_ctl_t *ctl, snd_ctl_elem_info_t *info) return ctl->ops->element_info(ctl, info); } +static bool validate_element_member_dimension(snd_ctl_elem_info_t *info) +{ + unsigned int members; + unsigned int i; + + if (info->dimen.d[0] == 0) + return true; + + members = 1; + for (i = 0; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] == 0) + break; + members *= info->dimen.d[i]; + + if (members > info->count) + return false; + } + + for (++i; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] > 0) + return false; + } + + return members == info->count; +} + /** * \brief Create and add some user-defined control elements of integer type. * \param ctl A handle of backend module for control interface. @@ -366,6 +393,9 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer.max = max; info->value.integer.step = step; + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); if (err < 0) return err; @@ -451,6 +481,9 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer64.max = max; info->value.integer64.step = step; + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); if (err < 0) return err; @@ -524,6 +557,9 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer.min = 0; info->value.integer.max = 1; + if (!validate_element_member_dimension(info)) + return -EINVAL; + return ctl->ops->element_add(ctl, info); } @@ -605,6 +641,9 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, p += strlen(labels[i]) + 1; } + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); free(buf); @@ -663,6 +702,9 @@ int snd_ctl_elem_add_bytes_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->owner = element_count; info->count = member_count; + if (!validate_element_member_dimension(info)) + return -EINVAL; + return ctl->ops->element_add(ctl, info); } -- 2.7.4

2 1

[alsa-devel] [alsa-lib][PATCH] ctl: add dimension validator
by Takashi Sakamoto 07 Jul '16

07 Jul '16

Linux 4.7 or former have no validator of dimension information. This can causes an issue related to user-defined element set. For example, When calculated total members in multi-dimensional matrix is larger than actual capacity of snd_ctl_elem_value_t, processes to handle the element can cause buffer-over-run. For backward portability of this userspace library, this commit adds a validator of dimension information. When userspace applications give invalid dimension information to APIs to add element set, they receive -EINVAL. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/src/control/control.c b/src/control/control.c index 2ad3e0d..ee6b1d8 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -91,6 +91,7 @@ I/O operations. #include <fcntl.h> #include <signal.h> #include <sys/poll.h> +#include <stdbool.h> #include "control_local.h" /** @@ -302,6 +303,32 @@ int snd_ctl_elem_info(snd_ctl_t *ctl, snd_ctl_elem_info_t *info) return ctl->ops->element_info(ctl, info); } +static bool validate_element_member_dimension(struct snd_ctl_elem_info *info) +{ + unsigned int members; + unsigned int i; + + if (info->dimen.d[0] == 0) + return true; + + members = 1; + for (i = 0; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] == 0) + break; + members *= info->dimen.d[i]; + + if (members > info->count) + return false; + } + + for (++i; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] > 0) + return false; + } + + return members == info->count; +} + /** * \brief Create and add some user-defined control elements of integer type. * \param ctl A handle of backend module for control interface. @@ -366,6 +393,9 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer.max = max; info->value.integer.step = step; + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); if (err < 0) return err; @@ -451,6 +481,9 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer64.max = max; info->value.integer64.step = step; + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); if (err < 0) return err; @@ -524,6 +557,9 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer.min = 0; info->value.integer.max = 1; + if (!validate_element_member_dimension(info)) + return -EINVAL; + return ctl->ops->element_add(ctl, info); } @@ -605,6 +641,9 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, p += strlen(labels[i]) + 1; } + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); free(buf); @@ -663,6 +702,9 @@ int snd_ctl_elem_add_bytes_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->owner = element_count; info->count = member_count; + if (!validate_element_member_dimension(info)) + return -EINVAL; + return ctl->ops->element_add(ctl, info); } -- 2.7.4

2 2

[alsa-devel] [alsa-lib][PATCH] ctl: fix a test for user-defined element set in a point of dimension
by Takashi Sakamoto 07 Jul '16

07 Jul '16

Currently, this test program has a bug related to dimension information. When testing an element set with byte type, the program adds 512 members to an element. This is a maximum number permitted for this type. On the other hand, dimension information of the elements consists of [8, 4, 8, 4]. Calculated members in the multi-dimensional matrix is 1024. This should be less than 512. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/user-ctl-element-set.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c index c346e03..0481300 100644 --- a/test/user-ctl-element-set.c +++ b/test/user-ctl-element-set.c @@ -513,7 +513,7 @@ int main(void) trial.dimension[0] = 8; trial.dimension[1] = 4; trial.dimension[2] = 8; - trial.dimension[3] = 4; + trial.dimension[3] = 2; trial.add_elem_set = add_bytes_elem_set; trial.check_elem_props = NULL; trial.change_elem_members = change_bytes_elem_members; -- 2.7.4

2 1

[alsa-devel] [PATCH alsa-lib 0/3] Misc fix and pcm-multi-thread test
by Takashi Iwai 07 Jul '16

07 Jul '16

Hi, this is an excerpt from my previous RFC patchset for thread safety, containing a trivial cleanup and an addition of pcm-multi-thread test program, where both can be applied generically without thread-safe patches. It's just a resend, as these have no changes from the previous post. Takashi === Takashi Iwai (3): pcm: Remove superfluous rmb() from PCM meter plugin test: Add pcm-multi-thread program Add pcm-multi-thread to .gitignore .gitignore | 1 + src/pcm/pcm_meter.c | 1 - test/Makefile.am | 3 +- test/pcm-multi-thread.c | 263 ++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 266 insertions(+), 2 deletions(-) create mode 100644 test/pcm-multi-thread.c -- 2.9.0

1 3

[alsa-devel] [PATCH v3] ALSA: control: add dimension validator for userspace elements
by Takashi Sakamoto 07 Jul '16

07 Jul '16

The 'dimen' field in struct snd_ctl_elem_info is used to compose all of members in the element as multi-dimensional matrix. The field has four members. Each member represents the width in each dimension level by element member unit. For example, if the members consist of typical two dimensional matrix, the dimen[0] represents the number of rows and dimen[1] represents the number of columns (or vise-versa). The total members in the matrix should be exactly the same as the number of members in the element, while current implementation has no validator of this information. In a view of userspace applications, the information must be valid so that it cannot cause any bugs such as buffer-over-run. This commit adds a validator of dimension information for userspace applications which add new element sets. When they add the element sets with wrong dimension information, they receive -EINVAL. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index a85d455..9ff081c 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -805,6 +805,36 @@ static int snd_ctl_elem_list(struct snd_card *card, return 0; } +static bool validate_element_member_dimension(struct snd_ctl_elem_info *info) +{ + unsigned int members; + unsigned int i; + + if (info->dimen.d[0] == 0) + return true; + + members = 1; + for (i = 0; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] == 0) + break; + members *= info->dimen.d[i]; + + /* + * info->count should be validated in advance, to guarantee + * calculation soundness. + */ + if (members > info->count) + return false; + } + + for (++i; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] > 0) + return false; + } + + return members == info->count; +} + static int snd_ctl_elem_info(struct snd_ctl_file *ctl, struct snd_ctl_elem_info *info) { @@ -1272,6 +1302,8 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1 || info->count > max_value_counts[info->type]) return -EINVAL; + if (!validate_element_member_dimension(info)) + return -EINVAL; private_size = value_sizes[info->type] * info->count; /* -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: rt5645: add DAC1 soft volume func control
by Bard Liao 07 Jul '16

07 Jul '16

This patch add an alsa control for DAC1 digital volume control function selection. The options are: 0: Gain update immediately 1: Gain update when a zero crossing 2: Gain update when a zero crossing with a soft ramp Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- v2 * Use SOC_ENUM --- sound/soc/codecs/rt5645.c | 11 +++++++++++ sound/soc/codecs/rt5645.h | 3 +++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9251720..aefce8f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -742,6 +742,14 @@ static int rt5645_spk_put_volsw(struct snd_kcontrol *kcontrol, return ret; } +static const char * const rt5645_dac1_vol_ctrl_mode_text[] = { + "immediately", "zero crossing", "soft ramp" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac1_vol_ctrl_mode, RT5645_PR_BASE, + RT5645_DA1_ZDET_SFT, rt5645_dac1_vol_ctrl_mode_text); + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -808,6 +816,9 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, 1, 1), RT5645_HWEQ("Speaker HWEQ"), + + /* Digital Soft Volume Control */ + SOC_ENUM("DAC1 Digital Volume Control Func", rt5645_dac1_vol_ctrl_mode), }; /** diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 205e071..cfc5f97 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2018,6 +2018,9 @@ /* Codec Private Register definition */ +/* DAC ADC Digital Volume (0x00) */ +#define RT5645_DA1_ZDET_SFT 6 + /* 3D Speaker Control (0x63) */ #define RT5645_3D_SPK_MASK (0x1 << 15) #define RT5645_3D_SPK_SFT 15 -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [patch] ASoC: mediatek: mt2701: fix some error handling in probe
by Dan Carpenter 07 Jul '16

07 Jul '16

The check for if the "afe" allocation failed was too late and there wasn't a check for "afe->platform_priv". Fixes: 43a6a7e71063 ('ASoC: mediatek: add mt2701 platform driver implementation.') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index c865ba1..fc5a87d 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1350,11 +1350,13 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev) ret = 0; afe = devm_kzalloc(&pdev->dev, sizeof(*afe), GFP_KERNEL); + if (!afe) + return -ENOMEM; afe->platform_priv = devm_kzalloc(&pdev->dev, sizeof(*afe_priv), GFP_KERNEL); - afe_priv = afe->platform_priv; - if (!afe) + if (!afe->platform_priv) return -ENOMEM; + afe_priv = afe->platform_priv; afe->dev = &pdev->dev; dev = afe->dev;

3 2

[alsa-devel] [PATCH 1/2] ASoC: rt5645: set RT5645_PRIV_INDEX as volatile
by Bard Liao 07 Jul '16

07 Jul '16

RT5645_PRIV_INDEX(0x6a) indicate the address of PR- registers. So, it should be volatile. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 97bf96e..9251720 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -441,6 +441,7 @@ static bool rt5645_volatile_register(struct device *dev, unsigned int reg) switch (reg) { case RT5645_RESET: + case RT5645_PRIV_INDEX: case RT5645_PRIV_DATA: case RT5645_IN1_CTRL1: case RT5645_IN1_CTRL2: -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: Fix DMA params for different SSC
by Peter Meerwald 07 Jul '16

07 Jul '16

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> follow-up patch from c706f2e55f ASoC: atmel_ssc_dai: distinguish the different SSC cpu_dai id is always 0, use platform_device id to distinguish DMA parameters of SSCs Signed-off-by: Peter Meerwald-Stadler <p.meerwald(a)bct-electronic.com> Cc: "Wu, Songjun" <songjun.wu(a)atmel.com> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.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 841d059..c7daea8 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -320,7 +320,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, return ret; } - dma_params = &ssc_dma_params[dai->id][dir]; + dma_params = &ssc_dma_params[pdev->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; -- 2.9.0

2 1

[alsa-devel] [PATCH 0/3 v3] ALSA: ctl: add dimension information validator
by Takashi Sakamoto 07 Jul '16

07 Jul '16

This patchset is revised version of my previous one: [alsa-devel] [PATCH 0/3 v2] ALSA: ctl: add dimension information validator http://mailman.alsa-project.org/pipermail/alsa-devel/2016-July/109830.html Changes: * Apply simpler prevention of arithmetic overflow. Takashi Sakamoto (3): ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members ALSA: control: add dimension validator for userspace element ALSA: control: add dimension validator for kernel driver sound/core/control.c | 78 ++++++++++++++++++++++++++++++++--------- sound/pci/echoaudio/echoaudio.c | 6 ++-- 2 files changed, 65 insertions(+), 19 deletions(-) -- 2.7.4

2 10

[alsa-devel] [PATCH] ALSA: hda/realtek - add new pin definition in alc225 pin quirk table
by Hui Wang 07 Jul '16

07 Jul '16

We have some Dell laptops which can't detect headset mic, the machines use the codec ALC225, they have some new pin configuration values, after adding them in the alc225 pin quirk table, they work well. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 45d4dc1..e25ef2b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5753,7 +5753,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {} }; #define ALC225_STANDARD_PINS \ - {0x12, 0xb7a60130}, \ {0x21, 0x04211020} #define ALC256_STANDARD_PINS \ @@ -5778,9 +5777,19 @@ static const struct hda_model_fixup alc269_fixup_models[] = { static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, {0x14, 0x901701a0}), SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, + {0x14, 0x901701b0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60150}, + {0x14, 0x901701a0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60150}, {0x14, 0x901701b0}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, {0x14, 0x90170110}, -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: nau8825: jack connection decision with different insertion logic
by John Hsu 06 Jul '16

06 Jul '16

The original design only covers the jack insertion logic is active low. Add more condition to cover no matter the logic is active low and high. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3f30e6e..a2f0d03 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1345,10 +1345,18 @@ EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); static bool nau8825_is_jack_inserted(struct regmap *regmap) { - int status; + int status, jkdet, res; regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); - return !(status & NAU8825_GPIO2JD1); + regmap_read(regmap, NAU8825_REG_JACK_DET_CTRL, &jkdet); + + /* return jack connection status according to jack insertion logic + * active high or active low. + */ + res = !(status & NAU8825_GPIO2JD1) * !(jkdet & NAU8825_JACK_POLARITY) + + (status & NAU8825_GPIO2JD1) * (jkdet & NAU8825_JACK_POLARITY); + + return res ? true : false; } static void nau8825_restart_jack_detection(struct regmap *regmap) -- 2.6.4

3 5

[alsa-devel] [PATCH 0/6] ASoC: simple-card: use asoc_simple_card_parse_xxx()
by Kuninori Morimoto 06 Jul '16

06 Jul '16

Hi Mark These are v3 of [PATCH 00/39 v2] ASoC: add simple-card-core and standardize "simple" card but as new patch series. These are separated into small pieces. Basically, these patches are util.c has new function -> simple-card use it. I will post next remain patches if previous patches was accepted. It will be 6 - 9 patches / 1 series. Now, util.c on mark/topic/simple branch has daifmt() function. So, 1) - 2) : use new daifmt() 3) - 5) : add new dailink_name() and use it. 6) : tidyup rsrc-card This existing rcar-card will be renamed to simple-dpcm-card in final patch. (Maybe we need to consider more better nameing) Kuninori Morimoto (6): 1) ASoC: simple-card: use asoc_simple_card_parse_daifmt() 2) ASoC: rsrc-card: use asoc_simple_card_parse_daifmt() 3) ASoC: simple-card-utils: add asoc_simple_card_parse_dailink_name() 4) ASoC: simple-card: use asoc_simple_card_parse_dailink_name() 5) ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name() 6) ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai include/sound/simple_card.h | 11 +---- include/sound/simple_card_utils.h | 12 +++++ sound/soc/generic/Kconfig | 1 + sound/soc/generic/simple-card-utils.c | 30 +++++++++++++ sound/soc/generic/simple-card.c | 60 ++----------------------- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/rsrc-card.c | 83 ++++++++--------------------------- 7 files changed, 67 insertions(+), 131 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 1/2] sound: lpass-cpu: add module licence and description
by Srinivas Kandagatla 06 Jul '16

06 Jul '16

This patch adds module licence to lpass-cpu driver, without this patch lpass-cpu module would taint with below error: snd_soc_lpass_cpu: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint snd_soc_lpass_cpu: Unknown symbol regmap_write (err 0) snd_soc_lpass_cpu: Unknown symbol devm_kmalloc (err 0) ... Acked-by: Kenneth Westfield <kwestfie(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 3cde9fb..eff3f9a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -586,3 +586,6 @@ int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev) return 0; } EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_platform_remove); + +MODULE_DESCRIPTION("QTi LPASS CPU Driver"); +MODULE_LICENSE("GPL v2"); -- 2.7.4

2 2

[alsa-devel] [PATCH v4 3/4] ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem
by Sylwester Nawrocki 05 Jul '16

05 Jul '16

This patch adds DT binding documentation for Exnos5433 based TM2 and TM2E boards sound subsystem. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v2: - none. Changes since initial version: - dropped clocks, clock-names properties, instead properties from the CODEC node will be used, - property renames: 'samsung,model' -> 'model', 'samsung,i2s-controller' -> 'i2s-controller', 'samsung,speaker-amplifier' -> 'audio-amplifier', - added 'audio-codec' property. --- .../bindings/sound/samsung,tm2-audio.txt | 36 ++++++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt new file mode 100644 index 0000000..3ae56f3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt @@ -0,0 +1,36 @@ +Samsung Exynos5433 TM2(E) audio complex with WM5110 codec + +Required properties: + + - compatible : "samsung,tm2-audio" + - model : the user-visible name of this sound complex + - audio-codec: the phandle of the wm5110 audio codec node, + as described in Documentation/devicetree/bindings/mfd/arizona.txt + - i2s-controller : the phandle of the I2S controller + - audio-amplifier : the phandle of the MAX98504 amplifier + - samsung,audio-routing : a list of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the WM5110's and MAX98504's + pins and the jacks on the board: + HP, SPK, Main Mic, Sub Mic, Third Mic, Headset Mic. + - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator + +Example: + +sound { + compatible = "samsung,tm2-audio"; + audio-codec = <&wm5110>; + i2s-controller = <&i2s0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3 2 0>; + model = "wm5110"; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; +}; -- 1.9.1

1 0

[alsa-devel] [PATCH v4 0/4] ASoC: samsung: Sound support for Exynos5433 TM2(E) boards
by Sylwester Nawrocki 05 Jul '16

05 Jul '16

This patch series adds a MFD driver for Samsung Exynos5433 SoC Low Power Audio Subsystem and an ASoC machine driver for Exynos5433 based TM2/TM2E boards. The changes since v3 are listed in individual patches, the main change was rewriting LPASS driver as a MFD driver. It seems much better now as the dependencies between LPASS Top block and its IP sub-blocks are properly taken care of. Sylwester Nawrocki (4): mfd: Add DT bindings documentation for Samsung Exynos LPASS mfd: Add Samsung Exynos Low Power Audio Subsystem driver ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem ASoC: samsung: Add machine driver for Exynos5433 based TM2 board .../bindings/mfd/samsung,exynos5433-lpass.txt | 21 + .../bindings/sound/samsung,tm2-audio.txt | 36 ++ drivers/mfd/Kconfig | 8 + drivers/mfd/Makefile | 1 + drivers/mfd/exynos-lpass.c | 185 +++++++ include/linux/mfd/syscon/exynos5-pmu.h | 4 +- sound/soc/samsung/Kconfig | 9 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/tm2_wm5110.c | 579 +++++++++++++++++++++ 9 files changed, 844 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 drivers/mfd/exynos-lpass.c create mode 100644 sound/soc/samsung/tm2_wm5110.c -- 1.9.1

1 0

[alsa-devel] [PATCH RFC alsa-lib 0/5] Add thread-safety to PCM API
by Takashi Iwai 05 Jul '16

05 Jul '16

Hi, here is an experimental patchset to add multi thread safety to PCM functions. Basically ALSA PCM functions are thread-unsafe, and applications are supposed to do the proper protection against racy accesses. The reality is, however, that application developers don't care such, as alsa-lib works in most cases. Of course, users still get occasionally mysterious crashes. As a workaround, this patchset adds the pthread mutex protection to most of exported PCM functions. This is slightly an overkill, but the biggest merit is that it's easy to implement; I just wrapped the functions and replaced the internal ones with unlocked versions. To be noted, there is an optimization for the direct hw PCM access. Performance-sensitive applications like JACK should work like before without any overhead. Another bonus by this addition is that we can finally get rid of home brew (and deadly smelling) atomic macros from the tree, since it's now more widely protected. I lightly tested on my local machines and it seems working well, so far. Comments and suggestions welcome. thanks, Takashi === Takashi Iwai (5): pcm: Add thread-safety to PCM API test: Add pcm-multi-thread program Add pcm-multi-thread to .gitignore pcm: Remove superfluous rmb() from PCM meter plugin pcm: Remove home brew atomic operations .gitignore | 1 + configure.ac | 15 ++ include/Makefile.am | 2 +- include/iatomic.h | 170 ------------------ src/pcm/Makefile.am | 2 +- src/pcm/atomic.c | 43 ----- src/pcm/pcm.c | 468 +++++++++++++++++++++++++++++++++++++++++------- src/pcm/pcm_direct.c | 4 +- src/pcm/pcm_dmix.c | 13 +- src/pcm/pcm_dshare.c | 13 +- src/pcm/pcm_dsnoop.c | 15 +- src/pcm/pcm_file.c | 21 ++- src/pcm/pcm_generic.c | 10 +- src/pcm/pcm_hw.c | 3 + src/pcm/pcm_ioplug.c | 11 +- src/pcm/pcm_local.h | 62 ++++++- src/pcm/pcm_meter.c | 1 - src/pcm/pcm_mmap.c | 16 +- src/pcm/pcm_params.c | 2 +- src/pcm/pcm_plugin.c | 72 ++------ src/pcm/pcm_plugin.h | 2 - src/pcm/pcm_rate.c | 41 ++--- src/pcm/pcm_route.c | 2 +- test/Makefile.am | 3 +- test/pcm-multi-thread.c | 263 +++++++++++++++++++++++++++ 25 files changed, 851 insertions(+), 404 deletions(-) delete mode 100644 include/iatomic.h delete mode 100644 src/pcm/atomic.c create mode 100644 test/pcm-multi-thread.c -- 2.9.0

2 7

[alsa-devel] [PATCH v2 0/6] Sound support for Exynos5433 TM2(E) boards
by Sylwester Nawrocki 05 Jul '16

05 Jul '16

This patch series adds sound supoort for the Exynos 5433 based TM2 and TM2E board. It includes minimal drivers for the Low Power Audio Subsystem (LPASS) and the MAX98504 class D speaker amplifier as well as the actual asoc machine driver. The changes are listed in individual patches, there were only v1 3/6, 4/6 patches (max98504), for others I skipped from RFC to v2. Inha Song (2): ASoC: samsung: Add Samsung Low Power Audio Subsystem driver ASoC: samsung: Add machine driver for Exynos5433 based TM2 board Sylwester Nawrocki (4): ASoC: samsung: Add DT bindings documentation for LPASS ASoC: Add DT bindings documentation for max98504 amplifier ASoC: max98504: Add max98504 speaker amplifier driver ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem .../devicetree/bindings/sound/max98504.txt | 26 + .../bindings/sound/samsung,exynos5433-lpass.txt | 8 + .../bindings/sound/samsung,tm2-audio.txt | 36 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98504.c | 356 +++++++++++++ sound/soc/codecs/max98504.h | 76 +++ sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 4 + sound/soc/samsung/lpass.c | 162 ++++++ sound/soc/samsung/lpass.h | 47 ++ sound/soc/samsung/tm2_wm5110.c | 579 +++++++++++++++++++++ 12 files changed, 1313 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,exynos5433-lpass.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 sound/soc/codecs/max98504.c create mode 100644 sound/soc/codecs/max98504.h create mode 100644 sound/soc/samsung/lpass.c create mode 100644 sound/soc/samsung/lpass.h create mode 100644 sound/soc/samsung/tm2_wm5110.c -- 1.9.1

3 20

Re: [alsa-devel] [PATCH] hda/realtek: Add Lenovo L460 to docking unit fixup
by Takashi Iwai 05 Jul '16

05 Jul '16

On Tue, 05 Jul 2016 10:40:22 +0200, Torsten Hilbrich wrote: > > This solves the issue that a headphone is not working on the docking > unit. > > Signed-off-by: Torsten Hilbrich <torsten.hilbrich(a)secunet.com> Applied, thanks. Takashi

1 0

[alsa-devel] [PATCH 0/4] ASoC: Mediatek: add MT2701 platform and machine driver
by Garlic Tseng 05 Jul '16

05 Jul '16

Some patches of the patchset "ASoC: Mediatek: Add support for MT2701 SOC" have been applied to the broonie tree, so I send a new patchset for the remaining patches. The patchset adds mt2701 support based on Mediatek AFE common structure. Change since the previous patchset: * remove unnecessary set_clkdiv and set_fmt ops in mt2701 platform driver * add another snd_soc_dai_driver for bt-sco codec to support 8k and 16k Garlic Tseng (4): ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- sound/soc/codecs/bt-sco.c | 52 +- sound/soc/mediatek/Kconfig | 21 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 9 - sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1654 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 412 +++++ 8 files changed, 2144 insertions(+), 26 deletions(-) create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c -- 1.9.1

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Re: [alsa-devel] ASUS STRIX sound card quiet microphone (like, really quiet ...)
by Clemens Ladisch 05 Jul '16

05 Jul '16

auspicious(a)inventati.org wrote: > On 05/07/2016 07:19, Clemens Ladisch wrote: >> auspicious(a)inventati.org wrote: >>> I currently am using an ASUS STRIX RAID DLX sound card. It contains the following processor: >>> C-Media USB2.0 6632AX High-Definition Sound Processor (Max. 384KHz / 24bit) >>> >>> the microphone bit has always been troublesome. >>> it started working but it is incredibly quiet. >> >> You did not mention mixer controls. >> Does it have any? (Try alsamixer.) > > Attached image in the attachment That "Input Gain" looks OK. Does it actually work? Does it make the mic input quieter? It's possible that there is some vendor-specific stuff that the USB audio driver does not know. Does it work in Windows with Microsoft's generic USB audio driver, or does it need a custom driver from Asus? Regards, Clemens

1 0

[alsa-devel] [PATCH] ASoC: rt5645: disable soft volume control
by Bard Liao 05 Jul '16

05 Jul '16

With soft volume control function, rt5645/rt5650 will start with 0dB volume and take a little time to adjust volume to given value. And it will cause a bad user experience. This patch disable it. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 97bf96e..c969f48 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -63,6 +63,7 @@ static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x20, 0x611f}, {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, + {RT5645_PR_BASE + 0x00, 0x0a2a}, {RT5645_ASRC_4, 0x0120}, }; -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] ASUS STRIX sound card quiet microphone (like, really quiet ...)
by auspicious＠inventati.org 05 Jul '16

05 Jul '16

Dear mailing list individuals, I currently am using an ASUS STRIX RAID DLX sound card. It contains the following processor: C-Media USB2.0 6632AX High-Definition Sound Processor (Max. 384KHz / 24bit) More information: https://www.asus.com/Sound-Cards/STRIX-RAID-DLX/specifications It gets detected as so on Debian testing with Linux kernel 4.6: card 1: DLX [STRIX RAID DLX], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DLX [STRIX RAID DLX], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DLX [STRIX RAID DLX], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 The output of the card has been wonderfully fine ever since I have had it, but the microphone bit has always been troublesome. On Linux Mint 17.3 it would not work (kernel 4.1 I believe). On Linux Mint 18 (kernel 4.4) it started working but it is incredibly quiet. On the loudest setting, I have to put the microphone against my mouth for anyone to hear. On the same system using on-board Intel audio, this issue does not occur. I am using Debian testing to have more recent software, but nothing made a difference. What can I do about this? I do not really know whether I need to contact the Linux kernel guys or ALSA. I am rather inexperienced with Linux driver things. Sincerely yours, Desperate Individual

2 1

[alsa-devel] SND_PCM_STREAM_CAPTURE and SND_PCM_NONBLOCK producing strange artifacts
by Trent Reed 05 Jul '16

05 Jul '16

Hey All, I've been banging my head against the wall for about a week on this bug. The following gist shows my sample reproduction code: https://gist.github.com/TReed0803/985c5d5c3d295245e19006a27be447c3 I'm simply opening up a non-blocking PCM capture stream and writing the contents of the reads to stdout. (Originally, I was writing to the playback stream, but I was hearing this strange static occasionally!) It's the static I'm trying to debug. It doesn't happen every time. In fact, sometimes I'll go a few consecutive executions without hearing it. I was able to capture some of the bad data, and I loaded it up in Audacity for visualization: https://drive.google.com/file/d/0B-1aumGKQcQTcUJoZzIwRWhYSFE/view?usp=shari… It looks like the internal buffer occasionally is sending me more data than it actually captured, and I end up either reading old PCM data from the internal ring buffer, or (at the very beginning) a bunch of zeros. Can anyone help me understand what is going on? What could cause these definitely incorrect samples to be recorded? (I get the same effect regardless of hardware, but I will list hardware just in case.) I hope I have all the information you might need: Hardware: Samson Meteor Mic (USB-Audio, USB Mixer) [Though, it even happens with my built-in microphone] ALSA version: Advanced Linux Sound Architecture Driver Version k4.4.0-28-generic. apt-cache policy (installed alsa-lib version): 1.1.0-0ubuntu1 Thanks, - Trent Reed

2 2

[alsa-devel] [PATCH v2 0/2] Add audio support for Microsoft Surface 3 tablet
by Stephen Just 05 Jul '16

05 Jul '16

This series adds audio support for the Surface 3 tablet. This tablet identifes an rt5645 audio codec using the ACPI identifer 10EC5640, which is already associated with the rt5640, thus a quirk is required to have the correct drivers loaded. Additionally, in order for jack-detection and the tablet's internal microphone to work, the Surface 3 must be configured in the audio codec driver similarly to other devices. Resolves: https://bugzilla.kernel.org/show_bug.cgi?id=98001 Changes in v2: Remove alsa-lib changes from series Stephen Just (2): ASoC: rt5645: Add support for Surface 3 tablet ASoC: Intel: Atom: add quirk for CHT w/ RT5645 sound/soc/codecs/rt5645.c | 7 +++++++ sound/soc/intel/atom/sst/sst_acpi.c | 29 ++++++++++++++++++++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 1 + 3 files changed, 36 insertions(+), 1 deletion(-) -- 2.7.4

3 6

[alsa-devel] [PATCH] [alsa-lib] conf/ucm: chtrt5645: Add UCM config for chtrt5645
by Stephen Just 05 Jul '16

05 Jul '16

Add a UCM configuration for the rt5645 codec on Intel's Cherry-Trail platform. Tested on the Microsoft Surface 3. Signed-off-by: Stephen Just <stephenjust(a)gmail.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/chtrt5645/HiFi.conf | 209 ++++++++++++++++++++++++++++++++++ src/conf/ucm/chtrt5645/Makefile.am | 4 + src/conf/ucm/chtrt5645/chtrt5645.conf | 5 + 5 files changed, 220 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/chtrt5645/HiFi.conf create mode 100644 src/conf/ucm/chtrt5645/Makefile.am create mode 100644 src/conf/ucm/chtrt5645/chtrt5645.conf diff --git a/configure.ac b/configure.ac index 53fca33..9eebd0a 100644 --- a/configure.ac +++ b/configure.ac @@ -659,6 +659,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/GoogleNyan/Makefile \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ + src/conf/ucm/chtrt5645/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 3cce36a..8865770 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf new file mode 100644 index 0000000..0c2c83c --- /dev/null +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -0,0 +1,209 @@ +SectionVerb { + # ALSA PCM + Value { + TQ "HiFi" + + # ALSA PCM device for HiFi + PlaybackPCM "hw:chtrt5645,0" + CapturePCM "hw:chtrt5645,0" + } + + EnableSequence [ + cdev "hw:chtrt5645" + + # Enable audio output path + cset "name='codec_out1 mix 0 pcm0_in Switch' on" + cset "name='media0_out mix 0 media1_in Switch' on" + + cset "name='media1_in Gain 0 Ramp Delay' 50" + cset "name='media1_in Gain 0 Switch' on" + cset "name='media1_in Gain 0 Volume' 80% 80%" + + cset "name='pcm0_in Gain 0 Ramp Delay' 50" + cset "name='pcm0_in Gain 0 Switch' on" + cset "name='pcm0_in Gain 0 Volume' 80% 80%" + + cset "name='codec_out1 Gain 0 Ramp Delay' 50" + cset "name='codec_out1 Gain 0 Switch' on" + cset "name='codec_out1 Gain 0 Volume' 70% 70%" + + # Enable audio input path + cset "name='pcm1_out mix 0 media_loop2_in Switch' on" + cset "name='media_loop2_out mix 0 codec_in0 Switch' on" + + cset "name='codec_in0 Gain 0 Ramp Delay' 50" + cset "name='codec_in0 Gain 0 Switch' on" + cset "name='codec_in0 Gain 0 Volume' 80% 80%" + + cset "name='media_loop2_out Gain 0 Ramp Delay' 50" + cset "name='media_loop2_out Gain 0 Switch' on" + cset "name='media_loop2_out Gain 0 Volume' 80% 80%" + + cset "name='pcm1_out Gain 0 Ramp Delay' 50" + cset "name='pcm1_out Gain 0 Switch' on" + cset "name='pcm1_out Gain 0 Volume' 80% 80%" + + # Output Configuration + cset "name='DAC L2 Mux' 'IF1 DAC'" + cset "name='DAC R2 Mux' 'IF1 DAC'" + cset "name='Mono DAC MIXL DAC L2 Switch' on" + cset "name='Mono DAC MIXR DAC R2 Switch' on" + cset "name='DAC2 Playback Switch' on" + + cset "name='HPOVOL MIXL DAC2 Switch' on" + cset "name='HPOVOL MIXR DAC2 Switch' on" + cset "name='HPO MIX HPVOL Switch' on" + cset "name='HPOVOL L Switch' on" + cset "name='HPOVOL R Switch' on" + + cset "name='SPK MIXL DAC L2 Switch' on" + cset "name='SPK MIXR DAC R2 Switch' on" + cset "name='SPOL MIX SPKVOL L Switch' on" + cset "name='SPOR MIX SPKVOL R Switch' on" + cset "name='SPKVOL L Switch' on" + cset "name='SPKVOL R Switch' on" + + # Input Configuration + cset "name='Stereo1 DMIC Mux' 0" + cset "name='Stereo1 ADC2 Mux' 1" + cset "name='ADC Capture Switch' on" + cset "name='ADC Capture Volume' 31" + cset "name='ADC Boost Capture Volume' 3" + cset "name='Mono ADC Capture Volume' 63" + cset "name='Mono ADC Boost Capture Volume' 2" + cset "name='IN Capture Volume' 63" + cset "name='I2S2 Func Switch' on" + + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + # Disable audio output path + cset "name='codec_out1 mix 0 pcm0_in Switch' off" + cset "name='media0_out mix 0 media1_in Switch' off" + + cset "name='media1_in Gain 0 Switch' off" + cset "name='pcm0_in Gain 0 Switch' off" + cset "name='codec_out1 Gain 0 Switch' off" + + # Disable audio input path + cset "name='pcm1_out mix 0 media_loop2_in Switch' off" + cset "name='media_loop2_out mix 0 codec_in0 Switch' off" + + cset "name='media_loop2_out Gain 0 Switch' off" + cset "name='pcm1_out Gain 0 Switch' off" + cset "name='codec_in0 Gain 0 Switch' off" + ] +} + +SectionDevice."Speaker" { + Comment "Speaker" + + Value { + PlaybackChannels "2" + } + + ConflictingDevice [ + "Headphone" + ] + + EnableSequence [ + cdev "hw:chtrt5645" + + cset "name='Ext Spk Switch' on" + cset "name='Speaker Channel Switch' on" + cset "name='Speaker Playback Volume' 39" + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + cset "name='Ext Spk Switch' off" + cset "name='Speaker Channel Switch' off" + ] +} + +SectionDevice."Headphone".0 { + Comment "Headphones" + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cdev "hw:chtrt5645" + + cset "name='Headphone Switch' on" + cset "name='Headphone Channel Switch' on" + cset "name='Headphone Playback Volume' 39" + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + cset "name='Headphone Switch' off" + cset "name='Headphone Channel Switch' off" + ] +} + +SectionDevice."Mic".0 { + Comment "Internal Microphone" + + Value { + CaptureChannels "2" + CapturePriority "150" + } + + EnableSequence [ + cdev "hw:chtrt5645" + + cset "name='Int Mic Switch' on" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + cset "name='Int Mic Switch' off" + ] +} + +SectionDevice."HSMic".0 { + Comment "Headset Microphone" + + Value { + CaptureChannels "2" + JackControl "Headset Mic Jack" + JackHWMute "Mic" + } + + EnableSequence [ + cdev "hw:chtrt5645" + + cset "name='Headset Mic Switch' on" + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + cset "name='Sto1 ADC MIXL ADC1 Switch' on" + cset "name='Sto1 ADC MIXR ADC1 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + cset "name='Headset Mic Switch' off" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='Sto1 ADC MIXL ADC1 Switch' off" + cset "name='Sto1 ADC MIXR ADC1 Switch' off" + ] +} diff --git a/src/conf/ucm/chtrt5645/Makefile.am b/src/conf/ucm/chtrt5645/Makefile.am new file mode 100644 index 0000000..b3a82f5 --- /dev/null +++ b/src/conf/ucm/chtrt5645/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/chtrt5645 +ucm_DATA = chtrt5645.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/chtrt5645/chtrt5645.conf b/src/conf/ucm/chtrt5645/chtrt5645.conf new file mode 100644 index 0000000..dfc36c8 --- /dev/null +++ b/src/conf/ucm/chtrt5645/chtrt5645.conf @@ -0,0 +1,5 @@ +Comment "Intel SoC Audio Device" +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} -- 2.9.0

2 1

[alsa-devel] [PATCH] ALSA: ppc/awacs: shut up maybe-uninitialized warning
by Arnd Bergmann 05 Jul '16

05 Jul '16

The awacs sound driver produces a false-positive warning in ppc64_defconfig: sound/ppc/awacs.c: In function 'snd_pmac_awacs_init': include/sound/control.h:219:9: warning: 'master_vol' may be used uninitialized in this function [-Wmaybe-uninitialized] I haven't come up with a good way to rewrite the code to avoid the warning, so here is a bad one: I initialize the variable before the conditionall initialization so gcc no longer has to worry about it. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/ppc/awacs.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 09da7b52bc2e..1468e4b7bf93 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -991,6 +991,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) if (err < 0) return err; } + master_vol = NULL; if (pm7500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500), -- 2.9.0

2 1

[alsa-devel] [PATCH v2 0/7] of_graph: prepare for ALSA graph support
by Kuninori Morimoto 05 Jul '16

05 Jul '16

Hi Rob These are v2 of of_graph patch-set Now OF graph is mainly used by V4L2 SoC, and ALSA SoC is using different style for SoC <-> Codec binding. But, for example, HDMI case, V4L2 <-> ALSA need to collaborate, and then ALSA SoC needs to adjust to OF graph. OTOH, V4L2's "OF graph" position is same as ALSA SoC "sound card" position. And ALSA SoC side want to keep existing supported feature on new OF graph style. I'm posting this on ALSA SoC ML now. Now, current of_graph is indicating port/endpoint, but there is no way to understand that it is for video port ? or sound port ? or other device port ? For example, HDMI has video port, and sound port. Because of this reason, ALSA SoC side can't handle OF graph correctly. Thus, this patch-set tries to add new "type" on OF graph. And this patch-set includes small feature which are useful for ALSA SoC side OF graph support. Kuninori Morimoto (7): Documentation: of: add type property of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_port_type_is() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_top_port() of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() of_graph: add of_graph_get_endpoint_count() Documentation/devicetree/bindings/graph.txt | 21 ++++ drivers/of/base.c | 164 ++++++++++++++++++++++++++-- include/linux/of_graph.h | 61 +++++++++++ 3 files changed, 236 insertions(+), 10 deletions(-) -- 1.9.1

3 12

[alsa-devel] [PATCH] ASoC: rockchip: implement system suspend/resume for spdif
by Sugar Zhang 05 Jul '16

05 Jul '16

restore hw registers after power loss during a suspend/resume cycle. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/rockchip_spdif.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 100781e..00ae0e5 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -392,9 +392,35 @@ static int rk_spdif_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int rockchip_spdif_suspend(struct device *dev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + + regcache_mark_dirty(spdif->regmap); + + return 0; +} + +static int rockchip_spdif_resume(struct device *dev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(spdif->regmap); + pm_runtime_put(dev); + + return ret; +} +#endif + static const struct dev_pm_ops rk_spdif_pm_ops = { SET_RUNTIME_PM_OPS(rk_spdif_runtime_suspend, rk_spdif_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(rockchip_spdif_suspend, rockchip_spdif_resume) }; static struct platform_driver rk_spdif_driver = { -- 1.9.1

1 0

[alsa-devel] atmel_ssc_dai.c, use of dai->id
by Peter Meerwald-Stadler 05 Jul '16

05 Jul '16

Hello, I am trying to use SSC1 (not SSC0 which might work) on a SAMA5D4-Xplained board (4.1 kernel, but close to mainline) c706f2e5 ASoC: atmel_ssc_dai: distinguish the different SSC fixes an issue with cpu_dai id being always 0, while the ssc id should have been used (or the platform_device id) I think there is a remaining issue with DMA parameters in atmel_ssc_startup(): line 324: dma_params = &ssc_dma_params[dai->id][dir]; I think it should be dma_params = &ssc_dma_params[pdev->id][dir]; I am new to this platform, please comment I can submit a formal patch later-on of course thanks, regards, p. -- Peter Meerwald-Stadler +43-664-2444418 (mobile)

2 1

[alsa-devel] [PATCH] drm: sti: fix prototypes after API change
by Arnd Bergmann 05 Jul '16

05 Jul '16

Like the mediatek hdmi driver, the sti driver now suffers from an API change in linux-next: drivers/gpu/drm/sti/sti_hdmi.c:1250:15: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] .hw_params = hdmi_audio_hw_params, ^~~~~~~~~~~~~~~~~~~~ drivers/gpu/drm/sti/sti_hdmi.c:1250:15: note: (near initialization for 'audio_codec_ops.hw_params') This follows the API change, adding the extra function arguments. I suppose we want this to be part of the ASoC merge in linux-next for now too. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 2c348e505328 ("drm: sti: Add ASoC generic hdmi codec support.") Fixes: efc9194bcff8 ("ASoC: hdmi-codec: callback function will be called with private data") Link: https://lkml.org/lkml/2016/6/29/441 --- drivers/gpu/drm/sti/sti_hdmi.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/gpu/drm/sti/sti_hdmi.c b/drivers/gpu/drm/sti/sti_hdmi.c index 927e2b2e5af2..fedc17f98d9b 100644 --- a/drivers/gpu/drm/sti/sti_hdmi.c +++ b/drivers/gpu/drm/sti/sti_hdmi.c @@ -1169,7 +1169,7 @@ static int hdmi_audio_configure(struct sti_hdmi *hdmi, return hdmi_audio_infoframe_config(hdmi); } -static void hdmi_audio_shutdown(struct device *dev) +static void hdmi_audio_shutdown(struct device *dev, void *data) { struct sti_hdmi *hdmi = dev_get_drvdata(dev); int audio_cfg; @@ -1186,6 +1186,7 @@ static void hdmi_audio_shutdown(struct device *dev) } static int hdmi_audio_hw_params(struct device *dev, + void *data, struct hdmi_codec_daifmt *daifmt, struct hdmi_codec_params *params) { @@ -1221,7 +1222,7 @@ static int hdmi_audio_hw_params(struct device *dev, return 0; } -static int hdmi_audio_digital_mute(struct device *dev, bool enable) +static int hdmi_audio_digital_mute(struct device *dev, void *data, bool enable) { struct sti_hdmi *hdmi = dev_get_drvdata(dev); @@ -1235,7 +1236,7 @@ static int hdmi_audio_digital_mute(struct device *dev, bool enable) return 0; } -static int hdmi_audio_get_eld(struct device *dev, uint8_t *buf, size_t len) +static int hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf, size_t len) { struct sti_hdmi *hdmi = dev_get_drvdata(dev); struct drm_connector *connector = hdmi->drm_connector; -- 2.9.0

5 8

[alsa-devel] Alsa driver for Tascam US-16x08
by Detlef Urban 04 Jul '16

04 Jul '16

Hi, I'm a former application developer at a big IT company. Most of my private time I spend with semi-professional studio recording using Ubuntu Studio (Ardour). I bought a nice external sound device from Tascam called US-16x08 and it works very well. This device also contains some signal processing capabilities like a compressor and a parametric equalizer. But unfortunately there was no Linux driver available from the manufacturer and so I cant use there functionalities. So I started with a nightmare of re-engineering by monitoring USB communication via usbmon and wireshark. Now after 3-4 weeks I'm able to completely control this device with a customized snd-usb-audio kernel module. Additionally I developed a tiny GUI mixer application to test my driver. But I'm not very familiar with non-coding aspects of driver deployment (documentation, deployment etc.). Can some please help me with some hints/links, so that other users can benefit of my work? regards Detlef

1 0

[alsa-devel] [PATCH -next] ASoC: cs35l33: Remove unused including <linux/version.h>
by weiyj_lk＠163.com 04 Jul '16

04 Jul '16

From: Wei Yongjun <weiyj.lk(a)gmail.com> Remove including <linux/version.h> that don't need it. Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/codecs/cs35l33.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 622a111..c20edd4 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -12,7 +12,6 @@ */ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h>

2 1

[alsa-devel] [PATCH 0/7] ASoC: rockchip: Fix audio on Veyron.
by Enric Balletbo i Serra 04 Jul '16

04 Jul '16

Hi all, This patch series adds support and fixes the audio support for Veyron devices. With these patches I'm able to playback from speakers and headphones, and record from the microphone on a Veyron Jerry Chromebook. Patches 1/7 and 2/7 were already sent by John Keeping,[1] [2], but I think it's interesting have them included here. Patches from 1/7 to 5/7 should go through the Mark Brown ASoC tree. Patches 6/7 and 7/7 should go through the Heiko Stuebner rockchip tree. [1] https://patchwork.kernel.org/patch/9017491/ [2] https://patchwork.kernel.org/patch/9017501/ Enric Balletbo i Serra (5): ASoC: rockchip-max98090: Fix NULL pointer de reference while accessing to jack ASoC: rockchip-max98090: Fix the Headset Mic route. ASoC: rockchip-max98090: Fix DAPM unknown pin for Headset Jack ARM: dts: rockchip: Add shared file for audio related nodes for veyron boards ARM: dts: rockchip: veyron: Add analog audio codecs. John Keeping (2): ASoC: rockchip: Revert "ASoC: rockchip: i2s: remove unused variables" ASoC: rockchip: Revert "ASoC: rockchip: i2s: separate capture and playback" arch/arm/boot/dts/rk3288-veyron-analog-audio.dtsi | 102 ++++++++++++++++++++++ arch/arm/boot/dts/rk3288-veyron-chromebook.dtsi | 1 + sound/soc/rockchip/rockchip_i2s.c | 87 +++++++++++------- sound/soc/rockchip/rockchip_max98090.c | 54 ++++++------ 4 files changed, 187 insertions(+), 57 deletions(-) create mode 100644 arch/arm/boot/dts/rk3288-veyron-analog-audio.dtsi -- 2.1.0

3 14

[alsa-devel] [PATCH -next] ASoC: sunxi: remove redundant dev_err call in sun4i_i2s_probe()
by weiyj_lk＠163.com 04 Jul '16

04 Jul '16

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> There is a error message within devm_ioremap_resource already, so remove the dev_err call to avoid redundant error message. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/sunxi/sun4i-i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index fab5234..687a8f8 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -599,10 +599,8 @@ static int sun4i_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) { - dev_err(&pdev->dev, "Can't request IO region\n"); + if (IS_ERR(regs)) return PTR_ERR(regs); - } irq = platform_get_irq(pdev, 0); if (irq < 0) {

1 0

[alsa-devel] [PATCH] ASOC: Atmel: ClassD: Simplify use of devm_ioremap_resource
by Amitoj Kaur Chawla 04 Jul '16

04 Jul '16

Remove unneeded error handling on the result of a call to platform_get_resource when the value is passed to devm_ioremap_resource. The Coccinelle semantic patch that makes this change is as follows: // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l; @@ - res = platform_get_resource(pdev, IORESOURCE_MEM, n); ... when != res - if (res == NULL) { ... $goto l;\|return ret;$ } ... when != res + res = platform_get_resource(pdev, IORESOURCE_MEM, n); e = devm_ioremap_resource(e1, res); // </smpl> Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/soc/atmel/atmel-classd.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 6107de9..6d9b8b4 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -593,11 +593,6 @@ static int atmel_classd_probe(struct platform_device *pdev) } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "no memory resource\n"); - return -ENXIO; - } - io_base = devm_ioremap_resource(dev, res); if (IS_ERR(io_base)) { ret = PTR_ERR(io_base); -- 1.9.1

3 6

[alsa-devel] [PATCH] ASOC: atmel-pdmic: Simplify use of devm_ioremap_resource
by Amitoj Kaur Chawla 04 Jul '16

04 Jul '16

Remove unneeded error handling on the result of a call to platform_get_resource when the value is passed to devm_ioremap_resource. The Coccinelle semantic patch that makes this change is as follows: // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l; @@ - res = platform_get_resource(pdev, IORESOURCE_MEM, n); ... when != res - if (res == NULL) { ... $goto l;\|return ret;$ } ... when != res + res = platform_get_resource(pdev, IORESOURCE_MEM, n); e = devm_ioremap_resource(e1, res); // </smpl> Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/soc/atmel/atmel-pdmic.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index aee4787..5f56da6 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -624,11 +624,6 @@ static int atmel_pdmic_probe(struct platform_device *pdev) } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "no memory resource\n"); - return -ENXIO; - } - io_base = devm_ioremap_resource(dev, res); if (IS_ERR(io_base)) { ret = PTR_ERR(io_base); -- 1.9.1

1 0

[alsa-devel] [PATCH] ALSA: timer: Fix negative queue usage by racy accesses
by Takashi Iwai 04 Jul '16

04 Jul '16

The user timer tu->qused counter may go to a negative value when multiple concurrent reads are performed since both the check and the decrement of tu->qused are done in two individual locked contexts. This results in bogus read outs, and the endless loop in the user-space side. The fix is to move the decrement of the tu->qused counter into the same spinlock context as the zero-check of the counter. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index e722022d325d..9a6157ea6881 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1955,6 +1955,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, qhead = tu->qhead++; tu->qhead %= tu->queue_size; + tu->qused--; spin_unlock_irq(&tu->qlock); if (tu->tread) { @@ -1968,7 +1969,6 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, } spin_lock_irq(&tu->qlock); - tu->qused--; if (err < 0) goto _error; result += unit; -- 2.9.0

1 0

[alsa-devel] [PATCH] ASoC: codecs: Allow user to select WM9713 codec
by Alexander Shiyan 04 Jul '16

04 Jul '16

Add simple description and necessary options to allow to select Wolfson WM9713 codec. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4d82a58..6721427 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1010,8 +1010,9 @@ config SND_SOC_WM9712 tristate config SND_SOC_WM9713 - tristate + tristate "Wolfson Microelectronics WM9713 codec" select REGMAP_AC97 + select SND_SOC_AC97_BUS # Amp config SND_SOC_LM4857 -- 2.4.9

2 1

Re: [alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by Mark Brown 04 Jul '16

04 Jul '16

On Mon, Jul 04, 2016 at 10:26:54AM +0800, KaiChieh Chuang wrote: > On Thu, 2016-06-30 at 03:08 +0800, Mark Brown wrote: > > On Sun, Jun 05, 2016 at 02:58:59PM +0800, Kai Chieh Chuang wrote: > > > The BE receiving SND_SOC_DAPM_STREAM_STOP event, while still used by a > > > FE who is in STREAM_START state. The DAPM will just inactive BE related > > > widget/dai without checking. This just doesn't seems right. > > Why not? The backend is still active and running, why would it expect > > to be stopped unless it's supposed to actually stop? > Yes, I don't expect it to stop. > But at this situation, the BE change to state STREAM_STOP while > still used by a FE in STREAM_START state. > Shouldn't the BE stay in the STREAM_START state? Yes, but that sounded like the situation you were describing? It's really hard to follow what you're trying to say here, perhaps just send the fix you're proposing?

