November 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Intel: common: Replace sst_acpi_check_hid with acpi_dev_present
by Jeremy Cline 10 Nov '17

10 Nov '17

Replace the sst_acpi_check_hid helper function added in commit 915ae2b9f0fe ("ASoC: Intel: Create a helper to search for matching machine") with the generic acpi_dev_present helper function and remove the now unused sst_acpi_check_hid function. This should have no functional change. Signed-off-by: Jeremy Cline <jeremy(a)jcline.org> --- sound/soc/intel/common/sst-acpi.h | 3 --- sound/soc/intel/common/sst-match-acpi.c | 32 ++------------------------------ 2 files changed, 2 insertions(+), 33 deletions(-) diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h index afe9b87b8bd5..d7facac83b7e 100644 --- a/sound/soc/intel/common/sst-acpi.h +++ b/sound/soc/intel/common/sst-acpi.h @@ -43,9 +43,6 @@ static inline bool sst_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], /* acpi match */ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines); -/* acpi check hid */ -bool sst_acpi_check_hid(const u8 hid[ACPI_ID_LEN]); - /* Descriptor for SST ASoC machine driver */ struct sst_acpi_mach { /* ACPI ID for the matching machine driver. Audio codec for instance */ diff --git a/sound/soc/intel/common/sst-match-acpi.c b/sound/soc/intel/common/sst-match-acpi.c index 56d26f36a3cb..f8e1b92de79a 100644 --- a/sound/soc/intel/common/sst-match-acpi.c +++ b/sound/soc/intel/common/sst-match-acpi.c @@ -49,40 +49,12 @@ const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) } EXPORT_SYMBOL_GPL(sst_acpi_find_name_from_hid); -static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, - void *context, void **ret) -{ - unsigned long long sta; - acpi_status status; - - *(bool *)context = true; - status = acpi_evaluate_integer(handle, "_STA", NULL, &sta); - if (ACPI_FAILURE(status) || !(sta & ACPI_STA_DEVICE_PRESENT)) - *(bool *)context = false; - - return AE_OK; -} - -bool sst_acpi_check_hid(const u8 hid[ACPI_ID_LEN]) -{ - acpi_status status; - bool found = false; - - status = acpi_get_devices(hid, sst_acpi_mach_match, &found, NULL); - - if (ACPI_FAILURE(status)) - return false; - - return found; -} -EXPORT_SYMBOL_GPL(sst_acpi_check_hid); - struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines) { struct sst_acpi_mach *mach; for (mach = machines; mach->id[0]; mach++) { - if (sst_acpi_check_hid(mach->id) == true) { + if (acpi_dev_present(mach->id, NULL, -1) == true) { if (mach->machine_quirk == NULL) return mach; @@ -161,7 +133,7 @@ struct sst_acpi_mach *sst_acpi_codec_list(void *arg) return mach; for (i = 0; i < codec_list->num_codecs; i++) { - if (sst_acpi_check_hid(codec_list->codecs[i]) != true) + if (acpi_dev_present(codec_list->codecs[i], NULL, -1) != true) return NULL; } -- 2.14.3