1 0

[alsa-devel] [PATCH] ASoC: bt-sco: add config prompt
by Garlic Tseng 04 Jul '16

04 Jul '16

Add config prompt for bt-sco codec driver Signed-off-by: Garlic Tseng <garlic.tseng(a)mediatek.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f2f8603..1cd6ab3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -383,7 +383,7 @@ config SND_SOC_ALC5632 tristate config SND_SOC_BT_SCO - tristate + tristate "Dummy BT SCO codec driver" config SND_SOC_CQ0093VC tristate -- 1.9.1

2 1

Re: [alsa-devel] [PATCH 3/3] [alsa-lib] conf/ucm: chtrt5645: Add UCM config for chtrt5645
by Mark Brown 04 Jul '16

04 Jul '16

On Sun, Jul 03, 2016 at 11:06:07PM -0600, Stephen Just wrote: > Add a UCM configuration for the rt5645 codec on Intel's Cherry-Trail > platform. Tested on the Microsoft Surface 3. This is not a kernel patch, it should be sent separately and sent to the alsa-lib maintainers (at least Taksahi).

1 0

[alsa-devel] [PATCH] clkdev: add devm_of_clk_get()
by Kuninori Morimoto 04 Jul '16

04 Jul '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This is based on devm_clk_get() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- drivers/clk/clkdev.c | 26 ++++++++++++++++++++++++++ include/linux/clk.h | 7 +++++++ 2 files changed, 33 insertions(+) diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 89cc700..93a613b 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -55,6 +55,32 @@ struct clk *of_clk_get(struct device_node *np, int index) } EXPORT_SYMBOL(of_clk_get); +static void devm_of_clk_release(struct device *dev, void *res) +{ + clk_put(*(struct clk **)res); +} + +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + struct clk **ptr, *clk; + + ptr = devres_alloc(devm_of_clk_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return ERR_PTR(-ENOMEM); + + clk = of_clk_get(np, index); + if (!IS_ERR(clk)) { + *ptr = clk; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return clk; +} +EXPORT_SYMBOL(devm_of_clk_get); + static struct clk *__of_clk_get_by_name(struct device_node *np, const char *dev_id, const char *name) diff --git a/include/linux/clk.h b/include/linux/clk.h index 834179f..01005e78 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -494,6 +494,8 @@ struct of_phandle_args; #if defined(CONFIG_OF) && defined(CONFIG_COMMON_CLK) struct clk *of_clk_get(struct device_node *np, int index); +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index); struct clk *of_clk_get_by_name(struct device_node *np, const char *name); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec); #else @@ -501,6 +503,11 @@ static inline struct clk *of_clk_get(struct device_node *np, int index) { return ERR_PTR(-ENOENT); } +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + return ERR_PTR(-ENOENT); +} static inline struct clk *of_clk_get_by_name(struct device_node *np, const char *name) { -- 1.9.1

2 2

[alsa-devel] [PATCH v5 0/9] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 04 Jul '16

04 Jul '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since v4: * remove SND_SOC_MEDIATEK config prompt * fix config prompt to meaningful one for SND_SOC_MT2701_CS42448 * remove some unnecessary random space change Change since v3: * fix comments style. Change since v2: * fix comments style. * replace EXPORT_SYMBOL with EXPORT_SYMBOL_GPL * reduce debug message * move mutex lock for irq-allocated to AFE struct * remove func mtk_simple_isr and call snd_pcm_period_elapsed directly * mt2701 driver probe will fail if clock initial error * fix dapm on/off switch control naming Change since v1: implement common private structure, fe dai and platform driver. let mt8173 and mt2701 use the common structure. Garlic Tseng (9): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: implement mediatek common structure ASoC: mediatek: let mt8173 use mediatek common structure ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add structure define and clock control for 2701 ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 37 +- sound/soc/mediatek/Makefile | 10 +- sound/soc/mediatek/common/Makefile | 16 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 379 +++++ sound/soc/mediatek/common/mtk-afe-fe-dai.h | 45 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 90 ++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 23 + sound/soc/mediatek/common/mtk-base-afe.h | 104 ++ sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 464 ++++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 38 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 181 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1698 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 422 +++++ sound/soc/mediatek/mt2701/mt2701-reg.h | 186 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 73 + .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 921 +++++------ sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 27 files changed, 4390 insertions(+), 634 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/common/Makefile create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.c create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.h create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.c create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.h create mode 100644 sound/soc/mediatek/common/mtk-base-afe.h create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (51%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

3 19

[alsa-devel] [asoc:for-next 9/16] cygnus-pcm.c:undefined reference to `bad_dma_ops'
by kbuild test robot 03 Jul '16

03 Jul '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 5664c75bd47efe7c701d1d46ee90c795ff8bc150 commit: ebe0ee8d716bcb3cb890e4feb6c6e07e0de11208 [9/16] Merge remote-tracking branches 'asoc/topic/cygnus', 'asoc/topic/da7219', 'asoc/topic/davinci' and 'asoc/topic/dpcm' into asoc-next config: m32r-allyesconfig (attached as .config) compiler: m32r-linux-gcc (GCC) 4.9.0 reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout ebe0ee8d716bcb3cb890e4feb6c6e07e0de11208 # save the attached .config to linux build tree make.cross ARCH=m32r All errors (new ones prefixed by >>): drivers/built-in.o: In function `zynq_fpga_ops_write': zynq-fpga.c:(.text+0xeff96c): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeff974): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeffa94): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcb4): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcbc): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): undefined reference to `dma_common_mmap' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): relocation truncated to fit: R_M32R_26_PCREL_RELA against undefined symbol `dma_common_mmap' sound/built-in.o: In function `cygnus_pcm_preallocate_dma_buffer': >> cygnus-pcm.c:(.text+0x10efa0): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10efa4): undefined reference to `bad_dma_ops' sound/built-in.o: In function `cygnus_dma_free_dma_buffers': cygnus-pcm.c:(.text+0x10f15c): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10f160): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10f16c): undefined reference to `bad_dma_ops' sound/built-in.o:cygnus-pcm.c:(.text+0x10f170): more undefined references to `bad_dma_ops' follow --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [GIT PULL] ASoC fixes for v4.7
by Mark Brown 03 Jul '16

03 Jul '16

The following changes since commit 4c2e07c6a29e0129e975727b9f57eede813eea85: Linux 4.7-rc5 (2016-06-26 17:52:03 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.7-rc5 for you to fetch changes up to 7c5eec4edf4693aa61f3ee78bf28621dfe33d9f0: Merge remote-tracking branches 'asoc/fix/rcar', 'asoc/fix/rt5670' and 'asoc/fix/wm8940' into asoc-linus (2016-07-01 18:05:34 +0200) ---------------------------------------------------------------- ASoC: Fixes for v4.7 A small clutch of hardware specific fixes for various ASoC devices, all small individually and important if you have that device but not otherwise. ---------------------------------------------------------------- Ajit Pandey (1): ASoC: wm5110: Add missing route from OUT3R to SYSCLK Alan Cox (1): ASoC: Intel: atom: fix missing breaks that would cause the wrong operation to execute Alexander Shiyan (1): ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode Arnd Bergmann (1): ASoC: hdmi-codec: select CONFIG_HDMI Bard Liao (2): ASoC: rt5670: fix HP Playback Volume control ASoC: rt5645: fix reg-2f default value. Charles Keepax (1): ASoC: wm5102: Correct supported channels on trace compressed DAI Fabio Estevam (1): ASoC: ak4613: Remove owner assignment from platform_driver Geert Uytterhoeven (2): ASoC: wm8940: Enable cache usage to fix crashes on resume ASoC: ak4613: Enable cache usage to fix crashes on resume Janusz Krzysztofik (1): ASoC: cx20442: set tty->receiver_room in v253_open Kuninori Morimoto (1): ASoC: rsnd: open 31bit of SSICKR mask Mark Brown (4): Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/ak4613', 'asoc/fix/arizona', 'asoc/fix/cx20442', 'asoc/fix/davinci', 'asoc/fix/fsl-ssi' and 'asoc/fix/hdmi' into asoc-linus Merge remote-tracking branches 'asoc/fix/rcar', 'asoc/fix/rt5670' and 'asoc/fix/wm8940' into asoc-linus Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix dra7 DMA offset when using CFG port Vinod Koul (2): ASoC: hdac_hdmi: Fix potential NULL dereference ASoC: Intel: Skylake: Initialize module list for Broxton sound/soc/codecs/Kconfig | 7 +-- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/cx20442.c | 1 + sound/soc/codecs/hdac_hdmi.c | 20 ++++++++ sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 1 + sound/soc/codecs/wm8940.c | 1 + sound/soc/davinci/davinci-mcasp.c | 56 +++++++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 4 +- sound/soc/fsl/fsl_ssi.c | 12 ++--- sound/soc/intel/atom/sst-mfld-platform-compress.c | 9 +++- sound/soc/intel/skylake/bxt-sst.c | 1 + sound/soc/sh/rcar/adg.c | 2 +- 15 files changed, 100 insertions(+), 22 deletions(-)

2 1

[alsa-devel] [PATCH] drm: mediatek: fix prototypes after API change
by Arnd Bergmann 02 Jul '16

02 Jul '16

The newly added mediatek HDMI driver clashes with an API change for struct hdmi_codec_ops, causing an 'allmodconfig' build to fail: drivers/gpu/drm/mediatek/mtk_hdmi.c:1653:15: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] drivers/gpu/drm/mediatek/mtk_hdmi.c:1654:19: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] drivers/gpu/drm/mediatek/mtk_hdmi.c:1655:20: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] drivers/gpu/drm/mediatek/mtk_hdmi.c:1656:18: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] drivers/gpu/drm/mediatek/mtk_hdmi.c:1657:13: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] This changes the API accordingly for the mediatek driver. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 8f83f26891e1 ("drm/mediatek: Add HDMI support") Fixes: efc9194bcff8 ("ASoC: hdmi-codec: callback function will be called with private data") --- The new driver is currently part of Dave Airlie's drm-next tree, while the API change is in Mark Brown's asoc-next tree, so my fix has to be done on top of both. I can see two ways out of this: a) Stephen Rothwell could add my fixup on top of linux-next as a merge fix, and the fix gets sent to Linus Torvalds along with the second of the two branches b) efc9194bcff8 gets merged into drm-next along with my change in the merge commit. It is currently a standalone patch in its own branch in the ASoC tree. --- drivers/gpu/drm/mediatek/mtk_hdmi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c index ba812ef2c9d1..334562d06731 100644 --- a/drivers/gpu/drm/mediatek/mtk_hdmi.c +++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c @@ -1535,7 +1535,7 @@ static int mtk_hdmi_dt_parse_pdata(struct mtk_hdmi *hdmi, * HDMI audio codec callbacks */ -static int mtk_hdmi_audio_hw_params(struct device *dev, +static int mtk_hdmi_audio_hw_params(struct device *dev, void *data, struct hdmi_codec_daifmt *daifmt, struct hdmi_codec_params *params) { @@ -1604,7 +1604,7 @@ static int mtk_hdmi_audio_hw_params(struct device *dev, return 0; } -static int mtk_hdmi_audio_startup(struct device *dev) +static int mtk_hdmi_audio_startup(struct device *dev, void *data) { struct mtk_hdmi *hdmi = dev_get_drvdata(dev); @@ -1615,7 +1615,7 @@ static int mtk_hdmi_audio_startup(struct device *dev) return 0; } -static void mtk_hdmi_audio_shutdown(struct device *dev) +static void mtk_hdmi_audio_shutdown(struct device *dev, void *data) { struct mtk_hdmi *hdmi = dev_get_drvdata(dev); @@ -1624,7 +1624,7 @@ static void mtk_hdmi_audio_shutdown(struct device *dev) mtk_hdmi_audio_disable(hdmi); } -int mtk_hdmi_audio_digital_mute(struct device *dev, bool enable) +int mtk_hdmi_audio_digital_mute(struct device *dev, void *data, bool enable) { struct mtk_hdmi *hdmi = dev_get_drvdata(dev); @@ -1638,7 +1638,7 @@ int mtk_hdmi_audio_digital_mute(struct device *dev, bool enable) return 0; } -static int mtk_hdmi_audio_get_eld(struct device *dev, uint8_t *buf, size_t len) +static int mtk_hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf, size_t len) { struct mtk_hdmi *hdmi = dev_get_drvdata(dev); -- 2.9.0

4 3

[alsa-devel] [PATCH] ASoC: cs4270: fix DAPM stream name mismatch
by murray foster 02 Jul '16

02 Jul '16

Mismatching stream names in DAPM route and widget definitions are causing compilation errors. Fixing these names allows the cs4270 driver to compile and function. Signed-off-by: Murray Foster <mrafoster(a)gmail.com> --- sound/soc/codecs/cs4270.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e07807d..4ffdca4 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -1,4 +1,4 @@ -/* +s/* * CS4270 ALSA SoC (ASoC) codec driver * * Author: Timur Tabi <timur(a)freescale.com> @@ -148,11 +148,11 @@ SND_SOC_DAPM_OUTPUT("AOUTR"), }; static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { - { "Capture", NULL, "AINA" }, - { "Capture", NULL, "AINB" }, + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, - { "AOUTA", NULL, "Playback" }, - { "AOUTB", NULL, "Playback" }, + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, }; /** -- 2.4.9

1 0

[alsa-devel] [GIT PULL] Broxton fw update
by Vinod Koul 01 Jul '16

01 Jul '16

Please PULL to receive the latest Broxton fw. The following changes since commit 3ef7857d511ce6a91c5ce609da76c4702651cfa5: amdgpu: update polaris ucode (2016-06-28 14:31:11 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git apl for you to fetch changes up to 040c24514b94fcdbd6308d986d1bde59a704a21a: linux-firmware: intel: Update Broxton audio firmware (2016-06-29 14:20:20 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Broxton audio firmware WHENCE | 4 ++-- intel/dsp_fw_bxtn.bin | 2 +- intel/dsp_fw_bxtn_v702.bin | Bin 0 -> 505608 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_bxtn_v702.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH 0/3 v2] ALSA: ctl: add dimension information validator
by Takashi Sakamoto 01 Jul '16

01 Jul '16

Hi, This patchset is revised version of my previous one. [alsa-devel] [PATCH 0/3] ALSA: add dimension information validator http://mailman.alsa-project.org/pipermail/alsa-devel/2016-June/109811.html The main purpose of this patchset is the same, just including some changes below. * Add arithmetic overflow check * Add negative value check of first dimension level * Change function name of the validator and name of local variable so that the aim is clear. Takashi Sakamoto (3): ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members ALSA: control: add dimension validator for userspace element ALSA: control: add dimension validator for kernel driver sound/core/control.c | 87 +++++++++++++++++++++++++++++++++-------- sound/pci/echoaudio/echoaudio.c | 6 +-- 2 files changed, 74 insertions(+), 19 deletions(-) -- 2.7.4

2 10

[alsa-devel] Assertion failure in alsa-lib crashing applications (snd_pcm_area_copy: Assertion `dst < src || dst >= src + bytes' failed)
by Baptiste Jonglez 01 Jul '16

01 Jul '16

Hi, When using Ring [1], an audio-video application, it sometimes crashes because of an assertion in libasound.so: pcm.c:2758: snd_pcm_area_copy: Assertion `dst < src || dst >= src + bytes' failed. The complete upstream bug report is at [2]. The (not-so-useful) stack trace is the following: #0 0x00007fc356e462a8 raise (libc.so.6) #1 0x00007fc356e4772a abort (libc.so.6) #2 0x00007fc356e3f1b7 __assert_fail_base (libc.so.6) #3 0x00007fc356e3f262 __assert_fail (libc.so.6) #4 0x00007fc35ba65c84 snd_pcm_area_copy (libasound.so.2) #5 0x00007fc35ba65ebc snd_pcm_areas_copy (libasound.so.2) #6 0x00007fc35baa6d95 n/a (libasound.so.2) #7 0x00007fc35ba77c1f n/a (libasound.so.2) #8 0x00007fc35ba77a59 n/a (libasound.so.2) #9 0x000000000051ba97 n/a (dring) #10 0x000000000051c118 n/a (dring) #11 0x000000000051cfad n/a (dring) #12 0x00007fc357787c40 execute_native_thread_routine (libstdc++.so.6) #13 0x00007fc359ec2424 start_thread (libpthread.so.0) #14 0x00007fc356efacbd __clone (libc.so.6) The next time it crashes, I will have debug symbols in dring. I can also recompile alsa-lib with debug symbols, what is the best way to do that? Looking around, it seems other projects have run into the same issue: https://aur.archlinux.org/packages/ultrastardx-git/?comments=all#comment-43… https://aur.archlinux.org/packages/zoom/#comment-544696 http://ubuntuforums.org/showthread.php?t=2248373 https://github.com/js-platform/node-webrtc/issues/110 https://fedorahosted.org/fldigi/ticket/70 The output of the alsa-info.sh script on my system is at [3]. What else can I provide to debug this issue further? Thanks, Baptiste [1] https://ring.cx [2] https://tuleap.ring.cx/plugins/tracker/?aid=502 [3] http://www.alsa-project.org/db/?f=0dd2ba1021b3d535f30f07c55dc18e2ef60db26d

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: patch reg-0x8a
by Bard Liao 01 Jul '16

01 Jul '16

reg-8a assign the tracking source for each ASRC tracker. The default value is 0x0000 which means all ASRC trackers will track LRCK1. But in most cases, we wish each ASRC tracker track the corresponding LRCK. i.e. ASRC1 tracks LRCK1, ASRC2 tracks LRCK2 and so on. So, we rewrite reg-8a as 0x0120. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1f457d3..c21e41c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -64,6 +64,7 @@ static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, {RT5645_PR_BASE + 0x00, 0x0a2a}, + {RT5645_ASRC_4, 0x0120}, }; static const struct reg_sequence rt5650_init_list[] = { @@ -159,7 +160,7 @@ static const struct reg_default rt5645_reg[] = { { 0x83, 0x0000 }, { 0x84, 0x0000 }, { 0x85, 0x0000 }, - { 0x8a, 0x0000 }, + { 0x8a, 0x0120 }, { 0x8e, 0x0004 }, { 0x8f, 0x1100 }, { 0x90, 0x0646 }, @@ -316,7 +317,7 @@ static const struct reg_default rt5650_reg[] = { { 0x83, 0x0000 }, { 0x84, 0x0000 }, { 0x85, 0x0000 }, - { 0x8a, 0x0000 }, + { 0x8a, 0x0120 }, { 0x8e, 0x0004 }, { 0x8f, 0x1100 }, { 0x90, 0x0646 }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2 1/2] ASoC: rt5645: fix reg-2f default value.
by Bard Liao 01 Jul '16

01 Jul '16

The default value of reg-2f in codec rt5650 is 0x5002, not 0x1002. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3c6594d..d70847c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -253,7 +253,7 @@ static const struct reg_default rt5650_reg[] = { { 0x2b, 0x5454 }, { 0x2c, 0xaaa0 }, { 0x2d, 0x0000 }, - { 0x2f, 0x1002 }, + { 0x2f, 0x5002 }, { 0x31, 0x5000 }, { 0x32, 0x0000 }, { 0x33, 0x0000 }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [asoc:topic/sunxi 2/2] sound/soc/sunxi/sun4i-i2s.c:624:2-3: Unneeded semicolon
by kbuild test robot 01 Jul '16

01 Jul '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/sunxi head: fa7c0d13cb26216f6dec5ef19e028e68b300530d commit: fa7c0d13cb26216f6dec5ef19e028e68b300530d [2/2] ASoC: sunxi: Add Allwinner A10 Digital Audio driver coccinelle warnings: (new ones prefixed by >>) >> sound/soc/sunxi/sun4i-i2s.c:624:2-3: Unneeded semicolon Please review and possibly fold the followup patch. --- 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 1/2] ASoC: cs35l33: Fix testing return value of devm_gpiod_get_optional
by Axel Lin 01 Jul '16

01 Jul '16

devm_gpiod_get_optional() returns NULL when the gpio is not assigned. So the if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) test is always false. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l33.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 622a111..aa5c0c5 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -1176,11 +1176,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, /* We could issue !RST or skip it based on AMP topology */ cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, "reset-gpios", GPIOD_OUT_HIGH); - - if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { - dev_warn(&i2c_client->dev, - "%s WARNING: No reset gpio assigned\n", __func__); - } else if (IS_ERR(cs35l33->reset_gpio)) { + if (IS_ERR(cs35l33->reset_gpio)) { dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", __func__); return PTR_ERR(cs35l33->reset_gpio); -- 2.5.0

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[alsa-devel] [PATCH 0/3] ASoC: Reduce audio related kernel spew.
by vedang.patel＠intel.com 01 Jul '16

01 Jul '16

From: Vedang Patel <vedang.patel(a)intel.com> There have been complaints regarding audio related kernel messages flooding dmesg. This series tries to address those complaints. Also, I have created 3 different patches for this because they touch different drivers. But, I can combine those to reduce the number of patches. Vedang Patel (3): ASoC: hdac_hdmi: Increase loglevel of hex dump printed ASoC: Intel: common: increase the loglevel of debug message. ASoC: Intel: Skylake: Increase loglevel of debug message. sound/soc/codecs/hdac_hdmi.c | 2 ++ sound/soc/intel/common/sst-dsp.c | 2 +- sound/soc/intel/skylake/skl-messages.c | 2 ++ sound/soc/intel/skylake/skl-sst-ipc.c | 7 +++++-- 4 files changed, 10 insertions(+), 3 deletions(-) -- 2.1.2

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Re: [alsa-devel] [PATCH 1/1] sound/usb: Add Asus Xonar U7 Mixer - input support
by Piotr G. 01 Jul '16

01 Jul '16

Hello Guys, I made some sniffing of USB protocol with Windows10 and Windows8 drivers for Asus Xonar U7. ALSA has no Capture Volume controls for Line Input and Mic Input. Mic doesn't work. According to Windows drivers when using Mic or Line in sliders: wValue: 0x8202 [L] -> 0x82 It's not declared in UAC2 0x8201 [R] -> 0x82 It's not declared in UAC2 AudioControl Interface Descriptor: bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 4 (Mic Terminal) bmaControls( 0) 0x00000003 Mute Control (read/write) bmaControls( 1) 0x00000000 (No Volume Control?) bmaControls( 2) 0x00000000 (No Volume Control?) iFeature 0 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 17 bSourceID 5 (Line in Terminal) bmaControls( 0) 0x00000003 Mute Control (read/write) bmaControls( 1) 0x00000000 (No Volume Control?) bmaControls( 2) 0x00000000 (No Volume Control?) iFeature 0 It's looks like firmware doesn't declare Volume or Input Gain in FU with 0x82 CS. How can I properly add it do the source? (redefine mixer_maps?) wIndex: 0x1100 (Line In) -> 0x1100 ALSA FU16 OK 0x1000 (Mic) -> 0x1000 ALSA FU17 OK In Alsamixer in Capture section we have only Line and Mic Controls where we can toggle On or Off. This is correct when parsing upper FU only Mute Control is declared Where or how I can manual declare that this controls have Volume Control with CS 0x82 (Control Selector)? WHat is the correct way? About Ranges: Line in Data: 0x00a6 (160) - 0x00ff (255) (Mixer from 0 (-96dB) to 99 (-0.15dB) log) 0x0000 (0) (Mixer 100 (0dB) Mic Data: >From 0x00a6 (160) - 0x0000 (0) (Mixer from 0 (-96dB) to 50 (0dB) log) >From 0x000 (0) - 0x0032 (50) (Mixer from 50 (0dB) to 100 (50dB) lin) Thanks for support. Piotr W dniu 10.09.2015 o 16:47, Piotr SQ9FK pisze: > Hello Takashi,Johan > My patches are unessesary since following last patch was applied to the > git: > ALSA: usb-audio: Change internal PCM order > > Capture working but there are no volume sliders for Line-In Capture or > Mic-In Capture (I can only toggle on off). Probably the RANGES are > broken?. From ASUS Windows driver I can see following ranges: > > Master Volume: (Phisycal Knob on Asus) > from 100 to 1 correspond 0dB to -70dB, 0 is -127dB > Range from 1 to 100 -> 0dB to -70dB (logarythmic) > > Mic-In: > from 100 to 1 correspond 30dB to -10dB, 0 is -192dB > Range from 1 to 10 -> -10dB to 10dB (linear?) > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Line-In: > from 100 to 1 correspond 0dB to -40dB, 0 is -192dB > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Actual Situation: > Input Gain Pad Control - 3 steps [0 - 50 - 100] > Double Slider. In Capture and Playback Categories, both of them are > controlled in the same time. > Controls Playback Volume only, Capture not. > > "Mic" and "Line" Switches [on/off] > PCM Cature Source [Line/Mic] > Electret Mic doesn't work. With all possible configurations - Silent. > When I pluged Line level signal, I can record the signal without problems. > PCM Capture Source [Line] - Line [on] Mic [off] - Sound > PCM Capture Source [Line] - Line [on] Mic [on] - Sound > PCM Capture Source [Line] - Line [off] Mic [off] - No Sound > PCM Capture Source [Line] - Line [off] Mic [on] - No Sound > The same with selected [Mic]. There is no Capture Volume control. > > PCM 1 [mute/on] > I think this is SPDIF switch. It doesn't work. When I switch it on -> > Blue Led on Xonar doesn't light up. > > I attached screen of alsamixer, usbmix from proc and my mixer_map from > my old patch. > I think there are is still bugs in parsing. Please point me how I can > add manually Volume control for Capture? > > Piotr Gaska > > W dniu 2015-07-27 o 14:10, Takashi Iwai pisze: >> On Sun, 26 Jul 2015 10:00:12 +0200, >> Piotr G. wrote: >>> Hello, >>> This is my first patch to alsa-devel, and I'm not very skilled >>> programmer. From lsusb -vvv, I've draw Xonar U7 map with all IT OT FU CS >>> in mixer_map.c I don't know why the driver by default doesn't parse all >>> info correctly. With this patch you can see Line Switch and Mic Switch >>> in alsamixer. I can record my voice in Audacity with all supported bit >>> and sampling rates in Mono or Stereo >>> This is temporary solution. Please correct me if the patch is not >>> properly made. >>> >>> What doesn't work: >>> - Cannot change Capture Volume for Line/Mic. No mixer control in >>> alsamixer >>> - Switching between Line/Mic doesn't work (Mute doesn't work) >>> - SPDIF not tested >>> As what I wrote at beginning I'm not skilled programmer so any >>> help is >>> aprreciated >>> >>> >>> proc-usbmixer -> cat /proc/asound/*/usbmix >>> usb.debug -> lsusb -vv (only Asus Xonar U7) >> The changes in mixer_maps.c look good, but what's the reason for >> uac2_audio_feature_info[]? It seems adding "Control" suffix to each >> entry, which doesn't look correct. >> >> >> thanks, >> >> Takashi >> >> >>> --- >>> sound/usb/mixer.c | 23 +++++++++++++++++++++-- >>> sound/usb/mixer_maps.c | 48 >>> ++++++++++++++++++++++++++++++++++++++++++++++++ >>> 2 files changed, 69 insertions(+), 2 deletions(-) >>> >>> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c >>> index 6b3acba..027d160 100644 >>> --- a/sound/usb/mixer.c >>> +++ b/sound/usb/mixer.c >>> @@ -818,6 +818,25 @@ static struct usb_feature_control_info >>> audio_feature_info[] = { >>> { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> }; >>> >>> +static struct usb_feature_control_info uac2_audio_feature_info[] = { >>> + /* UAC2 specific */ >>> + { "Mute Control", USB_MIXER_INV_BOOLEAN }, >>> + { "Volume Control", USB_MIXER_S16 }, >>> + { "Bass Control", USB_MIXER_S8 }, >>> + { "Mid Control", USB_MIXER_S8 }, >>> + { "Treble Control", USB_MIXER_S8 }, >>> + { "Graphic Equalizer Control", USB_MIXER_S8 }, >>> + { "Automatic Gain Control", USB_MIXER_BOOLEAN }, >>> + { "Delay Control", USB_MIXER_S16 }, /* FIXME: U32 in UAC2 */ >>> + { "Bass Boost Control", USB_MIXER_BOOLEAN }, >>> + { "Loudness Control", USB_MIXER_BOOLEAN }, >>> + { "Input Gain Control", USB_MIXER_S16 }, >>> + { "Input Gain Pad Control", USB_MIXER_S16 }, >>> + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> + { "Underflow Control", USB_MIXER_BOOLEAN }, >>> + { "Overflow Control", USB_MIXER_BOOLEAN }, >>> +}; >>> + >>> /* private_free callback */ >>> void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) >>> { >>> @@ -1240,7 +1259,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); >>> cval->control = control; >>> cval->cmask = ctl_mask; >>> - cval->val_type = audio_feature_info[control-1].type; >>> + cval->val_type = uac2_audio_feature_info[control-1].type; >>> if (ctl_mask == 0) { >>> cval->channels = 1; /* master channel */ >>> cval->master_readonly = readonly_mask; >>> @@ -1318,7 +1337,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> break; >>> default: >>> if (!len) >>> - strlcpy(kctl->id.name, audio_feature_info[control-1].name, >>> + strlcpy(kctl->id.name, >>> uac2_audio_feature_info[control-1].name, >>> sizeof(kctl->id.name)); >>> break; >>> } >>> diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c >>> index e5000da..07f510e 100644 >>> --- a/sound/usb/mixer_maps.c >>> +++ b/sound/usb/mixer_maps.c >>> @@ -341,6 +341,48 @@ static const struct usbmix_name_map >>> scms_usb3318_map[] = { >>> { 0 } >>> }; >>> >>> +/* ASUS Xonar U7 - USB CM6632A (ADC CS5361 // DAC Dolby CS4362A // DAC >>> Stereo CS4398 */ >>> +/* Topology: >>> + >>> +USB_IN[1]------->FU[13]-----------CS[18]--->Spk_OUT[7] >>> + >>> +Dig_IN[2]------->FU[14]-----------CS[19]--->Dig_OUT[8] >>> + >>> +Mic_IN[4]------->FU[16]---+ >>> + SU[20]--CS[22]--->USB_OUT[11] >>> +Line_IN[5]------>FU[17]---+ >>> + >>> +*/ >>> + >>> +static const struct usbmix_name_map xonar_u7_map[] = { >>> + /* 1: IT PCM Analog (USB Streaming) */ >>> + /* 2: IT PCM2 Digital (USB Streaming) */ >>> + /* 4: IT Microphone */ >>> + /* 5: IT Line IN */ >>> + /* 7: OT Speaker */ >>> + /* 8: OT SPDIF */ >>> + /* 11 OT Capture Playback (USB Streaming) */ >>> + { 13, "Speaker Playback" }, /* 13 FU Speaker Out Mute/Volume */ >>> + { 14, "IEC958 Playback" }, /* 14 FU SPDIF Out Mute/Volume */ >>> + { 16, "Mic Capture" }, /* 16 FU Capture Mic-in >>> Mute/Volume */ >>> + { 17, "Line Capture" }, /* 17 FU Capture Line-in Mute/Volume */ >>> + /* 18 CS for 1 PCM (USB Streaming) Clock Source */ >>> + /* 19 CS for 2 SPDIF (USB Streaming) Clock Source */ >>> + { 20, "Capture Source" }, /* 20 SU Capture Source Selector */ >>> + /* 22 CS Line-in and Mic-in Clock Source */ >>> + { 0 } >>> +}; >>> + >>> +static struct usbmix_selector_map xonar_u7_selectors[] = { >>> +{ >>> + .id = 20, >>> + .count = 2, >>> + .names = (const char*[]) {"Mic", "Line"} >>> + }, >>> + { 0 } /* terminator */ >>> + >>> +}; >>> + >>> /* >>> * Control map entries >>> */ >>> @@ -451,6 +493,12 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { >>> .id = USB_ID(0x25c4, 0x0003), >>> .map = scms_usb3318_map, >>> }, >>> + { >>> + /* Asus Xonar U7 */ >>> + .id = USB_ID(0x1043, 0x857c), >>> + .map = xonar_u7_map, >>> + .selector_map = xonar_u7_selectors, >>> + }, >>> { 0 } /* terminator */ >>> }; >>> >>> -- >>> 2.3.6 >>> >

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[alsa-devel] [PATCH 0/3] ALSA: add dimension information validator
by Takashi Sakamoto 30 Jun '16

30 Jun '16

Hi, This patchset is to add a validator of dimension information for drivers in kernel/userspace. The dimension information was added to ALSA control interface. With the information, members in an element compose multi-dimensional matrix. For example, this matrix can be used for row/column table with values. In this case, dimension level 1 and 2 are utilized to describe the number of rows and columns by element member unit. When dimension information has no contradictions to the number of members in an element, there's no problem. Once it has, it can cause an issue. In worst case, it can cause buffer-overrun in userspace. The aim of this patchset is to prevent this situation. Takashi Sakamoto (3): ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members ALSA: control: add dimension validator for userspace element ALSA: control: add dimension validator for kernel driver sound/core/control.c | 77 ++++++++++++++++++++++++++++++++--------- sound/pci/echoaudio/echoaudio.c | 6 ++-- 2 files changed, 64 insertions(+), 19 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 0/4] ASoC: sunxi: Add i2s controller support
by Maxime Ripard 30 Jun '16

30 Jun '16

Hi everyone, This is the second version of the I2S support for the controller found in the Allwinner A10 and later SoCs. Playback has been tested with an UDA1380 on an A20-Olinuxino. Capture is not implemented yet, but will come eventually. Let me know what you think, Maxime Changes from v1: - Fixed bogus indentation and spurious whitespaces / newlines - Moved the device initialisation in the probe to the startup function - Disabled the device in shutdown - Removed function to get wss and sr to inline them in hw_params - Changed the way to access the divider array - Checked clk_set_rate return code - Fixed condition for invalid bclk and mclk dividers - Added sound-dai-cells to the bindings documentation required properties Emilio López (1): ARM: sun7i: Add mod1 clock nodes Maxime Ripard (3): ASoC: sunxi: Add A10 I2S controller binding documentation ASoC: sunxi: Add Allwinner A10 Digital Audio driver ARM: sun7i: Add DAI nodes .../devicetree/bindings/sound/sun4i-i2s.txt | 34 + arch/arm/boot/dts/sun7i-a20.dtsi | 87 ++- sound/soc/sunxi/Kconfig | 9 + sound/soc/sunxi/Makefile | 2 +- sound/soc/sunxi/sun4i-i2s.c | 703 +++++++++++++++++++++ 5 files changed, 832 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun4i-i2s.txt create mode 100644 sound/soc/sunxi/sun4i-i2s.c -- 2.9.0

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[alsa-devel] [alsa-lib][PATCH 0/5] ctl: support extra information for user-defined element set
by Takashi Sakamoto 30 Jun '16

30 Jun '16

Hi, This patchset adds support extra information to the type-specific for user-defined element set. Currently, 'dimension' is such an extra information. In ALSA kernel/userspace interface, information to an element is described in this structure. struct snd_ctl_elem_info { struct snd_ctl_elem_id id; unsigned int access; unsigned int count; pid_t owner; union value; union dimen; unsigned char reserved[64 - 4 * sizeof(unsigned short)]; }; This structure includes reserved fields, thus it's possible to add more fields for future extension. Currently, APIs to add user-defined element set don't support such extra fields. Meanwhile, supporting just the dimension field is not good as stable library APIs. This patchset changes prototype of the APIs with 'snd_ctl_elem_info_t' instead of adding more parameters. Callers are expected to fill the parameter with identification information and the extra information. Takashi Sakamoto (5): ctl: support extra information in user-defined element set ctl: add an API to set dimension levels to element information ctl: optimize a test for user-defined element set to older kernels ctl: optimize a test for user-defined element set to changes of APIs ctl: support dimension test for user-defined element set include/control.h | 12 ++- src/control/control.c | 219 +++++++++++++++++++++++--------------------- src/pcm/pcm_softvol.c | 10 +- test/user-ctl-element-set.c | 122 ++++++++++++++++++------ 4 files changed, 220 insertions(+), 143 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: riptide: Use DIV_ROUND_UP
by Amitoj Kaur Chawla 30 Jun '16

30 Jun '16

The kernel.h macro DIV_ROUND_UP performs the computation (((n) + (d) - 1) /(d)) but is perhaps more readable. The Coccinelle script used to make this change is as follows: @haskernel@ @@ #include <linux/kernel.h> @depends on haskernel@ expression n,d; @@ ( - (n + d - 1) / d + DIV_ROUND_UP(n,d) | - (n + (d - 1)) / d + DIV_ROUND_UP(n,d) ) Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/pci/riptide/riptide.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 94639d6..067a912 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1496,7 +1496,7 @@ static int snd_riptide_prepare(struct snd_pcm_substream *substream) f = PAGE_SIZE; while ((size + (f >> 1) - 1) <= (f << 7) && (f << 1) > period) f = f >> 1; - pages = (size + f - 1) / f; + pages = DIV_ROUND_UP(size, f); data->size = size; data->pages = pages; snd_printdd -- 1.9.1

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Re: [alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode
by Nicolin Chen 29 Jun '16

29 Jun '16

On Wed, Jun 29, 2016 at 09:50:24PM +0300, Alexander Shiyan wrote: > >Среда, 29 июня 2016, 21:26 +03:00 от Nicolin Chen <nicoleotsuka(a)gmail.com>: > > > >On Wed, Jun 29, 2016 at 10:52:01AM +0300, Alexander Shiyan wrote: > > > >> >your failed case. Could you please tell me what were the FSclk, > >> >BITclk and size of your case? > > > >> Here is my oscillograms with/without patch: > >> Bclk=12 MHz, sound is just a "cat /dev/urandom > /dev/dsp", i.e. FS=8 KHz, MONO. > >> Upper (yellow) line is TXDATA, Lower (blue) is FS. > >> > >> Measurement without patch: https://cloud.mail.ru/public/Rcsv/xr88V5Sa1 > >> Measurement with patch: https://cloud.mail.ru/public/NCwn/DQnzioQEL > > > >Thanks for the reply. So in the case without patch, it looks like > >SSI just sent the data ignoring the slot size right? > > I could be wrong, but think that WL (word length) still point to number of valid > data bits within 32bit chunk (as defined for I2S-Master mode), > but since DC=0, this cause to send next word to the channel again and again. I see. Thanks for the fix.

1 0

[alsa-devel] [PATCH v3 0/6] ASoC: samsung: Sound support for Exynos5433 TM2(E) boards
by Sylwester Nawrocki 29 Jun '16

29 Jun '16

This patch series adds sound support for the Exynos 5433 based TM2 and TM2E board. It includes minimal drivers for the Low Power Audio Subsystem (LPASS) and the MAX98504 class D speaker amplifier as well as the actual asoc machine driver. The changes since v2 are listed in individual patches, I didn't add any Acked-by tag to the max98504 DT binding patch as there was significant change since v2. Inha Song (1): ASoC: samsung: Add Samsung Low Power Audio Subsystem driver Sylwester Nawrocki (5): ASoC: samsung: Add DT bindings documentation for LPASS ASoC: Add DT bindings documentation for max98504 amplifier ASoC: max98504: Add max98504 speaker amplifier driver ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem ASoC: samsung: Add machine driver for Exynos5433 based TM2 board .../devicetree/bindings/sound/max98504.txt | 44 ++ .../bindings/sound/samsung,exynos5433-lpass.txt | 8 + .../bindings/sound/samsung,tm2-audio.txt | 36 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98504.c | 383 ++++++++++++++ sound/soc/codecs/max98504.h | 59 +++ sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 4 + sound/soc/samsung/lpass.c | 162 ++++++ sound/soc/samsung/lpass.h | 47 ++ sound/soc/samsung/tm2_wm5110.c | 579 +++++++++++++++++++++ 12 files changed, 1341 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,exynos5433-lpass.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 sound/soc/codecs/max98504.c create mode 100644 sound/soc/codecs/max98504.h create mode 100644 sound/soc/samsung/lpass.c create mode 100644 sound/soc/samsung/lpass.h create mode 100644 sound/soc/samsung/tm2_wm5110.c -- 1.9.1

2 8

[alsa-devel] [PATCH v4 0/9] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 29 Jun '16

29 Jun '16

Sorry for submitting new version in a short time. I find some comment style error and fix it. please ignore v3 and directly review v4. This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since v3: * fix comments style. Change since v2: * fix comments style. * replace EXPORT_SYMBOL with EXPORT_SYMBOL_GPL * reduce debug message * move mutex lock for irq-allocated to AFE struct * remove func mtk_simple_isr and call snd_pcm_period_elapsed directly * mt2701 driver probe will fail if clock initial error * fix dapm on/off switch control naming Change since v1: implement common private structure, fe dai and platform driver. let mt8173 and mt2701 use the common structure. Garlic Tseng (9): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: implement mediatek common structure ASoC: mediatek: let mt8173 use mediatek common structure ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add structure define and clock control for 2701 ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 44 +- sound/soc/mediatek/Makefile | 11 +- sound/soc/mediatek/common/Makefile | 17 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 379 +++++ sound/soc/mediatek/common/mtk-afe-fe-dai.h | 45 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 90 ++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 23 + sound/soc/mediatek/common/mtk-base-afe.h | 104 ++ sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 464 ++++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 38 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 181 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1698 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 422 +++++ sound/soc/mediatek/mt2701/mt2701-reg.h | 186 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 73 + .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 921 +++++------ sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 27 files changed, 4399 insertions(+), 634 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/common/Makefile create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.c create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.h create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.c create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.h create mode 100644 sound/soc/mediatek/common/mtk-base-afe.h create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (51%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

3 14

[alsa-devel] [PATCH v2 0/9] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 29 Jun '16

29 Jun '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since v1: implement common private structure, fe dai and platform driver. let mt8173 and mt2701 use the common structure. Garlic Tseng (9): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: implement mediatek common structure ASoC: mediatek: let mt8173 use mediatek common structure ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add structure define and clock control for 2701 ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 44 +- sound/soc/mediatek/Makefile | 11 +- sound/soc/mediatek/common/Makefile | 17 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 387 +++++ sound/soc/mediatek/common/mtk-afe-fe-dai.h | 47 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 90 ++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 23 + sound/soc/mediatek/common/mtk-base-afe.h | 103 ++ sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 357 +++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 33 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 185 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1664 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 422 +++++ sound/soc/mediatek/mt2701/mt2701-reg.h | 186 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 73 + .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 923 +++++------ sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 27 files changed, 4267 insertions(+), 635 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/common/Makefile create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.c create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.h create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.c create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.h create mode 100644 sound/soc/mediatek/common/mtk-base-afe.h create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (51%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

2 18

[alsa-devel] [PATCH v3 0/9] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 29 Jun '16

29 Jun '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since v2: * fix comments style. * replace EXPORT_SYMBOL with EXPORT_SYMBOL_GPL * reduce debug message * move mutex lock for irq-allocated to AFE struct * remove func mtk_simple_isr and call snd_pcm_period_elapsed directly * mt2701 driver probe will fail if clock initial error * fix dapm on/off switch control naming Change since v1: implement common private structure, fe dai and platform driver. let mt8173 and mt2701 use the common structure. Garlic Tseng (9): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: implement mediatek common structure ASoC: mediatek: let mt8173 use mediatek common structure ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add structure define and clock control for 2701 ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 44 +- sound/soc/mediatek/Makefile | 11 +- sound/soc/mediatek/common/Makefile | 17 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 379 +++++ sound/soc/mediatek/common/mtk-afe-fe-dai.h | 45 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 90 ++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 23 + sound/soc/mediatek/common/mtk-base-afe.h | 104 ++ sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 464 ++++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 38 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 181 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1698 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 422 +++++ sound/soc/mediatek/mt2701/mt2701-reg.h | 186 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 73 + .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 921 +++++------ sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 27 files changed, 4399 insertions(+), 634 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/common/Makefile create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.c create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.h create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.c create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.h create mode 100644 sound/soc/mediatek/common/mtk-base-afe.h create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (51%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

2 10

Re: [alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by Kai Chieh Chuang 29 Jun '16

29 Jun '16

> If this isn't happening that seems like a bug, I'm a bit surprised > nobody else ran into it? Shouldn't the counts of stream events that > happen be symmetric (ie, we get as many stops as starts), or are we > possibly missing some from things being switched in and out or something? This idea is same as the DPCM BE state control, in int dpcm_be_dai_hw_free(..) { ...... /* do not free hw if this BE is used by other FE */ if (be->dpcm[stream].users > 1) continue; and int dpcm_be_dai_shutdown(...) { .... if (--be->dpcm[stream].users != 0) continue; The BE receiving SND_SOC_DAPM_STREAM_STOP event, while still used by a FE who is in STREAM_START state. The DAPM will just inactive BE related widget/dai without checking. This just doesn't seems right. And the last FE using the BE calls in to dpcm_fe_dai_shutdown(), will run the process normally. Actually, i trace the log, when one of the FE sent stop event to "still in use BE", the BE power state will not be changed. Until the system goes into SNDRV_CTL_POWER_D3cold, the BE will be shutdown (i did set the ignore_suspend=1). I didn't check why the BE is not turned off at the point when receiving the stop event. I'm also curious why no one seems to have the same issue. Probably there is not much DPCM architecture system yet on the Linux?

2 1

Re: [alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by Mark Brown 29 Jun '16

29 Jun '16

On Fri, Jun 24, 2016 at 02:21:34PM +0800, KaiChieh Chuang wrote: > Is there any comment on this? Please don't send content free pings and please allow a reasonable time for review. People get busy, go on holiday, attend conferences and so on so unless there is some reason for urgency (like critical bug fixes) please allow at least a couple of weeks for review. Sending content free pings just adds to the mail volume (if they are seen at all) and if something has gone wrong you'll have to resend the patches anyway.

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode
by Nicolin Chen 29 Jun '16

29 Jun '16

On Tue, Jun 28, 2016 at 10:59:32PM +0300, Alexander Shiyan wrote: > >Вторник, 28 июня 2016, 21:20 +03:00 от Nicolin Chen <nicoleotsuka(a)gmail.com>: > > > >On Sat, Jun 25, 2016 at 07:59:22AM +0300, Alexander Shiyan wrote: > >> The i.MX51 datasheet says: > >> Chapter 56.1.2.4 I2S Mode > >> ... > >> When I2S modes are entered (I2S master (01) or I2S slave (10)), > >> the following settings are recommended: > >> ... > >> - TX Frame Rate should be 2 i.e. (STCCR[12:8] = 1) > >> - RX Frame Rate should be 2 i.e. (SRCCR[12:8] = 1) > >> > >> Chapter 56.3.3.12 SSI Transmit and Receive Clock Control Registers (STCCR & SRCCR) > >> ... > >> Bits 12-8 DC4-DC0 > >> Frame Rate Divider Control. These bits are used to control the divide ratio > >> for the programmable frame rate dividers. The divide ratio works on the word > >> clock. In Normal mode, this ratio determines the word transfer rate. > >> In Network mode, this ratio sets the number of words per frame. The divide > >> ratio ranges from 1 to 32 in Normal mode and from 2 to 32 in Network mode. > >> In Normal mode, a divide ratio of 1 (DC=00000) provides continuous periodic > >> data word transfer. A bit-length frame sync must be used in this case. > >> > >> Function fsl_ssi_hw_params() setup Normal mode for MONO output, > >> so with DC=0, SSI enters to continuous periodic data word transfer. > >> To fix this, setup DC for any I2S mode. > > > >The change looks fine and didn't show any problem when I tested it. > >But I am wondering what's the failed user case here so we need this > >patch to fix? (The mono playback works without this patch as well.) > > Probably this works fine without patch if FSclk == BITclk * size. I acked this patch since it's more likely a correct configuration according to the reference manual while still want to know what's your failed case. Could you please tell me what were the FSclk, BITclk and size of your case? Thanks Nicolin

2 2

[alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode
by Alexander Shiyan 29 Jun '16

29 Jun '16

The i.MX51 datasheet says: Chapter 56.1.2.4 I2S Mode ... When I2S modes are entered (I2S master (01) or I2S slave (10)), the following settings are recommended: ... - TX Frame Rate should be 2 i.e. (STCCR[12:8] = 1) - RX Frame Rate should be 2 i.e. (SRCCR[12:8] = 1) Chapter 56.3.3.12 SSI Transmit and Receive Clock Control Registers (STCCR & SRCCR) ... Bits 12-8 DC4-DC0 Frame Rate Divider Control. These bits are used to control the divide ratio for the programmable frame rate dividers. The divide ratio works on the word clock. In Normal mode, this ratio determines the word transfer rate. In Network mode, this ratio sets the number of words per frame. The divide ratio ranges from 1 to 32 in Normal mode and from 2 to 32 in Network mode. In Normal mode, a divide ratio of 1 (DC=00000) provides continuous periodic data word transfer. A bit-length frame sync must be used in this case. Function fsl_ssi_hw_params() setup Normal mode for MONO output, so with DC=0, SSI enters to continuous periodic data word transfer. To fix this, setup DC for any I2S mode. Patch has tested on custom board based on Digi CCMX-51 module (i.MX51). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/fsl_ssi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 632ecc0..bedec4a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -952,16 +952,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, ssi_private->i2s_mode = CCSR_SSI_SCR_NET; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + regmap_update_bits(regs, CCSR_SSI_STCCR, + CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + regmap_update_bits(regs, CCSR_SSI_SRCCR, + CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFS: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER; - regmap_update_bits(regs, CCSR_SSI_STCCR, - CCSR_SSI_SxCCR_DC_MASK, - CCSR_SSI_SxCCR_DC(2)); - regmap_update_bits(regs, CCSR_SSI_SRCCR, - CCSR_SSI_SxCCR_DC_MASK, - CCSR_SSI_SxCCR_DC(2)); break; case SND_SOC_DAIFMT_CBM_CFM: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_SLAVE; -- 2.4.9

3 3

[alsa-devel] [PATCH v4 0/4] ASoC: MAX9860: new driver
by Peter Rosin 29 Jun '16

29 Jun '16

Hi! I kept the sidetone changes in separate patches since I don't know if I solved the DAPM-off issue cleanly enough. Also, a new driver may still have a chance to make 4.7, but maybe it's too late for the DAPM changes? Finally, I wonder if the device tree crowd prefers seeing the whole of a series such as this, or only the bindings patch? Cheers, Peter New in v4: - Split out the device tree bindings in its own patch. - Move format check from ->startup to ->set_fmt. - Don't spam the log on probe defer. - Follow-up patches for the backwards sidetone control. New in v3: - The updated bindings file went missing in v2. Sorry for the confusion. New in v2: - Add comment about fall through when Integer Clock Mode is not possible. - Drop export of max9860_probe. - Ignore clk docs and read the mclk rate w/o enabling the clock. - Manage the DVDDIO supply. Peter Rosin (4): dt-bindings: sound: add bindings for the max9860 codec ASoC: MAX9860: new driver ASoC: dapm: support mixer controls with mute at non-zero value ASoC: MAX9860: add sidetone mixer control .../devicetree/bindings/sound/max9860.txt | 28 + MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9860.c | 765 +++++++++++++++++++++ sound/soc/codecs/max9860.h | 162 +++++ sound/soc/soc-dapm.c | 38 +- 7 files changed, 1005 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/max9860.txt create mode 100644 sound/soc/codecs/max9860.c create mode 100644 sound/soc/codecs/max9860.h -- 2.1.4

2 10

[alsa-devel] [PATCH] ASoC: cs35l33: mark PM functions as __maybe_unused
by Arnd Bergmann 29 Jun '16

29 Jun '16

The newly added cs35l33 driver produces a harmless warning when CONFIG_PM is disabled: sound/soc/codecs/cs35l33.c:908:12: error: 'cs35l33_runtime_suspend' defined but not used [-Werror=unused-function] sound/soc/codecs/cs35l33.c:868:12: error: 'cs35l33_runtime_resume' defined but not used [-Werror=unused-function] This adds __maybe_unused annotations to shut up the warning regardless of the configuration. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cs35l33.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 622a1111b21c..d8b5fc3fc45d 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -862,7 +862,7 @@ static const struct regmap_config cs35l33_regmap = { .use_single_rw = true, }; -static int cs35l33_runtime_resume(struct device *dev) +static int __maybe_unused cs35l33_runtime_resume(struct device *dev) { struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); int ret; @@ -902,7 +902,7 @@ err: return ret; } -static int cs35l33_runtime_suspend(struct device *dev) +static int __maybe_unused cs35l33_runtime_suspend(struct device *dev) { struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); -- 2.9.0

3 3

[alsa-devel] [PATCH v3 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 29 Jun '16

29 Jun '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 58 +++++++++++++++++++++++++++++++++++----------- sound/soc/soc-pcm.c | 3 ++- 3 files changed, 51 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..8eed5c4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..a6976d5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,9 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); + if (custom_stop_condition && custom_stop_condition(widget, dir)) + return con; + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1113,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1126,37 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, it should return true if the walk should be + * stopped and false otherwise. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, it should return true if the walk should be + * stopped and false otherwise. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1164,24 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, it should return true if the walk should be + * stopped and false otherwise. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1201,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1300,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1960,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

2 2

[alsa-devel] old bug in au88x0_core.c
by Dan Carpenter 29 Jun '16

29 Jun '16

Hi ALSA devs, The patch 1da177e4c3f4: "Linux-2.6.12-rc2" from Apr 16, 2005, leads to the following static checker warning: sound/pci/au88x0/au88x0_core.c:1449 vortex_wtdma_bufshift() warn: mask and shift to zero sound/pci/au88x0/au88x0_core.c 1441 static int vortex_wtdma_bufshift(vortex_t * vortex, int wtdma) 1442 { 1443 stream_t *dma = &vortex->dma_wt[wtdma]; 1444 int page, p, pp, delta, i; 1445 1446 page = 1447 (hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)) & 1448 WT_SUBBUF_MASK) 1449 >> WT_SUBBUF_SHIFT; This is always zero because: #define WT_SUBBUF_MASK 0x3 #define WT_SUBBUF_SHIFT 0x15 1450 if (dma->nr_periods >= 4) 1451 delta = (page - dma->period_real) & 3; 1452 else { 1453 delta = (page - dma->period_real); 1454 if (delta < 0) 1455 delta += dma->nr_periods; 1456 } 1457 if (delta == 0) 1458 return 0; 1493 #if 0 1494 static void 1495 vortex_wtdma_getposition(vortex_t * vortex, int wtdma, int *subbuf, int *pos) 1496 { 1497 int temp; 1498 temp = hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)); 1499 *subbuf = (temp >> WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK; >From this ifdef zero code, it looks like we should shift first then do the mask. 1500 *pos = temp & POS_MASK; 1501 } 1502 1503 static int vortex_wtdma_getcursubuffer(vortex_t * vortex, int wtdma) 1504 { 1505 return ((hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)) >> 1506 POS_SHIFT) & POS_MASK); 1507 } 1508 #endif regards, dan carpenter

2 1

[alsa-devel] [PATCH] ctl: use consistent name for second argument of snd_ctl_elem_read/write
by Takashi Sakamoto 29 Jun '16

29 Jun '16

Second argument of these functions has three different names in documentation/header/code (obj/control/value). This easily causes users' confusion. This commit applies consistent names for the arguments. Fixes: 90020c05f886 ('ctl: improve comments for handling element data') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/control.h | 4 ++-- src/control/control.c | 16 ++++++++-------- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/include/control.h b/include/control.h index 0d4ae86..6931b1f 100644 --- a/include/control.h +++ b/include/control.h @@ -245,8 +245,8 @@ int snd_ctl_subscribe_events(snd_ctl_t *ctl, int subscribe); int snd_ctl_card_info(snd_ctl_t *ctl, snd_ctl_card_info_t *info); int snd_ctl_elem_list(snd_ctl_t *ctl, snd_ctl_elem_list_t *list); int snd_ctl_elem_info(snd_ctl_t *ctl, snd_ctl_elem_info_t *info); -int snd_ctl_elem_read(snd_ctl_t *ctl, snd_ctl_elem_value_t *value); -int snd_ctl_elem_write(snd_ctl_t *ctl, snd_ctl_elem_value_t *value); +int snd_ctl_elem_read(snd_ctl_t *ctl, snd_ctl_elem_value_t *data); +int snd_ctl_elem_write(snd_ctl_t *ctl, snd_ctl_elem_value_t *data); int snd_ctl_elem_lock(snd_ctl_t *ctl, snd_ctl_elem_id_t *id); int snd_ctl_elem_unlock(snd_ctl_t *ctl, snd_ctl_elem_id_t *id); int snd_ctl_elem_tlv_read(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, diff --git a/src/control/control.c b/src/control/control.c index 6090abd..bbb5a78 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -803,27 +803,27 @@ int snd_ctl_elem_remove(snd_ctl_t *ctl, snd_ctl_elem_id_t *id) /** * \brief Get CTL element value * \param ctl CTL handle - * \param obj Data of an element. + * \param data Data of an element. * \return 0 on success otherwise a negative error code */ -int snd_ctl_elem_read(snd_ctl_t *ctl, snd_ctl_elem_value_t *control) +int snd_ctl_elem_read(snd_ctl_t *ctl, snd_ctl_elem_value_t *data) { - assert(ctl && control && (control->id.name[0] || control->id.numid)); - return ctl->ops->element_read(ctl, control); + assert(ctl && data && (data->id.name[0] || data->id.numid)); + return ctl->ops->element_read(ctl, data); } /** * \brief Set CTL element value * \param ctl CTL handle - * \param obj Data of an element. + * \param data Data of an element. * \retval 0 on success * \retval >0 on success when value was changed * \retval <0 a negative error code */ -int snd_ctl_elem_write(snd_ctl_t *ctl, snd_ctl_elem_value_t *control) +int snd_ctl_elem_write(snd_ctl_t *ctl, snd_ctl_elem_value_t *data) { - assert(ctl && control && (control->id.name[0] || control->id.numid)); - return ctl->ops->element_write(ctl, control); + assert(ctl && data && (data->id.name[0] || data->id.numid)); + return ctl->ops->element_write(ctl, data); } static int snd_ctl_tlv_do(snd_ctl_t *ctl, int op_flag, -- 2.7.4

2 2

[alsa-devel] [PATCH] ALSA: hda - Add PCI ID for Kabylake-H
by Vinod Koul 29 Jun '16

29 Jun '16

Kabylake-H shows up as PCI ID 0xa2f0. We missed adding this earlier with other KBL IDs. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/hda_intel.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 94089fc71884..e320c44714b1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -367,9 +367,10 @@ enum { #define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) #define IS_KBL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa171) #define IS_KBL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d71) +#define IS_KBL_H(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa2f0) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) #define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) || \ - IS_KBL(pci) || IS_KBL_LP(pci) + IS_KBL(pci) || IS_KBL_LP(pci) || IS_KBL_H(pci) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -2190,6 +2191,9 @@ static const struct pci_device_id azx_ids[] = { /* Kabylake-LP */ { PCI_DEVICE(0x8086, 0x9d71), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, + /* Kabylake-H */ + { PCI_DEVICE(0x8086, 0xa2f0), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, -- 1.9.1

2 1

[alsa-devel] [PATCH v7 1/2] ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.
by Paul.Handrigan＠cirrus.com 28 Jun '16

28 Jun '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l33 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l33.h | 48 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l33.c | 1315 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l33.h | 221 ++++++++ 5 files changed, 1591 insertions(+) create mode 100644 include/sound/cs35l33.h create mode 100644 sound/soc/codecs/cs35l33.c create mode 100644 sound/soc/codecs/cs35l33.h diff --git a/include/sound/cs35l33.h b/include/sound/cs35l33.h new file mode 100644 index 0000000..b6eadce --- /dev/null +++ b/include/sound/cs35l33.h @@ -0,0 +1,48 @@ +/* + * linux/sound/cs35l33.h -- Platform data for CS35l33 + * + * Copyright (c) 2016 Cirrus Logic Inc. + * + * 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 __CS35L33_H +#define __CS35L33_H + +struct cs35l33_hg { + bool enable_hg_algo; + unsigned int mem_depth; + unsigned int release_rate; + unsigned int hd_rm; + unsigned int ldo_thld; + unsigned int ldo_path_disable; + unsigned int ldo_entry_delay; + bool vp_hg_auto; + unsigned int vp_hg; + unsigned int vp_hg_rate; + unsigned int vp_hg_va; +}; + +struct cs35l33_pdata { + /* Boost Controller Voltage Setting */ + unsigned int boost_ctl; + + /* Boost Controller Peak Current */ + unsigned int boost_ipk; + + /* Amplifier Drive Select */ + unsigned int amp_drv_sel; + + /* soft volume ramp */ + unsigned int ramp_rate; + + /* IMON adc scale */ + unsigned int imon_adc_scale; + + /* H/G algo configuration */ + struct cs35l33_hg hg_config; +}; + +#endif /* __CS35L33_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 60f8f13..02d2084 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C + select SND_SOC_CS35L33 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -391,6 +392,10 @@ config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C +config SND_SOC_CS35L33 + tristate "Cirrus Logic CS35L33 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1264fa4..abf2126 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -37,6 +37,7 @@ snd-soc-arizona-objs := arizona.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o +snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -257,6 +258,7 @@ obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o +obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c new file mode 100644 index 0000000..841374a --- /dev/null +++ b/sound/soc/codecs/cs35l33.c @@ -0,0 +1,1315 @@ +/* + * cs35l33.c -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.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/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.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 <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <sound/cs35l33.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/of_gpio.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> + +#include "cs35l33.h" + +#define CS35L33_BOOT_DELAY 50 + +struct cs35l33_private { + struct snd_soc_codec *codec; + struct cs35l33_pdata pdata; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + bool amp_cal; + int mclk_int; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + bool is_tdm_mode; + bool enable_soft_ramp; +}; + +static const struct reg_default cs35l33_reg[] = { + {CS35L33_PWRCTL1, 0x85}, + {CS35L33_PWRCTL2, 0xFE}, + {CS35L33_CLK_CTL, 0x0C}, + {CS35L33_BST_PEAK_CTL, 0x90}, + {CS35L33_PROTECT_CTL, 0x55}, + {CS35L33_BST_CTL1, 0x00}, + {CS35L33_BST_CTL2, 0x01}, + {CS35L33_ADSP_CTL, 0x00}, + {CS35L33_ADC_CTL, 0xC8}, + {CS35L33_DAC_CTL, 0x14}, + {CS35L33_DIG_VOL_CTL, 0x00}, + {CS35L33_CLASSD_CTL, 0x04}, + {CS35L33_AMP_CTL, 0x90}, + {CS35L33_INT_MASK_1, 0xFF}, + {CS35L33_INT_MASK_2, 0xFF}, + {CS35L33_DIAG_LOCK, 0x00}, + {CS35L33_DIAG_CTRL_1, 0x40}, + {CS35L33_DIAG_CTRL_2, 0x00}, + {CS35L33_HG_MEMLDO_CTL, 0x62}, + {CS35L33_HG_REL_RATE, 0x03}, + {CS35L33_LDO_DEL, 0x12}, + {CS35L33_HG_HEAD, 0x0A}, + {CS35L33_HG_EN, 0x05}, + {CS35L33_TX_VMON, 0x00}, + {CS35L33_TX_IMON, 0x03}, + {CS35L33_TX_VPMON, 0x02}, + {CS35L33_TX_VBSTMON, 0x05}, + {CS35L33_TX_FLAG, 0x06}, + {CS35L33_TX_EN1, 0x00}, + {CS35L33_TX_EN2, 0x00}, + {CS35L33_TX_EN3, 0x00}, + {CS35L33_TX_EN4, 0x00}, + {CS35L33_RX_AUD, 0x40}, + {CS35L33_RX_SPLY, 0x03}, + {CS35L33_RX_ALIVE, 0x04}, + {CS35L33_BST_CTL4, 0x63}, +}; + +static const struct reg_sequence cs35l33_patch[] = { + { 0x00, 0x99, 0 }, + { 0x59, 0x02, 0 }, + { 0x52, 0x30, 0 }, + { 0x39, 0x45, 0 }, + { 0x57, 0x30, 0 }, + { 0x2C, 0x68, 0 }, + { 0x00, 0x00, 0 }, +}; + +static bool cs35l33_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l33_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + /* these are read only registers */ + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return false; + default: + return true; + } +} + +static bool cs35l33_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_PWRCTL1: + case CS35L33_PWRCTL2: + case CS35L33_CLK_CTL: + case CS35L33_BST_PEAK_CTL: + case CS35L33_PROTECT_CTL: + case CS35L33_BST_CTL1: + case CS35L33_BST_CTL2: + case CS35L33_ADSP_CTL: + case CS35L33_ADC_CTL: + case CS35L33_DAC_CTL: + case CS35L33_DIG_VOL_CTL: + case CS35L33_CLASSD_CTL: + case CS35L33_AMP_CTL: + case CS35L33_INT_MASK_1: + case CS35L33_INT_MASK_2: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_DIAG_LOCK: + case CS35L33_DIAG_CTRL_1: + case CS35L33_DIAG_CTRL_2: + case CS35L33_HG_MEMLDO_CTL: + case CS35L33_HG_REL_RATE: + case CS35L33_LDO_DEL: + case CS35L33_HG_HEAD: + case CS35L33_HG_EN: + case CS35L33_TX_VMON: + case CS35L33_TX_IMON: + case CS35L33_TX_VPMON: + case CS35L33_TX_VBSTMON: + case CS35L33_TX_FLAG: + case CS35L33_TX_EN1: + case CS35L33_TX_EN2: + case CS35L33_TX_EN3: + case CS35L33_TX_EN4: + case CS35L33_RX_AUD: + case CS35L33_RX_SPLY: + case CS35L33_RX_ALIVE: + case CS35L33_BST_CTL4: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l33_snd_controls[] = { + + SOC_SINGLE_TLV("SPK Amp Volume", CS35L33_AMP_CTL, + 4, 0x09, 0, classd_ctl_tlv), + SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, + 0, 0x34, 0xE4, dac_tlv), +}; + +static int cs35l33_spkrdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!priv->amp_cal) { + usleep_range(8000, 9000); + priv->amp_cal = true; + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, 0); + dev_dbg(codec->dev, "Amp calibration done\n"); + } + dev_dbg(codec->dev, "Amp turned on\n"); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "Amp turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + break; + } + + return 0; +} + +static int cs35l33_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, 0); + val = priv->is_tdm_mode ? 0 : CS35L33_PDN_TDM; + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, val); + dev_dbg(codec->dev, "BST turned on\n"); + break; + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "SDIN turned on\n"); + if (!priv->amp_cal) { + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, CS35L33_AMP_CAL); + dev_dbg(codec->dev, "Amp calibration started\n"); + usleep_range(10000, 11000); + } + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, CS35L33_PDN_TDM); + usleep_range(4000, 4100); + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, CS35L33_PDN_BST); + dev_dbg(codec->dev, "BST and SDIN turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + + } + + return 0; +} + +static int cs35l33_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int mask = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + unsigned int mask2 = CS35L33_SDOUT_3ST_TDM; + unsigned int val, val2; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->is_tdm_mode) { + /* set sdout_3st_i2s and reset pdn_tdm */ + val = CS35L33_SDOUT_3ST_I2S; + /* reset sdout_3st_tdm */ + val2 = 0; + } else { + /* reset sdout_3st_i2s and set pdn_tdm */ + val = CS35L33_PDN_TDM; + /* set sdout_3st_tdm */ + val2 = CS35L33_SDOUT_3ST_TDM; + } + dev_dbg(codec->dev, "SDOUT turned on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + val = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + val2 = CS35L33_SDOUT_3ST_TDM; + dev_dbg(codec->dev, "SDOUT turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return 0; + } + + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + mask, val); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + mask2, val2); + + return 0; +} + +static const struct snd_soc_dapm_widget cs35l33_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUT_DRV_E("SPKDRV", CS35L33_PWRCTL1, 7, 1, NULL, 0, + cs35l33_spkrdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L33_PWRCTL2, + 2, 1, cs35l33_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("MON"), + + SND_SOC_DAPM_ADC("VMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VMON_SHIFT, 1), + SND_SOC_DAPM_ADC("IMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_IMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VPMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VPMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VBSTMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VBSTMON_SHIFT, 1), + + SND_SOC_DAPM_AIF_OUT_E("SDOUT", NULL, 0, SND_SOC_NOPM, 0, 0, + cs35l33_sdout_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l33_audio_map[] = { + {"SDIN", NULL, "CS35L33 Playback"}, + {"SPKDRV", NULL, "SDIN"}, + {"SPK", NULL, "SPKDRV"}, + + {"VMON", NULL, "MON"}, + {"IMON", NULL, "MON"}, + + {"SDOUT", NULL, "VMON"}, + {"SDOUT", NULL, "IMON"}, + {"CS35L33 Capture", NULL, "SDOUT"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vphg_auto_route[] = { + {"SPKDRV", NULL, "VPMON"}, + {"VPMON", NULL, "CS35L33 Playback"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vp_vbst_mon_route[] = { + {"SDOUT", NULL, "VPMON"}, + {"VPMON", NULL, "MON"}, + {"SDOUT", NULL, "VBSTMON"}, + {"VBSTMON", NULL, "MON"}, +}; + +static int cs35l33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int val; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, 0); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, CS35L33_PDN_ALL); + regmap_read(priv->regmap, CS35L33_INT_STATUS_2, &val); + usleep_range(1000, 1100); + if (val & CS35L33_PDN_DONE) + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, CS35L33_MCLKDIS); + break; + case SND_SOC_BIAS_OFF: + break; + default: + return -EINVAL; + } + + dapm->bias_level = level; + + return 0; +} + +struct cs35l33_mclk_div { + int mclk; + int srate; + u8 adsp_rate; + u8 int_fs_ratio; +}; + +static const struct cs35l33_mclk_div cs35l33_mclk_coeffs[] = { + /* MCLK, Sample Rate, adsp_rate, int_fs_ratio */ + {5644800, 11025, 0x4, CS35L33_INT_FS_RATE}, + {5644800, 22050, 0x8, CS35L33_INT_FS_RATE}, + {5644800, 44100, 0xC, CS35L33_INT_FS_RATE}, + + {6000000, 8000, 0x1, 0}, + {6000000, 11025, 0x2, 0}, + {6000000, 11029, 0x3, 0}, + {6000000, 12000, 0x4, 0}, + {6000000, 16000, 0x5, 0}, + {6000000, 22050, 0x6, 0}, + {6000000, 22059, 0x7, 0}, + {6000000, 24000, 0x8, 0}, + {6000000, 32000, 0x9, 0}, + {6000000, 44100, 0xA, 0}, + {6000000, 44118, 0xB, 0}, + {6000000, 48000, 0xC, 0}, + + {6144000, 8000, 0x1, CS35L33_INT_FS_RATE}, + {6144000, 12000, 0x4, CS35L33_INT_FS_RATE}, + {6144000, 16000, 0x5, CS35L33_INT_FS_RATE}, + {6144000, 24000, 0x8, CS35L33_INT_FS_RATE}, + {6144000, 32000, 0x9, CS35L33_INT_FS_RATE}, + {6144000, 48000, 0xC, CS35L33_INT_FS_RATE}, +}; + +static int cs35l33_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l33_mclk_coeffs); i++) { + if (cs35l33_mclk_coeffs[i].mclk == mclk && + cs35l33_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, CS35L33_MS_MASK); + dev_dbg(codec->dev, "Audio port in master mode\n"); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, 0); + dev_dbg(codec->dev, "Audio port in slave mode\n"); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + /* + * tdm mode in cs35l33 resembles dsp-a mode very + * closely, it is dsp-a with fsync shifted left by half bclk + */ + priv->is_tdm_mode = true; + dev_dbg(codec->dev, "Audio port in TDM mode\n"); + break; + case SND_SOC_DAIFMT_I2S: + priv->is_tdm_mode = false; + dev_dbg(codec->dev, "Audio port in I2S mode\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l33_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + int sample_size = params_width(params); + int coeff = cs35l33_get_mclk_coeff(priv->mclk_int, params_rate(params)); + + if (coeff < 0) + return coeff; + + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_ADSP_FS | CS35L33_INT_FS_RATE, + cs35l33_mclk_coeffs[coeff].int_fs_ratio + | cs35l33_mclk_coeffs[coeff].adsp_rate); + + if (priv->is_tdm_mode) { + sample_size = (sample_size / 8) - 1; + if (sample_size > 2) + sample_size = 2; + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_AUDIN_RX_DEPTH, + sample_size << CS35L33_AUDIN_RX_DEPTH_SHIFT); + } + + dev_dbg(codec->dev, "sample rate=%d, bits per sample=%d\n", + params_rate(params), params_width(params)); + + return 0; +} + +static const unsigned int cs35l33_src_rates[] = { + 8000, 11025, 11029, 12000, 16000, 22050, + 22059, 24000, 32000, 44100, 44118, 48000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l33_constraints = { + .count = ARRAY_SIZE(cs35l33_src_rates), + .list = cs35l33_src_rates, +}; + +static int cs35l33_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l33_constraints); + return 0; +} + +static int cs35l33_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (tristate) { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, CS35L33_SDOUT_3ST_I2S); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, CS35L33_SDOUT_3ST_TDM); + } else { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, 0); + } + + return 0; +} + +static int cs35l33_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos, i; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) { + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "Audio starts from slots %d", slot); + } + + /* + * scan tx_mask: vmon(2 slots); imon (2 slots); + * vpmon (1 slot) vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + for (i = 0; i < 2 ; i++) { + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON + i, + CS35L33_X_STATE | CS35L33_X_LOC, CS35L33_X_STATE + | CS35L33_X_LOC); + } + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route, + ARRAY_SIZE(cs35l33_vp_vbst_mon_route)); + + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) { + regmap_update_bits(priv->regmap, CS35L33_TX_VMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "VMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure IMON_TX_LOC */ + if (slot_num == 3) { + regmap_update_bits(priv->regmap, CS35L33_TX_IMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "IMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure VPMON_TX_LOC */ + if (slot_num == 4) { + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[0], 2); + dev_dbg(codec->dev, "VPMON enabled in slots %d", slot); + } + + /* configure VBSTMON_TX_LOC */ + if (slot_num == 5) { + regmap_update_bits(priv->regmap, CS35L33_TX_VBSTMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[2], 2); + dev_dbg(codec->dev, + "VBSTMON enabled in slots %d", slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L33_TX_EN4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + regmap_update_bits(priv->regmap, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l33_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS35L33_MCLK_5644: + case CS35L33_MCLK_6: + case CS35L33_MCLK_6144: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, 0); + cs35l33->mclk_int = freq; + break; + case CS35L33_MCLK_11289: + case CS35L33_MCLK_12: + case CS35L33_MCLK_12288: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, CS35L33_MCLKDIV2); + cs35l33->mclk_int = freq/2; + break; + default: + cs35l33->mclk_int = 0; + return -EINVAL; + } + + dev_dbg(codec->dev, "external mclk freq=%d, internal mclk freq=%d\n", + freq, cs35l33->mclk_int); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l33_ops = { + .startup = cs35l33_pcm_startup, + .set_tristate = cs35l33_set_tristate, + .set_fmt = cs35l33_set_dai_fmt, + .hw_params = cs35l33_pcm_hw_params, + .set_tdm_slot = cs35l33_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l33_dai = { + .name = "cs35l33-dai", + .id = 0, + .playback = { + .stream_name = "CS35L33 Playback", + .channels_min = 1, + .channels_max = 1, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .capture = { + .stream_name = "CS35L33 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .ops = &cs35l33_ops, + .symmetric_rates = 1, +}; + +static int cs35l33_set_hg_data(struct snd_soc_codec *codec, + struct cs35l33_pdata *pdata) +{ + struct cs35l33_hg *hg_config = &pdata->hg_config; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (hg_config->enable_hg_algo) { + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_MEM_DEPTH_MASK, + hg_config->mem_depth << CS35L33_MEM_DEPTH_SHIFT); + regmap_write(priv->regmap, CS35L33_HG_REL_RATE, + hg_config->release_rate); + regmap_update_bits(priv->regmap, CS35L33_HG_HEAD, + CS35L33_HD_RM_MASK, + hg_config->hd_rm << CS35L33_HD_RM_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_THLD_MASK, + hg_config->ldo_thld << CS35L33_LDO_THLD_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_DISABLE_MASK, + hg_config->ldo_path_disable << + CS35L33_LDO_DISABLE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_LDO_ENTRY_DELAY_MASK, + hg_config->ldo_entry_delay << + CS35L33_LDO_ENTRY_DELAY_SHIFT); + if (hg_config->vp_hg_auto) { + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_AUTO_MASK, + CS35L33_VP_HG_AUTO_MASK); + snd_soc_dapm_add_routes(dapm, cs35l33_vphg_auto_route, + ARRAY_SIZE(cs35l33_vphg_auto_route)); + } + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_MASK, + hg_config->vp_hg << CS35L33_VP_HG_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_RATE_MASK, + hg_config->vp_hg_rate << CS35L33_VP_HG_RATE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_VA_MASK, + hg_config->vp_hg_va << CS35L33_VP_HG_VA_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_CLASS_HG_EN_MASK, CS35L33_CLASS_HG_EN_MASK); + } + return 0; +} + +static int cs35l33_set_bst_ipk(struct snd_soc_codec *codec, unsigned int bst) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + int ret = 0, steps = 0; + + /* Boost current in uA */ + if (bst > 3600000 || bst < 1850000) { + dev_err(codec->dev, "Invalid boost current %d\n", bst); + ret = -EINVAL; + goto err; + } + + if (bst % 15625) { + dev_err(codec->dev, "Current not a multiple of 15625uA (%d)\n", + bst); + ret = -EINVAL; + goto err; + } + + while (bst > 1850000) { + bst -= 15625; + steps++; + } + + regmap_write(cs35l33->regmap, CS35L33_BST_PEAK_CTL, + steps+0x70); + +err: + return ret; +} + +static int cs35l33_probe(struct snd_soc_codec *codec) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + cs35l33->codec = codec; + pm_runtime_get_sync(codec->dev); + + regmap_update_bits(cs35l33->regmap, CS35L33_PROTECT_CTL, + CS35L33_ALIVE_WD_DIS, 0x8); + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL2, + CS35L33_ALIVE_WD_DIS2, + CS35L33_ALIVE_WD_DIS2); + + /* Set Platform Data */ + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL1, + CS35L33_BST_CTL_MASK, cs35l33->pdata.boost_ctl); + regmap_update_bits(cs35l33->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_DRV_SEL_MASK, + cs35l33->pdata.amp_drv_sel << CS35L33_AMP_DRV_SEL_SHIFT); + + if (cs35l33->pdata.boost_ipk) + cs35l33_set_bst_ipk(codec, cs35l33->pdata.boost_ipk); + + if (cs35l33->enable_soft_ramp) { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, CS35L33_DIGSFT); + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DSR_RATE, cs35l33->pdata.ramp_rate); + } else { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, 0); + } + + /* update IMON scaling rate if different from default of 0x8 */ + if (cs35l33->pdata.imon_adc_scale != 0x8) + snd_soc_update_bits(codec, CS35L33_ADC_CTL, + CS35L33_IMON_SCALE, cs35l33->pdata.imon_adc_scale); + + cs35l33_set_hg_data(codec, &(cs35l33->pdata)); + + /* + * unmask important interrupts that causes the chip to enter + * speaker safe mode and hence deserves user attention + */ + regmap_update_bits(cs35l33->regmap, CS35L33_INT_MASK_1, + CS35L33_M_OTE | CS35L33_M_OTW | CS35L33_M_AMP_SHORT | + CS35L33_M_CAL_ERR, 0); + + pm_runtime_put_sync(codec->dev); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { + .probe = cs35l33_probe, + + .set_bias_level = cs35l33_set_bias_level, + .set_sysclk = cs35l33_codec_set_sysclk, + + .dapm_widgets = cs35l33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l33_dapm_widgets), + .dapm_routes = cs35l33_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map), + .controls = cs35l33_snd_controls, + .num_controls = ARRAY_SIZE(cs35l33_snd_controls), + + .idle_bias_off = true, +}; + +static const struct regmap_config cs35l33_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L33_MAX_REGISTER, + .reg_defaults = cs35l33_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l33_reg), + .volatile_reg = cs35l33_volatile_register, + .readable_reg = cs35l33_readable_register, + .writeable_reg = cs35l33_writeable_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, +}; + +static int cs35l33_runtime_resume(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + int ret; + + dev_dbg(dev, "%s\n", __func__); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, false); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + + ret = regcache_sync(cs35l33->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + return 0; + +err: + regcache_cache_only(cs35l33->regmap, true); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return ret; +} + +static int cs35l33_runtime_suspend(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s\n", __func__); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + regcache_cache_only(cs35l33->regmap, true); + regcache_mark_dirty(cs35l33->regmap); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l33_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l33_runtime_suspend, + cs35l33_runtime_resume, + NULL) +}; + +static int cs35l33_get_hg_data(const struct device_node *np, + struct cs35l33_pdata *pdata) +{ + struct device_node *hg; + struct cs35l33_hg *hg_config = &pdata->hg_config; + u32 val32; + + hg = of_get_child_by_name(np, "cirrus,hg-algo"); + hg_config->enable_hg_algo = hg ? true : false; + + if (hg_config->enable_hg_algo) { + if (of_property_read_u32(hg, "cirrus,mem-depth", &val32) >= 0) + hg_config->mem_depth = val32; + if (of_property_read_u32(hg, "cirrus,release-rate", + &val32) >= 0) + hg_config->release_rate = val32; + if (of_property_read_u32(hg, "cirrus,ldo-thld", &val32) >= 0) + hg_config->ldo_thld = val32; + if (of_property_read_u32(hg, "cirrus,ldo-path-disable", + &val32) >= 0) + hg_config->ldo_path_disable = val32; + if (of_property_read_u32(hg, "cirrus,ldo-entry-delay", + &val32) >= 0) + hg_config->ldo_entry_delay = val32; + + hg_config->vp_hg_auto = of_property_read_bool(hg, + "cirrus,vp-hg-auto"); + + if (of_property_read_u32(hg, "cirrus,vp-hg", &val32) >= 0) + hg_config->vp_hg = val32; + if (of_property_read_u32(hg, "cirrus,vp-hg-rate", &val32) >= 0) + hg_config->vp_hg_rate = val32; + if (of_property_read_u32(hg, "cirrus,vp-hg-va", &val32) >= 0) + hg_config->vp_hg_va = val32; + } + + of_node_put(hg); + + return 0; +} + +static irqreturn_t cs35l33_irq_thread(int irq, void *data) +{ + struct cs35l33_private *cs35l33 = data; + struct snd_soc_codec *codec = cs35l33->codec; + unsigned int sticky_val1, sticky_val2, current_val, mask1, mask2; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2, + &sticky_val2); + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &sticky_val1); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_2, &mask2); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_1, &mask1); + + /* Check to see if the unmasked bits are active, + * if not then exit. + */ + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + return IRQ_NONE; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &current_val); + + /* handle the interrupts */ + + if (sticky_val1 & CS35L33_AMP_SHORT) { + dev_crit(codec->dev, "Amp short error\n"); + if (!(current_val & CS35L33_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, + CS35L33_AMP_SHORT_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_AMP_SHORT_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + if (!(current_val & CS35L33_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + CS35L33_CAL_ERR_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_OTE) { + dev_crit(codec->dev, "Over temperature error\n"); + if (!(current_val & CS35L33_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, + CS35L33_OTE_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + } + } + + if (sticky_val1 & CS35L33_OTW) { + dev_err(codec->dev, "Over temperature warning\n"); + if (!(current_val & CS35L33_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, + CS35L33_OTW_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + } + } + if (CS35L33_ALIVE_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: ADSPCLK Interrupt\n"); + + if (CS35L33_MCLK_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: MCLK Interrupt\n"); + + if (CS35L33_VMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VMON Overflow Interrupt\n"); + + if (CS35L33_IMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: IMON Overflow Interrupt\n"); + + if (CS35L33_VPMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + + return IRQ_HANDLED; +} + +static const char * const cs35l33_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l33_of_get_pdata(struct device *dev, + struct cs35l33_private *cs35l33) +{ + struct device_node *np = dev->of_node; + struct cs35l33_pdata *pdata = &cs35l33->pdata; + u32 val32; + + if (!np) + return 0; + + if (of_property_read_u32(np, "cirrus,boost-ctl", &val32) >= 0) { + pdata->boost_ctl = val32; + pdata->amp_drv_sel = 1; + } + + if (of_property_read_u32(np, "cirrus,ramp-rate", &val32) >= 0) { + pdata->ramp_rate = val32; + cs35l33->enable_soft_ramp = true; + } + + if (of_property_read_u32(np, "cirrus,boost-ipk", &val32) >= 0) + pdata->boost_ipk = val32; + + if (of_property_read_u32(np, "cirrus,imon-adc-scale", &val32) >= 0) { + if ((val32 == 0x0) || (val32 == 0x7) || (val32 == 0x6)) + pdata->imon_adc_scale = val32; + else + /* use default value */ + pdata->imon_adc_scale = 0x8; + } else { + /* use default value */ + pdata->imon_adc_scale = 0x8; + } + + cs35l33_get_hg_data(np, pdata); + + return 0; +} + +static int cs35l33_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l33_private *cs35l33; + struct cs35l33_pdata *pdata = dev_get_platdata(&i2c_client->dev); + int ret, devid, i; + unsigned int reg; + + cs35l33 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l33_private), + GFP_KERNEL); + if (!cs35l33) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l33); + cs35l33->regmap = devm_regmap_init_i2c(i2c_client, &cs35l33_regmap); + if (IS_ERR(cs35l33->regmap)) { + ret = PTR_ERR(cs35l33->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, true); + + for (i = 0; i < ARRAY_SIZE(cs35l33_core_supplies); i++) + cs35l33->core_supplies[i].supply + = cs35l33_core_supplies[i]; + cs35l33->num_core_supplies = ARRAY_SIZE(cs35l33_core_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l33->pdata = *pdata; + } else { + cs35l33_of_get_pdata(&i2c_client->dev, cs35l33); + pdata = &cs35l33->pdata; + } + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l33_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l33", cs35l33); + if (ret != 0) + dev_warn(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + /* We could issue !RST or skip it based on AMP topology */ + cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset-gpios", GPIOD_OUT_HIGH); + + if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { + dev_warn(&i2c_client->dev, + "%s WARNING: No reset gpio assigned\n", __func__); + } else if (IS_ERR(cs35l33->reset_gpio)) { + dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", + __func__); + return PTR_ERR(cs35l33->reset_gpio); + } + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + goto err_irq; + } + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + regcache_cache_only(cs35l33->regmap, false); + + /* initialize codec */ + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L33_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L33 Device ID (%X). Expected ID %X\n", + devid, CS35L33_CHIP_ID); + goto err_enable; + } + + ret = regmap_read(cs35l33->regmap, CS35L33_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_enable; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L33, Revision: %02X\n", ret & 0xFF); + + ret = regmap_register_patch(cs35l33->regmap, + cs35l33_patch, ARRAY_SIZE(cs35l33_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Error in applying regmap patch: %d\n", ret); + goto err_enable; + } + + /* disable mclk and tdm */ + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l33, &cs35l33_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, "%s: Register codec failed\n", + __func__); + goto err_irq; + } + + return 0; + +err_enable: + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); +err_irq: + free_irq(i2c_client->irq, cs35l33); + + return ret; +} + +static int cs35l33_i2c_remove(struct i2c_client *client) +{ + struct cs35l33_private *cs35l33 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + free_irq(client->irq, cs35l33); + + return 0; +} + +static const struct of_device_id cs35l33_of_match[] = { + { .compatible = "cirrus,cs35l33", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l33_of_match); + +static const struct i2c_device_id cs35l33_id[] = { + {"cs35l33", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l33_id); + +static struct i2c_driver cs35l33_i2c_driver = { + .driver = { + .name = "cs35l33", + .owner = THIS_MODULE, + .pm = &cs35l33_pm_ops, + .of_match_table = cs35l33_of_match, + + }, + .id_table = cs35l33_id, + .probe = cs35l33_i2c_probe, + .remove = cs35l33_i2c_remove, + +}; +module_i2c_driver(cs35l33_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L33 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l33.h b/sound/soc/codecs/cs35l33.h new file mode 100644 index 0000000..c045737 --- /dev/null +++ b/sound/soc/codecs/cs35l33.h @@ -0,0 +1,221 @@ +/* + * cs35l33.h -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.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 __CS35L33_H__ +#define __CS35L33_H__ + +#define CS35L33_CHIP_ID 0x00035A33 +#define CS35L33_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L33_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L33_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L33_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L33_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L33_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L33_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L33_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L33_BST_PEAK_CTL 0x09 /* Max Current for Boost */ +#define CS35L33_PROTECT_CTL 0x0A /* Amp Protection Parameters */ +#define CS35L33_BST_CTL1 0x0B /* Boost Converter CTL1 */ +#define CS35L33_BST_CTL2 0x0C /* Boost Converter CTL2 */ +#define CS35L33_ADSP_CTL 0x0D /* Serial Port Control */ +#define CS35L33_ADC_CTL 0x0E /* ADC Control */ +#define CS35L33_DAC_CTL 0x0F /* DAC Control */ +#define CS35L33_DIG_VOL_CTL 0x10 /* Digital Volume CTL */ +#define CS35L33_CLASSD_CTL 0x11 /* Class D Amp CTL */ +#define CS35L33_AMP_CTL 0x12 /* Amp Gain/Protecton Release CTL */ +#define CS35L33_INT_MASK_1 0x13 /* Interrupt Mask 1 */ +#define CS35L33_INT_MASK_2 0x14 /* Interrupt Mask 2 */ +#define CS35L33_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L33_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L33_DIAG_LOCK 0x17 /* Diagnostic Mode Register Lock */ +#define CS35L33_DIAG_CTRL_1 0x18 /* Diagnostic Mode Register Control */ +#define CS35L33_DIAG_CTRL_2 0x19 /* Diagnostic Mode Register Control 2 */ +#define CS35L33_HG_MEMLDO_CTL 0x23 /* H/G Memory/LDO CTL */ +#define CS35L33_HG_REL_RATE 0x24 /* H/G Release Rate */ +#define CS35L33_LDO_DEL 0x25 /* LDO Entry Delay/VPhg Control 1 */ +#define CS35L33_HG_HEAD 0x29 /* H/G Headroom */ +#define CS35L33_HG_EN 0x2A /* H/G Enable/VPhg CNT2 */ +#define CS35L33_TX_VMON 0x2D /* TDM TX Control 1 (VMON) */ +#define CS35L33_TX_IMON 0x2E /* TDM TX Control 2 (IMON) */ +#define CS35L33_TX_VPMON 0x2F /* TDM TX Control 3 (VPMON) */ +#define CS35L33_TX_VBSTMON 0x30 /* TDM TX Control 4 (VBSTMON) */ +#define CS35L33_TX_FLAG 0x31 /* TDM TX Control 5 (FLAG) */ +#define CS35L33_TX_EN1 0x32 /* TDM TX Enable 1 */ +#define CS35L33_TX_EN2 0x33 /* TDM TX Enable 2 */ +#define CS35L33_TX_EN3 0x34 /* TDM TX Enable 3 */ +#define CS35L33_TX_EN4 0x35 /* TDM TX Enable 4 */ +#define CS35L33_RX_AUD 0x36 /* TDM RX Control 1 */ +#define CS35L33_RX_SPLY 0x37 /* TDM RX Control 2 */ +#define CS35L33_RX_ALIVE 0x38 /* TDM RX Control 3 */ +#define CS35L33_BST_CTL4 0x39 /* Boost Converter Control 4 */ +#define CS35L33_HG_STATUS 0x3F /* H/G Status */ +#define CS35L33_MAX_REGISTER 0x59 + +#define CS35L33_MCLK_5644 5644800 +#define CS35L33_MCLK_6144 6144000 +#define CS35L33_MCLK_6 6000000 +#define CS35L33_MCLK_11289 11289600 +#define CS35L33_MCLK_12 12000000 +#define CS35L33_MCLK_12288 12288000 + +/* CS35L33_PWRCTL1 */ +#define CS35L33_PDN_AMP (1 << 7) +#define CS35L33_PDN_BST (1 << 2) +#define CS35L33_PDN_ALL 1 + +/* CS35L33_PWRCTL2 */ +#define CS35L33_PDN_VMON_SHIFT 7 +#define CS35L33_PDN_VMON (1 << CS35L33_PDN_VMON_SHIFT) +#define CS35L33_PDN_IMON_SHIFT 6 +#define CS35L33_PDN_IMON (1 << CS35L33_PDN_IMON_SHIFT) +#define CS35L33_PDN_VPMON_SHIFT 5 +#define CS35L33_PDN_VPMON (1 << CS35L33_PDN_VPMON_SHIFT) +#define CS35L33_PDN_VBSTMON_SHIFT 4 +#define CS35L33_PDN_VBSTMON (1 << CS35L33_PDN_VBSTMON_SHIFT) +#define CS35L33_SDOUT_3ST_I2S_SHIFT 3 +#define CS35L33_SDOUT_3ST_I2S (1 << CS35L33_SDOUT_3ST_I2S_SHIFT) +#define CS35L33_PDN_SDIN_SHIFT 2 +#define CS35L33_PDN_SDIN (1 << CS35L33_PDN_SDIN_SHIFT) +#define CS35L33_PDN_TDM_SHIFT 1 +#define CS35L33_PDN_TDM (1 << CS35L33_PDN_TDM_SHIFT) + +/* CS35L33_CLK_CTL */ +#define CS35L33_MCLKDIS (1 << 7) +#define CS35L33_MCLKDIV2 (1 << 6) +#define CS35L33_SDOUT_3ST_TDM (1 << 5) +#define CS35L33_INT_FS_RATE (1 << 4) +#define CS35L33_ADSP_FS 0xF + +/* CS35L33_PROTECT_CTL */ +#define CS35L33_ALIVE_WD_DIS (3 << 2) + +/* CS35L33_BST_CTL1 */ +#define CS35L33_BST_CTL_SRC (1 << 6) +#define CS35L33_BST_CTL_SHIFT (1 << 5) +#define CS35L33_BST_CTL_MASK 0x3F + +/* CS35L33_BST_CTL2 */ +#define CS35L33_TDM_WD_SEL (1 << 4) +#define CS35L33_ALIVE_WD_DIS2 (1 << 3) +#define CS35L33_VBST_SR_STEP 0x3 + +/* CS35L33_ADSP_CTL */ +#define CS35L33_ADSP_DRIVE (1 << 7) +#define CS35L33_MS_MASK (1 << 6) +#define CS35L33_SDIN_LOC (3 << 4) +#define CS35L33_ALIVE_RATE 0x3 + +/* CS35L33_ADC_CTL */ +#define CS35L33_INV_VMON (1 << 7) +#define CS35L33_INV_IMON (1 << 6) +#define CS35L33_ADC_NOTCH_DIS (1 << 5) +#define CS35L33_IMON_SCALE 0xF + +/* CS35L33_DAC_CTL */ +#define CS35L33_INV_DAC (1 << 7) +#define CS35L33_DAC_NOTCH_DIS (1 << 5) +#define CS35L33_DIGSFT (1 << 4) +#define CS35L33_DSR_RATE 0xF + +/* CS35L33_CLASSD_CTL */ +#define CS35L33_AMP_SD (1 << 6) +#define CS35L33_AMP_DRV_SEL_SRC (1 << 5) +#define CS35L33_AMP_DRV_SEL_MASK 0x10 +#define CS35L33_AMP_DRV_SEL_SHIFT 4 +#define CS35L33_AMP_CAL (1 << 3) +#define CS35L33_GAIN_CHG_ZC_MASK 0x04 +#define CS35L33_GAIN_CHG_ZC_SHIFT 2 +#define CS35L33_CLASS_D_CTL_MASK 0x3F + +/* CS35L33_AMP_CTL */ +#define CS35L33_AMP_GAIN 0xF0 +#define CS35L33_CAL_ERR_RLS (1 << 3) +#define CS35L33_AMP_SHORT_RLS (1 << 2) +#define CS35L33_OTW_RLS (1 << 1) +#define CS35L33_OTE_RLS 1 + +/* CS35L33_INT_MASK_1 */ +#define CS35L33_M_CAL_ERR_SHIFT 6 +#define CS35L33_M_CAL_ERR (1 << CS35L33_M_CAL_ERR_SHIFT) +#define CS35L33_M_ALIVE_ERR_SHIFT 5 +#define CS35L33_M_ALIVE_ERR (1 << CS35L33_M_ALIVE_ERR_SHIFT) +#define CS35L33_M_AMP_SHORT_SHIFT 2 +#define CS35L33_M_AMP_SHORT (1 << CS35L33_M_AMP_SHORT_SHIFT) +#define CS35L33_M_OTW_SHIFT 1 +#define CS35L33_M_OTW (1 << CS35L33_M_OTW_SHIFT) +#define CS35L33_M_OTE_SHIFT 0 +#define CS35L33_M_OTE (1 << CS35L33_M_OTE_SHIFT) + +/* CS35L33_INT_STATUS_1 */ +#define CS35L33_CAL_ERR (1 << 6) +#define CS35L33_ALIVE_ERR (1 << 5) +#define CS35L33_ADSPCLK_ERR (1 << 4) +#define CS35L33_MCLK_ERR (1 << 3) +#define CS35L33_AMP_SHORT (1 << 2) +#define CS35L33_OTW (1 << 1) +#define CS35L33_OTE (1 << 0) + +/* CS35L33_INT_STATUS_2 */ +#define CS35L33_VMON_OVFL (1 << 7) +#define CS35L33_IMON_OVFL (1 << 6) +#define CS35L33_VPMON_OVFL (1 << 5) +#define CS35L33_VBSTMON_OVFL (1 << 4) +#define CS35L33_PDN_DONE 1 + +/* CS35L33_BST_CTL4 */ +#define CS35L33_BST_RGS 0x70 +#define CS35L33_BST_COEFF3 0xF + +/* CS35L33_HG_MEMLDO_CTL */ +#define CS35L33_MEM_DEPTH_SHIFT 5 +#define CS35L33_MEM_DEPTH_MASK (0x3 << CS35L33_MEM_DEPTH_SHIFT) +#define CS35L33_LDO_THLD_SHIFT 1 +#define CS35L33_LDO_THLD_MASK (0xF << CS35L33_LDO_THLD_SHIFT) +#define CS35L33_LDO_DISABLE_SHIFT 0 +#define CS35L33_LDO_DISABLE_MASK (0x1 << CS35L33_LDO_DISABLE_SHIFT) + +/* CS35L33_LDO_DEL */ +#define CS35L33_VP_HG_VA_SHIFT 5 +#define CS35L33_VP_HG_VA_MASK (0x7 << CS35L33_VP_HG_VA_SHIFT) +#define CS35L33_LDO_ENTRY_DELAY_SHIFT 2 +#define CS35L33_LDO_ENTRY_DELAY_MASK (0x7 << CS35L33_LDO_ENTRY_DELAY_SHIFT) +#define CS35L33_VP_HG_RATE_SHIFT 0 +#define CS35L33_VP_HG_RATE_MASK (0x3 << CS35L33_VP_HG_RATE_SHIFT) + +/* CS35L33_HG_HEAD */ +#define CS35L33_HD_RM_SHIFT 0 +#define CS35L33_HD_RM_MASK (0x7F << CS35L33_HD_RM_SHIFT) + +/* CS35L33_HG_EN */ +#define CS35L33_CLASS_HG_ENA_SHIFT 7 +#define CS35L33_CLASS_HG_EN_MASK (0x1 << CS35L33_CLASS_HG_ENA_SHIFT) +#define CS35L33_VP_HG_AUTO_SHIFT 6 +#define CS35L33_VP_HG_AUTO_MASK (0x1 << 6) +#define CS35L33_VP_HG_SHIFT 0 +#define CS35L33_VP_HG_MASK (0x1F << CS35L33_VP_HG_SHIFT) + +#define CS35L33_RATES (SNDRV_PCM_RATE_8000_48000) +#define CS35L33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* CS35L33_{RX,TX}_X */ +#define CS35L33_X_STATE_SHIFT 7 +#define CS35L33_X_STATE (1 << CS35L33_X_STATE_SHIFT) +#define CS35L33_X_LOC_SHIFT 0 +#define CS35L33_X_LOC (0x1F << CS35L33_X_LOC_SHIFT) + +/* CS35L33_RX_AUD */ +#define CS35L33_AUDIN_RX_DEPTH_SHIFT 5 +#define CS35L33_AUDIN_RX_DEPTH (0x7 << CS35L33_AUDIN_RX_DEPTH_SHIFT) + +#endif -- 1.9.1