3 2

[alsa-devel] C-Media CM6533 limited sample rate
by Luigi Tarenga 10 Nov '17

10 Nov '17

hello everybody, I bought on Amazon a cheap usb soundcard based on CM6533 chip. For a amateur radio project I need a microphone able to sample at 96Khz and from the datasheet the CM6533 can sample up to this rate. Actually I have 2 different project, the first an echolink node that would live happily with a sample rate of 8Khz and another of a software modem that would work better with a sample rate of 96Khz. Unfortunately it seems I can't set this sample rate with arecord or sox. Actually I can only sample at 44100 or 48000 and only in stereo mode. The datasheet say the chip support PCM at 8K/11.025K/16K/22.05K/32K/ 44.1K/48K/96K Hz with 16 or 24 bit depth. (the datasheet can be found here https://www.cmedia.com.tw/support/download_center/25 ) with arecord the error I get is: $ arecord -D hw:2,0 -vv -f S16_LE -c 2 -r 96000 -t raw -d 10 > /dev/null Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 96000 Hz, Stereo Warning: rate is not accurate (requested = 96000Hz, got = 48000Hz) please, try the plug plugin Hardware PCM card 2 'USB Advanced Audio Device' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 #+ | 00%^CAborted by signal Interrupt... same error if I try 8000 Hz. note that another problem is that I have to use 2 channel for sampling while I would prefer to sample in mono since I would have to convert it anyway with an alsa plugin or in userspace and waste some cpu cicle. If I try to sample in mono: $ arecord -D hw:2,0 -vv -f S16_LE -c 1 -r 8000 -t raw -d 10 > /dev/null Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono arecord: set_params:1324: Channels count non available my software version: $ uname -a Linux bullet 4.13.11-1-ARCH #1 SMP PREEMPT Thu Nov 2 10:25:56 CET 2017 x86_64 GNU/Linux $ pacman -Q|grep alsa alsa-lib 1.1.4.1-2 alsa-tools 1.1.3-1 alsa-utils 1.1.4-1 zita-alsa-pcmi 0.2.0-3 lsusb give this output: lsusb -v -d 0d8c:0024 Bus 003 Device 007: ID 0d8c:0024 C-Media Electronics, Inc. Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 16 idVendor 0x0d8c C-Media Electronics, Inc. idProduct 0x0024 bcdDevice 1.11 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 393 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 95 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 6 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 8 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 7 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 8 bNrInPins 2 baSourceID( 0) 1 baSourceID( 1) 7 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 bmControls 0x00 iMixer 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x89 EP 9 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0002 1x 2 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x00c4 1x 196 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 88200 tSamFreq[ 1] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 2 Discrete tSamFreq[ 0] 88200 tSamFreq[ 1] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x00c4 1x 196 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.11 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 77 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 32 >From the output it seems it's reported support for only 44.1Khz or 48Khz. I'm trying to find out what could be the problem, is the snd_usb_audio driver that stick with a safe defaults? is the firmware on the chip that limit the hardware capabilities and when queried by the driver it report only 2 sample rates? The datasheet make few example of usb audio topology configured via firmware so maybe there is a configuration (actually undocumented) that limit the microphone sampe rate to only those 2 (just guessing). Reading the driver source (snd_usb_audio) I undestand that there are some quirks to customize the capabilities of different usb audio card, do you think adding a quirk for this chip and trying to overcome the firmware configuration and adding support for other sample rate can be a solution? If someone could help me going on track to modify the source I can try to prepare a patch to add quirks for this chip or if someone is interested in investigating into the problem I can sponsor a soundcard for testing and debugging (I'ts not much I know, very few dollars). If the firmware is the culprit of the problem does anyone know if there is alternative firmware available and the eeprom is still writeable after the OEM have wrote on it? many thanks Luigi

2 1

[alsa-devel] [PATCH 00/14] soundwire: Add a new SoundWire subsystem
by Vinod Koul 10 Nov '17