4 9

[alsa-devel] [PATCH] ASoC: cs35l33: Remove unnecessary free_irq call
by Axel Lin 28 Jun '16

28 Jun '16

Current code uses devm_request_threaded_irq() so it does not need to explicitly call free_irq() in .probe error path and .remove. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l33.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 55c1f75..90dc743 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -1195,7 +1195,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "Failed to enable core supplies: %d\n", ret); - goto err_irq; + return ret; } if (cs35l33->reset_gpio) @@ -1251,7 +1251,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, if (ret < 0) { dev_err(&i2c_client->dev, "%s: Register codec failed\n", __func__); - goto err_irq; + goto err_enable; } return 0; @@ -1259,8 +1259,6 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, err_enable: regulator_bulk_disable(cs35l33->num_core_supplies, cs35l33->core_supplies); -err_irq: - free_irq(i2c_client->irq, cs35l33); return ret; } @@ -1277,7 +1275,6 @@ static int cs35l33_i2c_remove(struct i2c_client *client) pm_runtime_disable(&client->dev); regulator_bulk_disable(cs35l33->num_core_supplies, cs35l33->core_supplies); - free_irq(client->irq, cs35l33); return 0; } -- 2.5.0

3 2

[alsa-devel] [PATCH] ASoC: cs35l33: Remove setting dapm->bias_level in cs35l33_set_bias_level
by Axel Lin 28 Jun '16

28 Jun '16

This is done by ASoC core now. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l33.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 90dc743..622a111 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -365,7 +365,6 @@ static int cs35l33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { unsigned int val; - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); switch (level) { @@ -392,8 +391,6 @@ static int cs35l33_set_bias_level(struct snd_soc_codec *codec, return -EINVAL; } - dapm->bias_level = level; - return 0; } -- 2.5.0

3 2

[alsa-devel] [PATCH] ASoC: fsl_spdif: fix spelling mistake: "receivce" -> "receive"
by Colin King 28 Jun '16

28 Jun '16

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 151849f..beec793 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -172,7 +172,7 @@ static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name) if (*pos >= size * 2) { *pos = 0; } else if (unlikely((*pos % size) + 3 > size)) { - dev_err(&pdev->dev, "User bit receivce buffer overflow\n"); + dev_err(&pdev->dev, "User bit receive buffer overflow\n"); return; } -- 2.8.1

2 2

[alsa-devel] [PATCH] ALSA: echoaudio: Fix memory allocation
by Christophe JAILLET 28 Jun '16

28 Jun '16

'commpage_bak' is allocated with 'sizeof(struct echoaudio)' bytes. We then copy 'sizeof(struct comm_page)' bytes in it. On my system, smatch complains because one is 2960 and the other is 3072. This would result in memory corruption or a oops. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/pci/echoaudio/echoaudio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1cb85ae..286f5e3 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2200,11 +2200,11 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL); if (commpage_bak == NULL) return -ENOMEM; commpage = chip->comm_page; - memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + memcpy(commpage_bak, commpage, sizeof(*commpage)); err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); if (err < 0) { -- 2.7.4

3 2

[alsa-devel] [alsa-lib][PATCH] ctl: use automatic variable instead of call of alloca(3)
by Takashi Sakamoto 27 Jun '16

27 Jun '16

Inner this library, layouts of all structures are public. At a compilation time, each size of the structures can be calculated. It means that we can use automatic variable instead of calling alloca(3) to program this library because in both ways storages are kept on stack frame of process VMA. Besides, the usage of automatic variables requires less instructions than calls of alloca(3). Furthermore, alloca(3) is not described in any C language standards. This commit replaces calls of alloca(3) just for structures with automatic variables, for control features. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 201 +++++++++++++++++++++++-------------------------- src/control/ctlparse.c | 7 +- src/control/namehint.c | 40 +++++----- src/control/setup.c | 5 +- src/control/tlv.c | 15 ++-- 5 files changed, 125 insertions(+), 143 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 0cb0105..5c0efc6 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -347,8 +347,8 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int member_count, long min, long max, long step) { - snd_ctl_elem_info_t *info; - snd_ctl_elem_value_t *data; + snd_ctl_elem_info_t info = {0}; + snd_ctl_elem_value_t data = {0}; unsigned int i; unsigned int j; unsigned int numid; @@ -356,38 +356,36 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_INTEGER; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; - info->value.integer.min = min; - info->value.integer.max = max; - info->value.integer.step = step; - - err = ctl->ops->element_add(ctl, info); + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_INTEGER; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; + info.value.integer.min = min; + info.value.integer.max = max; + info.value.integer.step = step; + + err = ctl->ops->element_add(ctl, &info); if (err < 0) return err; - numid = snd_ctl_elem_id_get_numid(&info->id); + numid = snd_ctl_elem_id_get_numid(&info.id); /* Set initial value to all of members in all of added elements. */ - snd_ctl_elem_value_alloca(&data); - data->id = info->id; + data.id = info.id; for (i = 0; i < element_count; i++) { - snd_ctl_elem_id_set_numid(&data->id, numid + i); + snd_ctl_elem_id_set_numid(&data.id, numid + i); for (j = 0; j < member_count; j++) - data->value.integer.value[j] = min; + data.value.integer.value[j] = min; - err = ctl->ops->element_write(ctl, data); + err = ctl->ops->element_write(ctl, &data); if (err < 0) return err; } - *id = info->id; + *id = info.id; return 0; } @@ -436,8 +434,8 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int member_count, long long min, long long max, long long step) { - snd_ctl_elem_info_t *info; - snd_ctl_elem_value_t *data; + snd_ctl_elem_info_t info = {0}; + snd_ctl_elem_value_t data = {0}; unsigned int i; unsigned int j; unsigned int numid; @@ -445,38 +443,36 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_INTEGER64; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; - info->value.integer64.min = min; - info->value.integer64.max = max; - info->value.integer64.step = step; - - err = ctl->ops->element_add(ctl, info); + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_INTEGER64; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; + info.value.integer64.min = min; + info.value.integer64.max = max; + info.value.integer64.step = step; + + err = ctl->ops->element_add(ctl, &info); if (err < 0) return err; - numid = snd_ctl_elem_id_get_numid(&info->id); + numid = snd_ctl_elem_id_get_numid(&info.id); /* Set initial value to all of members in all of added elements. */ - snd_ctl_elem_value_alloca(&data); - data->id = info->id; + data.id = info.id; for (i = 0; i < element_count; i++) { - snd_ctl_elem_id_set_numid(&data->id, numid + i); + snd_ctl_elem_id_set_numid(&data.id, numid + i); for (j = 0; j < member_count; j++) - data->value.integer64.value[j] = min; + data.value.integer64.value[j] = min; - err = ctl->ops->element_write(ctl, data); + err = ctl->ops->element_write(ctl, &data); if (err < 0) return err; } - *id = info->id; + *id = info.id; return 0; } @@ -520,25 +516,24 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int element_count, unsigned int member_count) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; int err; assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_BOOLEAN; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; - info->value.integer.min = 0; - info->value.integer.max = 1; - - err = ctl->ops->element_add(ctl, info); + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_BOOLEAN; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; + info.value.integer.min = 0; + info.value.integer.max = 1; + + err = ctl->ops->element_add(ctl, &info); if (err >= 0) - *id = info->id; + *id = info.id; return err; } @@ -590,22 +585,21 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int items, const char *const labels[]) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; unsigned int i, bytes; char *buf, *p; int err; assert(ctl && id && id->name[0] && labels); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_ENUMERATED; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; - info->value.enumerated.items = items; + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_ENUMERATED; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; + info.value.enumerated.items = items; bytes = 0; for (i = 0; i < items; ++i) @@ -615,17 +609,17 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, buf = malloc(bytes); if (buf == NULL) return -ENOMEM; - info->value.enumerated.names_ptr = (uintptr_t)buf; - info->value.enumerated.names_length = bytes; + info.value.enumerated.names_ptr = (uintptr_t)buf; + info.value.enumerated.names_length = bytes; p = buf; for (i = 0; i < items; ++i) { strcpy(p, labels[i]); p += strlen(labels[i]) + 1; } - err = ctl->ops->element_add(ctl, info); + err = ctl->ops->element_add(ctl, &info); if (err >= 0) - *id = info->id; + *id = info.id; free(buf); @@ -673,23 +667,22 @@ int snd_ctl_elem_add_bytes_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int element_count, unsigned int member_count) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; int err; assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_BYTES; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_BYTES; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; - err = ctl->ops->element_add(ctl, info); + err = ctl->ops->element_add(ctl, &info); if (err >= 0) - *id = info->id; + *id = info.id; return err; } @@ -705,12 +698,11 @@ int snd_ctl_elem_add_integer(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count, long min, long max, long step) { - snd_ctl_elem_id_t *local_id; + snd_ctl_elem_id_t local_id = {0}; - snd_ctl_elem_id_alloca(&local_id); - *local_id = *id; + local_id = *id; - return snd_ctl_elem_add_integer_set(ctl, local_id, 1, member_count, + return snd_ctl_elem_add_integer_set(ctl, &local_id, 1, member_count, min, max, step); } @@ -725,12 +717,11 @@ int snd_ctl_elem_add_integer64(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count, long long min, long long max, long long step) { - snd_ctl_elem_id_t *local_id; + snd_ctl_elem_id_t local_id = {0}; - snd_ctl_elem_id_alloca(&local_id); - *local_id = *id; + local_id = *id; - return snd_ctl_elem_add_integer64_set(ctl, local_id, 1, member_count, + return snd_ctl_elem_add_integer64_set(ctl, &local_id, 1, member_count, min, max, step); } @@ -744,12 +735,11 @@ int snd_ctl_elem_add_integer64(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, int snd_ctl_elem_add_boolean(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count) { - snd_ctl_elem_id_t *local_id; + snd_ctl_elem_id_t local_id = {0}; - snd_ctl_elem_id_alloca(&local_id); - *local_id = *id; + local_id = *id; - return snd_ctl_elem_add_boolean_set(ctl, local_id, 1, member_count); + return snd_ctl_elem_add_boolean_set(ctl, &local_id, 1, member_count); } /** @@ -765,12 +755,11 @@ int snd_ctl_elem_add_enumerated(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count, unsigned int items, const char *const labels[]) { - snd_ctl_elem_id_t *local_id; + snd_ctl_elem_id_t local_id = {0}; - snd_ctl_elem_id_alloca(&local_id); - *local_id = *id; + local_id = *id; - return snd_ctl_elem_add_enumerated_set(ctl, local_id, 1, member_count, + return snd_ctl_elem_add_enumerated_set(ctl, &local_id, 1, member_count, items, labels); } @@ -805,15 +794,15 @@ int snd_ctl_elem_add_enumerated(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, */ int snd_ctl_elem_add_iec958(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_IEC958; - info->owner = 1; - info->count = 1; - return ctl->ops->element_add(ctl, info); + + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_IEC958; + info.owner = 1; + info.count = 1; + return ctl->ops->element_add(ctl, &info); } /** diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index d38b44e..5c9a168 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -312,17 +312,16 @@ int snd_ctl_ascii_value_parse(snd_ctl_t *handle, const char *value) { const char *ptr = value; - snd_ctl_elem_id_t *myid; + snd_ctl_elem_id_t myid = {0}; snd_ctl_elem_type_t type; unsigned int idx, count; long tmp; long long tmp64; - snd_ctl_elem_id_alloca(&myid); - snd_ctl_elem_info_get_id(info, myid); + snd_ctl_elem_info_get_id(info, &myid); type = snd_ctl_elem_info_get_type(info); count = snd_ctl_elem_info_get_count(info); - snd_ctl_elem_value_set_id(dst, myid); + snd_ctl_elem_value_set_id(dst, &myid); for (idx = 0; idx < count && idx < 128 && ptr && *ptr; idx++) { if (*ptr == ',') diff --git a/src/control/namehint.c b/src/control/namehint.c index ad8dda3..4dbd216 100644 --- a/src/control/namehint.c +++ b/src/control/namehint.c @@ -97,45 +97,42 @@ static int get_dev_name1(struct hint_list *list, char **res, int device, #ifdef BUILD_HWDEP case SND_CTL_ELEM_IFACE_HWDEP: { - snd_hwdep_info_t *info; - snd_hwdep_info_alloca(&info); - snd_hwdep_info_set_device(info, device); - if (snd_ctl_hwdep_info(list->ctl, info) < 0) + snd_hwdep_info_t info = {0}; + snd_hwdep_info_set_device(&info, device); + if (snd_ctl_hwdep_info(list->ctl, &info) < 0) return 0; - *res = strdup(snd_hwdep_info_get_name(info)); + *res = strdup(snd_hwdep_info_get_name(&info)); return 0; } #endif #ifdef BUILD_PCM case SND_CTL_ELEM_IFACE_PCM: { - snd_pcm_info_t *info; - snd_pcm_info_alloca(&info); - snd_pcm_info_set_device(info, device); - snd_pcm_info_set_stream(info, stream ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK); - if (snd_ctl_pcm_info(list->ctl, info) < 0) + snd_pcm_info_t info = {0}; + snd_pcm_info_set_device(&info, device); + snd_pcm_info_set_stream(&info, stream ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK); + if (snd_ctl_pcm_info(list->ctl, &info) < 0) return 0; - switch (snd_pcm_info_get_class(info)) { + switch (snd_pcm_info_get_class(&info)) { case SND_PCM_CLASS_MODEM: case SND_PCM_CLASS_DIGITIZER: return -ENODEV; default: break; } - *res = strdup(snd_pcm_info_get_name(info)); + *res = strdup(snd_pcm_info_get_name(&info)); return 0; } #endif #ifdef BUILD_RAWMIDI case SND_CTL_ELEM_IFACE_RAWMIDI: { - snd_rawmidi_info_t *info; - snd_rawmidi_info_alloca(&info); - snd_rawmidi_info_set_device(info, device); - snd_rawmidi_info_set_stream(info, stream ? SND_RAWMIDI_STREAM_INPUT : SND_RAWMIDI_STREAM_OUTPUT); - if (snd_ctl_rawmidi_info(list->ctl, info) < 0) + snd_rawmidi_info_t info = {0}; + snd_rawmidi_info_set_device(&info, device); + snd_rawmidi_info_set_stream(&info, stream ? SND_RAWMIDI_STREAM_INPUT : SND_RAWMIDI_STREAM_OUTPUT); + if (snd_ctl_rawmidi_info(list->ctl, &info) < 0) return 0; - *res = strdup(snd_rawmidi_info_get_name(info)); + *res = strdup(snd_rawmidi_info_get_name(&info)); return 0; } #endif @@ -419,11 +416,10 @@ static int add_card(snd_config_t *config, snd_config_t *rw_config, struct hint_l snd_config_iterator_t i, next; const char *str; char ctl_name[16]; - snd_ctl_card_info_t *info; + snd_ctl_card_info_t info = {0}; int device, max_device = 0; - snd_ctl_card_info_alloca(&info); - list->info = info; + list->info = &info; err = snd_config_search(config, list->siface, &conf); if (err < 0) return err; @@ -431,7 +427,7 @@ static int add_card(snd_config_t *config, snd_config_t *rw_config, struct hint_l err = snd_ctl_open(&list->ctl, ctl_name, 0); if (err < 0) return err; - err = snd_ctl_card_info(list->ctl, info); + err = snd_ctl_card_info(list->ctl, &info); if (err < 0) goto __error; snd_config_for_each(i, next, conf) { diff --git a/src/control/setup.c b/src/control/setup.c index f23bf2c..c598c8b 100644 --- a/src/control/setup.c +++ b/src/control/setup.c @@ -252,13 +252,12 @@ static int snd_config_get_ctl_elem_value(snd_config_t *conf, { int err; snd_config_iterator_t i, next; - snd_ctl_elem_id_t *id; + snd_ctl_elem_id_t id = {0}; snd_ctl_elem_type_t type; unsigned int count; long v; long idx; - snd_ctl_elem_id_alloca(&id); - snd_ctl_elem_value_get_id(val, id); + snd_ctl_elem_value_get_id(val, &id); count = snd_ctl_elem_info_get_count(info); type = snd_ctl_elem_info_get_type(info); if (count == 1) { diff --git a/src/control/tlv.c b/src/control/tlv.c index b08d887..6ad64a1 100644 --- a/src/control/tlv.c +++ b/src/control/tlv.c @@ -409,20 +409,19 @@ struct tlv_info { static int get_tlv_info(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, struct tlv_info *rec) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; int err; - snd_ctl_elem_info_alloca(&info); - snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_info(ctl, info); + snd_ctl_elem_info_set_id(&info, id); + err = snd_ctl_elem_info(ctl, &info); if (err < 0) return err; - if (!snd_ctl_elem_info_is_tlv_readable(info)) + if (!snd_ctl_elem_info_is_tlv_readable(&info)) return -EINVAL; - if (snd_ctl_elem_info_get_type(info) != SND_CTL_ELEM_TYPE_INTEGER) + if (snd_ctl_elem_info_get_type(&info) != SND_CTL_ELEM_TYPE_INTEGER) return -EINVAL; - rec->minval = snd_ctl_elem_info_get_min(info); - rec->maxval = snd_ctl_elem_info_get_max(info); + rec->minval = snd_ctl_elem_info_get_min(&info); + rec->maxval = snd_ctl_elem_info_get_max(&info); err = snd_ctl_elem_tlv_read(ctl, id, rec->buf, sizeof(rec->buf)); if (err < 0) return err; -- 2.7.4

2 1

[alsa-devel] period time vs. buffer time
by Joël Krähemann 27 Jun '16

27 Jun '16

Hi all Do I need to call both functions in order to get good performance? snd_pcm_hw_params_set_period_time() snd_pcm_hw_params_set_buffer_time() Or is one enough? Since I call: snd_pcm_hw_params_set_buffer_size() I have developed an application called gsequencer I experience good throughput but playback collapses after a specified amount of time. The application is unusable after 60 seconds of playback. http://www.nongnu.org/gsequencer/ http://www.nongnu.org/gsequencer/api/ags/AgsThread.html - does the JIFFIEs http://www.nongnu.org/gsequencer/api/ags/AgsDevout.html - does setup audio HW http://www.nongnu.org/gsequencer/api/ags/AgsSoundcardThread.html - calls ags_soundcard_play() I do some kind of overclocking within thread frequency. I ceil() the frequency of the thread in Hertz jiffies and add one. Within this thread I call: snd_pcm_writei() In numbers: 2 audio channels 44100 samplerate 944 buffer size in samples = 2 * 944 frames = 2 * 944 * sizeof(signed short) 48 times per second call: snd_pcm_writei() I use the soundcard in non-blocking mode because I need the time to compute the rest of the tree within time. The question is how to handle lost precision? As I round values and do some overclocking to ensure alsa has enough frames to do the playback. What do I have to do in order to get great performance and throughput? Do I need to drop frames? Or even buffers? And what is alsa's job? So playback gets delay by nanosleep() as following: if(g_atomic_pointer_get(&(thread->parent)) == NULL){ gdouble time_spent, relative_time_spent; gdouble time_cycle; static const gdouble nsec_per_jiffie = NSEC_PER_SEC / AGS_THREAD_HERTZ_JIFFIE; static const gdouble lost_per_jiffie = nsec_per_jiffie / AGS_THREAD_MAX_PRECISION; gdouble lost_precision; if(thread->tic_delay == thread->delay){ struct timespec timed_sleep = { 0, 0, }; clock_gettime(CLOCK_MONOTONIC, &time_now); if(time_now.tv_sec == thread->computing_time->tv_sec + 1){ time_spent = (time_now.tv_nsec) + (NSEC_PER_SEC - thread->computing_time->tv_nsec); }else if(time_now.tv_sec > thread->computing_time->tv_sec + 1){ time_spent = (time_now.tv_sec - thread->computing_time->tv_sec) * NSEC_PER_SEC; time_spent += (time_now.tv_nsec - thread->computing_time->tv_nsec); }else{ time_spent = time_now.tv_nsec - thread->computing_time->tv_nsec; } //FIXME:JK: do optimization time_cycle = ((gdouble) NSEC_PER_SEC / thread->freq); relative_time_spent = time_cycle - time_spent; if(relative_time_spent > 0.0 && relative_time_spent < time_cycle){ timed_sleep.tv_nsec = (long) relative_time_spent; /* lost precision */ //NOTE:JK: this is experimental lost_precision = lost_per_jiffie / (1.0 + (1.0 / thread->freq)); if(relative_time_spent - lost_precision > 0.0){ timed_sleep.tv_nsec = (long) (relative_time_spent - lost_precision); }else{ g_warning("ags_thread-posix.c - not optimized tree\0"); } /* sleep the rest of available time of the period */ nanosleep(&timed_sleep, NULL); } clock_gettime(CLOCK_MONOTONIC, thread->computing_time); } Here's a piece of soundcard initialization: /* set the period time */ period_time = (USEC_PER_SEC / (devout->samplerate)); dir = -1; err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, &dir); if (err < 0) { pthread_mutex_unlock(mutex); str = snd_strerror(err); g_warning("Unable to set period time %i for playback: %s\n", period_time, str); free(str); return; } My problem with alsa is that as I don't tell the exact values I get good throughput but playback collapses. When I do exact values the sound is soften and loses volume. Strange is I can sometimes hear some crackling and I'm unsure what to set as period time. Does it need the time to do periodic things or should it be reduced or overloaded to have accurate time? Bests, Joël -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

1 0

[alsa-devel] [alsa-lib][PATCH] ctl: fix returning zero for dimension level 4
by Takashi Sakamoto 27 Jun '16

27 Jun '16

In ALSA kernel/userspace interfaces, 'struct snd_ctl_elem_info' has a 'dimension' parameter. This parameter consists of an array with four elements. Each element represents the number of members in corresponding dimension level to construct matrix. In current implementation, a get function, 'snd_ctl_elem_info_get_dimension()' is hardcoded to return zero to level 4, against actual value. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/control.c b/src/control/control.c index d515cff..0cb0105 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2483,7 +2483,7 @@ int snd_ctl_elem_info_get_dimension(const snd_ctl_elem_info_t *obj, unsigned int #endif { assert(obj); - if (idx >= 3) + if (idx > 3) return 0; return obj->dimen.d[idx]; } -- 2.7.4

2 1

[alsa-devel] [PATCH 00/39 v2][RESEND] ASoC: add simple-card-core and standardize "simple" card
by Kuninori Morimoto 27 Jun '16

27 Jun '16

Hi Mark These are resend of v2 of simple-card cleanup patches. My previous patch-set was sent almost 1month ago. Some of them are aleady accepted, but nothing happen for almost all other patches. Because of this stagnation, I already have almost 90 patches for simple-card + graph driver in locally... I would like to post them as soon as possible. OTOH, I know you don't like super big patchset, Thus, I will post small batch patch set only today. I will post next small batch patch set if 1st batch was accepted. I listed my local patch set below. It is for simple-card + graph support, and HDMI support for Renesas. This time I will post * mark patches. These are for simple-card cleanup. Kuninori Morimoto (90): * ASoC: add new simple-card-utils.c * ASoC: simple-card-utils: add asoc_simple_card_parse_tdm() * ASoC: simple-card-utils: add asoc_simple_card_parse_dailink_name() * ASoC: simple-card-utils: add asoc_simple_card_parse_card_name() * ASoC: simple-card-utils: add asoc_simple_card_parse_card_prefix() * ASoC: simple-card-utils: add asoc_simple_card_parse_clk() * ASoC: simple-card-utils: add asoc_simple_card_parse_endpoint() * ASoC: simple-card-utils: add asoc_simple_card_init_dai() * ASoC: simple-card-utils: add asoc_simple_card_canonicalize_dailink() * ASoC: simple-card-utils: add asoc_simple_card_canonicalize_cpu() * ASoC: simple-card-utils: add asoc_simple_card_clean_reference() ASoC: simple-card: use common PREFIX for each DT property ASoC: simple-card: add new asoc_simple_jack and use it ASoC: simple-card: use asoc_simple_card_parse_daifmt() ASoC: simple-card: use asoc_simple_card_parse_clk() ASoC: simple-card: use asoc_simple_card_parse_endpoint() ASoC: simple-card: use asoc_simple_card_parse_tdm() ASoC: simple-card: use asoc_simple_card_parse_card_name() ASoC: simple-card: use asoc_simple_card_parse_dailink_name() ASoC: simple-card: use asoc_simple_card_init_dai() ASoC: simple-card: use asoc_simple_card_canonicalize_dailink() ASoC: simple-card: use asoc_simple_card_canonicalize_cpu() ASoC: simple-card: use asoc_simple_card_clean_reference() ASoC: rsrc-card: use asoc_simple_card_parse_daifmt() ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name() ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai ASoC: rsrc-card: use asoc_simple_card_parse_clk() ASoC: rsrc-card: use asoc_simple_card_parse_endpoint() ASoC: rsrc-card: use asoc_simple_card_parse_card_name() ASoC: rsrc-card: use asoc_simple_card_parse_card_prefix() ASoC: rsrc-card: use asoc_simple_card_parse_tdm() ASoC: rsrc-card: use asoc_simple_card_init_dai() ASoC: rsrc-card: use asoc_simple_card_canonicalize_cpu() ASoC: rsrc-card: use asoc_simple_card_canonicalize_dailink() ASoC: rsrc-card: use asoc_simple_card_clean_reference() ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase1 ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase2 ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase3 Documentation: of: add type property of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_port/endpoint_type_is() of_graph: add of_graph_get_endpoint_count() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_top_port() of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() drm: bridge: add DesignWare HDMI I2S audio support ASoC: hdmi-codec: callback function will be called with private data ASoC: hdmi-codec: enable multi probe for same device ASoC: soc-core: snd_soc_get_dai_name() become non static ASoC: soc-core: adjust for graph on snd_soc_of_parse_card_name ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_simple_widgets ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_routing ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_prefix ASoC: simple-card-utils: rename asoc_simple_card_parse_endpoint to dai ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: simple-card-utils: adjust for graph on asoc_simple_card_parse_card_name ASoC: simple-card-utils: add asoc_simple_card_try_to_probe_graph_card() ASoC: simple-card-utils: adjust for graph on asoc_simple_card_parse_card_prefix ASoC: simple-card: call of_node_put() for dai_link ASoC: simple-card: tidyup mclk-fs of_property_read_u32() usage ASoC: simple-card: code sync: tidyup simple_priv_to_xxx() macro ASoC: simple-card: code sync: use simple_priv_to_props() macro ASoC: simple-card: is GPL v2 ASoC: simple-card: code sync: tidyup white line ASoC: simple-card: use kzalloc() for dai_props / dai_link ASoC: simple-dpcm-card: tidyup codec daifmt handling method ASoC: simple-dpcm-card: tidyup asoc_simple_card_parse_links() method ASoC: simple-dpcm-card: depends on CONFIG_OF ASoC: simple-dpcm-card: add 2 CPU 1 Codec example ASoC: simple-dpcm-card: code sync: follow to simple family style ASoC: simple-dpcm-card: is GPL v2 ASoC: simple-dpcm-card: code sync: move asoc_simple_card_of_match ASoC: simple-dpcm-card: remove unused definitions ASoC: simple-dpcm-card: code sync: rename asoc_simple_card_priv ASoC: simple-dpcm-card: code sync: tidyup props/link naming ASoC: add simple-graph-card document ASoC: add simple-graph-card support ASoC: add simple-graph-dpcm-card document ASoC: add simple-graph-dpcm-card support ASoC: rsnd: remove duplicate define of rsnd_dvc_of_node() ASoC: rsnd: amend .probe/.remove call for DPCM ASoC: rsnd: add rsnd_mod_next() for for_each_rsnd_mod_xxx() ASoC: rsnd: use for_each_rsnd_mod_xxx() on rsnd_dai_call() ASoC: rsnd: use for_each_rsnd_mod_xxx() on rsnd_rdai_continuance_probe() ASoC: rsnd: add rsnd_parse_of_node() and integrate rsnd_xxx_of_node ASoC: rsnd: add support graph base DT phase 1 ASoC: rsnd: add support graph base DT phase 2 ASoC: rsnd: depends on OF ASoC: rsnd: use simple-graph-card if possible ASoC: rsnd: add HDMI output support Documentation/devicetree/bindings/graph.txt | 26 + .../devicetree/bindings/sound/renesas,rsnd.txt | 2 + .../bindings/sound/renesas,rsrc-card.txt | 75 --- .../devicetree/bindings/sound/simple-dpcm-card.txt | 109 +++++ .../bindings/sound/simple-graph-card.txt | 65 +++ .../bindings/sound/simple-graph-dpcm-card.txt | 53 +++ drivers/gpu/drm/bridge/Kconfig | 8 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 + drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 123 +++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 +- drivers/gpu/drm/bridge/dw-hdmi.h | 21 + drivers/of/base.c | 197 +++++++- include/linux/of_graph.h | 67 +++ include/sound/hdmi-codec.h | 13 +- include/sound/simple_card.h | 11 +- include/sound/simple_card_utils.h | 92 ++++ include/sound/soc.h | 31 +- sound/soc/codecs/hdmi-codec.c | 81 +++- sound/soc/generic/Kconfig | 25 + sound/soc/generic/Makefile | 12 +- sound/soc/generic/simple-card-utils.c | 387 +++++++++++++++ sound/soc/generic/simple-card.c | 501 +++++++------------- sound/soc/generic/simple-dpcm-card.c | 323 +++++++++++++ sound/soc/generic/simple-graph-card.c | 452 ++++++++++++++++++ sound/soc/generic/simple-graph-dpcm-card.c | 409 ++++++++++++++++ sound/soc/sh/Kconfig | 8 +- sound/soc/sh/rcar/Makefile | 3 - sound/soc/sh/rcar/core.c | 318 ++++++++----- sound/soc/sh/rcar/dma.c | 11 + sound/soc/sh/rcar/dvc.c | 2 - sound/soc/sh/rcar/gen.c | 2 + sound/soc/sh/rcar/rsnd.h | 62 ++- sound/soc/sh/rcar/rsrc-card.c | 527 --------------------- sound/soc/sh/rcar/ssi.c | 27 +- sound/soc/sh/rcar/ssiu.c | 37 ++ sound/soc/soc-core.c | 39 +- 37 files changed, 3003 insertions(+), 1146 deletions(-)

3 16

[alsa-devel] [PATCH] ASOC: wm8753: Remove unneeded header file
by Amitoj Kaur Chawla 26 Jun '16

26 Jun '16

Drop redundant include of moduleparam.h The Coccinelle semantic patch used to make this change is as follows: @ includesmodule @ @@ #include <linux/module.h> @ depends on includesmodule @ @@ - #include <linux/moduleparam.h> Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/soc/codecs/wm8753.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6f1024f..b4e6893 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -32,7 +32,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -- 1.9.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: codecs: wm8753: Replace magic number
by Alexander Shiyan 26 Jun '16

26 Jun '16

Use SND_SOC_NOPM constant, instead of -1. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/wm8753.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6f1024f..9f851b8 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -486,7 +486,7 @@ SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", WM8753_PWR1, 4, 0), SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_MUX("Mono 2 Mux", SND_SOC_NOPM, 0, 0, &wm8753_mono2_controls), SND_SOC_DAPM_OUTPUT("MONO2"), -SND_SOC_DAPM_MIXER("Out3 Left + Right", -1, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Out3 Left + Right", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out3_controls), SND_SOC_DAPM_PGA("Out 3", WM8753_PWR3, 4, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("OUT3"), -- 2.4.9

2 2

[alsa-devel] [PATCH] SoC: dwc: trivial fix of spelling mistake "unsuppted" -> "unsupported"
by Colin King 26 Jun '16

26 Jun '16

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/dwc/designware_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 0db69b7..1223ba6 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -278,7 +278,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, break; default: - dev_err(dev->dev, "designware-i2s: unsuppted PCM fmt"); + dev_err(dev->dev, "designware-i2s: unsupported PCM fmt"); return -EINVAL; } -- 2.8.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: tpa6130a2: Add support for multiple instances
by Helen Koike 26 Jun '16

26 Jun '16

The current tpa6130a2 driver supports only a single instance. This patch series add support for multiple instances by removing the global variable that holds the instance. This is performed by using the component API, regmap, the snd_soc_{info,put,get}_volsw API and DAPM. This patch series also touches code from the Nokia RX51 which I didn't tested (as I am testing the tpa6130a2 in another board that is not upstream). I would appreciate is if someone who possesses the Nokia RX51 (n900) could please test the code. This patch series is based on git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Available at https://git.collabora.com/cgit/user/koike/linux.git/log/?h=sound/review/tpa… Helen Koike (4): ASoC: tpa6130a2: Register component ASoC: tap6130a2: Use regmap ASoC: tpa6130a2: Use snd soc volsw functions ASoC: tpa6130a2: Add DAPM support sound/soc/codecs/tpa6130a2.c | 382 ++++++++++++------------------------------- sound/soc/codecs/tpa6130a2.h | 14 +- sound/soc/omap/rx51.c | 45 ++--- 3 files changed, 130 insertions(+), 311 deletions(-) -- 1.9.1

5 35

[alsa-devel] [PATCH] ASoC: Samsung: s3c-i2s-v2: fix spelling mistake: "unknwon" -> "unknown"
by Colin King 26 Jun '16

26 Jun '16

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in pr_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/samsung/s3c-i2s-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index b6ab3fc..bf8ae79 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -268,7 +268,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod &= ~S3C2412_IISMOD_SLAVE; break; default: - pr_err("unknwon master/slave format\n"); + pr_err("unknown master/slave format\n"); return -EINVAL; } -- 2.8.1

2 1

[alsa-devel] [PATCH v4 0/2] ASoC: da7219: Convert driver to use generic FW functions
by Adam Thomson 26 Jun '16

26 Jun '16

This patch set converts the da7219 codec driver to use device/fwnode functions to access properties, instead of the DT only of_* functions, allowing ACPI to be used as well. The DT bindings for da7219 have a device node for the main codec properties, and a named child node (da7219_aad), which contains all of the accessory detection related properties for the device. ACPI also supports this kind of FW hierarchy (data only sub-nodes), but some support in the kernel needs to be added to take make use of this in driver code. The first patch adds functions to allow searching for a named child node of a device, for both DT and ACPI, and the second patch updates the codec driver to use the standard device/fwnode calls, including this new function. These changes are based on the v4.7-rc4 kernel. Changes in v4: - Rebase to v4.7-rc4 - Use strcmp() in acpi_data_node_match() for matching ACPI data nodes. Changes in v3: - Use of_node_cmp() in device_get_named_child_node() to match DT node. Changes in v2: - Rebase to v4.7-rc1 - Small updates to codec patch based on previous reviewer comments Adam Thomson (2): device property: Add function to search for named child of device ASoC: da7219: Convert driver to use generic device/fwnode functions drivers/base/property.c | 28 ++++++++++++ include/acpi/acpi_bus.h | 7 +++ include/linux/acpi.h | 6 +++ include/linux/of.h | 14 +++--- include/linux/property.h | 3 ++ sound/soc/codecs/da7219-aad.c | 103 +++++++++++++++++++++--------------------- sound/soc/codecs/da7219.c | 34 +++++++------- 7 files changed, 119 insertions(+), 76 deletions(-) -- 1.9.3

5 9

[alsa-devel] [PATCH] ALSA: hda - fix read before array start
by Bob Copeland 26 Jun '16

26 Jun '16

UBSAN reports the following warning from accessing path->path[-1] in set_path_power(): [ 16.078040] ================================================================================ [ 16.078124] UBSAN: Undefined behaviour in sound/pci/hda/hda_generic.c:3981:17 [ 16.078198] index -1 is out of range for type 'hda_nid_t [10]' [ 16.078270] CPU: 2 PID: 1738 Comm: modprobe Not tainted 4.7.0-rc1-wt+ #47 [ 16.078274] Hardware name: LENOVO 3443CTO/3443CTO, BIOS G6ET23WW (1.02 ) 08/14/2012 [ 16.078278] ffff8800cb246000 ffff8800cb3638b8 ffffffff815c4fe3 0000000000000032 [ 16.078286] ffff8800cb3638e0 ffffffffffffffff ffff8800cb3638d0 ffffffff8162443d [ 16.078294] ffffffffa0894200 ffff8800cb363920 ffffffff81624af7 0000000000000292 [ 16.078302] Call Trace: [ 16.078311] [<ffffffff815c4fe3>] dump_stack+0x86/0xd3 [ 16.078317] [<ffffffff8162443d>] ubsan_epilogue+0xd/0x40 [ 16.078324] [<ffffffff81624af7>] __ubsan_handle_out_of_bounds+0x67/0x70 [ 16.078335] [<ffffffffa087665f>] set_path_power+0x1bf/0x230 [snd_hda_codec_generic] [ 16.078344] [<ffffffffa087880d>] add_pin_power_ctls+0x8d/0xc0 [snd_hda_codec_generic] [ 16.078352] [<ffffffffa087f190>] ? pin_power_down_callback+0x20/0x20 [snd_hda_codec_generic] [ 16.078360] [<ffffffffa0878947>] add_all_pin_power_ctls+0x107/0x150 [snd_hda_codec_generic] [ 16.078370] [<ffffffffa08842b3>] snd_hda_gen_parse_auto_config+0x2d73/0x49e0 [snd_hda_codec_generic] [ 16.078376] [<ffffffff81173360>] ? trace_hardirqs_on_caller+0x1b0/0x2c0 [ 16.078390] [<ffffffffa089df27>] alc_parse_auto_config+0x147/0x310 [snd_hda_codec_realtek] [ 16.078402] [<ffffffffa08a332a>] patch_alc269+0x23a/0x560 [snd_hda_codec_realtek] [ 16.078417] [<ffffffffa0838644>] hda_codec_driver_probe+0xa4/0x1a0 [snd_hda_codec] [ 16.078424] [<ffffffff817bbac1>] driver_probe_device+0x101/0x380 [ 16.078430] [<ffffffff817bbdf9>] __driver_attach+0xb9/0x100 [ 16.078438] [<ffffffff817bbd40>] ? driver_probe_device+0x380/0x380 [ 16.078444] [<ffffffff817b8d20>] bus_for_each_dev+0x70/0xc0 [ 16.078449] [<ffffffff817bb087>] driver_attach+0x27/0x50 [ 16.078454] [<ffffffff817ba956>] bus_add_driver+0x166/0x2c0 [ 16.078460] [<ffffffffa0369000>] ? 0xffffffffa0369000 [ 16.078465] [<ffffffff817bd13d>] driver_register+0x7d/0x130 [ 16.078477] [<ffffffffa083816f>] __hda_codec_driver_register+0x6f/0x90 [snd_hda_codec] [ 16.078488] [<ffffffffa036901e>] realtek_driver_init+0x1e/0x1000 [snd_hda_codec_realtek] [ 16.078493] [<ffffffff8100215e>] do_one_initcall+0x4e/0x1d0 [ 16.078499] [<ffffffff8119f54d>] ? rcu_read_lock_sched_held+0x6d/0x80 [ 16.078504] [<ffffffff813701b1>] ? kmem_cache_alloc_trace+0x391/0x560 [ 16.078510] [<ffffffff812bb314>] ? do_init_module+0x28/0x273 [ 16.078515] [<ffffffff812bb387>] do_init_module+0x9b/0x273 [ 16.078522] [<ffffffff811e3782>] load_module+0x20b2/0x3410 [ 16.078527] [<ffffffff811df140>] ? m_show+0x210/0x210 [ 16.078533] [<ffffffff813b2b26>] ? kernel_read+0x66/0xe0 [ 16.078541] [<ffffffff811e4cfa>] SYSC_finit_module+0xba/0xc0 [ 16.078547] [<ffffffff811e4d1e>] SyS_finit_module+0xe/0x10 [ 16.078552] [<ffffffff81a860fc>] entry_SYSCALL_64_fastpath+0x1f/0xbd [ 16.078556] ================================================================================ Fix by checking path->depth before use. Signed-off-by: Bob Copeland <me(a)bobcopeland.com> --- sound/pci/hda/hda_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 320445f3bf73..79c7b340acc2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3977,6 +3977,8 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, for (n = 0; n < spec->paths.used; n++) { path = snd_array_elem(&spec->paths, n); + if (!path->depth) + continue; if (path->path[0] == nid || path->path[path->depth - 1] == nid) { bool pin_old = path->pin_enabled; -- 2.6.1

2 1

[alsa-devel] UBSAN bug in hda_generic.c
by Bob Copeland 25 Jun '16

25 Jun '16

Hi, I have UBSAN reporting an out-of-bounds array access (see below) on my machine. The following patch fixes the warning for me, but not sure if that is just papering over some other bug. Thanks in advance for looking! >From 551e904f7a7aea9e9c03c439e554100643239c5c Mon Sep 17 00:00:00 2001 From: Bob Copeland <me(a)bobcopeland.com> Date: Sat, 25 Jun 2016 06:53:44 -0400 Subject: [PATCH] ALSA: hda - fix read before array start UBSAN reports the following warning from accessing path->path[-1] in set_path_power(): [ 16.078040] ================================================================================ [ 16.078124] UBSAN: Undefined behaviour in sound/pci/hda/hda_generic.c:3981:17 [ 16.078198] index -1 is out of range for type 'hda_nid_t [10]' [ 16.078270] CPU: 2 PID: 1738 Comm: modprobe Not tainted 4.7.0-rc1-wt+ #47 [ 16.078274] Hardware name: LENOVO 3443CTO/3443CTO, BIOS G6ET23WW (1.02 ) 08/14/2012 [ 16.078278] ffff8800cb246000 ffff8800cb3638b8 ffffffff815c4fe3 0000000000000032 [ 16.078286] ffff8800cb3638e0 ffffffffffffffff ffff8800cb3638d0 ffffffff8162443d [ 16.078294] ffffffffa0894200 ffff8800cb363920 ffffffff81624af7 0000000000000292 [ 16.078302] Call Trace: [ 16.078311] [<ffffffff815c4fe3>] dump_stack+0x86/0xd3 [ 16.078317] [<ffffffff8162443d>] ubsan_epilogue+0xd/0x40 [ 16.078324] [<ffffffff81624af7>] __ubsan_handle_out_of_bounds+0x67/0x70 [ 16.078335] [<ffffffffa087665f>] set_path_power+0x1bf/0x230 [snd_hda_codec_generic] [ 16.078344] [<ffffffffa087880d>] add_pin_power_ctls+0x8d/0xc0 [snd_hda_codec_generic] [ 16.078352] [<ffffffffa087f190>] ? pin_power_down_callback+0x20/0x20 [snd_hda_codec_generic] [ 16.078360] [<ffffffffa0878947>] add_all_pin_power_ctls+0x107/0x150 [snd_hda_codec_generic] [ 16.078370] [<ffffffffa08842b3>] snd_hda_gen_parse_auto_config+0x2d73/0x49e0 [snd_hda_codec_generic] [ 16.078376] [<ffffffff81173360>] ? trace_hardirqs_on_caller+0x1b0/0x2c0 [ 16.078390] [<ffffffffa089df27>] alc_parse_auto_config+0x147/0x310 [snd_hda_codec_realtek] [ 16.078402] [<ffffffffa08a332a>] patch_alc269+0x23a/0x560 [snd_hda_codec_realtek] [ 16.078417] [<ffffffffa0838644>] hda_codec_driver_probe+0xa4/0x1a0 [snd_hda_codec] [ 16.078424] [<ffffffff817bbac1>] driver_probe_device+0x101/0x380 [ 16.078430] [<ffffffff817bbdf9>] __driver_attach+0xb9/0x100 [ 16.078438] [<ffffffff817bbd40>] ? driver_probe_device+0x380/0x380 [ 16.078444] [<ffffffff817b8d20>] bus_for_each_dev+0x70/0xc0 [ 16.078449] [<ffffffff817bb087>] driver_attach+0x27/0x50 [ 16.078454] [<ffffffff817ba956>] bus_add_driver+0x166/0x2c0 [ 16.078460] [<ffffffffa0369000>] ? 0xffffffffa0369000 [ 16.078465] [<ffffffff817bd13d>] driver_register+0x7d/0x130 [ 16.078477] [<ffffffffa083816f>] __hda_codec_driver_register+0x6f/0x90 [snd_hda_codec] [ 16.078488] [<ffffffffa036901e>] realtek_driver_init+0x1e/0x1000 [snd_hda_codec_realtek] [ 16.078493] [<ffffffff8100215e>] do_one_initcall+0x4e/0x1d0 [ 16.078499] [<ffffffff8119f54d>] ? rcu_read_lock_sched_held+0x6d/0x80 [ 16.078504] [<ffffffff813701b1>] ? kmem_cache_alloc_trace+0x391/0x560 [ 16.078510] [<ffffffff812bb314>] ? do_init_module+0x28/0x273 [ 16.078515] [<ffffffff812bb387>] do_init_module+0x9b/0x273 [ 16.078522] [<ffffffff811e3782>] load_module+0x20b2/0x3410 [ 16.078527] [<ffffffff811df140>] ? m_show+0x210/0x210 [ 16.078533] [<ffffffff813b2b26>] ? kernel_read+0x66/0xe0 [ 16.078541] [<ffffffff811e4cfa>] SYSC_finit_module+0xba/0xc0 [ 16.078547] [<ffffffff811e4d1e>] SyS_finit_module+0xe/0x10 [ 16.078552] [<ffffffff81a860fc>] entry_SYSCALL_64_fastpath+0x1f/0xbd [ 16.078556] ================================================================================ Fix by checking path->depth before use. Signed-off-by: Bob Copeland <me(a)bobcopeland.com> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 320445f3bf73..bc23a9d8e7b3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3978,7 +3978,7 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, for (n = 0; n < spec->paths.used; n++) { path = snd_array_elem(&spec->paths, n); if (path->path[0] == nid || - path->path[path->depth - 1] == nid) { + (path->depth > 0 && path->path[path->depth - 1] == nid)) { bool pin_old = path->pin_enabled; bool stream_old = path->stream_enabled; -- 2.6.1 -- Bob Copeland %% http://bobcopeland.com/

2 2

[alsa-devel] snd_pcm_open() leaks error
by Joël Krähemann 25 Jun '16

25 Jun '16

Hi all I have some issues as opening soundcard multiple times the error is handled but it leaks memory: http://pastebin.com/aQfAQV2q The problematic function is snd_pcm_open() but I think it doesn't have any performance impact on ̀gsequencer` http://www.nongnu.org/gsequencer/ Bests, Joël -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

1 0

[alsa-devel] [PATCH v8 1/2] ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.
by Paul.Handrigan＠cirrus.com 24 Jun '16

24 Jun '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l33 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l33.h | 48 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l33.c | 1314 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l33.h | 221 ++++++++ 5 files changed, 1590 insertions(+) create mode 100644 include/sound/cs35l33.h create mode 100644 sound/soc/codecs/cs35l33.c create mode 100644 sound/soc/codecs/cs35l33.h diff --git a/include/sound/cs35l33.h b/include/sound/cs35l33.h new file mode 100644 index 0000000..b6eadce --- /dev/null +++ b/include/sound/cs35l33.h @@ -0,0 +1,48 @@ +/* + * linux/sound/cs35l33.h -- Platform data for CS35l33 + * + * Copyright (c) 2016 Cirrus Logic Inc. + * + * 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 __CS35L33_H +#define __CS35L33_H + +struct cs35l33_hg { + bool enable_hg_algo; + unsigned int mem_depth; + unsigned int release_rate; + unsigned int hd_rm; + unsigned int ldo_thld; + unsigned int ldo_path_disable; + unsigned int ldo_entry_delay; + bool vp_hg_auto; + unsigned int vp_hg; + unsigned int vp_hg_rate; + unsigned int vp_hg_va; +}; + +struct cs35l33_pdata { + /* Boost Controller Voltage Setting */ + unsigned int boost_ctl; + + /* Boost Controller Peak Current */ + unsigned int boost_ipk; + + /* Amplifier Drive Select */ + unsigned int amp_drv_sel; + + /* soft volume ramp */ + unsigned int ramp_rate; + + /* IMON adc scale */ + unsigned int imon_adc_scale; + + /* H/G algo configuration */ + struct cs35l33_hg hg_config; +}; + +#endif /* __CS35L33_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 60f8f13..02d2084 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C + select SND_SOC_CS35L33 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -391,6 +392,10 @@ config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C +config SND_SOC_CS35L33 + tristate "Cirrus Logic CS35L33 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1264fa4..abf2126 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -37,6 +37,7 @@ snd-soc-arizona-objs := arizona.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o +snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -257,6 +258,7 @@ obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o +obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c new file mode 100644 index 0000000..55c1f75 --- /dev/null +++ b/sound/soc/codecs/cs35l33.c @@ -0,0 +1,1314 @@ +/* + * cs35l33.c -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.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/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.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 <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <sound/cs35l33.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/of_gpio.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> + +#include "cs35l33.h" + +#define CS35L33_BOOT_DELAY 50 + +struct cs35l33_private { + struct snd_soc_codec *codec; + struct cs35l33_pdata pdata; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + bool amp_cal; + int mclk_int; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + bool is_tdm_mode; + bool enable_soft_ramp; +}; + +static const struct reg_default cs35l33_reg[] = { + {CS35L33_PWRCTL1, 0x85}, + {CS35L33_PWRCTL2, 0xFE}, + {CS35L33_CLK_CTL, 0x0C}, + {CS35L33_BST_PEAK_CTL, 0x90}, + {CS35L33_PROTECT_CTL, 0x55}, + {CS35L33_BST_CTL1, 0x00}, + {CS35L33_BST_CTL2, 0x01}, + {CS35L33_ADSP_CTL, 0x00}, + {CS35L33_ADC_CTL, 0xC8}, + {CS35L33_DAC_CTL, 0x14}, + {CS35L33_DIG_VOL_CTL, 0x00}, + {CS35L33_CLASSD_CTL, 0x04}, + {CS35L33_AMP_CTL, 0x90}, + {CS35L33_INT_MASK_1, 0xFF}, + {CS35L33_INT_MASK_2, 0xFF}, + {CS35L33_DIAG_LOCK, 0x00}, + {CS35L33_DIAG_CTRL_1, 0x40}, + {CS35L33_DIAG_CTRL_2, 0x00}, + {CS35L33_HG_MEMLDO_CTL, 0x62}, + {CS35L33_HG_REL_RATE, 0x03}, + {CS35L33_LDO_DEL, 0x12}, + {CS35L33_HG_HEAD, 0x0A}, + {CS35L33_HG_EN, 0x05}, + {CS35L33_TX_VMON, 0x00}, + {CS35L33_TX_IMON, 0x03}, + {CS35L33_TX_VPMON, 0x02}, + {CS35L33_TX_VBSTMON, 0x05}, + {CS35L33_TX_FLAG, 0x06}, + {CS35L33_TX_EN1, 0x00}, + {CS35L33_TX_EN2, 0x00}, + {CS35L33_TX_EN3, 0x00}, + {CS35L33_TX_EN4, 0x00}, + {CS35L33_RX_AUD, 0x40}, + {CS35L33_RX_SPLY, 0x03}, + {CS35L33_RX_ALIVE, 0x04}, + {CS35L33_BST_CTL4, 0x63}, +}; + +static const struct reg_sequence cs35l33_patch[] = { + { 0x00, 0x99, 0 }, + { 0x59, 0x02, 0 }, + { 0x52, 0x30, 0 }, + { 0x39, 0x45, 0 }, + { 0x57, 0x30, 0 }, + { 0x2C, 0x68, 0 }, + { 0x00, 0x00, 0 }, +}; + +static bool cs35l33_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l33_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + /* these are read only registers */ + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return false; + default: + return true; + } +} + +static bool cs35l33_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_PWRCTL1: + case CS35L33_PWRCTL2: + case CS35L33_CLK_CTL: + case CS35L33_BST_PEAK_CTL: + case CS35L33_PROTECT_CTL: + case CS35L33_BST_CTL1: + case CS35L33_BST_CTL2: + case CS35L33_ADSP_CTL: + case CS35L33_ADC_CTL: + case CS35L33_DAC_CTL: + case CS35L33_DIG_VOL_CTL: + case CS35L33_CLASSD_CTL: + case CS35L33_AMP_CTL: + case CS35L33_INT_MASK_1: + case CS35L33_INT_MASK_2: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_DIAG_LOCK: + case CS35L33_DIAG_CTRL_1: + case CS35L33_DIAG_CTRL_2: + case CS35L33_HG_MEMLDO_CTL: + case CS35L33_HG_REL_RATE: + case CS35L33_LDO_DEL: + case CS35L33_HG_HEAD: + case CS35L33_HG_EN: + case CS35L33_TX_VMON: + case CS35L33_TX_IMON: + case CS35L33_TX_VPMON: + case CS35L33_TX_VBSTMON: + case CS35L33_TX_FLAG: + case CS35L33_TX_EN1: + case CS35L33_TX_EN2: + case CS35L33_TX_EN3: + case CS35L33_TX_EN4: + case CS35L33_RX_AUD: + case CS35L33_RX_SPLY: + case CS35L33_RX_ALIVE: + case CS35L33_BST_CTL4: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l33_snd_controls[] = { + + SOC_SINGLE_TLV("SPK Amp Volume", CS35L33_AMP_CTL, + 4, 0x09, 0, classd_ctl_tlv), + SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, + 0, 0x34, 0xE4, dac_tlv), +}; + +static int cs35l33_spkrdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!priv->amp_cal) { + usleep_range(8000, 9000); + priv->amp_cal = true; + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, 0); + dev_dbg(codec->dev, "Amp calibration done\n"); + } + dev_dbg(codec->dev, "Amp turned on\n"); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "Amp turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + break; + } + + return 0; +} + +static int cs35l33_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, 0); + val = priv->is_tdm_mode ? 0 : CS35L33_PDN_TDM; + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, val); + dev_dbg(codec->dev, "BST turned on\n"); + break; + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "SDIN turned on\n"); + if (!priv->amp_cal) { + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, CS35L33_AMP_CAL); + dev_dbg(codec->dev, "Amp calibration started\n"); + usleep_range(10000, 11000); + } + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, CS35L33_PDN_TDM); + usleep_range(4000, 4100); + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, CS35L33_PDN_BST); + dev_dbg(codec->dev, "BST and SDIN turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + + } + + return 0; +} + +static int cs35l33_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int mask = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + unsigned int mask2 = CS35L33_SDOUT_3ST_TDM; + unsigned int val, val2; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->is_tdm_mode) { + /* set sdout_3st_i2s and reset pdn_tdm */ + val = CS35L33_SDOUT_3ST_I2S; + /* reset sdout_3st_tdm */ + val2 = 0; + } else { + /* reset sdout_3st_i2s and set pdn_tdm */ + val = CS35L33_PDN_TDM; + /* set sdout_3st_tdm */ + val2 = CS35L33_SDOUT_3ST_TDM; + } + dev_dbg(codec->dev, "SDOUT turned on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + val = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + val2 = CS35L33_SDOUT_3ST_TDM; + dev_dbg(codec->dev, "SDOUT turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return 0; + } + + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + mask, val); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + mask2, val2); + + return 0; +} + +static const struct snd_soc_dapm_widget cs35l33_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUT_DRV_E("SPKDRV", CS35L33_PWRCTL1, 7, 1, NULL, 0, + cs35l33_spkrdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L33_PWRCTL2, + 2, 1, cs35l33_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("MON"), + + SND_SOC_DAPM_ADC("VMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VMON_SHIFT, 1), + SND_SOC_DAPM_ADC("IMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_IMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VPMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VPMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VBSTMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VBSTMON_SHIFT, 1), + + SND_SOC_DAPM_AIF_OUT_E("SDOUT", NULL, 0, SND_SOC_NOPM, 0, 0, + cs35l33_sdout_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l33_audio_map[] = { + {"SDIN", NULL, "CS35L33 Playback"}, + {"SPKDRV", NULL, "SDIN"}, + {"SPK", NULL, "SPKDRV"}, + + {"VMON", NULL, "MON"}, + {"IMON", NULL, "MON"}, + + {"SDOUT", NULL, "VMON"}, + {"SDOUT", NULL, "IMON"}, + {"CS35L33 Capture", NULL, "SDOUT"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vphg_auto_route[] = { + {"SPKDRV", NULL, "VPMON"}, + {"VPMON", NULL, "CS35L33 Playback"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vp_vbst_mon_route[] = { + {"SDOUT", NULL, "VPMON"}, + {"VPMON", NULL, "MON"}, + {"SDOUT", NULL, "VBSTMON"}, + {"VBSTMON", NULL, "MON"}, +}; + +static int cs35l33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int val; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, 0); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, CS35L33_PDN_ALL); + regmap_read(priv->regmap, CS35L33_INT_STATUS_2, &val); + usleep_range(1000, 1100); + if (val & CS35L33_PDN_DONE) + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, CS35L33_MCLKDIS); + break; + case SND_SOC_BIAS_OFF: + break; + default: + return -EINVAL; + } + + dapm->bias_level = level; + + return 0; +} + +struct cs35l33_mclk_div { + int mclk; + int srate; + u8 adsp_rate; + u8 int_fs_ratio; +}; + +static const struct cs35l33_mclk_div cs35l33_mclk_coeffs[] = { + /* MCLK, Sample Rate, adsp_rate, int_fs_ratio */ + {5644800, 11025, 0x4, CS35L33_INT_FS_RATE}, + {5644800, 22050, 0x8, CS35L33_INT_FS_RATE}, + {5644800, 44100, 0xC, CS35L33_INT_FS_RATE}, + + {6000000, 8000, 0x1, 0}, + {6000000, 11025, 0x2, 0}, + {6000000, 11029, 0x3, 0}, + {6000000, 12000, 0x4, 0}, + {6000000, 16000, 0x5, 0}, + {6000000, 22050, 0x6, 0}, + {6000000, 22059, 0x7, 0}, + {6000000, 24000, 0x8, 0}, + {6000000, 32000, 0x9, 0}, + {6000000, 44100, 0xA, 0}, + {6000000, 44118, 0xB, 0}, + {6000000, 48000, 0xC, 0}, + + {6144000, 8000, 0x1, CS35L33_INT_FS_RATE}, + {6144000, 12000, 0x4, CS35L33_INT_FS_RATE}, + {6144000, 16000, 0x5, CS35L33_INT_FS_RATE}, + {6144000, 24000, 0x8, CS35L33_INT_FS_RATE}, + {6144000, 32000, 0x9, CS35L33_INT_FS_RATE}, + {6144000, 48000, 0xC, CS35L33_INT_FS_RATE}, +}; + +static int cs35l33_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l33_mclk_coeffs); i++) { + if (cs35l33_mclk_coeffs[i].mclk == mclk && + cs35l33_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, CS35L33_MS_MASK); + dev_dbg(codec->dev, "Audio port in master mode\n"); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, 0); + dev_dbg(codec->dev, "Audio port in slave mode\n"); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + /* + * tdm mode in cs35l33 resembles dsp-a mode very + * closely, it is dsp-a with fsync shifted left by half bclk + */ + priv->is_tdm_mode = true; + dev_dbg(codec->dev, "Audio port in TDM mode\n"); + break; + case SND_SOC_DAIFMT_I2S: + priv->is_tdm_mode = false; + dev_dbg(codec->dev, "Audio port in I2S mode\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l33_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + int sample_size = params_width(params); + int coeff = cs35l33_get_mclk_coeff(priv->mclk_int, params_rate(params)); + + if (coeff < 0) + return coeff; + + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_ADSP_FS | CS35L33_INT_FS_RATE, + cs35l33_mclk_coeffs[coeff].int_fs_ratio + | cs35l33_mclk_coeffs[coeff].adsp_rate); + + if (priv->is_tdm_mode) { + sample_size = (sample_size / 8) - 1; + if (sample_size > 2) + sample_size = 2; + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_AUDIN_RX_DEPTH, + sample_size << CS35L33_AUDIN_RX_DEPTH_SHIFT); + } + + dev_dbg(codec->dev, "sample rate=%d, bits per sample=%d\n", + params_rate(params), params_width(params)); + + return 0; +} + +static const unsigned int cs35l33_src_rates[] = { + 8000, 11025, 11029, 12000, 16000, 22050, + 22059, 24000, 32000, 44100, 44118, 48000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l33_constraints = { + .count = ARRAY_SIZE(cs35l33_src_rates), + .list = cs35l33_src_rates, +}; + +static int cs35l33_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l33_constraints); + return 0; +} + +static int cs35l33_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (tristate) { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, CS35L33_SDOUT_3ST_I2S); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, CS35L33_SDOUT_3ST_TDM); + } else { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, 0); + } + + return 0; +} + +static int cs35l33_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos, i; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) { + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "Audio starts from slots %d", slot); + } + + /* + * scan tx_mask: vmon(2 slots); imon (2 slots); + * vpmon (1 slot) vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + for (i = 0; i < 2 ; i++) { + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON + i, + CS35L33_X_STATE | CS35L33_X_LOC, CS35L33_X_STATE + | CS35L33_X_LOC); + } + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route, + ARRAY_SIZE(cs35l33_vp_vbst_mon_route)); + + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) { + regmap_update_bits(priv->regmap, CS35L33_TX_VMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "VMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure IMON_TX_LOC */ + if (slot_num == 3) { + regmap_update_bits(priv->regmap, CS35L33_TX_IMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "IMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure VPMON_TX_LOC */ + if (slot_num == 4) { + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[0], 2); + dev_dbg(codec->dev, "VPMON enabled in slots %d", slot); + } + + /* configure VBSTMON_TX_LOC */ + if (slot_num == 5) { + regmap_update_bits(priv->regmap, CS35L33_TX_VBSTMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[2], 2); + dev_dbg(codec->dev, + "VBSTMON enabled in slots %d", slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L33_TX_EN4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + regmap_update_bits(priv->regmap, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l33_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS35L33_MCLK_5644: + case CS35L33_MCLK_6: + case CS35L33_MCLK_6144: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, 0); + cs35l33->mclk_int = freq; + break; + case CS35L33_MCLK_11289: + case CS35L33_MCLK_12: + case CS35L33_MCLK_12288: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, CS35L33_MCLKDIV2); + cs35l33->mclk_int = freq/2; + break; + default: + cs35l33->mclk_int = 0; + return -EINVAL; + } + + dev_dbg(codec->dev, "external mclk freq=%d, internal mclk freq=%d\n", + freq, cs35l33->mclk_int); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l33_ops = { + .startup = cs35l33_pcm_startup, + .set_tristate = cs35l33_set_tristate, + .set_fmt = cs35l33_set_dai_fmt, + .hw_params = cs35l33_pcm_hw_params, + .set_tdm_slot = cs35l33_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l33_dai = { + .name = "cs35l33-dai", + .id = 0, + .playback = { + .stream_name = "CS35L33 Playback", + .channels_min = 1, + .channels_max = 1, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .capture = { + .stream_name = "CS35L33 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .ops = &cs35l33_ops, + .symmetric_rates = 1, +}; + +static int cs35l33_set_hg_data(struct snd_soc_codec *codec, + struct cs35l33_pdata *pdata) +{ + struct cs35l33_hg *hg_config = &pdata->hg_config; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (hg_config->enable_hg_algo) { + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_MEM_DEPTH_MASK, + hg_config->mem_depth << CS35L33_MEM_DEPTH_SHIFT); + regmap_write(priv->regmap, CS35L33_HG_REL_RATE, + hg_config->release_rate); + regmap_update_bits(priv->regmap, CS35L33_HG_HEAD, + CS35L33_HD_RM_MASK, + hg_config->hd_rm << CS35L33_HD_RM_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_THLD_MASK, + hg_config->ldo_thld << CS35L33_LDO_THLD_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_DISABLE_MASK, + hg_config->ldo_path_disable << + CS35L33_LDO_DISABLE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_LDO_ENTRY_DELAY_MASK, + hg_config->ldo_entry_delay << + CS35L33_LDO_ENTRY_DELAY_SHIFT); + if (hg_config->vp_hg_auto) { + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_AUTO_MASK, + CS35L33_VP_HG_AUTO_MASK); + snd_soc_dapm_add_routes(dapm, cs35l33_vphg_auto_route, + ARRAY_SIZE(cs35l33_vphg_auto_route)); + } + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_MASK, + hg_config->vp_hg << CS35L33_VP_HG_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_RATE_MASK, + hg_config->vp_hg_rate << CS35L33_VP_HG_RATE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_VA_MASK, + hg_config->vp_hg_va << CS35L33_VP_HG_VA_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_CLASS_HG_EN_MASK, CS35L33_CLASS_HG_EN_MASK); + } + return 0; +} + +static int cs35l33_set_bst_ipk(struct snd_soc_codec *codec, unsigned int bst) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + int ret = 0, steps = 0; + + /* Boost current in uA */ + if (bst > 3600000 || bst < 1850000) { + dev_err(codec->dev, "Invalid boost current %d\n", bst); + ret = -EINVAL; + goto err; + } + + if (bst % 15625) { + dev_err(codec->dev, "Current not a multiple of 15625uA (%d)\n", + bst); + ret = -EINVAL; + goto err; + } + + while (bst > 1850000) { + bst -= 15625; + steps++; + } + + regmap_write(cs35l33->regmap, CS35L33_BST_PEAK_CTL, + steps+0x70); + +err: + return ret; +} + +static int cs35l33_probe(struct snd_soc_codec *codec) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + cs35l33->codec = codec; + pm_runtime_get_sync(codec->dev); + + regmap_update_bits(cs35l33->regmap, CS35L33_PROTECT_CTL, + CS35L33_ALIVE_WD_DIS, 0x8); + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL2, + CS35L33_ALIVE_WD_DIS2, + CS35L33_ALIVE_WD_DIS2); + + /* Set Platform Data */ + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL1, + CS35L33_BST_CTL_MASK, cs35l33->pdata.boost_ctl); + regmap_update_bits(cs35l33->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_DRV_SEL_MASK, + cs35l33->pdata.amp_drv_sel << CS35L33_AMP_DRV_SEL_SHIFT); + + if (cs35l33->pdata.boost_ipk) + cs35l33_set_bst_ipk(codec, cs35l33->pdata.boost_ipk); + + if (cs35l33->enable_soft_ramp) { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, CS35L33_DIGSFT); + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DSR_RATE, cs35l33->pdata.ramp_rate); + } else { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, 0); + } + + /* update IMON scaling rate if different from default of 0x8 */ + if (cs35l33->pdata.imon_adc_scale != 0x8) + snd_soc_update_bits(codec, CS35L33_ADC_CTL, + CS35L33_IMON_SCALE, cs35l33->pdata.imon_adc_scale); + + cs35l33_set_hg_data(codec, &(cs35l33->pdata)); + + /* + * unmask important interrupts that causes the chip to enter + * speaker safe mode and hence deserves user attention + */ + regmap_update_bits(cs35l33->regmap, CS35L33_INT_MASK_1, + CS35L33_M_OTE | CS35L33_M_OTW | CS35L33_M_AMP_SHORT | + CS35L33_M_CAL_ERR, 0); + + pm_runtime_put_sync(codec->dev); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { + .probe = cs35l33_probe, + + .set_bias_level = cs35l33_set_bias_level, + .set_sysclk = cs35l33_codec_set_sysclk, + + .dapm_widgets = cs35l33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l33_dapm_widgets), + .dapm_routes = cs35l33_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map), + .controls = cs35l33_snd_controls, + .num_controls = ARRAY_SIZE(cs35l33_snd_controls), + + .idle_bias_off = true, +}; + +static const struct regmap_config cs35l33_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L33_MAX_REGISTER, + .reg_defaults = cs35l33_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l33_reg), + .volatile_reg = cs35l33_volatile_register, + .readable_reg = cs35l33_readable_register, + .writeable_reg = cs35l33_writeable_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, +}; + +static int cs35l33_runtime_resume(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + int ret; + + dev_dbg(dev, "%s\n", __func__); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, false); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + + ret = regcache_sync(cs35l33->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + return 0; + +err: + regcache_cache_only(cs35l33->regmap, true); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return ret; +} + +static int cs35l33_runtime_suspend(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s\n", __func__); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + regcache_cache_only(cs35l33->regmap, true); + regcache_mark_dirty(cs35l33->regmap); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l33_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l33_runtime_suspend, + cs35l33_runtime_resume, + NULL) +}; + +static int cs35l33_get_hg_data(const struct device_node *np, + struct cs35l33_pdata *pdata) +{ + struct device_node *hg; + struct cs35l33_hg *hg_config = &pdata->hg_config; + u32 val32; + + hg = of_get_child_by_name(np, "cirrus,hg-algo"); + hg_config->enable_hg_algo = hg ? true : false; + + if (hg_config->enable_hg_algo) { + if (of_property_read_u32(hg, "cirrus,mem-depth", &val32) >= 0) + hg_config->mem_depth = val32; + if (of_property_read_u32(hg, "cirrus,release-rate", + &val32) >= 0) + hg_config->release_rate = val32; + if (of_property_read_u32(hg, "cirrus,ldo-thld", &val32) >= 0) + hg_config->ldo_thld = val32; + if (of_property_read_u32(hg, "cirrus,ldo-path-disable", + &val32) >= 0) + hg_config->ldo_path_disable = val32; + if (of_property_read_u32(hg, "cirrus,ldo-entry-delay", + &val32) >= 0) + hg_config->ldo_entry_delay = val32; + + hg_config->vp_hg_auto = of_property_read_bool(hg, + "cirrus,vp-hg-auto"); + + if (of_property_read_u32(hg, "cirrus,vp-hg", &val32) >= 0) + hg_config->vp_hg = val32; + if (of_property_read_u32(hg, "cirrus,vp-hg-rate", &val32) >= 0) + hg_config->vp_hg_rate = val32; + if (of_property_read_u32(hg, "cirrus,vp-hg-va", &val32) >= 0) + hg_config->vp_hg_va = val32; + } + + of_node_put(hg); + + return 0; +} + +static irqreturn_t cs35l33_irq_thread(int irq, void *data) +{ + struct cs35l33_private *cs35l33 = data; + struct snd_soc_codec *codec = cs35l33->codec; + unsigned int sticky_val1, sticky_val2, current_val, mask1, mask2; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2, + &sticky_val2); + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &sticky_val1); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_2, &mask2); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_1, &mask1); + + /* Check to see if the unmasked bits are active, + * if not then exit. + */ + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + return IRQ_NONE; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &current_val); + + /* handle the interrupts */ + + if (sticky_val1 & CS35L33_AMP_SHORT) { + dev_crit(codec->dev, "Amp short error\n"); + if (!(current_val & CS35L33_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, + CS35L33_AMP_SHORT_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_AMP_SHORT_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + if (!(current_val & CS35L33_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + CS35L33_CAL_ERR_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_OTE) { + dev_crit(codec->dev, "Over temperature error\n"); + if (!(current_val & CS35L33_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, + CS35L33_OTE_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + } + } + + if (sticky_val1 & CS35L33_OTW) { + dev_err(codec->dev, "Over temperature warning\n"); + if (!(current_val & CS35L33_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, + CS35L33_OTW_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + } + } + if (CS35L33_ALIVE_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: ADSPCLK Interrupt\n"); + + if (CS35L33_MCLK_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: MCLK Interrupt\n"); + + if (CS35L33_VMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VMON Overflow Interrupt\n"); + + if (CS35L33_IMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: IMON Overflow Interrupt\n"); + + if (CS35L33_VPMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + + return IRQ_HANDLED; +} + +static const char * const cs35l33_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l33_of_get_pdata(struct device *dev, + struct cs35l33_private *cs35l33) +{ + struct device_node *np = dev->of_node; + struct cs35l33_pdata *pdata = &cs35l33->pdata; + u32 val32; + + if (!np) + return 0; + + if (of_property_read_u32(np, "cirrus,boost-ctl", &val32) >= 0) { + pdata->boost_ctl = val32; + pdata->amp_drv_sel = 1; + } + + if (of_property_read_u32(np, "cirrus,ramp-rate", &val32) >= 0) { + pdata->ramp_rate = val32; + cs35l33->enable_soft_ramp = true; + } + + if (of_property_read_u32(np, "cirrus,boost-ipk", &val32) >= 0) + pdata->boost_ipk = val32; + + if (of_property_read_u32(np, "cirrus,imon-adc-scale", &val32) >= 0) { + if ((val32 == 0x0) || (val32 == 0x7) || (val32 == 0x6)) + pdata->imon_adc_scale = val32; + else + /* use default value */ + pdata->imon_adc_scale = 0x8; + } else { + /* use default value */ + pdata->imon_adc_scale = 0x8; + } + + cs35l33_get_hg_data(np, pdata); + + return 0; +} + +static int cs35l33_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l33_private *cs35l33; + struct cs35l33_pdata *pdata = dev_get_platdata(&i2c_client->dev); + int ret, devid, i; + unsigned int reg; + + cs35l33 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l33_private), + GFP_KERNEL); + if (!cs35l33) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l33); + cs35l33->regmap = devm_regmap_init_i2c(i2c_client, &cs35l33_regmap); + if (IS_ERR(cs35l33->regmap)) { + ret = PTR_ERR(cs35l33->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, true); + + for (i = 0; i < ARRAY_SIZE(cs35l33_core_supplies); i++) + cs35l33->core_supplies[i].supply + = cs35l33_core_supplies[i]; + cs35l33->num_core_supplies = ARRAY_SIZE(cs35l33_core_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l33->pdata = *pdata; + } else { + cs35l33_of_get_pdata(&i2c_client->dev, cs35l33); + pdata = &cs35l33->pdata; + } + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l33_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l33", cs35l33); + if (ret != 0) + dev_warn(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + /* We could issue !RST or skip it based on AMP topology */ + cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset-gpios", GPIOD_OUT_HIGH); + + if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { + dev_warn(&i2c_client->dev, + "%s WARNING: No reset gpio assigned\n", __func__); + } else if (IS_ERR(cs35l33->reset_gpio)) { + dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", + __func__); + return PTR_ERR(cs35l33->reset_gpio); + } + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + goto err_irq; + } + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + regcache_cache_only(cs35l33->regmap, false); + + /* initialize codec */ + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L33_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L33 Device ID (%X). Expected ID %X\n", + devid, CS35L33_CHIP_ID); + goto err_enable; + } + + ret = regmap_read(cs35l33->regmap, CS35L33_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_enable; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L33, Revision: %02X\n", ret & 0xFF); + + ret = regmap_register_patch(cs35l33->regmap, + cs35l33_patch, ARRAY_SIZE(cs35l33_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Error in applying regmap patch: %d\n", ret); + goto err_enable; + } + + /* disable mclk and tdm */ + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l33, &cs35l33_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, "%s: Register codec failed\n", + __func__); + goto err_irq; + } + + return 0; + +err_enable: + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); +err_irq: + free_irq(i2c_client->irq, cs35l33); + + return ret; +} + +static int cs35l33_i2c_remove(struct i2c_client *client) +{ + struct cs35l33_private *cs35l33 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + free_irq(client->irq, cs35l33); + + return 0; +} + +static const struct of_device_id cs35l33_of_match[] = { + { .compatible = "cirrus,cs35l33", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l33_of_match); + +static const struct i2c_device_id cs35l33_id[] = { + {"cs35l33", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l33_id); + +static struct i2c_driver cs35l33_i2c_driver = { + .driver = { + .name = "cs35l33", + .pm = &cs35l33_pm_ops, + .of_match_table = cs35l33_of_match, + + }, + .id_table = cs35l33_id, + .probe = cs35l33_i2c_probe, + .remove = cs35l33_i2c_remove, + +}; +module_i2c_driver(cs35l33_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L33 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l33.h b/sound/soc/codecs/cs35l33.h new file mode 100644 index 0000000..c045737 --- /dev/null +++ b/sound/soc/codecs/cs35l33.h @@ -0,0 +1,221 @@ +/* + * cs35l33.h -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.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 __CS35L33_H__ +#define __CS35L33_H__ + +#define CS35L33_CHIP_ID 0x00035A33 +#define CS35L33_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L33_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L33_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L33_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L33_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L33_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L33_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L33_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L33_BST_PEAK_CTL 0x09 /* Max Current for Boost */ +#define CS35L33_PROTECT_CTL 0x0A /* Amp Protection Parameters */ +#define CS35L33_BST_CTL1 0x0B /* Boost Converter CTL1 */ +#define CS35L33_BST_CTL2 0x0C /* Boost Converter CTL2 */ +#define CS35L33_ADSP_CTL 0x0D /* Serial Port Control */ +#define CS35L33_ADC_CTL 0x0E /* ADC Control */ +#define CS35L33_DAC_CTL 0x0F /* DAC Control */ +#define CS35L33_DIG_VOL_CTL 0x10 /* Digital Volume CTL */ +#define CS35L33_CLASSD_CTL 0x11 /* Class D Amp CTL */ +#define CS35L33_AMP_CTL 0x12 /* Amp Gain/Protecton Release CTL */ +#define CS35L33_INT_MASK_1 0x13 /* Interrupt Mask 1 */ +#define CS35L33_INT_MASK_2 0x14 /* Interrupt Mask 2 */ +#define CS35L33_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L33_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L33_DIAG_LOCK 0x17 /* Diagnostic Mode Register Lock */ +#define CS35L33_DIAG_CTRL_1 0x18 /* Diagnostic Mode Register Control */ +#define CS35L33_DIAG_CTRL_2 0x19 /* Diagnostic Mode Register Control 2 */ +#define CS35L33_HG_MEMLDO_CTL 0x23 /* H/G Memory/LDO CTL */ +#define CS35L33_HG_REL_RATE 0x24 /* H/G Release Rate */ +#define CS35L33_LDO_DEL 0x25 /* LDO Entry Delay/VPhg Control 1 */ +#define CS35L33_HG_HEAD 0x29 /* H/G Headroom */ +#define CS35L33_HG_EN 0x2A /* H/G Enable/VPhg CNT2 */ +#define CS35L33_TX_VMON 0x2D /* TDM TX Control 1 (VMON) */ +#define CS35L33_TX_IMON 0x2E /* TDM TX Control 2 (IMON) */ +#define CS35L33_TX_VPMON 0x2F /* TDM TX Control 3 (VPMON) */ +#define CS35L33_TX_VBSTMON 0x30 /* TDM TX Control 4 (VBSTMON) */ +#define CS35L33_TX_FLAG 0x31 /* TDM TX Control 5 (FLAG) */ +#define CS35L33_TX_EN1 0x32 /* TDM TX Enable 1 */ +#define CS35L33_TX_EN2 0x33 /* TDM TX Enable 2 */ +#define CS35L33_TX_EN3 0x34 /* TDM TX Enable 3 */ +#define CS35L33_TX_EN4 0x35 /* TDM TX Enable 4 */ +#define CS35L33_RX_AUD 0x36 /* TDM RX Control 1 */ +#define CS35L33_RX_SPLY 0x37 /* TDM RX Control 2 */ +#define CS35L33_RX_ALIVE 0x38 /* TDM RX Control 3 */ +#define CS35L33_BST_CTL4 0x39 /* Boost Converter Control 4 */ +#define CS35L33_HG_STATUS 0x3F /* H/G Status */ +#define CS35L33_MAX_REGISTER 0x59 + +#define CS35L33_MCLK_5644 5644800 +#define CS35L33_MCLK_6144 6144000 +#define CS35L33_MCLK_6 6000000 +#define CS35L33_MCLK_11289 11289600 +#define CS35L33_MCLK_12 12000000 +#define CS35L33_MCLK_12288 12288000 + +/* CS35L33_PWRCTL1 */ +#define CS35L33_PDN_AMP (1 << 7) +#define CS35L33_PDN_BST (1 << 2) +#define CS35L33_PDN_ALL 1 + +/* CS35L33_PWRCTL2 */ +#define CS35L33_PDN_VMON_SHIFT 7 +#define CS35L33_PDN_VMON (1 << CS35L33_PDN_VMON_SHIFT) +#define CS35L33_PDN_IMON_SHIFT 6 +#define CS35L33_PDN_IMON (1 << CS35L33_PDN_IMON_SHIFT) +#define CS35L33_PDN_VPMON_SHIFT 5 +#define CS35L33_PDN_VPMON (1 << CS35L33_PDN_VPMON_SHIFT) +#define CS35L33_PDN_VBSTMON_SHIFT 4 +#define CS35L33_PDN_VBSTMON (1 << CS35L33_PDN_VBSTMON_SHIFT) +#define CS35L33_SDOUT_3ST_I2S_SHIFT 3 +#define CS35L33_SDOUT_3ST_I2S (1 << CS35L33_SDOUT_3ST_I2S_SHIFT) +#define CS35L33_PDN_SDIN_SHIFT 2 +#define CS35L33_PDN_SDIN (1 << CS35L33_PDN_SDIN_SHIFT) +#define CS35L33_PDN_TDM_SHIFT 1 +#define CS35L33_PDN_TDM (1 << CS35L33_PDN_TDM_SHIFT) + +/* CS35L33_CLK_CTL */ +#define CS35L33_MCLKDIS (1 << 7) +#define CS35L33_MCLKDIV2 (1 << 6) +#define CS35L33_SDOUT_3ST_TDM (1 << 5) +#define CS35L33_INT_FS_RATE (1 << 4) +#define CS35L33_ADSP_FS 0xF + +/* CS35L33_PROTECT_CTL */ +#define CS35L33_ALIVE_WD_DIS (3 << 2) + +/* CS35L33_BST_CTL1 */ +#define CS35L33_BST_CTL_SRC (1 << 6) +#define CS35L33_BST_CTL_SHIFT (1 << 5) +#define CS35L33_BST_CTL_MASK 0x3F + +/* CS35L33_BST_CTL2 */ +#define CS35L33_TDM_WD_SEL (1 << 4) +#define CS35L33_ALIVE_WD_DIS2 (1 << 3) +#define CS35L33_VBST_SR_STEP 0x3 + +/* CS35L33_ADSP_CTL */ +#define CS35L33_ADSP_DRIVE (1 << 7) +#define CS35L33_MS_MASK (1 << 6) +#define CS35L33_SDIN_LOC (3 << 4) +#define CS35L33_ALIVE_RATE 0x3 + +/* CS35L33_ADC_CTL */ +#define CS35L33_INV_VMON (1 << 7) +#define CS35L33_INV_IMON (1 << 6) +#define CS35L33_ADC_NOTCH_DIS (1 << 5) +#define CS35L33_IMON_SCALE 0xF + +/* CS35L33_DAC_CTL */ +#define CS35L33_INV_DAC (1 << 7) +#define CS35L33_DAC_NOTCH_DIS (1 << 5) +#define CS35L33_DIGSFT (1 << 4) +#define CS35L33_DSR_RATE 0xF + +/* CS35L33_CLASSD_CTL */ +#define CS35L33_AMP_SD (1 << 6) +#define CS35L33_AMP_DRV_SEL_SRC (1 << 5) +#define CS35L33_AMP_DRV_SEL_MASK 0x10 +#define CS35L33_AMP_DRV_SEL_SHIFT 4 +#define CS35L33_AMP_CAL (1 << 3) +#define CS35L33_GAIN_CHG_ZC_MASK 0x04 +#define CS35L33_GAIN_CHG_ZC_SHIFT 2 +#define CS35L33_CLASS_D_CTL_MASK 0x3F + +/* CS35L33_AMP_CTL */ +#define CS35L33_AMP_GAIN 0xF0 +#define CS35L33_CAL_ERR_RLS (1 << 3) +#define CS35L33_AMP_SHORT_RLS (1 << 2) +#define CS35L33_OTW_RLS (1 << 1) +#define CS35L33_OTE_RLS 1 + +/* CS35L33_INT_MASK_1 */ +#define CS35L33_M_CAL_ERR_SHIFT 6 +#define CS35L33_M_CAL_ERR (1 << CS35L33_M_CAL_ERR_SHIFT) +#define CS35L33_M_ALIVE_ERR_SHIFT 5 +#define CS35L33_M_ALIVE_ERR (1 << CS35L33_M_ALIVE_ERR_SHIFT) +#define CS35L33_M_AMP_SHORT_SHIFT 2 +#define CS35L33_M_AMP_SHORT (1 << CS35L33_M_AMP_SHORT_SHIFT) +#define CS35L33_M_OTW_SHIFT 1 +#define CS35L33_M_OTW (1 << CS35L33_M_OTW_SHIFT) +#define CS35L33_M_OTE_SHIFT 0 +#define CS35L33_M_OTE (1 << CS35L33_M_OTE_SHIFT) + +/* CS35L33_INT_STATUS_1 */ +#define CS35L33_CAL_ERR (1 << 6) +#define CS35L33_ALIVE_ERR (1 << 5) +#define CS35L33_ADSPCLK_ERR (1 << 4) +#define CS35L33_MCLK_ERR (1 << 3) +#define CS35L33_AMP_SHORT (1 << 2) +#define CS35L33_OTW (1 << 1) +#define CS35L33_OTE (1 << 0) + +/* CS35L33_INT_STATUS_2 */ +#define CS35L33_VMON_OVFL (1 << 7) +#define CS35L33_IMON_OVFL (1 << 6) +#define CS35L33_VPMON_OVFL (1 << 5) +#define CS35L33_VBSTMON_OVFL (1 << 4) +#define CS35L33_PDN_DONE 1 + +/* CS35L33_BST_CTL4 */ +#define CS35L33_BST_RGS 0x70 +#define CS35L33_BST_COEFF3 0xF + +/* CS35L33_HG_MEMLDO_CTL */ +#define CS35L33_MEM_DEPTH_SHIFT 5 +#define CS35L33_MEM_DEPTH_MASK (0x3 << CS35L33_MEM_DEPTH_SHIFT) +#define CS35L33_LDO_THLD_SHIFT 1 +#define CS35L33_LDO_THLD_MASK (0xF << CS35L33_LDO_THLD_SHIFT) +#define CS35L33_LDO_DISABLE_SHIFT 0 +#define CS35L33_LDO_DISABLE_MASK (0x1 << CS35L33_LDO_DISABLE_SHIFT) + +/* CS35L33_LDO_DEL */ +#define CS35L33_VP_HG_VA_SHIFT 5 +#define CS35L33_VP_HG_VA_MASK (0x7 << CS35L33_VP_HG_VA_SHIFT) +#define CS35L33_LDO_ENTRY_DELAY_SHIFT 2 +#define CS35L33_LDO_ENTRY_DELAY_MASK (0x7 << CS35L33_LDO_ENTRY_DELAY_SHIFT) +#define CS35L33_VP_HG_RATE_SHIFT 0 +#define CS35L33_VP_HG_RATE_MASK (0x3 << CS35L33_VP_HG_RATE_SHIFT) + +/* CS35L33_HG_HEAD */ +#define CS35L33_HD_RM_SHIFT 0 +#define CS35L33_HD_RM_MASK (0x7F << CS35L33_HD_RM_SHIFT) + +/* CS35L33_HG_EN */ +#define CS35L33_CLASS_HG_ENA_SHIFT 7 +#define CS35L33_CLASS_HG_EN_MASK (0x1 << CS35L33_CLASS_HG_ENA_SHIFT) +#define CS35L33_VP_HG_AUTO_SHIFT 6 +#define CS35L33_VP_HG_AUTO_MASK (0x1 << 6) +#define CS35L33_VP_HG_SHIFT 0 +#define CS35L33_VP_HG_MASK (0x1F << CS35L33_VP_HG_SHIFT) + +#define CS35L33_RATES (SNDRV_PCM_RATE_8000_48000) +#define CS35L33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* CS35L33_{RX,TX}_X */ +#define CS35L33_X_STATE_SHIFT 7 +#define CS35L33_X_STATE (1 << CS35L33_X_STATE_SHIFT) +#define CS35L33_X_LOC_SHIFT 0 +#define CS35L33_X_LOC (0x1F << CS35L33_X_LOC_SHIFT) + +/* CS35L33_RX_AUD */ +#define CS35L33_AUDIN_RX_DEPTH_SHIFT 5 +#define CS35L33_AUDIN_RX_DEPTH (0x7 << CS35L33_AUDIN_RX_DEPTH_SHIFT) + +#endif -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: hda - Fix the headset mic jack detection on Dell machine
by woodrow.shen＠canonical.com 24 Jun '16

24 Jun '16

From: Woodrow Shen <woodrow.shen(a)gmail.com> The new Dell laptop with codec 3246 can't detect headset mic when headset was inserted on the machine. So adding pin configurations into quirk table makes headset mic work correctly. Codec: Realtek ALC3246 Vendor Id: 0x10ec0256 Subsystem Id: 0x10280781 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1f652ca1..ac0d5bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5847,6 +5847,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170120}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60170}, + {0x14, 0x90170120}, + {0x21, 0x02211030}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, -- 2.7.4

3 2

[alsa-devel] sound: use-after-free in hrtimer_cancel
by Dmitry Vyukov 24 Jun '16

24 Jun '16

Hello, The following program triggers use-after-free: // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> long r[20]; void* thr(void* arg) { switch ((long)arg) { case 0: break; case 1: r[2] = syscall(SYS_open, "/dev/audio", 0xa40ul, 0, 0, 0); break; case 2: *(uint32_t*)0x2001dde8 = r[2]; *(uint16_t*)0x2001ddec = (uint16_t)0x0; *(uint16_t*)0x2001ddee = (uint16_t)0x2; *(uint32_t*)0x2001ddf0 = r[2]; *(uint16_t*)0x2001ddf4 = (uint16_t)0xfffffffff3a15aea; *(uint16_t*)0x2001ddf6 = (uint16_t)0x1; *(uint32_t*)0x2001ddf8 = r[2]; *(uint16_t*)0x2001ddfc = (uint16_t)0x836a; *(uint16_t*)0x2001ddfe = (uint16_t)0x1ff; *(uint32_t*)0x2001de00 = r[2]; *(uint16_t*)0x2001de04 = (uint16_t)0x0; *(uint16_t*)0x2001de06 = (uint16_t)0x20; r[15] = syscall(SYS_poll, 0x2001dde8ul, 0x4ul, 0x8ul, 0, 0, 0); break; case 3: *(uint64_t*)0x20dc13c0 = (uint64_t)0x20dc1f27; *(uint64_t*)0x20dc13c8 = (uint64_t)0xd9; r[18] = syscall(SYS_readv, r[2], 0x20dc13c0ul, 0x1ul, 0, 0, 0); break; case 4: r[19] = syscall(SYS_read, r[2], 0x20dbefe0ul, 0x20ul, 0, 0, 0); break; } return 0; } int main() { long i; pthread_t th[10]; syscall(SYS_mmap, 0x20000000ul, 0xde0000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); srand(getpid()); memset(r, -1, sizeof(r)); for (i = 0; i < 5; i++) { pthread_create(&th[i], 0, thr, (void*)i); usleep(rand()%5000); } for (i = 0; i < 5; i++) { pthread_create(&th[5+i], 0, thr, (void*)i); if (rand()%2) usleep(rand()%5000); } for (i = 0; i < 10; i++) { pthread_join(th[i], 0); } return 0; } Here are 2 reports (note they are slightly different): ================================================================== BUG: KASAN: use-after-free in rb_erase+0x1b17/0x2010 at addr ffff88005e5b6f68 Read of size 8 by task syz-executor/8984 ============================================================================= BUG kmalloc-192 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- Disabling lock debugging due to kernel taint INFO: Allocated in 0xbbbbbbbbbbbbbbbb age=18446705582212484632 cpu=2172593693 pid=-1 [< inline >] kmalloc include/linux/slab.h:478 [< inline >] kzalloc include/linux/slab.h:622 [< none >] dummy_hrtimer_create+0x49/0x1a0 sound/drivers/dummy.c:464 [< none >] ___slab_alloc+0x55d/0x5a0 mm/slub.c:2476 [< none >] __slab_alloc+0x68/0xc0 mm/slub.c:2505 [< inline >] slab_alloc_node mm/slub.c:2568 [< inline >] slab_alloc mm/slub.c:2610 [< none >] kmem_cache_alloc_trace+0x263/0x3d0 mm/slub.c:2627 [< inline >] kmalloc include/linux/slab.h:478 [< inline >] kzalloc include/linux/slab.h:622 [< none >] dummy_hrtimer_create+0x49/0x1a0 sound/drivers/dummy.c:464 [< none >] dummy_pcm_open+0xcd/0x5e0 sound/drivers/dummy.c:574 [< none >] snd_pcm_open_substream+0x188/0x430 sound/core/pcm_native.c:2276 [< inline >] snd_pcm_oss_open_file sound/core/oss/pcm_oss.c:2346 [< none >] snd_pcm_oss_open.part.17+0x5a4/0x1110 sound/core/oss/pcm_oss.c:2428 [< none >] snd_pcm_oss_open+0x35/0x50 sound/core/oss/pcm_oss.c:2392 [< none >] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [< none >] chrdev_open+0x22a/0x4c0 fs/char_dev.c:388 [< none >] do_dentry_open+0x6a2/0xcb0 fs/open.c:736 [< none >] vfs_open+0x17b/0x1f0 fs/open.c:853 [< inline >] do_last fs/namei.c:3238 [< none >] path_openat+0x51bb/0x5ce0 fs/namei.c:3374 [< none >] do_filp_open+0x18e/0x250 fs/namei.c:3409 [< none >] do_sys_open+0x1fc/0x420 fs/open.c:1020 INFO: Freed in 0xfffd8e09 age=18446705496313138713 cpu=2164287125 pid=-1 [< none >] dummy_hrtimer_free+0x68/0x80 sound/drivers/dummy.c:481 [< none >] __slab_free+0x1e8/0x300 mm/slub.c:2687 [< inline >] slab_free mm/slub.c:2840 [< none >] kfree+0x2fc/0x370 mm/slub.c:3691 [< none >] dummy_hrtimer_free+0x68/0x80 sound/drivers/dummy.c:481 [< none >] dummy_pcm_close+0x9c/0xd0 sound/drivers/dummy.c:607 [< none >] snd_pcm_release_substream.part.37+0x169/0x2f0 sound/core/pcm_native.c:2241 [< none >] snd_pcm_release_substream+0x59/0x70 sound/core/pcm_native.c:2251 [< none >] snd_pcm_oss_release_file+0x7b/0xb0 sound/core/oss/pcm_oss.c:2305 [< none >] snd_pcm_oss_release+0xfa/0x280 sound/core/oss/pcm_oss.c:2485 [< none >] __fput+0x236/0x780 fs/file_table.c:208 [< none >] ____fput+0x15/0x20 fs/file_table.c:244 [< none >] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [< none >] do_exit+0x874/0x2d80 kernel/exit.c:748 [< none >] do_group_exit+0x108/0x330 kernel/exit.c:878 [< none >] get_signal+0x634/0x15e0 kernel/signal.c:2307 [< none >] do_signal+0x7f/0x1cf0 arch/x86/kernel/signal.c:784 INFO: Indirect in 0xfffd8e17 age=4294807066 cpu=0 pid=0 INFO: Slab 0xffffea0001796d00 objects=24 used=15 fp=0xffff88005e5b71e8 flags=0x4fffe0000004080 INFO: Object 0xffff88005e5b6f40 @offset=12096 fp=0xbbbbbbbbbbbbbbbb CPU: 1 PID: 8984 Comm: syz-executor Tainted: G B 4.6.0-rc6+ #355 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff880059fd7378 ffffffff82c9062f ffffffff5e5b4000 fffffbfff0fd64b8 ffff88003e804d40 ffff88005e5b6f40 ffffea0001796d00 ffff88005e5b4000 0000000000000000 ffff880059fd73a8 ffffffff81793e1d Call Trace: [<ffffffff8179e59e>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:333 [< inline >] rb_set_parent include/linux/rbtree_augmented.h:111 [< inline >] __rb_erase_augmented include/linux/rbtree_augmented.h:218 [<ffffffff82ca5787>] rb_erase+0x1b17/0x2010 lib/rbtree.c:427 [<ffffffff82cb02e8>] timerqueue_del+0x78/0x170 lib/timerqueue.c:86 [<ffffffff814d0c80>] __remove_hrtimer+0x90/0x220 kernel/time/hrtimer.c:903 [< inline >] remove_hrtimer kernel/time/hrtimer.c:945 [<ffffffff814d23da>] hrtimer_try_to_cancel+0x22a/0x570 kernel/time/hrtimer.c:1046 [<ffffffff814d2742>] hrtimer_cancel+0x22/0x40 kernel/time/hrtimer.c:1066 [<ffffffff85420531>] dummy_hrtimer_stop+0x91/0xb0 sound/drivers/dummy.c:417 [<ffffffff854228bf>] dummy_pcm_trigger+0x17f/0x1e0 sound/drivers/dummy.c:507 [<ffffffff85392170>] snd_pcm_do_stop+0x160/0x1b0 sound/core/pcm_native.c:1106 [<ffffffff85391b26>] snd_pcm_action_single+0x76/0x120 sound/core/pcm_native.c:956 [<ffffffff85391e01>] snd_pcm_action+0x231/0x290 sound/core/pcm_native.c:974 [< inline >] snd_pcm_stop sound/core/pcm_native.c:1139 [<ffffffff8539754d>] snd_pcm_drop+0x12d/0x1d0 sound/core/pcm_native.c:1784 [<ffffffff8539d3be>] snd_pcm_common_ioctl1+0xfae/0x2150 sound/core/pcm_native.c:2805 [<ffffffff8539ee91>] snd_pcm_capture_ioctl1+0x2a1/0x5e0 sound/core/pcm_native.c:2976 [<ffffffff8539f2ec>] snd_pcm_kernel_ioctl+0x11c/0x160 sound/core/pcm_native.c:3020 [<ffffffff853d9a44>] snd_pcm_oss_sync+0x3a4/0xa30 sound/core/oss/pcm_oss.c:1693 [<ffffffff853da27d>] snd_pcm_oss_release+0x1ad/0x280 sound/core/oss/pcm_oss.c:2483 [<ffffffff817fc066>] __fput+0x236/0x780 fs/file_table.c:208 [<ffffffff817fc635>] ____fput+0x15/0x20 fs/file_table.c:244 [<ffffffff813c91d0>] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [<ffffffff81373e64>] do_exit+0x874/0x2d80 kernel/exit.c:748 [<ffffffff813764e8>] do_group_exit+0x108/0x330 kernel/exit.c:878 [<ffffffff81399674>] get_signal+0x634/0x15e0 kernel/signal.c:2307 [<ffffffff811f40af>] do_signal+0x7f/0x1cf0 arch/x86/kernel/signal.c:784 [<ffffffff81006695>] exit_to_usermode_loop+0x1a5/0x210 arch/x86/entry/common.c:229 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:264 [<ffffffff8100868f>] syscall_return_slowpath+0x2bf/0x340 arch/x86/entry/common.c:329 [<ffffffff867c56dc>] entry_SYSCALL_64_fastpath+0xbf/0xc1 arch/x86/entry/entry_64.S:241 ================================================================== ================================================================== BUG: KASAN: use-after-free in timerqueue_add+0x29a/0x2a0 at addr ffff880062134580 Read of size 8 by task a.out/12711 ============================================================================= BUG kmalloc-192 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- Disabling lock debugging due to kernel taint INFO: Allocated in 0xbbbbbbbbbbbbbbbb age=18446689703718641197 cpu=2172593693 pid=-1 [< inline >] kmalloc include/linux/slab.h:478 [< inline >] kzalloc include/linux/slab.h:622 [< none >] dummy_hrtimer_create+0x49/0x1a0 sound/drivers/dummy.c:464 [< none >] ___slab_alloc+0x55d/0x5a0 mm/slub.c:2476 [< none >] __slab_alloc+0x68/0xc0 mm/slub.c:2505 [< inline >] slab_alloc_node mm/slub.c:2568 [< inline >] slab_alloc mm/slub.c:2610 [< none >] kmem_cache_alloc_trace+0x263/0x3d0 mm/slub.c:2627 [< inline >] kmalloc include/linux/slab.h:478 [< inline >] kzalloc include/linux/slab.h:622 [< none >] dummy_hrtimer_create+0x49/0x1a0 sound/drivers/dummy.c:464 [< none >] dummy_pcm_open+0xcd/0x5e0 sound/drivers/dummy.c:574 [< none >] snd_pcm_open_substream+0x188/0x430 sound/core/pcm_native.c:2276 [< inline >] snd_pcm_oss_open_file sound/core/oss/pcm_oss.c:2346 [< none >] snd_pcm_oss_open.part.17+0x5a4/0x1110 sound/core/oss/pcm_oss.c:2428 [< none >] snd_pcm_oss_open+0x35/0x50 sound/core/oss/pcm_oss.c:2392 [< none >] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [< none >] chrdev_open+0x22a/0x4c0 fs/char_dev.c:388 [< none >] do_dentry_open+0x6a2/0xcb0 fs/open.c:736 [< none >] vfs_open+0x17b/0x1f0 fs/open.c:853 [< inline >] do_last fs/namei.c:3238 [< none >] path_openat+0x51bb/0x5ce0 fs/namei.c:3374 [< none >] do_filp_open+0x18e/0x250 fs/namei.c:3409 [< none >] do_sys_open+0x1fc/0x420 fs/open.c:1020 INFO: Freed in 0x100015e1b age=18446689836862627373 cpu=0 pid=0 [< none >] dummy_hrtimer_free+0x68/0x80 sound/drivers/dummy.c:481 [< none >] __slab_free+0x1e8/0x300 mm/slub.c:2687 [< inline >] slab_free mm/slub.c:2840 [< none >] kfree+0x2fc/0x370 mm/slub.c:3691 [< none >] dummy_hrtimer_free+0x68/0x80 sound/drivers/dummy.c:481 [< none >] dummy_pcm_close+0x9c/0xd0 sound/drivers/dummy.c:607 [< none >] snd_pcm_release_substream.part.37+0x169/0x2f0 sound/core/pcm_native.c:2241 [< none >] snd_pcm_release_substream+0x59/0x70 sound/core/pcm_native.c:2251 [< none >] snd_pcm_oss_release_file+0x7b/0xb0 sound/core/oss/pcm_oss.c:2305 [< none >] snd_pcm_oss_release+0xfa/0x280 sound/core/oss/pcm_oss.c:2485 [< none >] __fput+0x236/0x780 fs/file_table.c:208 [< none >] ____fput+0x15/0x20 fs/file_table.c:244 [< none >] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [< none >] do_exit+0x874/0x2d80 kernel/exit.c:748 [< none >] do_group_exit+0x108/0x330 kernel/exit.c:878 [< inline >] SYSC_exit_group kernel/exit.c:889 [< none >] SyS_exit_group+0x1d/0x20 kernel/exit.c:887 [< none >] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207 INFO: Indirect in 0x100015e2e age=4295056943 cpu=0 pid=0 INFO: Slab 0xffffea0001884d00 objects=24 used=10 fp=0xffff880062135a48 flags=0x4fffe0000004080 INFO: Object 0xffff880062134540 @offset=1344 fp=0xbbbbbbbbbbbbbbbb CPU: 2 PID: 12711 Comm: a.out Tainted: G B 4.6.0-rc6+ #355 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff88006d407c60 ffffffff82c9062f ffffffff62134000 fffffbfff0fd64b8 ffff88003e804d40 ffff880062134540 ffffea0001884d00 ffff880062134000 ffff88006d417f10 ffff88006d407c90 ffffffff81793e1d Call Trace: [<ffffffff8179e59e>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:333 [<ffffffff82cb022a>] timerqueue_add+0x29a/0x2a0 lib/timerqueue.c:51 [<ffffffff814d1066>] enqueue_hrtimer+0x116/0x3d0 kernel/time/hrtimer.c:879 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1257 [<ffffffff814d1c5e>] __hrtimer_run_queues+0x93e/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d4022>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa62>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c7f59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c62ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 [<ffffffff813c93d3>] __kernel_text_address+0x73/0xa0 kernel/extable.c:103 [<ffffffff811fd73e>] print_context_stack+0x6e/0xc0 arch/x86/kernel/dumpstack.c:107 [<ffffffff811fc984>] dump_trace+0x124/0x320 arch/x86/kernel/dumpstack_64.c:243 [<ffffffff8121d396>] save_stack_trace+0x26/0x50 arch/x86/kernel/stacktrace.c:67 [< inline >] __save_stack_trace mm/kmemleak.c:530 [<ffffffff817e2467>] create_object+0x137/0x2d0 mm/kmemleak.c:585 [<ffffffff867a7d43>] kmemleak_alloc+0x63/0xa0 mm/kmemleak.c:915 [< inline >] kmemleak_alloc_recursive include/linux/kmemleak.h:47 [< inline >] slab_post_alloc_hook mm/slab.h:406 [< inline >] slab_alloc_node mm/slub.c:2602 [< inline >] slab_alloc mm/slub.c:2610 [<ffffffff81798236>] kmem_cache_alloc+0x166/0x3c0 mm/slub.c:2615 [<ffffffff81727df0>] ptlock_alloc+0x20/0x80 mm/memory.c:3968 [< inline >] ptlock_init include/linux/mm.h:1625 [< inline >] pgtable_page_ctor include/linux/mm.h:1659 [<ffffffff81299ba9>] pte_alloc_one+0x59/0x100 arch/x86/mm/pgtable.c:31 [<ffffffff81716c68>] __pte_alloc+0x28/0x2a0 mm/memory.c:569 [< inline >] __handle_mm_fault mm/memory.c:3470 [<ffffffff81721a6a>] handle_mm_fault+0xc0a/0x11a0 mm/memory.c:3522 [<ffffffff8128ad47>] __do_page_fault+0x457/0xbb0 arch/x86/mm/fault.c:1351 [<ffffffff8128b5ef>] trace_do_page_fault+0xdf/0x5b0 arch/x86/mm/fault.c:1444 [<ffffffff8127bd84>] do_async_page_fault+0x14/0xd0 arch/x86/kernel/kvm.c:265 [<ffffffff867c77f8>] async_page_fault+0x28/0x30 arch/x86/entry/entry_64.S:920 ================================================================== On commit 83858a701cf3271f81dd321c2a81e5666c6ca8f4 (note it is not super fresh: May 3).

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[alsa-devel] [PATCH] ALSA: hda / realtek - add two more Thinkpad IDs (5050, 5053) for tpt460 fixup
by Jaroslav Kysela 24 Jun '16

24 Jun '16

See: https://bugzilla.redhat.com/show_bug.cgi?id=1349539 See: https://bugzilla.kernel.org/show_bug.cgi?id=120961 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- 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 ac0d5bf..999620d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5665,6 +5665,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x5050, "Thinkpad T560p", ALC292_FIXUP_TPT460), + SND_PCI_QUIRK(0x17aa, 0x5053, "Thinkpad T460", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 2.5.5

2 1

Re: [alsa-devel] [PATCH 6/7] of_graph: add of_graph_get_top_port()
by Kuninori Morimoto 24 Jun '16

24 Jun '16

Hi OK, I will post v2 patch to solve this issue. But it will be after patch-review. > [auto build test WARNING on robh/for-next] > [also build test WARNING on v4.7-rc4 next-20160623] > [cannot apply to glikely/devicetree/next] > [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/Kuninori-Morimoto/of_graph-prepare… > base: https://git.kernel.org/pub/scm/linux/kernel/git/robh/linux for-next > config: arm-at91_dt_defconfig (attached as .config) > compiler: arm-linux-gnueabi-gcc (Debian 5.3.1-8) 5.3.1 20160205 > reproduce: > wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # save the attached .config to linux build tree > make.cross ARCH=arm > > All warnings (new ones prefixed by >>): > > In file included from drivers/gpu/drm/atmel-hlcdc/atmel_hlcdc_output.c:22:0: > >> include/linux/of_graph.h:54:50: warning: 'struct device' declared inside parameter list > struct device_node *of_graph_get_top_port(struct device *dev); > ^ > >> include/linux/of_graph.h:54:50: warning: its scope is only this definition or declaration, which is probably not what you want > > vim +54 include/linux/of_graph.h > > 38 */ > 39 #define for_each_endpoint_of_node(parent, child) \ > 40 for (child = of_graph_get_next_endpoint(parent, NULL); child != NULL; \ > 41 child = of_graph_get_next_endpoint(parent, child)) > 42 > 43 #define of_graph_port_type_is_sound(n) of_graph_port_type_is(n, "sound") > 44 #define of_graph_endpoint_type_is_sound(n) of_graph_endpoint_type_is(n, "sound") > 45 #define of_graph_get_sound_endpoint_count(n) of_graph_get_endpoint_count(n, "sound") > 46 > 47 #ifdef CONFIG_OF > 48 int of_graph_parse_endpoint(const struct device_node *node, > 49 struct of_endpoint *endpoint); > 50 bool of_graph_port_type_is(struct device_node *port, char *type); > 51 bool of_graph_endpoint_type_is(struct device_node *ep, char *type); > 52 int of_graph_get_endpoint_count(const struct device_node *np, char *type); > 53 struct device_node *of_graph_get_port_by_id(struct device_node *node, u32 id); > > 54 struct device_node *of_graph_get_top_port(struct device *dev); > 55 struct device_node *of_graph_get_next_endpoint(const struct device_node *parent, > 56 struct device_node *previous); > 57 struct device_node *of_graph_get_endpoint_by_regs( > 58 const struct device_node *parent, int port_reg, int reg); > 59 struct device_node *of_graph_get_remote_endpoint( > 60 const struct device_node *node); > 61 struct device_node *of_graph_get_port_parent(struct device_node *node); > 62 struct device_node *of_graph_get_remote_port_parent( > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation > [2 .config.gz <application/octet-stream (base64)>] >

1 0

[alsa-devel] S/PDIF and track markers (echo3g)
by Mark Hills 24 Jun '16

24 Jun '16

Hi, Is there any provision in ALSA (either within the kernel or libasound) for extracting track marker information from S/PDIF? I have a large number of MiniDisc recordings to preserve as FLAC format. Losing track markers would be a huge inconvenience, though I don't care for the rest of the metadata (track names etc..) There's enough recordings to be worth developing specific software for the archival if it can access this information. I don't really know which path is likely to be successful: * An ALSA API to access the information; though I can't see anything specific to track markers * Whether a timecode could be used to detect track change; but I can't see any in /proc/asound/Layla3G/status * Does ALSA provide access to raw AES streams, so I could decode it in userland? I'm un-sure if this relates to hardware AC3 decoding. * Maybe this sort of status is only available to the firmware of the interface and never makes it to userland (in my case, echo3g); though I can see some constants relating to 'non audio' S/PDIF. The hardware I have here is Echo Layla 3G (snd-echo3g) and a Pioneer MJ-D707 MiniDisc recorder. Any advice appreciated -- many thanks. -- Mark

2 2

[alsa-devel] [PATCH alsa-lib] doxygen: Suppress timestamp
by Takashi Iwai 23 Jun '16

23 Jun '16

The build timestamp in the doc is bad for packaging in general, so let's disable it as default. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- doc/doxygen.cfg.in | 2 ++ 1 file changed, 2 insertions(+) diff --git a/doc/doxygen.cfg.in b/doc/doxygen.cfg.in index 92bd52ba67a2..11740d342596 100644 --- a/doc/doxygen.cfg.in +++ b/doc/doxygen.cfg.in @@ -124,3 +124,5 @@ TYPEDEF_HIDES_STRUCT = YES # needed in doxygen >= 1.5.4 #INPUT_FILTER = inputfilter #FILTER_SOURCE_FILES = YES + +HTML_TIMESTAMP = NO -- 2.9.0

1 0

[alsa-devel] [very-RFC 0/8] TSN driver for the kernel
by Henrik Austad 23 Jun '16

23 Jun '16

Hi all (series based on v4.7-rc2, now with the correct netdev) This is a *very* early RFC for a TSN-driver in the kernel. It has been floating around in my repo for a while and I would appreciate some feedback on the overall design to avoid doing some major blunders. TSN: Time Sensitive Networking, formely known as AVB (Audio/Video Bridging). There are at least one AVB-driver (the AV-part of TSN) in the kernel already, however this driver aims to solve a wider scope as TSN can do much more than just audio. A very basic ALSA-driver is added to the end that allows you to play music between 2 machines using aplay in one end and arecord | aplay on the other (some fiddling required) We have plans for doing the same for v4l2 eventually (but there are other fishes to fry first). The same goes for a TSN_SOCK type approach as well. TSN is all about providing infrastructure. Allthough there are a few very interesting uses for TSN (reliable, deterministic network for audio and video), once you have that reliable link, you can do a lot more. Some notes on the design: The driver is directed via ConfigFS as we need userspace to handle stream-reservation (MSRP), discovery and enumeration (IEEE 1722.1) and whatever other management is needed. Once we have all the required attributes, we can create link using mkdir, and use write() to set the attributes. Once ready, specify the 'shim' (basically a thin wrapper between TSN and another subsystem) and we start pushing out frames. The network part: it ties directly into the rx-handler for receive and writes skb's using netdev_start_xmit(). This could probably be improved. 2 new fields in netdev_ops have been introduced, and the Intel igb-driver has been updated (as this is available as a PCI-e card). The igb-driver works-ish What remains - tie to (g)PTP properly, currently using ktime_get() for presentation time - get time from shim into TSN and vice versa - let shim create/manage buffer Henrik Austad (8): TSN: add documentation TSN: Add the standard formerly known as AVB to the kernel Adding TSN-driver to Intel I210 controller Add TSN header for the driver Add TSN machinery to drive the traffic from a shim over the network Add TSN event-tracing AVB ALSA - Add ALSA shim for TSN MAINTAINERS: add TSN/AVB-entries Documentation/TSN/tsn.txt | 147 +++++ MAINTAINERS | 14 + drivers/media/Kconfig | 15 + drivers/media/Makefile | 3 +- drivers/media/avb/Makefile | 5 + drivers/media/avb/avb_alsa.c | 742 +++++++++++++++++++++++ drivers/media/avb/tsn_iec61883.h | 124 ++++ drivers/net/ethernet/intel/Kconfig | 18 + drivers/net/ethernet/intel/igb/Makefile | 2 +- drivers/net/ethernet/intel/igb/igb.h | 19 + drivers/net/ethernet/intel/igb/igb_main.c | 10 +- drivers/net/ethernet/intel/igb/igb_tsn.c | 396 ++++++++++++ include/linux/netdevice.h | 32 + include/linux/tsn.h | 806 ++++++++++++++++++++++++ include/trace/events/tsn.h | 349 +++++++++++ net/Kconfig | 1 + net/Makefile | 1 + net/tsn/Kconfig | 32 + net/tsn/Makefile | 6 + net/tsn/tsn_configfs.c | 623 +++++++++++++++++++ net/tsn/tsn_core.c | 975 ++++++++++++++++++++++++++++++ net/tsn/tsn_header.c | 203 +++++++ net/tsn/tsn_internal.h | 383 ++++++++++++ net/tsn/tsn_net.c | 403 ++++++++++++ 24 files changed, 5306 insertions(+), 3 deletions(-) create mode 100644 Documentation/TSN/tsn.txt create mode 100644 drivers/media/avb/Makefile create mode 100644 drivers/media/avb/avb_alsa.c create mode 100644 drivers/media/avb/tsn_iec61883.h create mode 100644 drivers/net/ethernet/intel/igb/igb_tsn.c create mode 100644 include/linux/tsn.h create mode 100644 include/trace/events/tsn.h create mode 100644 net/tsn/Kconfig create mode 100644 net/tsn/Makefile create mode 100644 net/tsn/tsn_configfs.c create mode 100644 net/tsn/tsn_core.c create mode 100644 net/tsn/tsn_header.c create mode 100644 net/tsn/tsn_internal.h create mode 100644 net/tsn/tsn_net.c -- 2.7.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

9 63

[alsa-devel] [PATCH v2] ASoC: cs53l30: Add MUTE pin control support via GPIO
by Nicolin Chen 23 Jun '16

23 Jun '16

The codec chip has a physical MUTE pin to let users control it via GPIO. So this patch add a mute control support to the driver. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changelog: v1->v2 * Revise the DT binding part regarding the active state of the mute pin .../devicetree/bindings/sound/cs53l30.txt | 4 +++ sound/soc/codecs/cs53l30.c | 30 ++++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 1 + 3 files changed, 35 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt index 18d6b99..4dbfb82 100644 --- a/Documentation/devicetree/bindings/sound/cs53l30.txt +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -13,6 +13,10 @@ Optional properties: - reset-gpios : a GPIO spec for the reset pin. + - mute-gpios : a GPIO spec for the MUTE pin. The active state can be either + GPIO_ACTIVE_HIGH or GPIO_ACTIVE_LOW, which would be handled + by the driver automatically. + - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin. 0 = Hi-Z 1 = 1.80 V diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 384a3f7..e46b7de 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -35,6 +35,7 @@ struct cs53l30_private { struct regulator_bulk_data supplies[CS53L30_NUM_SUPPLIES]; struct regmap *regmap; struct gpio_desc *reset_gpio; + struct gpio_desc *mute_gpio; struct clk *mclk; bool use_sdout2; u32 mclk_rate; @@ -833,6 +834,16 @@ static int cs53l30_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int cs53l30_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec); + + if (priv->mute_gpio) + gpiod_set_value_cansleep(priv->mute_gpio, mute); + + return 0; +} + /* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ #define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) @@ -846,6 +857,7 @@ static const struct snd_soc_dai_ops cs53l30_ops = { .set_sysclk = cs53l30_set_sysclk, .set_tristate = cs53l30_set_tristate, .set_tdm_slot = cs53l30_set_dai_tdm_slot, + .mute_stream = cs53l30_mute_stream, }; static struct snd_soc_dai_driver cs53l30_dai = { @@ -990,6 +1002,24 @@ static int cs53l30_i2c_probe(struct i2c_client *client, cs53l30->mclk = NULL; } + /* Fetch the MUTE control */ + cs53l30->mute_gpio = devm_gpiod_get_optional(dev, "mute", + GPIOD_OUT_HIGH); + if (IS_ERR(cs53l30->mute_gpio)) { + ret = PTR_ERR(cs53l30->mute_gpio); + goto error; + } + + if (cs53l30->mute_gpio) { + /* Enable MUTE controls via MUTE pin */ + regmap_write(cs53l30->regmap, CS53L30_MUTEP_CTL1, + CS53L30_MUTEP_CTL1_MUTEALL); + /* Flip the polarity of MUTE pin */ + if (gpiod_is_active_low(cs53l30->mute_gpio)) + regmap_update_bits(cs53l30->regmap, CS53L30_MUTEP_CTL2, + CS53L30_MUTE_PIN_POLARITY, 0); + } + if (!of_property_read_u8(np, "cirrus,micbias-lvl", &val)) regmap_update_bits(cs53l30->regmap, CS53L30_MICBIAS_CTL, CS53L30_MIC_BIAS_CTRL_MASK, val); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h index 0dd4afb..5e39da5 100644 --- a/sound/soc/codecs/cs53l30.h +++ b/sound/soc/codecs/cs53l30.h @@ -253,6 +253,7 @@ #define CS53L30_MUTE_MB_ALL_PDN_MASK (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) #define CS53L30_MUTE_MB_ALL_PDN (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) +#define CS53L30_MUTEP_CTL1_MUTEALL (0xdf) #define CS53L30_MUTEP_CTL1_DEFAULT (0) /* R32 (0x20) CS53L30_MUTEP_CTL2 - MUTE Pin Control 2 */ -- 2.1.4

3 2

[alsa-devel] [PATCH 1/4] ASoC: compress: Pass error out of soc_compr_pointer
by Charles Keepax 22 Jun '16

22 Jun '16

The soc_compr_pointer does not correctly pass any errors returned by the driver callback back up the stack. This patch corrects this issue. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 875733c..d2df46c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -530,14 +530,15 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) - platform->driver->compr_ops->pointer(cstream, tstamp); + ret = platform->driver->compr_ops->pointer(cstream, tstamp); mutex_unlock(&rtd->pcm_mutex); - return 0; + return ret; } static int soc_compr_copy(struct snd_compr_stream *cstream, -- 2.1.4

4 15

[alsa-devel] [PATCH 1/2] ASoC: Intel: Kconfig: fix build when ACPI is not enabled
by Vinod Koul 22 Jun '16

22 Jun '16

Randy reported following error when ACPI is not enabled: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_BYTCR_RT5651_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI +which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) causing these build errors: In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:40:0: ../include/acpi/acpi_bus.h:65:20: error: conflicting types for 'acpi_evaluate_dsm' union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:31:0: ../include/linux/acpi.h:676:34: note: previous definition of 'acpi_evaluate_dsm' was here static inline union acpi_object *acpi_evaluate_dsm(acpi_handle handle, CONFIG_SND_SST_IPC_ACPI was already dependent upon ACPI, but that was not solving it. So move the depends up to machine drivers and remove from CONFIG_SND_SST_IPC_ACPI. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 3875425a9693..63e4c14ebe9f 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -25,7 +25,6 @@ config SND_SST_IPC_ACPI tristate select SND_SST_IPC select SND_SOC_INTEL_SST - depends on ACPI config SND_SOC_INTEL_SST tristate @@ -128,7 +127,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5640 codec" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -141,7 +140,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5651 codec" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5651 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -154,7 +153,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -167,7 +166,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI -- 1.9.1

2 3

[alsa-devel] [PATCH 1/2] ASoC: rt5514: Fix the issue that the variable dereferenced before checking
by Oder Chiou 22 Jun '16

22 Jun '16

The patch fixes the issue that variable dereferenced before checking 'rt5514_dsp->substream'. Move the assignment to after the variable checking of 'rt5514_dsp->substream'. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 8a9382e..743f509 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -80,7 +80,7 @@ static void rt5514_spi_copy_work(struct work_struct *work) { struct rt5514_dsp *rt5514_dsp = container_of(work, struct rt5514_dsp, copy_work.work); - struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; + struct snd_pcm_runtime *runtime; size_t period_bytes, truncated_bytes = 0; mutex_lock(&rt5514_dsp->dma_lock); @@ -89,6 +89,7 @@ static void rt5514_spi_copy_work(struct work_struct *work) goto done; } + runtime = rt5514_dsp->substream->runtime; period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset < period_bytes) -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v2 0/4] ASoC: Intel: Skylake: Disable async suspend
by Vinod Koul 22 Jun '16

22 Jun '16

Skylake does not support async suspend, so mark it and update the machine driver. Also revert older machine patch which worked around this by using pm.complete. Also update a style inconsistency while at it. Vinod Koul (4): ASoC: Intel: Revert "ASoC: Intel: Add support for PM ops in bxt-rt298" ASoC: Intel: Skylake: Disable async suspend ASoC: Intel: Skylake: Add pm ops for broxton-rt298 machine ASoC: Intel: Skylake: Update comment style sound/soc/intel/boards/bxt_rt298.c | 24 +----------------------- sound/soc/intel/skylake/skl.c | 4 +++- 2 files changed, 4 insertions(+), 24 deletions(-) Thanks -- 1.9.1

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[alsa-devel] [PATCH] ASoC: max9877: Remove unused function declaration
by Helen Koike 22 Jun '16

22 Jun '16

Remove unused function declaration from header Signed-off-by: Helen Koike <helen.koike(a)collabora.co.uk> --- sound/soc/codecs/max9877.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/max9877.h b/sound/soc/codecs/max9877.h index 6da7229..368343f 100644 --- a/sound/soc/codecs/max9877.h +++ b/sound/soc/codecs/max9877.h @@ -32,6 +32,4 @@ #define MAX9877_BYPASS (1 << 6) #define MAX9877_SHDN (1 << 7) -extern int max9877_add_controls(struct snd_soc_codec *codec); - #endif -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: Intel: Skylake: Add DSP muti-core support
by Vinod Koul 22 Jun '16

22 Jun '16

This series adds support for managing multiple DSP cores. SW is responsible for powering up/down cores in DSP and we do that using refcount. 1st patch introduces core infrastructure change in driver and then adds support in SKL and BXT Changes in v2: - fix error report by kbuild bot Jayachandran B (3): ASoC: Intel: Skylake: Add DSP muti-core infrastructure ASoC: Intel: Skylake: Support multi-core in Skylake ASoC: Intel: Skylake: Support multi-core in Broxton sound/soc/intel/skylake/bxt-sst.c | 160 ++++++++++++++------- sound/soc/intel/skylake/skl-sst-dsp.c | 253 ++++++++++++++++++++++++---------- sound/soc/intel/skylake/skl-sst-dsp.h | 96 ++++++++----- sound/soc/intel/skylake/skl-sst-ipc.h | 11 ++ sound/soc/intel/skylake/skl-sst.c | 91 +++++++----- 5 files changed, 425 insertions(+), 186 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Add DSP muti-core support
by Vinod Koul 22 Jun '16

22 Jun '16

This series adds support for managing multiple DSP cores. SW is responsible for powering up/down cores in DSP and we do that using refcount. 1st patch introduces core infrastructure change in driver and then adds support in SKL and BXT Jayachandran B (3): ASoC: Intel: Skylake: Add DSP muti-core infrastructure ASoC: Intel: Skylake: Support multi-core in Skylake ASoC: Intel: Skylake: Support multi-core in Broxton sound/soc/intel/skylake/bxt-sst.c | 160 ++++++++++++++------- sound/soc/intel/skylake/skl-sst-dsp.c | 252 ++++++++++++++++++++++++---------- sound/soc/intel/skylake/skl-sst-dsp.h | 96 ++++++++----- sound/soc/intel/skylake/skl-sst-ipc.h | 11 ++ sound/soc/intel/skylake/skl-sst.c | 91 +++++++----- 5 files changed, 424 insertions(+), 186 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: wm_adsp: Disable DMAs before clearing the transfer length
by Charles Keepax 22 Jun '16

22 Jun '16

From: Simon Trimmer <simont(a)opensource.wolfsonmicro.com> This patch reorders the clearing of the DMA masks to avoid potential artefacts being introduced. Signed-off-by: Simon Trimmer <simont(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 430bb8b..21fbe7d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2368,13 +2368,15 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA | ADSP2_CORE_ENA | - ADSP2_START, 0); + ADSP2_CORE_ENA | ADSP2_START, 0); /* Make sure DMAs are quiesced */ + regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0); regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_1, 0); regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_2, 0); - regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0); + + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA, 0); list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: hda/tegra: iomem fixups for sparse warnings
by Ben Dooks 22 Jun '16

22 Jun '16

The readl/writel are not being passed __iomem annotated variables, so fix the following sparse warnings by adding __iomem in: sound/pci/hda/hda_tegra.c:120:9: warning: incorrect type in argument 2 (different address spaces) sound/pci/hda/hda_tegra.c:120:9: expected void volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:120:9: got unsigned int [usertype] *addr sound/pci/hda/hda_tegra.c:125:16: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:125:16: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:125:16: got unsigned int [usertype] *addr sound/pci/hda/hda_tegra.c:134:13: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:134:13: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:134:13: got void *dword_addr sound/pci/hda/hda_tegra.c:137:9: warning: incorrect type in argument 2 (different address spaces) sound/pci/hda/hda_tegra.c:137:9: expected void volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:137:9: got void *dword_addr sound/pci/hda/hda_tegra.c:146:13: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:146:13: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:146:13: got void *dword_addr sound/pci/hda/hda_tegra.c:156:13: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:156:13: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:156:13: got void *dword_addr sound/pci/hda/hda_tegra.c:159:9: warning: incorrect type in argument 2 (different address spaces) sound/pci/hda/hda_tegra.c:159:9: expected void volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:159:9: got void *dword_addr sound/pci/hda/hda_tegra.c:168:13: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:168:13: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:168:13: got void *dword_addr sound/pci/hda/hda_tegra.c:173:23: warning: incorrect type in initializer (incompatible argument 2 (different address spaces)) sound/pci/hda/hda_tegra.c:173:23: expected void ( *reg_writel )( ... ) sound/pci/hda/hda_tegra.c:173:23: got void ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:174:22: warning: incorrect type in initializer (incompatible argument 1 (different address spaces)) sound/pci/hda/hda_tegra.c:174:22: expected unsigned int ( *reg_readl )( ... ) sound/pci/hda/hda_tegra.c:174:22: got unsigned int ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:175:23: warning: incorrect type in initializer (incompatible argument 2 (different address spaces)) sound/pci/hda/hda_tegra.c:175:23: expected void ( *reg_writew )( ... ) sound/pci/hda/hda_tegra.c:175:23: got void ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:176:22: warning: incorrect type in initializer (incompatible argument 1 (different address spaces)) sound/pci/hda/hda_tegra.c:176:22: expected unsigned short ( *reg_readw )( ... ) sound/pci/hda/hda_tegra.c:176:22: got unsigned short ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:177:23: warning: incorrect type in initializer (incompatible argument 2 (different address spaces)) sound/pci/hda/hda_tegra.c:177:23: expected void ( *reg_writeb )( ... ) sound/pci/hda/hda_tegra.c:177:23: got void ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:178:22: warning: incorrect type in initializer (incompatible argument 1 (different address spaces)) sound/pci/hda/hda_tegra.c:178:22: expected unsigned char ( *reg_readb )( ... ) sound/pci/hda/hda_tegra.c:178:22: got unsigned char ( static [toplevel] *<noident> )( ... ) Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: Alexandre Courbot <gnurou(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-tegra(a)vger.kernel.org --- sound/pci/hda/hda_tegra.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 17fd817..0621920 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -115,20 +115,20 @@ static int substream_free_pages(struct azx *chip, /* * Register access ops. Tegra HDA register access is DWORD only. */ -static void hda_tegra_writel(u32 value, u32 *addr) +static void hda_tegra_writel(u32 value, u32 __iomem *addr) { writel(value, addr); } -static u32 hda_tegra_readl(u32 *addr) +static u32 hda_tegra_readl(u32 __iomem *addr) { return readl(addr); } -static void hda_tegra_writew(u16 value, u16 *addr) +static void hda_tegra_writew(u16 value, u16 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); @@ -137,20 +137,20 @@ static void hda_tegra_writew(u16 value, u16 *addr) writel(v, dword_addr); } -static u16 hda_tegra_readw(u16 *addr) +static u16 hda_tegra_readw(u16 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); return (v >> shift) & 0xffff; } -static void hda_tegra_writeb(u8 value, u8 *addr) +static void hda_tegra_writeb(u8 value, u8 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); @@ -159,10 +159,10 @@ static void hda_tegra_writeb(u8 value, u8 *addr) writel(v, dword_addr); } -static u8 hda_tegra_readb(u8 *addr) +static u8 hda_tegra_readb(u8 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); -- 2.8.1

3 3

Re: [alsa-devel] ASoC: nau8825: cross talk suppression measurement function
by Dan Carpenter 22 Jun '16

22 Jun '16

Hello John Hsu, The patch b50455fab459: "ASoC: nau8825: cross talk suppression measurement function" from Jun 7, 2016, leads to the following static checker warning: sound/soc/codecs/nau8825.c:265 nau8825_sema_acquire() warn: 'sem:&nau8825->xtalk_sem' is sometimes locked here and sometimes unlocked. sound/soc/codecs/nau8825.c 240 /** 241 * nau8825_sema_acquire - acquire the semaphore of nau88l25 242 * @nau8825: component to register the codec private data with 243 * @timeout: how long in jiffies to wait before failure or zero to wait 244 * until release 245 * 246 * Attempts to acquire the semaphore with number of jiffies. If no more 247 * tasks are allowed to acquire the semaphore, calling this function will 248 * put the task to sleep. If the semaphore is not released within the 249 * specified number of jiffies, this function returns. 250 * Acquires the semaphore without jiffies. If no more tasks are allowed 251 * to acquire the semaphore, calling this function will put the task to 252 * sleep until the semaphore is released. 253 * It returns if the semaphore was acquired. Sometimes it returns without the lock held. 254 */ 255 static void nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) 256 { 257 int ret; 258 259 if (timeout) 260 ret = down_timeout(&nau8825->xtalk_sem, timeout); 261 else 262 ret = down_interruptible(&nau8825->xtalk_sem); 263 264 if (ret < 0) 265 dev_warn(nau8825->dev, "Acquire semaphone fail\n"); This function makes me sad. Warn and corrupt is my least favourite anti-pattern. 266 } regards, dan carpenter

2 3

[alsa-devel] [PATCH] ASoC: cs53l30: Add MUTE pin control support via GPIO
by Nicolin Chen 22 Jun '16

22 Jun '16

The codec chip has a physical MUTE pin to let users control it via GPIO. So this patch add a mute control support to the driver. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- .../devicetree/bindings/sound/cs53l30.txt | 2 ++ sound/soc/codecs/cs53l30.c | 30 ++++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 1 + 3 files changed, 33 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt index 18d6b99..6f7d3c8 100644 --- a/Documentation/devicetree/bindings/sound/cs53l30.txt +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -13,6 +13,8 @@ Optional properties: - reset-gpios : a GPIO spec for the reset pin. + - mute-gpios : a GPIO spec for the MUTE pin. + - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin. 0 = Hi-Z 1 = 1.80 V diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index b0a64a1..5988b5c 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -35,6 +35,7 @@ struct cs53l30_private { struct regulator_bulk_data supplies[CS53L30_NUM_SUPPLIES]; struct regmap *regmap; struct gpio_desc *reset_gpio; + struct gpio_desc *mute_gpio; struct clk *mclk; bool use_sdout2; u32 mclk_rate; @@ -833,6 +834,16 @@ static int cs53l30_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int cs53l30_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec); + + if (priv->mute_gpio) + gpiod_set_value_cansleep(priv->mute_gpio, mute); + + return 0; +} + /* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ #define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) @@ -846,6 +857,7 @@ static const struct snd_soc_dai_ops cs53l30_ops = { .set_sysclk = cs53l30_set_sysclk, .set_tristate = cs53l30_set_tristate, .set_tdm_slot = cs53l30_set_dai_tdm_slot, + .mute_stream = cs53l30_mute_stream, }; static struct snd_soc_dai_driver cs53l30_dai = { @@ -991,6 +1003,24 @@ static int cs53l30_i2c_probe(struct i2c_client *client, cs53l30->mclk = NULL; } + /* Fetch the MUTE control */ + cs53l30->mute_gpio = devm_gpiod_get_optional(dev, "mute", + GPIOD_OUT_HIGH); + if (IS_ERR(cs53l30->mute_gpio)) { + ret = PTR_ERR(cs53l30->mute_gpio); + goto error; + } + + if (cs53l30->mute_gpio) { + /* Enable MUTE controls via MUTE pin */ + regmap_write(cs53l30->regmap, CS53L30_MUTEP_CTL1, + CS53L30_MUTEP_CTL1_MUTEALL); + /* Flip the polarity of MUTE pin */ + if (gpiod_is_active_low(cs53l30->mute_gpio)) + regmap_update_bits(cs53l30->regmap, CS53L30_MUTEP_CTL2, + CS53L30_MUTE_PIN_POLARITY, 0); + } + if (!of_property_read_u8(np, "cirrus,micbias-lvl", &val)) regmap_update_bits(cs53l30->regmap, CS53L30_MICBIAS_CTL, CS53L30_MIC_BIAS_CTRL_MASK, val); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h index 0dd4afb..5e39da5 100644 --- a/sound/soc/codecs/cs53l30.h +++ b/sound/soc/codecs/cs53l30.h @@ -253,6 +253,7 @@ #define CS53L30_MUTE_MB_ALL_PDN_MASK (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) #define CS53L30_MUTE_MB_ALL_PDN (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) +#define CS53L30_MUTEP_CTL1_MUTEALL (0xdf) #define CS53L30_MUTEP_CTL1_DEFAULT (0) /* R32 (0x20) CS53L30_MUTEP_CTL2 - MUTE Pin Control 2 */ -- 2.1.4

2 2

[alsa-devel] [PATCH -next] ASoC: max9860: fix non static symbol warnings
by weiyj_lk＠163.com 21 Jun '16

21 Jun '16

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warnings: sound/soc/codecs/max9860.c:120:28: warning: symbol 'max9860_regmap' was not declared. Should it be static? sound/soc/codecs/max9860.c:596:25: warning: symbol 'max9860_pm_ops' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/max9860.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index 2b0dd6a..68074c9 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -117,7 +117,7 @@ static bool max9860_precious(struct device *dev, unsigned int reg) return false; } -const struct regmap_config max9860_regmap = { +static const struct regmap_config max9860_regmap = { .reg_bits = 8, .val_bits = 8, @@ -593,7 +593,7 @@ static int max9860_resume(struct device *dev) } #endif -const struct dev_pm_ops max9860_pm_ops = { +static const struct dev_pm_ops max9860_pm_ops = { SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL) };

2 1

[alsa-devel] [PATCH -next] ASoC: cs53l30: Fix non static symbol warnings
by weiyj_lk＠163.com 21 Jun '16

21 Jun '16

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warnings: sound/soc/codecs/cs53l30.c:182:20: warning: symbol 'input1_sel_values' was not declared. Should it be static? sound/soc/codecs/cs53l30.c:202:20: warning: symbol 'input2_sel_values' was not declared. Should it be static? sound/soc/codecs/cs53l30.c:734:20: warning: symbol 'cs53l30_src_rates' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/cs53l30.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 714e579..a9949ad 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -179,7 +179,7 @@ static const char * const input1_sel_text[] = { "DMIC1 Off ADC1 Off", }; -unsigned int const input1_sel_values[] = { +static unsigned int const input1_sel_values[] = { CS53L30_CH_TYPE, CS53L30_ADCxB_PDN | CS53L30_CH_TYPE, CS53L30_ADCxA_PDN | CS53L30_CH_TYPE, @@ -199,7 +199,7 @@ static const char * const input2_sel_text[] = { "DMIC2 Off ADC2 Off", }; -unsigned int const input2_sel_values[] = { +static unsigned int const input2_sel_values[] = { 0x0, CS53L30_ADCxB_PDN, CS53L30_ADCxA_PDN, @@ -731,7 +731,7 @@ static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) CS53L30_ASP_3ST_MASK, val); } -unsigned int const cs53l30_src_rates[] = { +static unsigned int const cs53l30_src_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 };

4 3

[alsa-devel] [PATCH] ASoC: cs53l30: Set idle_bias_off true
by Nicolin Chen 21 Jun '16

21 Jun '16

The driver is using the set_bias_level to control the power on and off so it should get SND_SOC_BIAS_OFF in order to proceed normal powering sequences. This patch enables the idle_bias_off option so the DAPM core will set the bias level to SND_SOC_BIAS_OFF instead of stopping at SND_SOC_BIAS_STANDBY. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/cs53l30.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index a5976f3..b0a64a1 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -879,6 +879,7 @@ static int cs53l30_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver cs53l30_driver = { .probe = cs53l30_codec_probe, .set_bias_level = cs53l30_set_bias_level, + .idle_bias_off = true, .dapm_widgets = cs53l30_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), -- 2.1.4

3 2

[alsa-devel] [PATCH] ASoC: dwc: make pcm support built-in when necessary
by Arnd Bergmann 21 Jun '16

21 Jun '16

The new PIO mode for the dwc audio driver causes a link failure when it is built as a loadable module but the audio driver is built-in: sound/built-in.o: In function `i2s_irq_handler': :(.text+0x58c64): undefined reference to `dw_pcm_push_tx' sound/built-in.o: In function `dw_i2s_probe': :(.text+0x593dc): undefined reference to `dw_pcm_register' We could link both into a single module, but apparently the author intended them to be separate, so this instead changes the Makefile to force the pcm module to be built-in if the base module is. This is a bit hacky but not as bad as trying to work around it in Kconfig language. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 79361b2b98b7 ("ASoC: dwc: Add PIO PCM extension") --- sound/soc/dwc/Makefile | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile index 1b48bcccbc51..38f1ca31c5fa 100644 --- a/sound/soc/dwc/Makefile +++ b/sound/soc/dwc/Makefile @@ -1,4 +1,5 @@ # SYNOPSYS Platform Support obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o -obj-$(CONFIG_SND_DESIGNWARE_PCM) += designware_pcm.o - +ifdef CONFIG_SND_DESIGNWARE_PCM +obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_pcm.o +endif -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: rt5670: patch reg-0x8a
by Bard Liao 21 Jun '16

21 Jun '16

reg-8a assign the tracking source for each ASRC tracker. The default value is 0x0000 which means all ASRC trackers will track LRCK1. But in most case, we wish each ASRC tracker track the corresponding LRCK. i.e. ASRC1 tracks LRCK1, ASRC2 tracks LRCK2 and so on. So, we rewrite reg-8a as 0x0123. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0af5ddb..8ef467f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -55,6 +55,7 @@ static const struct reg_sequence init_list[] = { { RT5670_PR_BASE + 0x14, 0x9a8a }, { RT5670_PR_BASE + 0x38, 0x3ba1 }, { RT5670_PR_BASE + 0x3d, 0x3640 }, + { 0x8a, 0x0123 }, }; static const struct reg_default rt5670_reg[] = { @@ -131,7 +132,7 @@ static const struct reg_default rt5670_reg[] = { { 0x87, 0x0000 }, { 0x88, 0x0000 }, { 0x89, 0x0000 }, - { 0x8a, 0x0000 }, + { 0x8a, 0x0123 }, { 0x8b, 0x0000 }, { 0x8c, 0x0003 }, { 0x8d, 0x0000 }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [RESEND PATCH v3 0/2] ASoC: da7219: Convert driver to use generic FW functions
by Adam Thomson 21 Jun '16

21 Jun '16

This patch set converts the da7219 codec driver to use device/fwnode functions to access properties, instead of the DT only of_* functions, allowing ACPI to be used as well. The DT bindings for da7219 have a device node for the main codec properties, and a named child node (da7219_aad), which contains all of the accessory detection related properties for the device. ACPI also supports this kind of FW hierarchy (data only sub-nodes), but some support in the kernel needs to be added to take make use of this in driver code. The first patch adds functions to allow searching for a named child node of a device, for both DT and ACPI, and the second patch updates the codec driver to use the standard device/fwnode calls, including this new function. These changes are based on the v4.7-rc3 kernel. Changes in v3: - Use of_node_cmp() in device_get_named_child_node() to match DT node. Changes in v2: - Rebase to v4.7-rc1 - Small updates to codec patch based on previous reviewer comments Adam Thomson (2): device property: Add function to search for named child of device ASoC: da7219: Convert driver to use generic device/fwnode functions drivers/base/property.c | 28 ++++++++++++ include/acpi/acpi_bus.h | 7 +++ include/linux/acpi.h | 6 +++ include/linux/of.h | 14 +++--- include/linux/property.h | 3 ++ sound/soc/codecs/da7219-aad.c | 103 +++++++++++++++++++++--------------------- sound/soc/codecs/da7219.c | 34 +++++++------- 7 files changed, 119 insertions(+), 76 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH] gcc6 reporting error building alsa-tools on PowerPC
by Michel Normand 21 Jun '16

21 Jun '16

Hello there, While trying to build alsa-tools with gcc6, many reported error that are solved by proposed patch. I do not know if sending to this mailing list is the correct way to provide it to you. Michel Normand (1): gcc6 narrowing error hdspmixer/src/channelmap.cxx | 158 +++++++++++++++++++++---------------------- us428control/Cus428Midi.cc | 30 ++++---- 2 files changed, 94 insertions(+), 94 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] INSTALL: update text and drop reference to Hg
by Henrik Austad 21 Jun '16

21 Jun '16

This is really just nitpick, but it annoyed me for a few seconds, so I thought I'd just fix it. In d7534b2ceec7 (hgcompile -> gitcompile) the filename was updated but INSTALL was not, leaving outdated docs. Signed-off-by: Henrik Austad <henrik(a)austad.us> --- INSTALL | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/INSTALL b/INSTALL index 47086e3..64f3d09 100644 --- a/INSTALL +++ b/INSTALL @@ -11,13 +11,19 @@ For installation you can use these commands: make install -Compilation from HG sources +Compilation from Git-sources --------------------------- -You need also GNU packages automake and libtool installed in your system -to compile HG (Mercurial) sources of alsa-lib package. +Requirements: + * automake + * libtool -For compilation you can use these commands: +To compile alsa-lib from source: + + ./gitcompile + +If you do not want to use the supplied gitcompile-script, you can +compile alsa-lib manually by using these steps: libtoolize --force --copy --automake aclocal @@ -27,7 +33,6 @@ For compilation you can use these commands: ./configure make -The included hgcompile script does this job for you. Note: Some automake packages have missing aclocal program. Use newer version in the case. -- 2.1.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

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Re: [alsa-devel] 答复: [PATCH 1/1] sound: Fix compile error of seq_device.c
by Takashi Iwai 21 Jun '16

21 Jun '16

On Tue, 21 Jun 2016 10:48:30 +0200, 高峰 wrote: > > The lastest CentOS7 platform, and the gcc version is gcc version 4.8.5 > 20150623 (Red Hat 4.8.5-4) (GCC). > It could pass compilation after append the lost semicolon. > > from sound/core/seq/seq_device.c:39: > include/linux/init.h:216:2: error: expected ‘,’ or ‘;’ before ‘static’ > static exitcall_t __exitcall_##fn __exit_call = fn > ^ > include/linux/init.h:279:24: note: in expansion of macro ‘__exitcall’ > #define module_exit(x) __exitcall(x); > ^ > sound/core/seq/seq_device.c:315:1: note: in expansion of macro > ‘module_exit’ > module_exit(alsa_seq_device_exit) > ^ > make[3]: *** [sound/core/seq/seq_device.o] Error 1 > make[2]: *** [sound/core/seq] Error 2 > make[1]: *** [sound/core] Error 2 > make: *** [sound] Error 2 It's a RH specific issue. The upstream code has a different definition of module_init(), thus no such an error would occur. Note that I'm fine to apply the patch, it's a trivial change. But you need to give the clear reason why to apply it. In this case, it's no fault of the upstream code. But if it would make someone's life a bit easier, it's OK to apply such a change. That being said, please resubmit the patch with a more explanation. thanks, Takashi > > -----邮件原件----- > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > 发送时间: 2016年6月21日 16:45 > 收件人: fgao(a)ikuai8.com > 抄送: alsa-devel(a)alsa-project.org; perex(a)perex.cz; linux-kernel(a)vger.kernel. > org; gfree.wind(a)gmail.com > 主题: Re: [PATCH 1/1] sound: Fix compile error of seq_device.c > > On Tue, 21 Jun 2016 10:34:25 +0200, > <fgao(a)ikuai8.com> wrote: > > > > From: Gao Feng <fgao(a)ikuai8.com> > > > > Signed-off-by: Gao Feng <fgao(a)ikuai8.com> > > What compile error did you get? > > > Takashi > > > --- > > sound/core/seq/seq_device.c | 4 ++-- > > 1 file changed, 2 insertions(+), 2 deletions(-) > > > > diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c > > index c4acf17..4e859e4 100644 > > --- a/sound/core/seq/seq_device.c > > +++ b/sound/core/seq/seq_device.c > > @@ -311,5 +311,5 @@ static void __exit alsa_seq_device_exit(void) > > bus_unregister(&snd_seq_bus_type); > > } > > > > -subsys_initcall(alsa_seq_device_init) > > -module_exit(alsa_seq_device_exit) > > +subsys_initcall(alsa_seq_device_init); > > +module_exit(alsa_seq_device_exit); > > -- > > 1.9.1 > > > > > > > > > > > > >

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[alsa-devel] [PATCH RFC 0/6] Sound support for Exynos5433 TM2(E) boards
by Sylwester Nawrocki 21 Jun '16

21 Jun '16

This patch series adds sound supoort for the Exynos 5433 based TM2(E) boards. It includes minimal drivers for the Low Power Audio Subsystem (LPASS) and the MAX98504 class D speaker amplifier as well as the actual asoc machine driver. Inha Song (2): ASoC: samsung: Add Samsung Low Power Audio Subsystem driver ASoC: samsung: Add machine driver for Exynos5433 based TM2 board Sylwester Nawrocki (4): ASoC: samsung: Add DT bindings documentation for LPASS ASoC: max98504: Add max98504 speaker amplifier driver ASoC: Add DT bindings documentation for max98504 amplifier ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem .../devicetree/bindings/sound/max98504.txt | 44 ++ .../bindings/sound/samsung,exynos5433-lpass.txt | 8 + .../bindings/sound/samsung,tm2-wm5110.txt | 39 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98504.c | 241 +++++++++ sound/soc/codecs/max98504.h | 94 ++++ sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 4 + sound/soc/samsung/lpass.c | 162 ++++++ sound/soc/samsung/lpass.h | 47 ++ sound/soc/samsung/tm2_wm5110.c | 553 +++++++++++++++++++++ 12 files changed, 1211 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,exynos5433-lpass.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-wm5110.txt create mode 100644 sound/soc/codecs/max98504.c create mode 100644 sound/soc/codecs/max98504.h create mode 100644 sound/soc/samsung/lpass.c create mode 100644 sound/soc/samsung/lpass.h create mode 100644 sound/soc/samsung/tm2_wm5110.c -- 1.9.1

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[alsa-devel] [PATCH 1/1] sound: Fix one compile error of seq_device.c
by fgao＠ikuai8.com 21 Jun '16

21 Jun '16

From: Gao Feng <fgao(a)ikuai8.com> The gcc reports the following compile error on CentOS7. include/linux/init.h:216:2: error: expected ',' or ';' before 'static' static exitcall_t __exitcall_##fn __exit_call = fn ^ include/linux/init.h:279:24: note: in expansion of macro '__exitcall' #define module_exit(x) __exitcall(x); ^ sound/core/seq/seq_device.c:315:1: note: in expansion of macro 'module_exit' module_exit(alsa_seq_device_exit) ^ make[3]: *** [sound/core/seq/seq_device.o] Error 1 make[2]: *** [sound/core/seq] Error 2 make[1]: *** [sound/core] Error 2 make: *** [sound] Error 2 Signed-off-by: Gao Feng <fgao(a)ikuai8.com> --- sound/core/seq/seq_device.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index c4acf17..4e859e4 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -311,5 +311,5 @@ static void __exit alsa_seq_device_exit(void) bus_unregister(&snd_seq_bus_type); } -subsys_initcall(alsa_seq_device_init) -module_exit(alsa_seq_device_exit) +subsys_initcall(alsa_seq_device_init); +module_exit(alsa_seq_device_exit); -- 1.9.1

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[alsa-devel] [PATCH 1/1] sound: Fix compile error of seq_device.c
by fgao＠ikuai8.com 21 Jun '16

21 Jun '16

From: Gao Feng <fgao(a)ikuai8.com> Signed-off-by: Gao Feng <fgao(a)ikuai8.com> --- sound/core/seq/seq_device.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index c4acf17..4e859e4 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -311,5 +311,5 @@ static void __exit alsa_seq_device_exit(void) bus_unregister(&snd_seq_bus_type); } -subsys_initcall(alsa_seq_device_init) -module_exit(alsa_seq_device_exit) +subsys_initcall(alsa_seq_device_init); +module_exit(alsa_seq_device_exit); -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Remove compilation warning
by Helen Koike 21 Jun '16

21 Jun '16

Remove the warning: warning: ‘err’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Helen Koike <helen.koike(a)collabora.co.uk> --- sound/hda/hdac_regmap.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 87041dd..7be6ca1 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -279,7 +279,7 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) { struct hdac_device *codec = context; unsigned int verb; - int i, bytes, err; + int i, bytes, err = 0; int pm_lock = 0; if (codec->caps_overwriting) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: intel: fix build when ACPI is not enabled
by Randy Dunlap 21 Jun '16

21 Jun '16

From: Randy Dunlap <rdunlap(a)infradead.org> kconfig tools generate the following warning when CONFIG_ACPI is not enabled: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_BYTCR_RT5651_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) causing these build errors: In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:40:0: ../include/acpi/acpi_bus.h:65:20: error: conflicting types for 'acpi_evaluate_dsm' union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:31:0: ../include/linux/acpi.h:676:34: note: previous definition of 'acpi_evaluate_dsm' was here static inline union acpi_object *acpi_evaluate_dsm(acpi_handle handle, I am told that ACPI is a requirement for these drivers, so make that explicit. Also end help text sentences with a period. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/Kconfig | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) Should SND_SOC_INTEL_SST and possibly other similar drivers get this same change? --- linux-next-20160607.orig/sound/soc/intel/Kconfig +++ linux-next-20160607/sound/soc/intel/Kconfig @@ -128,28 +128,28 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5640 codec" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI - select SND_SOC_INTEL_SST_MATCH if ACPI + select SND_SOC_INTEL_SST_MATCH help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR platforms with RT5640 audio codec. - Say Y if you have such a device + Say Y if you have such a device. If unsure select "N". config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5651 codec" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5651 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI - select SND_SOC_INTEL_SST_MATCH if ACPI + select SND_SOC_INTEL_SST_MATCH help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR platforms with RT5651 audio codec. - Say Y if you have such a device + Say Y if you have such a device. If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5672_MACH

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[alsa-devel] How does ALSALIB commuciates with low level kernel API
by Chemsi Mehdi 20 Jun '16

20 Jun '16

Dear experts, I am too new in ALSA, trying to read a little bit and understand. I want to underatnd how ALSA userspace API calls kernel space code ? for example when I call snd_pcm_open() which code is called in kernel sound code ? and how the link is made ? via IOCTL/write ? Please clarify Thanks a lot in advance. Mehdi

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Re: [alsa-devel] [PATCH] ASoC: rt5668: add rt5668 codec driver
by Mark Brown 20 Jun '16

20 Jun '16

On Thu, Jun 16, 2016 at 11:31:07AM +0000, Jack Yu wrote: > [Mark] Given that the interrupt handler just schedules a work item there's no > need for it to be a threaded interrupt, it can just be a normal > interrupt and save the bother of masking the hardware interrupt and > running a thread. > Do you mean that it's not appropriate to use " queue_delayed_work" here, we should > use other method for handling work item ? No, when I said it shouldn't be a threaded interrupt I meant that using request_threaded_irq() doesn't seem appropriate.

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[alsa-devel] [PATCH] ASoC: core: Remove selects of regmap buses
by Mark Brown 20 Jun '16

20 Jun '16

We used to have code wrapping the regmap I2C and SPI buses as part of the conversion of ASoC drivers from using the ASoC level I/O to the regmap level I/O. Since all drivers have now been converted and the core code removed we can remove the per-bus selects and instead just select the regmap core. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 182d92efc7c8..4935e23ff3d6 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -7,8 +7,7 @@ menuconfig SND_SOC select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS select SND_JACK - select REGMAP_I2C if I2C - select REGMAP_SPI if SPI_MASTER + select REGMAP ---help--- If you want ASoC support, you should say Y here and also to the -- 2.8.1

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[alsa-devel] [GIT PULL]: Skylake audio firmware update
by Vinod Koul 20 Jun '16

20 Jun '16

Hi, Please PULL to receive the latest Skylake audio firmware. The following changes since commit a4bbc8112cc04fcc0e0861e8b72197b6cb83ab83: linux-firmware: Update firmware file for Intel Bluetooth 8260 (2016-06-09 08:11:22 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to 80aae2270140807c4d8e304c39def1fe9f93edb9: linux-firmware: intel: Update Skylake audio firmware (2016-06-17 14:01:52 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v948.bin | Bin 0 -> 221184 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_release_v948.bin Thanks -- ~Vinod

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[alsa-devel] [PATCH] ASoC: Intel: Add support for PM ops in bxt-da7219_max98357a
by Vinod Koul 20 Jun '16

20 Jun '16

We need card to be early suspended and late resumed, so use prepare and complete for card suspend and resume. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Tested-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3774b117d365..df5f269a9b67 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -441,11 +441,30 @@ static int broxton_audio_probe(struct platform_device *pdev) return devm_snd_soc_register_card(&pdev->dev, &broxton_audio_card); } +#ifdef CONFIG_PM_SLEEP + +static void broxton_complete(struct device *dev) +{ + snd_soc_resume(dev); +} + +#else +#define broxton_complete NULL +#endif + +static const struct dev_pm_ops broxton_pm_ops = { + .prepare = snd_soc_suspend, + .complete = broxton_complete, + .freeze = snd_soc_suspend, + .thaw = snd_soc_resume, + .poweroff = snd_soc_poweroff, + .restore = snd_soc_resume, +}; static struct platform_driver broxton_audio = { .probe = broxton_audio_probe, .driver = { .name = "bxt_da7219_max98357a_i2s", - .pm = &snd_soc_pm_ops, + .pm = &broxton_pm_ops, }, }; module_platform_driver(broxton_audio) -- 1.9.1

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[alsa-devel] [PATCH v1 1/2] ASoC: Add DT bindings documentation for max98504 amplifier
by Sylwester Nawrocki 20 Jun '16

20 Jun '16

This patch adds DT bindings documentation for Maxim MAX98504 speaker amplifier. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since initial version: - dropped all optional properties used for setting up TDM and speaker data source, - added regulator supply properties. --- .../devicetree/bindings/sound/max98504.txt | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt diff --git a/Documentation/devicetree/bindings/sound/max98504.txt b/Documentation/devicetree/bindings/sound/max98504.txt new file mode 100644 index 0000000..7aa5b93 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98504.txt @@ -0,0 +1,22 @@ +Maxim MAX98504 class D mono speaker amplifier + +This device supports I2C control interface and an IRQ output signal. It features +a PCM and PDM digital audio interface (DAI) and a differential analog input. + +Required properties: + + - compatible : "maxim,max98504" + - reg : should contain the I2C slave device address + - DVDD-supply, DIOVDD-supply, PVDD-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + + max98504: max98504@31 { + compatible = "maxim,max98504"; + reg = <0x31>; + + DVDD-supply = <&regulator>; + DIOVDD-supply = <&regulator>; + PVDD-supply = <&regulator>; +}; -- 1.9.1

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[alsa-devel] [PATCH] alsa-tools gcc6 narrowing error
by Michel Normand 20 Jun '16

20 Jun '16

Trying to build alsa-tools with gcc6 for PowerPC architecture. It failed as detailed in proposed patch of next mail Michel Normand (1): gcc6 narrowing error hdspmixer/src/channelmap.cxx | 158 +++++++++++++++++++++---------------------- us428control/Cus428Midi.cc | 30 ++++---- 2 files changed, 94 insertions(+), 94 deletions(-) -- 1.9.1

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Re: [alsa-devel] [PATCH 2/2] ASoc: wm8731: add normal mode with 12MHz XTAL
by Matt Flax 20 Jun '16

20 Jun '16

I have realised that this second patch may not be necessary. It is also likely that it will skew the actual sample rate. Charles, can you please advise me on the following... I notice that I can probably get the correct sample rates by spoofing the crystal frequencies in my machine driver. For example ... The ratio between the USB and Normal mode fs multipliers with BOSR=0 is : 250/256 = 0.97656 Similarly, I can match this with the following crystal frequency ratio (crystal f in MHz) : 12/12.288 = 0.97656 By induction, I assume that I can spoof a 12.288 clock with BOSR=0 for sample rate = 48kHz, 96 kHz when using a 12 MHz crystal. Again, I can match ratios between the following crystal frequecies (in MHz) : 12/16.9344 = 0.70862 With BOSR = 1, I assume that can get 44.1 kHz and 88.2 kHz because the fs ratio between normal and usb mode is : 272/384 = 0.70833 Now these two ratios are skew by 0.41% which is equvalent to the skew in 12 MHz USB mode clocks of 44.1kHz to 44.118kHz as reported in the wm8731 datasheet. Can you advise me whether my idea of spoofing clocks in the machine driver is perhaps a better way of acquiring the correct sample rates, rather then this patch ? thanks Matt On 10/06/16 10:11, Matt Flax wrote: > This patch allows a machine to set normal mode when using a > 12 MHz crystal. In USB_MODE the 12 MHz crystal signal is output on > the BCLK pin. In NORMAL_MODE the bit clock is output on the BCLK pin. > The previously ignored direction variable input to the > wm8731_set_dai_sysclk function is used to indicate whether crystal > or bit clock is output on the BCLK codec pin. > I have ensured that this does not effect the db1200.c machine driver > (the only other driver to use a 12 MHz crystal in USB_MODE). It also > does not effect the other machine drivers which use the wm8731 codec > as they don't use USB_MODE nor 12 MHz crystals. > > When driving the wm8731 in normal mode with a 12 MHz crystal, the > DSP filters are expected to be effected. If the filters are activated > then it is expected that the filter cut offs are shifted by 2.3% at > 250 fs and 29.1% at 272 fs. The assumption is that the filters are > linear, which implies a linear shift in rate scaling effects the > filter cut off frequencies linearly. If the filters are not active, > there will be no effects. > > Signed-off-by: Matt Flax <flatmax(a)flatmax.org> > --- > sound/soc/codecs/wm8731.c | 14 ++++++++++++-- > 1 file changed, 12 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c > index d18261a..ec98d5a 100644 > --- a/sound/soc/codecs/wm8731.c > +++ b/sound/soc/codecs/wm8731.c > @@ -5,6 +5,7 @@ > * Copyright 2006-12 Wolfson Microelectronics, plc > * > * Author: Richard Purdie <richard(a)openedhand.com> > + * Author: Matt Flax <flatmax(a)flatmax.com> 12Mhz XTAL normal mode and 32bit mode. > * > * Based on wm8753.c by Liam Girdwood > * > @@ -53,6 +54,7 @@ struct wm8731_priv { > int sysclk_type; > int playback_fs; > bool deemph; > + bool bclk_12_mhz; > > struct mutex lock; > }; > @@ -341,8 +343,11 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, > struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); > u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3; > int i = get_coeff(wm8731->sysclk, params_rate(params)); > - u16 srate = (coeff_div[i].sr << 2) | > - (coeff_div[i].bosr << 1) | coeff_div[i].usb; > + u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1); > + > + /* this determines whether to output bit or crystal clk */ > + if (wm8731->bclk_12_mhz) > + srate |= coeff_div[i].usb; > > wm8731->playback_fs = params_rate(params); > > @@ -420,6 +425,11 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, > > wm8731->sysclk = freq; > > + if (dir == SND_SOC_CLOCK_IN) /* output the 12 MHz clock */ > + wm8731->bclk_12_mhz = 1; > + else /* SND_SOC_CLOCK_OUT, output the bit clock */ > + wm8731->bclk_12_mhz = 0; > + > snd_soc_dapm_sync(dapm); > > return 0;

3 9

[alsa-devel] [PATCH] ASoC: compress: Pass error out of soc_compr_pointer
by Charles Keepax 20 Jun '16

20 Jun '16

Both soc_compr_pointer and the platform driver pointer callback return ints but current soc_compr_pointer always returns 0. Update this so we return the actual value from the platform driver callback. This doesn't fix any issues simply makes the code more consistent. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-compress.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 875733c..d2df46c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -530,14 +530,15 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) - platform->driver->compr_ops->pointer(cstream, tstamp); + ret = platform->driver->compr_ops->pointer(cstream, tstamp); mutex_unlock(&rtd->pcm_mutex); - return 0; + return ret; } static int soc_compr_copy(struct snd_compr_stream *cstream, -- 2.1.4

1 0

[alsa-devel] [PATCH 1/1] ASoC: pcm5102a: Support 352k8 and 384k sample rates
by Clive Messer 19 Jun '16

19 Jun '16

Add support for 352k8 and 384k samples rates to pcm5102a codec driver, with SNDRV_PCM_RATE_KNOT and a constraint list. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/codecs/pcm5102a.c | 38 +++++++++++++++++++++++++++++++++++++- 1 file changed, 37 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c index ed51567..53dfc99 100644 --- a/sound/soc/codecs/pcm5102a.c +++ b/sound/soc/codecs/pcm5102a.c @@ -20,16 +20,52 @@ #include <sound/soc.h> +static const u32 pcm5102a_rates[] = { + 8000, 16000, 32000, 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000, +}; + +static const struct snd_pcm_hw_constraint_list pcm5102a_constraint_rates = { + .count = ARRAY_SIZE(pcm5102a_rates), + .list = pcm5102a_rates, +}; + +static int pcm5102a_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm5102a_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + dev_dbg(codec->dev, "%s: set rates (8k-384k) constraint\n", __func__); + + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &pcm5102a_constraint_rates); + if (ret != 0) { + dev_err(codec->dev, "%s: Failed to set rates constraint: %d\n", + __func__, ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_dai_ops pcm5102a_dai_ops = { + .startup = pcm5102a_dai_startup, +}; + static struct snd_soc_dai_driver pcm5102a_dai = { .name = "pcm5102a-hifi", .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, + .ops = &pcm5102a_dai_ops, }; static struct snd_soc_codec_driver soc_codec_dev_pcm5102a; -- 2.5.5

1 1

[alsa-devel] [PATCH 1/1] ASoC: pcm512x: Add 352k8 sample rate support
by Clive Messer 19 Jun '16

19 Jun '16

Add 352k8 sample rate to the slave constraint list. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/codecs/pcm512x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 047c489..eca6648 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -480,7 +480,7 @@ static unsigned long pcm512x_ncp_target(struct pcm512x_priv *pcm512x, static const u32 pcm512x_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, - 88200, 96000, 176400, 192000, 384000, + 88200, 96000, 176400, 192000, 352800, 384000, }; static const struct snd_pcm_hw_constraint_list constraints_slave = { -- 2.5.5

1 0

[alsa-devel] [asoc:for-next 8/14] cygnus-pcm.c:undefined reference to `bad_dma_ops'
by kbuild test robot 19 Jun '16

19 Jun '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 1a7d46a9dfeb578ece0a707f9d97f7bd097cd4a4 commit: 49bf948b5c0b766a8ecd30537aae302997a23487 [8/14] Merge remote-tracking branches 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/cs53l30' and 'asoc/topic/cygnus' into asoc-next config: m32r-allyesconfig (attached as .config) compiler: m32r-linux-gcc (GCC) 4.9.0 reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 49bf948b5c0b766a8ecd30537aae302997a23487 # save the attached .config to linux build tree make.cross ARCH=m32r All errors (new ones prefixed by >>): drivers/built-in.o: In function `zynq_fpga_ops_write': zynq-fpga.c:(.text+0xeff4ec): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeff4f4): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeff614): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcb4): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcbc): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): undefined reference to `dma_common_mmap' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): relocation truncated to fit: R_M32R_26_PCREL_RELA against undefined symbol `dma_common_mmap' sound/built-in.o: In function `cygnus_pcm_preallocate_dma_buffer': >> cygnus-pcm.c:(.text+0x10d2b0): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10d2b4): undefined reference to `bad_dma_ops' sound/built-in.o: In function `cygnus_dma_free_dma_buffers': cygnus-pcm.c:(.text+0x10d46c): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10d470): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10d47c): undefined reference to `bad_dma_ops' sound/built-in.o:cygnus-pcm.c:(.text+0x10d480): more undefined references to `bad_dma_ops' follow --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: seq_timer: use monotonic times internally
by Arnd Bergmann 17 Jun '16

17 Jun '16

The sequencer client manager reports timestamps in units of unsigned 32-bit seconds/nanoseconds, but that does not suffer from the y2038 overflow because it stores only the delta since the 'last_update' time was recorded. However, the use of the do_gettimeofday() function is problematic and we have to replace it to avoid the overflow on on 32-bit architectures. This uses 'struct timespec64' to record 'last_update', and changes the code to use monotonic timestamps that do not suffer from leap seconds and settimeofday updates. As a side-effect, the code can now use the timespec64_sub() helper and become more readable and also avoid a multiplication to convert from microseconds to nanoseconds. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/core/seq/seq_timer.c | 23 +++++++++-------------- sound/core/seq/seq_timer.h | 2 +- 2 files changed, 10 insertions(+), 15 deletions(-) diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 293104926098..dcc102813aef 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -165,7 +165,7 @@ static void snd_seq_timer_interrupt(struct snd_timer_instance *timeri, snd_seq_timer_update_tick(&tmr->tick, resolution); /* register actual time of this timer update */ - do_gettimeofday(&tmr->last_update); + ktime_get_ts64(&tmr->last_update); spin_unlock_irqrestore(&tmr->lock, flags); @@ -392,7 +392,7 @@ static int seq_timer_start(struct snd_seq_timer *tmr) return -EINVAL; snd_timer_start(tmr->timeri, tmr->ticks); tmr->running = 1; - do_gettimeofday(&tmr->last_update); + ktime_get_ts64(&tmr->last_update); return 0; } @@ -420,7 +420,7 @@ static int seq_timer_continue(struct snd_seq_timer *tmr) } snd_timer_start(tmr->timeri, tmr->ticks); tmr->running = 1; - do_gettimeofday(&tmr->last_update); + ktime_get_ts64(&tmr->last_update); return 0; } @@ -444,17 +444,12 @@ snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr) spin_lock_irqsave(&tmr->lock, flags); cur_time = tmr->cur_time; if (tmr->running) { - struct timeval tm; - int usec; - do_gettimeofday(&tm); - usec = (int)(tm.tv_usec - tmr->last_update.tv_usec); - if (usec < 0) { - cur_time.tv_nsec += (1000000 + usec) * 1000; - cur_time.tv_sec += tm.tv_sec - tmr->last_update.tv_sec - 1; - } else { - cur_time.tv_nsec += usec * 1000; - cur_time.tv_sec += tm.tv_sec - tmr->last_update.tv_sec; - } + struct timespec64 tm; + + ktime_get_ts64(&tm); + tm = timespec64_sub(tm, tmr->last_update); + cur_time.tv_nsec = tm.tv_nsec; + cur_time.tv_sec = tm.tv_sec; snd_seq_sanity_real_time(&cur_time); } spin_unlock_irqrestore(&tmr->lock, flags); diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h index 88dfb71805ae..9506b661fe5b 100644 --- a/sound/core/seq/seq_timer.h +++ b/sound/core/seq/seq_timer.h @@ -52,7 +52,7 @@ struct snd_seq_timer { unsigned int skew; unsigned int skew_base; - struct timeval last_update; /* time of last clock update, used for interpolation */ + struct timespec64 last_update; /* time of last clock update, used for interpolation */ spinlock_t lock; }; -- 2.9.0

2 1

[alsa-devel] [PATCH] ALSA: ctxfi: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 17 Jun '16

17 Jun '16

For readability and to allow for structure randomisation, replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/pci/ctxfi/cthw20k2.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 9dc2950..6414ecf 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1615,23 +1615,23 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info) int i; struct regs_cs4382 cs_read = {0}; struct regs_cs4382 cs_def = { - 0x00000001, /* Mode Control 1 */ - 0x00000000, /* Mode Control 2 */ - 0x00000084, /* Mode Control 3 */ - 0x00000000, /* Filter Control */ - 0x00000000, /* Invert Control */ - 0x00000024, /* Mixing Control Pair 1 */ - 0x00000000, /* Vol Control A1 */ - 0x00000000, /* Vol Control B1 */ - 0x00000024, /* Mixing Control Pair 2 */ - 0x00000000, /* Vol Control A2 */ - 0x00000000, /* Vol Control B2 */ - 0x00000024, /* Mixing Control Pair 3 */ - 0x00000000, /* Vol Control A3 */ - 0x00000000, /* Vol Control B3 */ - 0x00000024, /* Mixing Control Pair 4 */ - 0x00000000, /* Vol Control A4 */ - 0x00000000 /* Vol Control B4 */ + .mode_control_1 = 0x00000001, /* Mode Control 1 */ + .mode_control_2 = 0x00000000, /* Mode Control 2 */ + .mode_control_3 = 0x00000084, /* Mode Control 3 */ + .filter_control = 0x00000000, /* Filter Control */ + .invert_control = 0x00000000, /* Invert Control */ + .mix_control_P1 = 0x00000024, /* Mixing Control Pair 1 */ + .vol_control_A1 = 0x00000000, /* Vol Control A1 */ + .vol_control_B1 = 0x00000000, /* Vol Control B1 */ + .mix_control_P2 = 0x00000024, /* Mixing Control Pair 2 */ + .vol_control_A2 = 0x00000000, /* Vol Control A2 */ + .vol_control_B2 = 0x00000000, /* Vol Control B2 */ + .mix_control_P3 = 0x00000024, /* Mixing Control Pair 3 */ + .vol_control_A3 = 0x00000000, /* Vol Control A3 */ + .vol_control_B3 = 0x00000000, /* Vol Control B3 */ + .mix_control_P4 = 0x00000024, /* Mixing Control Pair 4 */ + .vol_control_A4 = 0x00000000, /* Vol Control A4 */ + .vol_control_B4 = 0x00000000 /* Vol Control B4 */ }; if (hw->model == CTSB1270) { -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ALSA: usb-audio: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 17 Jun '16

17 Jun '16

To allow for structure randomisation, replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- Changes in v2: -Modified commit message sound/usb/mixer_maps.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 1f8fb0d9..9038b2e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -107,8 +107,10 @@ static struct usbmix_name_map extigy_map[] = { * e.g. no Master and fake PCM volume * Pavel Mihaylov <bin(a)bash.info> */ -static struct usbmix_dB_map mp3plus_dB_1 = {-4781, 0}; /* just guess */ -static struct usbmix_dB_map mp3plus_dB_2 = {-1781, 618}; /* just guess */ +static struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0}; + /* just guess */ +static struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618}; + /* just guess */ static struct usbmix_name_map mp3plus_map[] = { /* 1: IT pcm */ -- 1.9.1

2 1

[alsa-devel] [PATCH] oss: avoid time_t usage
by Arnd Bergmann 17 Jun '16

17 Jun '16

We want to remove all time_t users from the kernel because of y2038 compatibility. This particular instance does not even use time_t to store a seconds value, so we can simply use 'unsigned int', which seems more fitting anywhere. The same code is used in two OSS files. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/oss/sound_timer.c | 2 +- sound/oss/sys_timer.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index 8021c85f076d..3a444a6f10eb 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -17,7 +17,7 @@ #include "sound_config.h" static volatile int initialized, opened, tmr_running; -static volatile time_t tmr_offs, tmr_ctr; +static volatile unsigned int tmr_offs, tmr_ctr; static volatile unsigned long ticks_offs; static volatile int curr_tempo, curr_timebase; static volatile unsigned long curr_ticks; diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c index 2226dda0eff0..d17019d25b99 100644 --- a/sound/oss/sys_timer.c +++ b/sound/oss/sys_timer.c @@ -19,7 +19,7 @@ #include "sound_config.h" static volatile int opened, tmr_running; -static volatile time_t tmr_offs, tmr_ctr; +static volatile unsigned int tmr_offs, tmr_ctr; static volatile unsigned long ticks_offs; static volatile int curr_tempo, curr_timebase; static volatile unsigned long curr_ticks; -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: max9867: Fix unix permissions for source files.
by Enric Balletbo i Serra 17 Jun '16

17 Jun '16

Change file permissions of source files max9867.c/h from 0755 to 0644. Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- sound/soc/codecs/max9867.c | 0 sound/soc/codecs/max9867.h | 0 2 files changed, 0 insertions(+), 0 deletions(-) mode change 100755 => 100644 sound/soc/codecs/max9867.c mode change 100755 => 100644 sound/soc/codecs/max9867.h diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h old mode 100755 new mode 100644 -- 2.1.0

1 0

[alsa-devel] [alsa-lib][PATCH v4] ctl: add test program for control element set
by Takashi Sakamoto 17 Jun '16

17 Jun '16

The feature of control element set has been abandoned for a long time since firstly introduced in 2003. Furthermore, there's few applications to utilize this feature. These situations bring a hard work to persons who need the feature. Especially, a lack of test program make it harder to fix much bugs in this feature. This commit adds a test program as a sample of the feature. This program adds element sets of each element type to 'hw:0' in this order; boolean, integer, enumerated, bytes, IEC958 and integer64. Each iteration includes below scheme: 1. add an element set with 900 elements. Each of them has maximum number of members allowed by ALSA ctl core. 2. check all of events generated by above operation. 3. retrieve information of each element, then validate it. 4. unlock each member of all elements because they're initially locked. 5. write to all of members in all elements and read. 6. check all of events generated by above operation. 7. write information for threshold level to the element set and read it. 8. check all of events generated by above operation. 9. remove the element set. 10.check all of events generated by above operation. When any of these operations fail, it means regression occurs. Then, added elements still remain in a certain sound card. In this case, unloading drivers corresponding to the card is an easy way to recover. Besides, this program doesn't perform below element operations of ALSA ctl feature: - list - lock - replace Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/Makefile.am | 5 +- test/user-ctl-element-set.c | 573 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 577 insertions(+), 1 deletion(-) create mode 100644 test/user-ctl-element-set.c diff --git a/test/Makefile.am b/test/Makefile.am index ce6e521..cbf2ab4 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -3,7 +3,7 @@ SUBDIRS=. lsb check_PROGRAMS=control pcm pcm_min latency seq \ playmidi1 timer rawmidi midiloop \ oldapi queue_timer namehint client_event_filter \ - chmap audio_time + chmap audio_time user-ctl-element-set control_LDADD=../src/libasound.la pcm_LDADD=../src/libasound.la @@ -24,6 +24,9 @@ code_CFLAGS=-Wall -pipe -g -O2 chmap_LDADD=../src/libasound.la audio_time_LDADD=../src/libasound.la +user_ctl_element_set_LDADD=../src/libasound.la +user_ctl_element_set_CFLAGS=-Wall -g + AM_CPPFLAGS=-I$(top_srcdir)/include AM_CFLAGS=-Wall -pipe -g diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c new file mode 100644 index 0000000..09c7929 --- /dev/null +++ b/test/user-ctl-element-set.c @@ -0,0 +1,573 @@ +/* + * user-control-element-set.c - a program to test in-kernel implementation of + * user-defined control element set. + * + * Copyright (c) 2015-2016 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "../include/asoundlib.h" + +struct elem_set_trial { + snd_ctl_t *handle; + + snd_ctl_elem_type_t type; + unsigned int member_count; + unsigned int element_count; + + snd_ctl_elem_id_t *id; + + int (*add_elem_set)(struct elem_set_trial *trial); + int (*check_elem_props)(struct elem_set_trial *trial, + snd_ctl_elem_info_t *info); + void (*change_elem_members)(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data); +}; + +/* Operations for elements in an element set with boolean type. */ +static int add_bool_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_boolean_set(trial->handle, trial->id, + trial->element_count, trial->member_count); +} + +static void change_bool_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + int val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_boolean(elem_data, i); + snd_ctl_elem_value_set_boolean(elem_data, i, !val); + } +} + +/* Operations for elements in an element set with integer type. */ +static int add_int_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_integer_set(trial->handle, trial->id, + trial->element_count, trial->member_count, + 0, 99, 1); +} + +static int check_int_elem_props(struct elem_set_trial *trial, + snd_ctl_elem_info_t *info) +{ + if (snd_ctl_elem_info_get_min(info) != 0) + return -EIO; + if (snd_ctl_elem_info_get_max(info) != 99) + return -EIO; + if (snd_ctl_elem_info_get_step(info) != 1) + return -EIO; + + return 0; +} + +static void change_int_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + long val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_integer(elem_data, i); + snd_ctl_elem_value_set_integer(elem_data, i, ++val); + } +} + +/* Operations for elements in an element set with enumerated type. */ +static const char *const labels[] = { + "trusty", + "utopic", + "vivid", + "willy", + "xenial", +}; + +static int add_enum_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_enumerated_set(trial->handle, trial->id, + trial->element_count, trial->member_count, + sizeof(labels) / sizeof(labels[0]), + labels); +} + +static int check_enum_elem_props(struct elem_set_trial *trial, + snd_ctl_elem_info_t *info) +{ + unsigned int items; + unsigned int i; + const char *label; + int err; + + items = snd_ctl_elem_info_get_items(info); + if (items != sizeof(labels) / sizeof(labels[0])) + return -EIO; + + /* Enumerate and validate all of labels registered to this element. */ + for (i = 0; i < items; ++i) { + snd_ctl_elem_info_set_item(info, i); + err = snd_ctl_elem_info(trial->handle, info); + if (err < 0) + return err; + + label = snd_ctl_elem_info_get_item_name(info); + if (strncmp(label, labels[i], strlen(labels[i])) != 0) + return -EIO; + } + + return 0; +} + +static void change_enum_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + unsigned int val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_enumerated(elem_data, i); + snd_ctl_elem_value_set_enumerated(elem_data, i, ++val); + } +} + +/* Operations for elements in an element set with bytes type. */ +static int add_bytes_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_bytes_set(trial->handle, trial->id, + trial->element_count, trial->member_count); +} + +static void change_bytes_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + unsigned char val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_byte(elem_data, i); + snd_ctl_elem_value_set_byte(elem_data, i, ++val); + } +} + +/* Operations for elements in an element set with iec958 type. */ +static int add_iec958_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_iec958(trial->handle, trial->id); +} + +static void change_iec958_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + snd_aes_iec958_t data; + + /* To suppress GCC warnings. */ + trial->element_count = 1; + + snd_ctl_elem_value_get_iec958(elem_data, &data); + /* This is an arbitrary number. */ + data.pad = 10; + snd_ctl_elem_value_set_iec958(elem_data, &data); +} + +/* Operations for elements in an element set with integer64 type. */ +static int add_int64_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_integer64_set(trial->handle, trial->id, + trial->element_count, trial->member_count, + 100, 10000, 30); +} + +static int check_int64_elem_props(struct elem_set_trial *trial, + snd_ctl_elem_info_t *info) +{ + if (snd_ctl_elem_info_get_min64(info) != 100) + return -EIO; + if (snd_ctl_elem_info_get_max64(info) != 10000) + return -EIO; + if (snd_ctl_elem_info_get_step64(info) != 30) + return -EIO; + + return 0; +} + +static void change_int64_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + long long val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_integer64(elem_data, i); + snd_ctl_elem_value_set_integer64(elem_data, i, ++val); + } +} + +/* Common operations. */ +static int add_elem_set(struct elem_set_trial *trial) +{ + char name[64] = {0}; + + snprintf(name, 64, "userspace-control-element-%s", + snd_ctl_elem_type_name(trial->type)); + + memset(trial->id, 0, snd_ctl_elem_id_sizeof()); + snd_ctl_elem_id_set_interface(trial->id, SND_CTL_ELEM_IFACE_MIXER); + snd_ctl_elem_id_set_name(trial->id, name); + + return trial->add_elem_set(trial); +} + +static int check_event(struct elem_set_trial *trial, unsigned int mask, + unsigned int expected_count) +{ + struct pollfd pfds; + int count; + unsigned short revents; + snd_ctl_event_t *event; + int err; + + snd_ctl_event_alloca(&event); + + if (snd_ctl_poll_descriptors_count(trial->handle) != 1) + return -ENXIO; + + if (snd_ctl_poll_descriptors(trial->handle, &pfds, 1) != 1) + return -ENXIO; + + while (expected_count > 0) { + count = poll(&pfds, 1, 1000); + if (count < 0) + return errno; + /* Some events are already supplied. */ + if (count == 0) + return -ETIMEDOUT; + + err = snd_ctl_poll_descriptors_revents(trial->handle, &pfds, + count, &revents); + if (err < 0) + return err; + if (revents & POLLERR) + return -EIO; + if (!(revents & POLLIN)) + continue; + + err = snd_ctl_read(trial->handle, event); + if (err < 0) + return err; + if (snd_ctl_event_get_type(event) != SND_CTL_EVENT_ELEM) + continue; + /* + * I expect each event is generated separately to the same + * element. + */ + if (!(snd_ctl_event_elem_get_mask(event) & mask)) + continue; + --expected_count; + } + + if (expected_count != 0) + return -EIO; + + return 0; +} + +static int check_elem_set_props(struct elem_set_trial *trial) +{ + snd_ctl_elem_id_t *id; + snd_ctl_elem_info_t *info; + unsigned int numid; + unsigned int i; + int err; + + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_info_alloca(&info); + + snd_ctl_elem_info_set_id(info, trial->id); + numid = snd_ctl_elem_info_get_numid(info); + + for (i = 0; i < trial->element_count; ++i) { + snd_ctl_elem_info_set_numid(info, numid + i); + + err = snd_ctl_elem_info(trial->handle, info); + if (err < 0) + return err; + + /* Check some common properties. */ + if (snd_ctl_elem_info_get_type(info) != trial->type) + return -EIO; + if (snd_ctl_elem_info_get_count(info) != trial->member_count) + return -EIO; + + /* + * In a case of IPC, this is the others. But in this case, + * it's myself. + */ + if (snd_ctl_elem_info_get_owner(info) != getpid()) + return -EIO; + + /* + * Just adding an element set by userspace applications, + * included elements are initially locked. + */ + if (!snd_ctl_elem_info_is_locked(info)) + return -EIO; + + /* Check type-specific properties. */ + if (trial->check_elem_props != NULL) { + err = trial->check_elem_props(trial, info); + if (err < 0) + return err; + } + + snd_ctl_elem_info_get_id(info, id); + err = snd_ctl_elem_unlock(trial->handle, id); + if (err < 0) + return err; + } + + return 0; +} + +static int check_elems(struct elem_set_trial *trial) +{ + snd_ctl_elem_value_t *data; + unsigned int numid; + unsigned int i; + int err; + + snd_ctl_elem_value_alloca(&data); + + snd_ctl_elem_value_set_id(data, trial->id); + + /* + * Get the value of numid field in identical information for the first + * element of this element set. + */ + numid = snd_ctl_elem_value_get_numid(data); + + for (i = 0; i < trial->element_count; ++i) { + /* + * Here, I increment numid, while we can also increment offset + * to enumerate each element in this element set. + */ + snd_ctl_elem_value_set_numid(data, numid + i); + + err = snd_ctl_elem_read(trial->handle, data); + if (err < 0) + return err; + + /* Change members of an element in this element set. */ + trial->change_elem_members(trial, data); + + err = snd_ctl_elem_write(trial->handle, data); + if (err < 0) + return err; + } + + return 0; +} + +static int check_tlv(struct elem_set_trial *trial) +{ + unsigned int orig[8], curr[8]; + int err; + + /* + * See a layout of 'struct snd_ctl_tlv'. I don't know the reason to + * construct this buffer with the same layout. It should be abstracted + * inner userspace library... + */ + orig[0] = snd_ctl_elem_id_get_numid(trial->id); + orig[1] = 6 * sizeof(orig[0]); + orig[2] = 'a'; + orig[3] = 'b'; + orig[4] = 'c'; + orig[5] = 'd'; + orig[6] = 'e'; + orig[7] = 'f'; + + /* + * In in-kernel implementation, write and command operations are the + * same for an element set added by userspace applications. Here, I + * use write. + */ + err = snd_ctl_elem_tlv_write(trial->handle, trial->id, + (const unsigned int *)orig); + if (err < 0) + return err; + + err = snd_ctl_elem_tlv_read(trial->handle, trial->id, curr, + sizeof(curr)); + if (err < 0) + return err; + + if (memcmp(curr, orig, sizeof(orig)) != 0) + return -EIO; + + return 0; +} + +int main(void) +{ + struct elem_set_trial trial = {0}; + unsigned int i; + int err; + + snd_ctl_elem_id_alloca(&trial.id); + + err = snd_ctl_open(&trial.handle, "hw:0", 0); + if (err < 0) + return EXIT_FAILURE; + + err = snd_ctl_subscribe_events(trial.handle, 1); + if (err < 0) + return EXIT_FAILURE; + + /* Test all of types. */ + for (i = 0; i < SND_CTL_ELEM_TYPE_LAST; ++i) { + trial.type = i + 1; + + /* Assign type-dependent operations. */ + switch (trial.type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + trial.element_count = 900; + trial.member_count = 128; + trial.add_elem_set = add_bool_elem_set; + trial.check_elem_props = NULL; + trial.change_elem_members = change_bool_elem_members; + break; + case SND_CTL_ELEM_TYPE_INTEGER: + trial.element_count = 900; + trial.member_count = 128; + trial.add_elem_set = add_int_elem_set; + trial.check_elem_props = check_int_elem_props; + trial.change_elem_members = change_int_elem_members; + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + trial.element_count = 900; + trial.member_count = 128; + trial.add_elem_set = add_enum_elem_set; + trial.check_elem_props = check_enum_elem_props; + trial.change_elem_members = change_enum_elem_members; + break; + case SND_CTL_ELEM_TYPE_BYTES: + trial.element_count = 900; + trial.member_count = 512; + trial.add_elem_set = add_bytes_elem_set; + trial.check_elem_props = NULL; + trial.change_elem_members = change_bytes_elem_members; + break; + case SND_CTL_ELEM_TYPE_IEC958: + trial.element_count = 1; + trial.member_count = 1; + trial.add_elem_set = add_iec958_elem_set; + trial.check_elem_props = NULL; + trial.change_elem_members = change_iec958_elem_members; + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + default: + trial.element_count = 900; + trial.member_count = 64; + trial.add_elem_set = add_int64_elem_set; + trial.check_elem_props = check_int64_elem_props; + trial.change_elem_members = change_int64_elem_members; + break; + } + + /* Test an operation to add an element set. */ + err = add_elem_set(&trial); + if (err < 0) { + printf("Fail to add an element set with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + err = check_event(&trial, SND_CTL_EVENT_MASK_ADD, + trial.element_count); + if (err < 0) { + printf("Fail to check some events to add elements with " + "%s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + + /* Check properties of each element in this element set. */ + err = check_elem_set_props(&trial); + if (err < 0) { + printf("Fail to check propetries of each element with " + "%s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + + /* + * Test operations to change the state of members in each + * element in the element set. + */ + err = check_elems(&trial); + if (err < 0) { + printf("Fail to change status of each element with %s " + "type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + err = check_event(&trial, SND_CTL_EVENT_MASK_VALUE, + trial.element_count); + if (err < 0) { + printf("Fail to check some events to change status of " + "each elements with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + + /* + * Test an operation to change threshold data of this element set, + * except for IEC958 type. + */ + if (trial.type != SND_CTL_ELEM_TYPE_IEC958) { + err = check_tlv(&trial); + if (err < 0) { + printf("Fail to change threshold level of an " + "element set with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + err = check_event(&trial, SND_CTL_EVENT_MASK_TLV, 1); + if (err < 0) { + printf("Fail to check an event to change " + "threshold level of an an element set " + "with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + } + + /* Test an operation to remove elements in this element set. */ + err = snd_ctl_elem_remove(trial.handle, trial.id); + if (err < 0) { + printf("Fail to remove elements with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + err = check_event(&trial, SND_CTL_EVENT_MASK_REMOVE, + trial.element_count); + if (err < 0) { + printf("Fail to check some events to remove each " + "element with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + } + + if (err < 0) { + printf("%s\n", snd_strerror(err)); + + /* To ensure. */ + snd_ctl_elem_remove(trial.handle, trial.id); + return EXIT_FAILURE; + } + + return EXIT_SUCCESS; +} -- 2.7.4

2 1

[alsa-devel] [PATCH v5] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 17 Jun '16

17 Jun '16

Because we need to support the multiple codecs (MAX98357A/RT5514/DA7219) on the RK3399 GRU boards, this patch can help us to support these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v5: - fix the build warning for missing I2C/CLKDEV_LOOKUP dependency Changes in v4: - the compatible name needs to be based on a single reference design, and the name of the board is called "GRU" - use one i2sX card "rockchip,cpu = <&i2s0>" instead of array that is based on GRU - rename DOC to rockchip,rk3399-gru-sound.txt - rename compatible to rockchip,rk3399-gru-sound - rename source code to rk3399_gru_sound.c Changes in v3: - rename DOC to rockchip,rk3399-max98357a-rt5514-da7219.txt - rename compatible to rockchip,rk3399-max98357a-rt5514-da7219 - rename source code to rk3399_max98357a_rt5514_da7219.c Changes in v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../bindings/sound/rockchip,rk3399-gru-sound.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 339 ++++++++++++++++++++ 4 files changed, 367 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt create mode 100644 sound/soc/rockchip/rk3399_gru_sound.c diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 0000000..f19b6c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..6d39032 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_RK3399_GRU_SOUND + tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support multiple codecs for SoC + audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..84e5c7c 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_RK3399_GRU_SOUND) += snd-soc-rk3399-gru-sound.o diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c new file mode 100644 index 0000000..cc9b943 --- /dev/null +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -0,0 +1,339 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rk3399-gru-sound" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speakers"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + unsigned int mclk; + int ret; + + snd_soc_dapm_disable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speakers"); + snd_soc_dapm_sync(dapm); + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + unsigned int mclk; + int ret; + + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Int Mic"); + snd_soc_dapm_sync(dapm); + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + int mclk, ret; + + snd_soc_dapm_disable_pin(dapm, "Speakers"); + snd_soc_dapm_disable_pin(dapm, "Int Mic"); + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_sync(dapm); + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rk3399-gru-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); + if (!cpu_node) { + dev_err(&pdev->dev, "Property 'rockchip,cpu' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rk3399-gru-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

4 5

[alsa-devel] [PATCH] ALSA: hdac_regmap - fix the register access for runtime PM
by Jaroslav Kysela 17 Jun '16

17 Jun '16

Call path: 1) snd_hdac_power_up_pm() 2) snd_hdac_power_up() 3) pm_runtime_get_sync() 4) __pm_runtime_resume() 5) rpm_resume() The rpm_resume() returns 1 when the device is already active. Because the return value is unmodified, the hdac regmap read/write functions should allow this value for the retry I/O operation, too. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/hda/hdac_regmap.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 87041dd..47a358f 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -444,7 +444,7 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, err = reg_raw_write(codec, reg, val); if (err == -EAGAIN) { err = snd_hdac_power_up_pm(codec); - if (!err) + if (err >= 0) err = reg_raw_write(codec, reg, val); snd_hdac_power_down_pm(codec); } @@ -470,7 +470,7 @@ static int __snd_hdac_regmap_read_raw(struct hdac_device *codec, err = reg_raw_read(codec, reg, val, uncached); if (err == -EAGAIN) { err = snd_hdac_power_up_pm(codec); - if (!err) + if (err >= 0) err = reg_raw_read(codec, reg, val, uncached); snd_hdac_power_down_pm(codec); } -- 2.5.5

2 1

[alsa-devel] [PATCH] ASoC fix up SND_SOC_WM8985 dependency
by Arnd Bergmann 17 Jun '16

17 Jun '16

I just added an I2C dependency to the wm8985 driver to work around a build failure, but it turns out that was premature: we actually need to depend on SND_SOC_I2C_AND_SPI, as the driver can work with either of the two, and we only need to prevent a configuration that has I2C=m and SND_SOC_WM8985=y. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 05252513fbb9 ("ASoC: wm8985: add i2c dependency") --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d0e8150c7170..60f8f13f0d34 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -994,7 +994,7 @@ config SND_SOC_WM8983 config SND_SOC_WM8985 tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver" - depends on I2C + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8988 tristate -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoc: wm8731: add 32bit mode.
by Matt Flax 17 Jun '16

17 Jun '16

This patch adds 32 bit word capability to the wm8731 driver. The wm8731 codec is capable of handling 32 bit word sizes, however that has not previously been activated in the codec driver. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/wm8731.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 4bcf5f8..d18261a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -358,6 +358,9 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, case 24: iface |= 0x0008; break; + case 32: + iface |= 0x000c; + break; } wm8731_set_deemph(codec); @@ -541,7 +544,7 @@ static int wm8731_startup(struct snd_pcm_substream *substream, #define WM8731_RATES SNDRV_PCM_RATE_8000_96000 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8731_dai_ops = { .startup = wm8731_startup, -- 2.7.4

3 2

[alsa-devel] [PATCH 0/3] ASoC: ak4613/wm8940: Regmap cache fixes
by Geert Uytterhoeven 17 Jun '16

17 Jun '16

Hi Liam, Mark, This series contains 3 fixes for regmap cache handling in the ak4613 and wm8940 drivers. All fixes are based on recent similar fixes for the ak4642 driver. The first patch fixes a crash I've seen on r8a7795/Salvator-X. The last patch is completely untested. The issue was discovered using 'git grep -Lw cache_type -- $(git grep -lw regcache_sync -- $(git grep -lw regmap_config))'' Thanks! Geert Uytterhoeven (3): ASoC: ak4613: Enable cache usage to fix crashes on resume ASoC: ak4613: Implement suspend callback ASoC: wm8940: Enable cache usage to fix crashes on resume sound/soc/codecs/ak4613.c | 13 ++++++++++++- sound/soc/codecs/wm8940.c | 1 + 2 files changed, 13 insertions(+), 1 deletion(-) -- 1.9.1 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

3 7

[alsa-devel] [PATCH] snd: soc: cx20442 codec: set tty->receiver_room in v253_open
by Janusz Krzysztofik 17 Jun '16

17 Jun '16

Commit 79901317ce80 ("n_tty: Don't flush buffer when closing ldisc"), introduced in v3.10, revealed a bug in the cx20442 codec driver which has never been setting tty->receive_room on line discipline open as it should from the beginning. Fix it. Created and tested on Amstrad Delta against Linux-4.7-rc3 Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- sound/soc/codecs/cx20442.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d6f4abb..fb3885f 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -226,6 +226,7 @@ static int v253_open(struct tty_struct *tty) if (!tty->disc_data) return -ENODEV; + tty->receive_room = 16; if (tty->ops->write(tty, v253_init, len) != len) { ret = -EIO; goto err; -- 2.7.3

2 2

[alsa-devel] [PATCH] ASoC: codecs: Remove "select REGMAP_SPI/REGMAP_I2C"
by Alexander Shiyan 17 Jun '16

17 Jun '16

REGMAP_SPI and REGMAP_I2C is selected automatically by SND_SOC, so no additional actions is required. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/Kconfig | 21 --------------------- 1 file changed, 21 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4d82a58..1e03fa0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -288,12 +288,10 @@ config SND_SOC_ADAU1761 config SND_SOC_ADAU1761_I2C tristate select SND_SOC_ADAU1761 - select REGMAP_I2C config SND_SOC_ADAU1761_SPI tristate select SND_SOC_ADAU1761 - select REGMAP_SPI config SND_SOC_ADAU1781 select SND_SOC_ADAU17X1 @@ -302,12 +300,10 @@ config SND_SOC_ADAU1781 config SND_SOC_ADAU1781_I2C tristate select SND_SOC_ADAU1781 - select REGMAP_I2C config SND_SOC_ADAU1781_SPI tristate select SND_SOC_ADAU1781 - select REGMAP_SPI config SND_SOC_ADAU1977 tristate @@ -315,12 +311,10 @@ config SND_SOC_ADAU1977 config SND_SOC_ADAU1977_SPI tristate select SND_SOC_ADAU1977 - select REGMAP_SPI config SND_SOC_ADAU1977_I2C tristate select SND_SOC_ADAU1977 - select REGMAP_I2C config SND_SOC_ADAV80X tristate @@ -403,7 +397,6 @@ config SND_SOC_CS42L73 config SND_SOC_CS4265 tristate "Cirrus Logic CS4265 CODEC" depends on I2C - select REGMAP_I2C # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 @@ -425,13 +418,11 @@ config SND_SOC_CS4271_I2C tristate "Cirrus Logic CS4271 CODEC (I2C)" depends on I2C select SND_SOC_CS4271 - select REGMAP_I2C config SND_SOC_CS4271_SPI tristate "Cirrus Logic CS4271 CODEC (SPI)" depends on SPI_MASTER select SND_SOC_CS4271 - select REGMAP_SPI config SND_SOC_CS42XX8 tristate @@ -440,7 +431,6 @@ config SND_SOC_CS42XX8_I2C tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)" depends on I2C select SND_SOC_CS42XX8 - select REGMAP_I2C # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 @@ -580,13 +570,11 @@ config SND_SOC_PCM3168A_I2C tristate "Texas Instruments PCM3168A CODEC - I2C" depends on I2C select SND_SOC_PCM3168A - select REGMAP_I2C config SND_SOC_PCM3168A_SPI tristate "Texas Instruments PCM3168A CODEC - SPI" depends on SPI_MASTER select SND_SOC_PCM3168A - select REGMAP_SPI config SND_SOC_PCM5102A tristate @@ -598,13 +586,11 @@ config SND_SOC_PCM512x_I2C tristate "Texas Instruments PCM512x CODECs - I2C" depends on I2C select SND_SOC_PCM512x - select REGMAP_I2C config SND_SOC_PCM512x_SPI tristate "Texas Instruments PCM512x CODECs - SPI" depends on SPI_MASTER select SND_SOC_PCM512x - select REGMAP_SPI config SND_SOC_RL6231 tristate @@ -668,7 +654,6 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate - select REGMAP_I2C select REGMAP_IRQ config SND_SOC_RT5677_SPI @@ -715,13 +700,11 @@ config SND_SOC_SSM2602_SPI tristate "Analog Devices SSM2602 CODEC - SPI" depends on SPI_MASTER select SND_SOC_SSM2602 - select REGMAP_SPI config SND_SOC_SSM2602_I2C tristate "Analog Devices SSM2602 CODEC - I2C" depends on I2C select SND_SOC_SSM2602 - select REGMAP_I2C config SND_SOC_SSM4567 tristate "Analog Devices ssm4567 amplifier driver support" @@ -730,7 +713,6 @@ config SND_SOC_SSM4567 config SND_SOC_STA32X tristate "STA326, STA328 and STA329 speaker amplifier" depends on I2C - select REGMAP_I2C config SND_SOC_STA350 tristate "STA350 speaker amplifier" @@ -788,7 +770,6 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC31XX tristate "Texas Instruments TLV320AIC31xx CODECs" depends on I2C - select REGMAP_I2C config SND_SOC_TLV320AIC32X4 tristate @@ -918,13 +899,11 @@ config SND_SOC_WM8804_I2C tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C" depends on I2C select SND_SOC_WM8804 - select REGMAP_I2C config SND_SOC_WM8804_SPI tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI" depends on SPI_MASTER select SND_SOC_WM8804 - select REGMAP_SPI config SND_SOC_WM8900 tristate -- 2.4.9

3 2

[alsa-devel] [alsa-lib][PATCH 0/9 v3] ctl: add APIs for control element set
by Takashi Sakamoto 16 Jun '16

16 Jun '16

Hi, This patchset is a revised version of below one: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-June/109041.html Changes: - Drop a patch for deprecating old APIs. - In detail, see our discussion: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-June/109117.html - Revise comments for TLV feature. - I realized it better to use vendor-neutral representation, after reading this discussion: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069483.h… Again, ALSA developers in PulseAudio side, test program in the last patch causes an abort to pulseaudio process as a result of kicking assertion of hcontrol in alsa-lib. In detail, please refer to my previous patchset. Regards Takashi Sakamoto (9): ctl: add an overview for design of ALSA control interface ctl: improve comments for handling element data ctl: add functions to add an element set ctl: improve comments for API to add an element of IEC958 type ctl: change former APIs as wrapper functions of element set APIs pcm: use new APIs to add a control element set for softvol plugin ctl: add explanation about threshold level feature ctl: improve API documentation for threshold level operations ctl: add test program for control element set include/control.h | 21 ++ src/control/control.c | 813 ++++++++++++++++++++++++++++++++------------ src/pcm/pcm_softvol.c | 7 +- test/Makefile.am | 5 +- test/user-ctl-element-set.c | 561 ++++++++++++++++++++++++++++++ 5 files changed, 1189 insertions(+), 218 deletions(-) create mode 100644 test/user-ctl-element-set.c -- 2.7.4

2 12

[alsa-devel] [PATCH v3 0/2] ASoC: Add support to Qualcomm msm8916-wcd codec
by Srinivas Kandagatla 16 Jun '16

16 Jun '16

This patchset aims at adding msm8916-wcd codec support. msm8916-wcd codec is found in Qualcomm msm8916 and apq8016 processors. This codec IP is split in to two parts(Digital & Analog), Analog part is integrated in to PMIC PM8916 and the digital part is integrated into Application processor. Register access to the analog part is done via SPMI interface to PMIC, and registers on the Application processor are memory mapped. Data transfer between Analog and Digital Die is done via a internal bus called PDM. Most of this driver is ported from downstream Andriod v3.10 kernel. Jack support and Headset button detection code is not added in to this series. This codec support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary Mic. - 2 Digital Microphones. - 2 Mic Bias Circuits. - Earpiece - Headset - Loud Speaker. - Jack Detect. - Headset Button detection circuit. Tested this code on v4.6-rc2 with lpass driver capture support patches for Headset (Mic and speakers), Secondary Mic and Loudspeaker on DB410C. DMIC should work too. Thanks, srini Changes since RFC v2: - Fixed error handling on regluator_enable suggested by Kenneth. - Fixed compatible strings suggested by Kenneth - make max/min rates inline with actual supported rates, spotted by Kenneth - Few minor suggestions by Kenneth. Changes since RFC v1 (https://lkml.org/lkml/2016/2/16/710) - Lot of code cleanup as suggest by Mark Brown. - Fixed bindings as suggested by Mark Rutland. - merged all the patches to one as spliting the patches did not help review. Srinivas Kandagatla (2): ASoC: msm8916: Add codec Device Tree bindings. ASoC: msm8916: Add msm8916-wcd codec driver .../devicetree/bindings/sound/qcom,msm8916-wcd.txt | 103 ++ include/dt-bindings/sound/msm8916-wcd.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 +- sound/soc/codecs/msm8916-wcd-registers.h | 710 +++++++++ sound/soc/codecs/msm8916-wcd.c | 1575 ++++++++++++++++++++ sound/soc/codecs/msm8916-wcd.h | 308 ++++ 7 files changed, 2709 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd.txt create mode 100644 include/dt-bindings/sound/msm8916-wcd.h create mode 100644 sound/soc/codecs/msm8916-wcd-registers.h create mode 100644 sound/soc/codecs/msm8916-wcd.c create mode 100644 sound/soc/codecs/msm8916-wcd.h -- 2.8.3

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Re: [alsa-devel] ASoC: rt5514: add rt5514 SPI driver
by Dan Carpenter 16 Jun '16

16 Jun '16

Hello Oder Chiou, This is a semi-automatic email about new static checker warnings. The patch 6eebf35b0e4a: "ASoC: rt5514: add rt5514 SPI driver" from Jun 6, 2016, leads to the following Smatch complaint: sound/soc/codecs/rt5514-spi.c:87 rt5514_spi_copy_work() warn: variable dereferenced before check 'rt5514_dsp->substream' (see line 83) sound/soc/codecs/rt5514-spi.c 82 container_of(work, struct rt5514_dsp, copy_work.work); 83 struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; ^^^^^^^^^^^^^^^^^^^^^^^ Dereference. 84 size_t period_bytes, truncated_bytes = 0; 85 86 mutex_lock(&rt5514_dsp->dma_lock); 87 if (!rt5514_dsp->substream) { ^^^^^^^^^^^^^^^^^^^^^ Checked too late. 88 dev_err(rt5514_dsp->dev, "No pcm substream\n"); 89 goto done; regards, dan carpenter

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[alsa-devel] [CX2075] No sound after suspend/resume from front speaker
by Max Zhao 16 Jun '16

16 Jun '16

Hi, I have recently been using Archlinux with standard kernel 4.6.2 on a Toshiba Z20T-B (Intel Broadwell). After using suspend to RAM (S3) no sound is output from the laptop speakers. (systemctl suspend) The headphone jack behaves normally. Even after suspend. Restarting pulseaudio, killing it etc does not work. Pulseaudio registers no errors and tries to output sound to speakers as normal. aplay -D front:PCH also reports no errors, but fails silently. Suspend on Win10 works with sound resume, so it should not be a hardware or UEFI bug. Best regards, Max Zhao lspci -vvnn 00:1b.0 Audio device [0403]: Intel Corporation Wildcat Point-LP High Definition Audio Controller [8086:9ca0] (rev 03) Subsystem: Toshiba America Info Systems Device [1179:0001] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 49 Region 0: Memory at f1234000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel 00:03.0 Audio device [0403]: Intel Corporation Broadwell-U Audio Controller [8086:160c] (rev 09) Subsystem: Toshiba America Info Systems Device [1179:0001] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 50 Region 0: Memory at f1230000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel 00:1b.0 should be the main audio card. 00:03.0 controls hdmi output. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: CX20753/4 Analog [CX20753/4 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default ALSA Output (currently PulseAudio Sound Server) hdmi:CARD=HDMI,DEV=0 HDA Intel HDMI, HDMI 0 HDMI Audio Output hdmi:CARD=HDMI,DEV=1 HDA Intel HDMI, HDMI 1 HDMI Audio Output hdmi:CARD=HDMI,DEV=2 HDA Intel HDMI, HDMI 2 HDMI Audio Output sysdefault:CARD=PCH HDA Intel PCH, CX20753/4 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers amixer -c1 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 74 Mono: Playback 65 [88%] [-9.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 0 [0%] [-74.00dB] [off] Front Right: Playback 0 [0%] [-74.00dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 7 Mono: Playback 4 [57%] [-12.00dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 80 Front Left: Capture 80 [100%] [6.00dB] [on] Front Right: Capture 80 [100%] [6.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [24.00dB] Front Right: 2 [67%] [24.00dB] /proc/asound/card1/codec#0 Codec: Conexant CX20753/4 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f15111 Subsystem Id: 0x11790694 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold S3D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20753/4 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=0, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] Converter: stream=0, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x12 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x04] Node 0x13 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Device: name="CX20753/4 Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x50 0x50] [0x50 0x50] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 4 0x18 0x1a* 0x19 0x1e Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 4 0x19* 0x1a 0x15 0x1e Node 0x15 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x16 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x04211040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x17 [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000010: OUT Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001124: IN Detect Vref caps: HIZ 80 Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001124: IN Detect Vref caps: HIZ 80 Pin Default 0x04a11030: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00000020: IN Pin Default 0x95a60120: [Fixed] Mic at Int Top Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x1e [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000024: IN Detect Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0

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Re: [alsa-devel] [PATCH v5] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 16 Jun '16

16 Jun '16

// + Adam, On 2016年06月16日 03:43, Steve Barber wrote: > On Mon, Jun 13, 2016 at 1:42 AM, Xing Zheng<zhengxing(a)rock-chips.com> wrote: >> + >> +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, >> + struct snd_pcm_hw_params *params) >> +{ >> + struct snd_soc_pcm_runtime *rtd = substream->private_data; >> + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; >> + struct snd_soc_dai *codec_dai = rtd->codec_dai; >> + struct snd_soc_dapm_context *dapm =&rtd->card->dapm; >> + int mclk, ret; >> + >> + snd_soc_dapm_disable_pin(dapm, "Speakers"); >> + snd_soc_dapm_disable_pin(dapm, "Int Mic"); >> + snd_soc_dapm_enable_pin(dapm, "Headphones"); >> + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); >> + snd_soc_dapm_sync(dapm); >> + >> + /* in bypass mode, the mclk has to be one of the frequencies below */ >> + switch (params_rate(params)) { >> + case 8000: >> + case 16000: >> + case 24000: >> + case 32000: >> + case 48000: >> + case 64000: >> + case 96000: >> + mclk = 12288000; >> + break; >> + case 11025: >> + case 22050: >> + case 44100: >> + case 88200: >> + mclk = 11289600; >> + break; >> + default: >> + return -EINVAL; >> + } >> + >> + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, >> + SND_SOC_CLOCK_OUT); >> + if (ret< 0) { >> + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); >> + return ret; >> + } >> + >> + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, >> + SND_SOC_CLOCK_IN); >> + if (ret< 0) { >> + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); >> + return ret; >> + } >> + >> + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); >> + if (ret< 0) { >> + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); >> + return ret; >> + } >> + >> + return 0; >> +} >> + >> +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) >> +{ >> + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; >> + int ret; >> + >> + /* Enable Headset and 4 Buttons Jack detection */ >> + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", >> + SND_JACK_HEADSET | > Should this also include SND_JACK_LINEOUT? da7219 differentiates > between the two so not including lineout means, for example, we can > miss jack insertion/removal events. > Hi Adam, Could you please confirm Steve says that about SND_JACK_LINEOUT? Thanks. -- - Xing Zheng

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[alsa-devel] [PATCH] ALSA: hda: fix some klockwork scan warnings
by libin.yang＠linux.intel.com 16 Jun '16

16 Jun '16

From: Libin Yang <libin.yang(a)linux.intel.com> This patch fixes some warnings from klockwork. These warnings are not the real issues. The patch adds the sanity check. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/hda/hdmi_chmap.c | 28 +++++++++++++++++++++++++--- sound/pci/hda/hda_codec.c | 8 +++++++- sound/pci/hda/patch_hdmi.c | 5 +++++ 3 files changed, 37 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index c6c75e7..81acc20 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -353,7 +353,8 @@ static void hdmi_std_setup_channel_mapping(struct hdac_chmap *chmap, int hdmi_slot = 0; /* fill actual channel mappings in ALSA channel (i) order */ for (i = 0; i < ch_alloc->channels; i++) { - while (!ch_alloc->speakers[7 - hdmi_slot] && !WARN_ON(hdmi_slot >= 8)) + while (!WARN_ON(hdmi_slot >= 8) && + !ch_alloc->speakers[7 - hdmi_slot]) hdmi_slot++; /* skip zero slots */ hdmi_channel_mapping[ca][i] = (i << 4) | hdmi_slot++; @@ -430,6 +431,12 @@ static int to_cea_slot(int ordered_ca, unsigned char pos) int mask = snd_hdac_chmap_to_spk_mask(pos); int i; + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (ordered_ca >= ARRAY_SIZE(channel_allocations)) + return -1; + if (mask) { for (i = 0; i < 8; i++) { if (channel_allocations[ordered_ca].speakers[7 - i] == mask) @@ -456,7 +463,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_spk_to_chmap); /* from CEA slot to ALSA API channel position */ static int from_cea_slot(int ordered_ca, unsigned char slot) { - int mask = channel_allocations[ordered_ca].speakers[7 - slot]; + int mask; + + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (slot >= 8) + return 0; + + mask = channel_allocations[ordered_ca].speakers[7 - slot]; return snd_hdac_spk_to_chmap(mask); } @@ -523,7 +538,8 @@ static void hdmi_setup_fake_chmap(unsigned char *map, int ca) int ordered_ca = get_channel_allocation_order(ca); for (i = 0; i < 8; i++) { - if (i < channel_allocations[ordered_ca].channels) + if (ordered_ca < ARRAY_SIZE(channel_allocations) && + i < channel_allocations[ordered_ca].channels) map[i] = from_cea_slot(ordered_ca, hdmi_channel_mapping[ca][i] & 0x0f); else map[i] = 0; @@ -551,6 +567,12 @@ int snd_hdac_get_active_channels(int ca) { int ordered_ca = get_channel_allocation_order(ca); + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (ordered_ca >= ARRAY_SIZE(channel_allocations)) + ordered_ca = 0; + return channel_allocations[ordered_ca].channels; } EXPORT_SYMBOL_GPL(snd_hdac_get_active_channels); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8374188..9913be8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3584,6 +3584,12 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, bool reset; spdif = snd_hda_spdif_out_of_nid(codec, nid); + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (WARN_ON(spdif == NULL)) + return; + curr_fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); reset = codec->spdif_status_reset && @@ -3768,7 +3774,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, spdif = snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid); if (mout->dig_out_nid && mout->share_spdif && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { - if (chs == 2 && + if (chs == 2 && spdif != NULL && snd_hda_is_supported_format(codec, mout->dig_out_nid, format) && !(spdif->status & IEC958_AES0_NONAUDIO)) { diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d0d5ad8..56e5204 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1680,6 +1680,11 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) mutex_lock(&codec->spdif_mutex); spdif = snd_hda_spdif_out_of_nid(codec, cvt_nid); + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (WARN_ON(spdif == NULL)) + return true; non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); mutex_unlock(&codec->spdif_mutex); return non_pcm; -- 1.9.1

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[alsa-devel] Wolfson 5102 on Beaglebone Black using ALSA
by Clark Dunson 15 Jun '16

15 Jun '16

Hey everybody! We have been working on integrating the Wolfson 5102 codec with the Beaglebone Black. We’ve made pretty decent progress, got the arizona code in kernel 4.4 registering the wm5102 as ALSA sound card (both player and recorder), have McAsp0 as 24.576 MHz clock source, and have 12 IRQs showing up in /proc/interrupts. We have been working with sound{compatible = "simple-audio-card”}, but now have hit our first real question. sysclk is hard coded to ‘0’ under simple-audio-card, and we had to hack the kernel and change that to ‘1’ to achieve ALSA sound card registration. We’d all like for this work to proceed without having to hack the kernel, and wondered if one of you might recommend another approach or selection for device tree sound{compatible} that might work out better? Thanks! Clark

3 10

[alsa-devel] [PATCH] ASoC: dapm: Fix to return correct path list in is_connected_ep.
by jeeja.kp＠intel.com 15 Jun '16

15 Jun '16

From: Jeeja KP <jeeja.kp(a)intel.com> In is_connected_ep, when custom_stop_condition is true, need to return the correct paths instead of con which is 0. Fixes: 6742064aef7f('ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets') Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-dapm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index db781f6..3c3f027 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1092,8 +1092,10 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); - if (custom_stop_condition && custom_stop_condition(widget, dir)) - return con; + if (custom_stop_condition && custom_stop_condition(widget, dir)) { + widget->endpoints[dir] = 1; + return widget->endpoints[dir]; + } if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); -- 2.5.0

2 1

[alsa-devel] [PATCH 0/6] ASoC: sgtl5000: fix use of regulators and internal LDO
by Clemens Gruber 15 Jun '16

15 Jun '16

This patch set addresses a structural problem in the handling of regulators for VDDIO, VDDA, and VDDD in the SGTL5000 driver. The first two of these power rails must be powered on prior to any I2C communication, and yet the regulators were tied to the codec, which is instantiated only after a fair amount of I2C communication takes place. In other words, these regulators could never have function, and we can surmise that no user of this driver has switched power supply rails connected to them. The third power rail (VDDD) can be derived internally (by using I2C registers) though the data sheet says that if an external VDDD is used, it should be enabled before MCLK is started and I2C activity begins. [I rebased Eric's patches from Feb 2015, fixed a few warnings, URLs, etc. and squashed two patches into one. - Clemens] Clemens Gruber (1): ASoC: sgtl5000: Remove misleading comment Eric Nelson (5): ASoC: sgtl5000: Fix regulator support ASoC: sgtl5000: Write all default registers ASoC: sgtl5000: Initialize CHIP_ANA_POWER to power-on defaults ASoC: sgtl5000: Disable internal PLL early ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF sound/soc/codecs/sgtl5000.c | 421 +++++++++++--------------------------------- sound/soc/codecs/sgtl5000.h | 2 + 2 files changed, 102 insertions(+), 321 deletions(-) -- 2.8.3

4 17

[alsa-devel] [PATCH RFC 0/6] ASoC: sgtl5000: fix use of regulators and internal LDO
by Eric Nelson 15 Jun '16

15 Jun '16

This patch set addresses a structural problem in the handling of regulators for VDDIO, VDDA, and VDDD in the SGTL5000 driver. The first two of these power rails must be powered on prior to any I2C communication, and yet the regulators were tied to the codec, which is instantiated only after a fair amount of I2C communication takes place. In other words, these regulators could never have function, and we can surmise that no user of this driver has switched power supply rails connected to them. The third power rail (VDDD) can be derived internally (by using I2C registers) though the data sheet says that if an external VDDD is used, it should be enabled before MCLK is started and I2C activity begins. Patch 1 moves the regulators to the I2C device and initializes them before I2C activity. Patch 2 addresses an issue found during development of the patch and is somewhat unrelated to regulators. It is included for discussion Patch 3 removes the root cause of patch 2 Patch 4 adds return value checks when setting the LINREG and ANA_POWER registers Patch 5 addresses the cause of follow-on failures and the source of the error in patch 3. Patch 6 removes the naive use of the supplies to lower power consumption and replaces it with something more modest. The patch set is being sent as an RFC for discussion, and should probably be squashed into two or three patches covering the regulator updates (patches 1, 3-5 and perhaps 6), initialization error handling (patch 2). Eric Nelson (6): ASoC: sgtl5000: fix regulator support ASoC: sgtl5000: write all default registers ASoC: sgtl5000: initialize CHIP_ANA_POWER to power-on defaults ASoC: sgtl5000: check return values ASoC: sgtl5000: disable internal PLL early ASoC: sgtl5000: Don't disable regulators in SND_SOC_BIAS_OFF sound/soc/codecs/sgtl5000.c | 417 +++++++++++--------------------------------- sound/soc/codecs/sgtl5000.h | 2 + 2 files changed, 102 insertions(+), 317 deletions(-) -- 1.9.1

3 28

[alsa-devel] [PATCH v2] ASoC: adau17x1: Add support for specifying the MCLK using the CCF
by Lars-Peter Clausen 15 Jun '16

15 Jun '16

The devices from the ADAU17X1 family all have a MCLK clock input which supplies the master clock for the device. The master clock is used as the input clock for the PLL. Currently the MCLK rate as well as the desired PLL output frequency need to be supplied by calling snd_soc_dai_set_pll() form a machine driver. Add support for specifying the MCLK using the common clock framework. In addition to that also automatically configure the PLL to a suitable rate if the master clock was provided using the CCW. This allows to use the CODEC driver without any special configuration requirements from the machine driver. While the PLL output frequency can be configured over a (more or less) continuous range the narrowness of the range and the other constraints of the clocking tree usually only result in two output frequencies that will actually be chosen. One for 44.1kHz based rates and one for 48kHz based rates, these are the rates that the automatic PLL configuration will use. For the rare case where a non-standard setup is required a machine driver can disable the auto-configuration and configure a custom frequency using the existing mechanisms. If the common clock framework is not enabled clk_get() will return NULL and the driver will function as before and the clock rate needs to be configured manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Change numerical value of ADAU17X1_CLK_SRC_PLL_AUTO to 0, since that is what the simple-card driver will call snd_soc_dai_set_sysclk() with when the CODEC node has a clock property. So this is really the only way to make this work with the simple-card driver. --- .../devicetree/bindings/sound/adi,adau17x1.txt | 8 + sound/soc/codecs/adau1761-i2c.c | 2 +- sound/soc/codecs/adau1761-spi.c | 2 +- sound/soc/codecs/adau1781-i2c.c | 2 +- sound/soc/codecs/adau1781-spi.c | 2 +- sound/soc/codecs/adau17x1.c | 219 +++++++++++++++------ sound/soc/codecs/adau17x1.h | 6 + 7 files changed, 179 insertions(+), 62 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt index 8dbce0e..1447dec 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt @@ -13,6 +13,11 @@ Required properties: - reg: The i2c address. Value depends on the state of ADDR0 and ADDR1, as wired in hardware. +Optional properties: + - clock-names: If provided must be "mclk". + - clocks: phandle + clock-specifiers for the clock that provides + the audio master clock for the device. + Examples: #include <dt-bindings/sound/adau17x1.h> @@ -20,5 +25,8 @@ Examples: adau1361@38 { compatible = "adi,adau1761"; reg = <0x38>; + + clock-names = "mclk"; + clocks = <&audio_clock>; }; }; diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 8de010f..9e7f257 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -31,7 +31,7 @@ static int adau1761_i2c_probe(struct i2c_client *client, static int adau1761_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c index d917124..a0b214b 100644 --- a/sound/soc/codecs/adau1761-spi.c +++ b/sound/soc/codecs/adau1761-spi.c @@ -48,7 +48,7 @@ static int adau1761_spi_probe(struct spi_device *spi) static int adau1761_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c index 06cbca8..7b9d180 100644 --- a/sound/soc/codecs/adau1781-i2c.c +++ b/sound/soc/codecs/adau1781-i2c.c @@ -31,7 +31,7 @@ static int adau1781_i2c_probe(struct i2c_client *client, static int adau1781_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-spi.c b/sound/soc/codecs/adau1781-spi.c index 3d965a0..9b23354 100644 --- a/sound/soc/codecs/adau1781-spi.c +++ b/sound/soc/codecs/adau1781-spi.c @@ -48,7 +48,7 @@ static int adau1781_spi_probe(struct spi_device *spi) static int adau1781_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 66a6e06..439aa3f 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -9,6 +9,7 @@ #include <linux/module.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> #include <sound/core.h> @@ -303,6 +304,116 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); + if (ret < 0) + return ret; + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + if (ret) + return ret; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + bool is_pll; + bool was_pll; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + is_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL_AUTO: + if (!adau->mclk) + return -EINVAL; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: + is_pll = true; + break; + default: + return -EINVAL; + } + + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_MCLK: + was_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL: + case ADAU17X1_CLK_SRC_PLL_AUTO: + was_pll = true; + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (is_pll != was_pll) { + if (is_pll) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_auto_pll(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct adau *adau = snd_soc_dai_get_drvdata(dai); + unsigned int pll_rate; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 96000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + case 88200: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + return adau17x1_set_dai_pll(dai, ADAU17X1_PLL, ADAU17X1_PLL_SRC_MCLK, + clk_get_rate(adau->mclk), pll_rate); +} + static int adau17x1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -312,10 +423,19 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, unsigned int freq; int ret; - if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_PLL_AUTO: + ret = adau17x1_auto_pll(dai, params); + if (ret) + return ret; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: freq = adau->pll_freq; - else + break; + default: freq = adau->sysclk; + break; + } if (freq % params_rate(params) != 0) return -EINVAL; @@ -387,62 +507,6 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); } -static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct snd_soc_codec *codec = dai->codec; - struct adau *adau = snd_soc_codec_get_drvdata(codec); - int ret; - - if (freq_in < 8000000 || freq_in > 27000000) - return -EINVAL; - - ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); - if (ret < 0) - return ret; - - /* The PLL register is 6 bytes long and can only be written at once. */ - ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, - adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); - if (ret) - return ret; - - adau->pll_freq = freq_out; - - return 0; -} - -static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); - struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); - - switch (clk_id) { - case ADAU17X1_CLK_SRC_MCLK: - case ADAU17X1_CLK_SRC_PLL: - break; - default: - return -EINVAL; - } - - adau->sysclk = freq; - - if (adau->clk_src != clk_id) { - if (clk_id == ADAU17X1_CLK_SRC_PLL) { - snd_soc_dapm_add_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } else { - snd_soc_dapm_del_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } - } - - adau->clk_src = clk_id; - - return 0; -} - static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { @@ -827,6 +891,10 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes, ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); } + + if (adau->clk_src != ADAU17X1_CLK_SRC_MCLK) + snd_soc_dapm_add_routes(dapm, &adau17x1_dapm_pll_route, 1); + return ret; } EXPORT_SYMBOL_GPL(adau17x1_add_routes); @@ -849,6 +917,7 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, const char *firmware_name) { struct adau *adau; + int ret; if (IS_ERR(regmap)) return PTR_ERR(regmap); @@ -857,6 +926,30 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, if (!adau) return -ENOMEM; + adau->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(adau->mclk)) { + if (PTR_ERR(adau->mclk) != -ENOENT) + return PTR_ERR(adau->mclk); + /* Clock is optional (for the driver) */ + adau->mclk = NULL; + } else if (adau->mclk) { + adau->clk_src = ADAU17X1_CLK_SRC_PLL_AUTO; + + /* + * Any valid PLL output rate will work at this point, use one + * that is likely to be chosen later as well. The register will + * be written when the PLL is powered up for the first time. + */ + ret = adau_calc_pll_cfg(clk_get_rate(adau->mclk), 48000 * 1024, + adau->pll_regs); + if (ret < 0) + return ret; + + ret = clk_prepare_enable(adau->mclk); + if (ret) + return ret; + } + adau->regmap = regmap; adau->switch_mode = switch_mode; adau->type = type; @@ -880,6 +973,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL_GPL(adau17x1_probe); +void adau17x1_remove(struct device *dev) +{ + struct adau *adau = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + if (adau->mclk) + clk_disable_unprepare(adau->mclk); +} +EXPORT_SYMBOL_GPL(adau17x1_remove); + MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 5ae87a0..bf04b7e 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -22,13 +22,18 @@ enum adau17x1_pll_src { }; enum adau17x1_clk_src { + /* Automatically configure PLL based on the sample rate */ + ADAU17X1_CLK_SRC_PLL_AUTO, ADAU17X1_CLK_SRC_MCLK, ADAU17X1_CLK_SRC_PLL, }; +struct clk; + struct adau { unsigned int sysclk; unsigned int pll_freq; + struct clk *mclk; enum adau17x1_clk_src clk_src; enum adau17x1_type type; @@ -52,6 +57,7 @@ int adau17x1_add_routes(struct snd_soc_codec *codec); int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name); +void adau17x1_remove(struct device *dev); int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 15 Jun '16

15 Jun '16

Replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/usb/mixer_maps.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 1f8fb0d9..9038b2e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -107,8 +107,10 @@ static struct usbmix_name_map extigy_map[] = { * e.g. no Master and fake PCM volume * Pavel Mihaylov <bin(a)bash.info> */ -static struct usbmix_dB_map mp3plus_dB_1 = {-4781, 0}; /* just guess */ -static struct usbmix_dB_map mp3plus_dB_2 = {-1781, 618}; /* just guess */ +static struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0}; + /* just guess */ +static struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618}; + /* just guess */ static struct usbmix_name_map mp3plus_map[] = { /* 1: IT pcm */ -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: adau17x1: Add support for specifying the MCLK using the CCF
by Lars-Peter Clausen 15 Jun '16

15 Jun '16

The devices from the ADAU17X1 family all have a MCLK clock input which supplies the master clock for the device. The master clock is used as the input clock for the PLL. Currently the MCLK rate as well as the desired PLL output frequency need to be supplied by calling snd_soc_dai_set_pll() form a machine driver. Add support for specifying the MCLK using the common clock framework. In addition to that also automatically configure the PLL to a suitable rate if the master clock was provided using the CCW. This allows to use the CODEC driver without any special configuration requirements from the machine driver. While the PLL output frequency can be configured over a (more or less) continuous range the narrowness of the range and the other constraints of the clocking tree usually only result in two output frequencies that will actually be chosen. One for 44.1kHz based rates and one for 48kHz based rates, these are the rates that the automatic PLL configuration will use. For the rare case where a non-standard setup is required a machine driver can disable the auto-configuration and configure a custom frequency using the existing mechanisms. If the common clock framework is not enabled clk_get() will return NULL and the driver will function as before and the clock rate needs to be configured manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- .../devicetree/bindings/sound/adi,adau17x1.txt | 8 + sound/soc/codecs/adau1761-i2c.c | 2 +- sound/soc/codecs/adau1761-spi.c | 2 +- sound/soc/codecs/adau1781-i2c.c | 2 +- sound/soc/codecs/adau1781-spi.c | 2 +- sound/soc/codecs/adau17x1.c | 219 +++++++++++++++------ sound/soc/codecs/adau17x1.h | 6 + 7 files changed, 179 insertions(+), 62 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt index 8dbce0e..1447dec 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt @@ -13,6 +13,11 @@ Required properties: - reg: The i2c address. Value depends on the state of ADDR0 and ADDR1, as wired in hardware. +Optional properties: + - clock-names: If provided must be "mclk". + - clocks: phandle + clock-specifiers for the clock that provides + the audio master clock for the device. + Examples: #include <dt-bindings/sound/adau17x1.h> @@ -20,5 +25,8 @@ Examples: adau1361@38 { compatible = "adi,adau1761"; reg = <0x38>; + + clock-names = "mclk"; + clocks = <&audio_clock>; }; }; diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 8de010f..9e7f257 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -31,7 +31,7 @@ static int adau1761_i2c_probe(struct i2c_client *client, static int adau1761_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c index d917124..a0b214b 100644 --- a/sound/soc/codecs/adau1761-spi.c +++ b/sound/soc/codecs/adau1761-spi.c @@ -48,7 +48,7 @@ static int adau1761_spi_probe(struct spi_device *spi) static int adau1761_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c index 06cbca8..7b9d180 100644 --- a/sound/soc/codecs/adau1781-i2c.c +++ b/sound/soc/codecs/adau1781-i2c.c @@ -31,7 +31,7 @@ static int adau1781_i2c_probe(struct i2c_client *client, static int adau1781_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-spi.c b/sound/soc/codecs/adau1781-spi.c index 3d965a0..9b23354 100644 --- a/sound/soc/codecs/adau1781-spi.c +++ b/sound/soc/codecs/adau1781-spi.c @@ -48,7 +48,7 @@ static int adau1781_spi_probe(struct spi_device *spi) static int adau1781_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 66a6e06..439aa3f 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -9,6 +9,7 @@ #include <linux/module.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> #include <sound/core.h> @@ -303,6 +304,116 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); + if (ret < 0) + return ret; + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + if (ret) + return ret; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + bool is_pll; + bool was_pll; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + is_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL_AUTO: + if (!adau->mclk) + return -EINVAL; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: + is_pll = true; + break; + default: + return -EINVAL; + } + + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_MCLK: + was_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL: + case ADAU17X1_CLK_SRC_PLL_AUTO: + was_pll = true; + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (is_pll != was_pll) { + if (is_pll) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_auto_pll(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct adau *adau = snd_soc_dai_get_drvdata(dai); + unsigned int pll_rate; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 96000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + case 88200: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + return adau17x1_set_dai_pll(dai, ADAU17X1_PLL, ADAU17X1_PLL_SRC_MCLK, + clk_get_rate(adau->mclk), pll_rate); +} + static int adau17x1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -312,10 +423,19 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, unsigned int freq; int ret; - if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_PLL_AUTO: + ret = adau17x1_auto_pll(dai, params); + if (ret) + return ret; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: freq = adau->pll_freq; - else + break; + default: freq = adau->sysclk; + break; + } if (freq % params_rate(params) != 0) return -EINVAL; @@ -387,62 +507,6 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); } -static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct snd_soc_codec *codec = dai->codec; - struct adau *adau = snd_soc_codec_get_drvdata(codec); - int ret; - - if (freq_in < 8000000 || freq_in > 27000000) - return -EINVAL; - - ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); - if (ret < 0) - return ret; - - /* The PLL register is 6 bytes long and can only be written at once. */ - ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, - adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); - if (ret) - return ret; - - adau->pll_freq = freq_out; - - return 0; -} - -static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); - struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); - - switch (clk_id) { - case ADAU17X1_CLK_SRC_MCLK: - case ADAU17X1_CLK_SRC_PLL: - break; - default: - return -EINVAL; - } - - adau->sysclk = freq; - - if (adau->clk_src != clk_id) { - if (clk_id == ADAU17X1_CLK_SRC_PLL) { - snd_soc_dapm_add_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } else { - snd_soc_dapm_del_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } - } - - adau->clk_src = clk_id; - - return 0; -} - static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { @@ -827,6 +891,10 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes, ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); } + + if (adau->clk_src != ADAU17X1_CLK_SRC_MCLK) + snd_soc_dapm_add_routes(dapm, &adau17x1_dapm_pll_route, 1); + return ret; } EXPORT_SYMBOL_GPL(adau17x1_add_routes); @@ -849,6 +917,7 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, const char *firmware_name) { struct adau *adau; + int ret; if (IS_ERR(regmap)) return PTR_ERR(regmap); @@ -857,6 +926,30 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, if (!adau) return -ENOMEM; + adau->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(adau->mclk)) { + if (PTR_ERR(adau->mclk) != -ENOENT) + return PTR_ERR(adau->mclk); + /* Clock is optional (for the driver) */ + adau->mclk = NULL; + } else if (adau->mclk) { + adau->clk_src = ADAU17X1_CLK_SRC_PLL_AUTO; + + /* + * Any valid PLL output rate will work at this point, use one + * that is likely to be chosen later as well. The register will + * be written when the PLL is powered up for the first time. + */ + ret = adau_calc_pll_cfg(clk_get_rate(adau->mclk), 48000 * 1024, + adau->pll_regs); + if (ret < 0) + return ret; + + ret = clk_prepare_enable(adau->mclk); + if (ret) + return ret; + } + adau->regmap = regmap; adau->switch_mode = switch_mode; adau->type = type; @@ -880,6 +973,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL_GPL(adau17x1_probe); +void adau17x1_remove(struct device *dev) +{ + struct adau *adau = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + if (adau->mclk) + clk_disable_unprepare(adau->mclk); +} +EXPORT_SYMBOL_GPL(adau17x1_remove); + MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 5ae87a0..e9fbc3b 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -24,11 +24,16 @@ enum adau17x1_pll_src { enum adau17x1_clk_src { ADAU17X1_CLK_SRC_MCLK, ADAU17X1_CLK_SRC_PLL, + /* Automatically configure PLL based on the sample rate */ + ADAU17X1_CLK_SRC_PLL_AUTO, }; +struct clk; + struct adau { unsigned int sysclk; unsigned int pll_freq; + struct clk *mclk; enum adau17x1_clk_src clk_src; enum adau17x1_type type; @@ -52,6 +57,7 @@ int adau17x1_add_routes(struct snd_soc_codec *codec); int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name); +void adau17x1_remove(struct device *dev); int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); -- 2.1.4

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[alsa-devel] [PATCH] ALSA: seq_oss: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 15 Jun '16

15 Jun '16

Replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Also, removed some unnecessary comments. Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/core/seq/oss/seq_oss_synth.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index b16dbef..cd0e0eb 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -70,11 +70,11 @@ struct seq_oss_synth { static int max_synth_devs; static struct seq_oss_synth *synth_devs[SNDRV_SEQ_OSS_MAX_SYNTH_DEVS]; static struct seq_oss_synth midi_synth_dev = { - -1, /* seq_device */ - SYNTH_TYPE_MIDI, /* synth_type */ - 0, /* synth_subtype */ - 16, /* nr_voices */ - "MIDI", /* name */ + .seq_device = -1, + .synth_type = SYNTH_TYPE_MIDI, + .synth_subtype = 0, + .nr_voices = 16, + .name = "MIDI", }; static DEFINE_SPINLOCK(register_lock); -- 1.9.1

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Re: [alsa-devel] Support Dell headset mode for ALC891
by Takashi Iwai 15 Jun '16

15 Jun '16

On Wed, 08 Jun 2016 09:17:12 +0200, Kailang wrote: > > Hi Takashi, > > This patch is supported Dell headset mode for ALC891. > It is only support I-phone type headset. > I think this function is only support for DELL. > So, headset mode and platform support is merged. > > This patch is test passed by Ubuntu team. > > Many Thanks. (snip) > Date: Tue, 7 Jun 2016 11:31:34 +0800 > Subject: [PATCH] ALSA: hda/realtek - ALC891 headset mode for Dell > > New headset mode of ALC891 for Dell. > > Signed-off-by: Kailang Yang <kailang(a)realtek.com> Could you put more comments in the changelog, as what you gave in the mail? > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 7960316..9dc641e 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -3718,6 +3718,9 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > case 0x10ec0295: > alc_process_coef_fw(codec, coef0225); > break; > + case 0x10ec0867: > + alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0); > + break; > } > codec_dbg(codec, "Headset jack set to unplugged mode.\n"); > } > @@ -3805,6 +3808,8 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > alc_process_coef_fw(codec, coef0293); > snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); > break; > + case 0x10ec0867: > + alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14); > case 0x10ec0662: No break between 0867 and 0662? thanks, Takashi

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[alsa-devel] [alsa-lib][PATCH 00/10 v2] ctl: add APIs for control element set
by Takashi Sakamoto 15 Jun '16

15 Jun '16

Hi, This patchset is revised version of below one: [alsa-devel] [alsa-lib][PATCH 0/6] control: add APIs for control element set http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104795.h… In this time, I add a program to test the feature. Changes: - Use a model which consists of element set/element/member/value. - Add an description about the design of ALSA ctl feature. - Obsolete old APIs due to the lack of const semantics. - Add an description about threshold level feature and topology framework. - Improve comments to threshold level APIs. - Add a test program. Before starting any discussions, please read description of the model, to stand on the same basis of understanding. I note that execution of the test program causes an abort to PulseAudio. I can see this line below. pulseaudio: hcontrol.c:745: snd_hctl_handle_event: Assertion `res >= 0 && dir == 0' failed. I guess that quick addition/removal of control element set causes this assertion, but I have little time to investigate the cause. I wish to get helps of ALSA developers in PulseAudio side. If you need an practical example of the feature, please test one of Python 3 samples in my 'alsa-gi'. https://github.com/takaswie/alsa-gi/ This works without alsa-lib, by using kernel/userspace interface directly. Therefore, this patchet doesn't affect the implementation. Takashi Sakamoto (10): ctl: add an overview for design of ALSA control interface ctl: improve comments for handling element data ctl: add functions to add an element set ctl: improve comments for API to add an element of IEC958 type ctl: change former APIs as wrapper functions of element set APIs ctl: deprecate APIs to add an element set with a single element pcm: use new APIs to add a control element set for softvol plugin ctl: add explaination about threshold level feature ctl: improve API documentation for threshold level operations ctl: add test program for control element set include/control.h | 36 +- src/control/control.c | 829 ++++++++++++++++++++++++++++++++------------ src/pcm/pcm_softvol.c | 7 +- test/Makefile.am | 5 +- test/user-ctl-element-set.c | 561 ++++++++++++++++++++++++++++++ 5 files changed, 1216 insertions(+), 222 deletions(-) create mode 100644 test/user-ctl-element-set.c -- 2.7.4

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[alsa-devel] [PATCH] sound: aedsp16: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 15 Jun '16

15 Jun '16

Replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/oss/aedsp16.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c index 35b5912..bb477d5 100644 --- a/sound/oss/aedsp16.c +++ b/sound/oss/aedsp16.c @@ -482,13 +482,13 @@ static struct orVals orDMA[] __initdata = { }; static struct aedsp16_info ae_config = { - DEF_AEDSP16_IOB, - DEF_AEDSP16_IRQ, - DEF_AEDSP16_MRQ, - DEF_AEDSP16_DMA, - -1, - -1, - INIT_NONE + .base_io = DEF_AEDSP16_IOB, + .irq = DEF_AEDSP16_IRQ, + .mpu_irq = DEF_AEDSP16_MRQ, + .dma = DEF_AEDSP16_DMA, + .mss_base = -1, + .mpu_base = -1, + .init = INIT_NONE }; /* -- 1.9.1

2 1

[alsa-devel] [PATCH v2 1/2] sound: lpass-cpu: add module licence and description
by Srinivas Kandagatla 14 Jun '16

14 Jun '16

This patch adds module licence to lpass-cpu driver, without this patch lpass-cpu module would taint with below error: snd_soc_lpass_cpu: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint snd_soc_lpass_cpu: Unknown symbol regmap_write (err 0) snd_soc_lpass_cpu: Unknown symbol devm_kmalloc (err 0) ... Acked-by: Kenneth Westfield <kwestfie(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 3cde9fb..eff3f9a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -586,3 +586,6 @@ int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev) return 0; } EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_platform_remove); + +MODULE_DESCRIPTION("QTi LPASS CPU Driver"); +MODULE_LICENSE("GPL v2"); -- 2.8.3

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