10 Nov '17

So today we celebrate Diwali, the festival of lights in this part of the world. On this occasion I would like to add a new Linux subsystem. This patch series adds a new SoundWire subsystem which implements a new MIPI bus protocol 'SoundWire'. The SoundWire protocol is a robust, scalable, low complexity, low power, low latency, two-pin (clock and data) multi-drop bus that allows for the transfer of multiple audio streams and embedded control/commands. SoundWire provides synchronization capabilities and supports both PCM and PDM, multichannel data, isochronous and asynchronous modes. This series adds SoundWire Bus, IO transfers, DisCo (Discovery and Configuration) sysfs interface, regmap and Documentation summary This patch series is also available on git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git topic/patch_v1 This is first version of the patch which incorporates lots of changes based on the RFC series posted last year https://lkml.org/lkml/2016/10/21/395 Sanyog Kale (4): Documentation: Add SoundWire summary soundwire: Add SoundWire MIPI defined registers soundwire: Add Slave status handling helpers soundwire: cdns: Add sdw_master_ops and IO transfer support Vinod Koul (10): soundwire: Add SoundWire bus type soundwire: Add Master registration soundwire: Add MIPI DisCo property helpers soundwire: Add IO transfer regmap: Add SoundWire bus support soundwire: Add slave status handling soundwire: Add sysfs for SoundWire DisCo properties soundwire: cdns: Add cadence module soundwire: intel: Add Intel Master driver soundwire: intel: Add Intel init module Documentation/soundwire/summary.txt | 192 +++++++ drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw.c | 146 +++++ drivers/soundwire/Kconfig | 37 ++ drivers/soundwire/Makefile | 18 + drivers/soundwire/bus.c | 983 ++++++++++++++++++++++++++++++++ drivers/soundwire/bus.h | 119 ++++ drivers/soundwire/bus_type.c | 245 ++++++++ drivers/soundwire/cadence_master.c | 793 ++++++++++++++++++++++++++ drivers/soundwire/cadence_master.h | 94 +++ drivers/soundwire/intel.c | 395 +++++++++++++ drivers/soundwire/intel.h | 70 +++ drivers/soundwire/intel_init.c | 230 ++++++++ drivers/soundwire/mipi_disco.c | 421 ++++++++++++++ drivers/soundwire/slave.c | 173 ++++++ drivers/soundwire/sysfs.c | 396 +++++++++++++ include/linux/mod_devicetable.h | 7 + include/linux/regmap.h | 37 ++ include/linux/soundwire/sdw.h | 564 ++++++++++++++++++ include/linux/soundwire/sdw_intel.h | 72 +++ include/linux/soundwire/sdw_registers.h | 238 ++++++++ scripts/mod/devicetable-offsets.c | 5 + scripts/mod/file2alias.c | 16 + 26 files changed, 5259 insertions(+) create mode 100644 Documentation/soundwire/summary.txt create mode 100644 drivers/base/regmap/regmap-sdw.c create mode 100644 drivers/soundwire/Kconfig create mode 100644 drivers/soundwire/Makefile create mode 100644 drivers/soundwire/bus.c create mode 100644 drivers/soundwire/bus.h create mode 100644 drivers/soundwire/bus_type.c create mode 100644 drivers/soundwire/cadence_master.c create mode 100644 drivers/soundwire/cadence_master.h create mode 100644 drivers/soundwire/intel.c create mode 100644 drivers/soundwire/intel.h create mode 100644 drivers/soundwire/intel_init.c create mode 100644 drivers/soundwire/mipi_disco.c create mode 100644 drivers/soundwire/slave.c create mode 100644 drivers/soundwire/sysfs.c create mode 100644 include/linux/soundwire/sdw.h create mode 100644 include/linux/soundwire/sdw_intel.h create mode 100644 include/linux/soundwire/sdw_registers.h -- 2.7.4

9 93

[alsa-devel] [RFC PATCH v2 0/7] Fix year 2038 issue for sound subsystem
by Baolin Wang 10 Nov '17

10 Nov '17

Since many structures will use timespec type variables to record time stamp in uapi/asound.h, which are not year 2038 safe on 32bit system. This patchset tries to introduce new structures removing timespec type to compatible native mode and compat mode. Moreover this patchset also converts the internal structrures to use timespec64 type and related APIs. Changes since v1: - Flat all structues to make them more clear. - Re-modify the convertion for struct snd_timer_tread. - Use same coding style when converting struct snd_rawmidi_status. - Use struct snd_timer_user_status64 in compat mode directly when converting struct snd_timer_status. - Add #ifdef __KERNEL__ when converting struct snd_ctl_elem_value. - Re-modify the convertion for struct snd_pcm_sync_ptr. - Fix some building errors. Appreciated for any comments, especially for patch6 which I am not sure if it is correct. Baolin Wang (7): sound: Replace timespec with timespec64 sound: core: Avoid using timespec for struct snd_pcm_status sound: core: Avoid using timespec for struct snd_rawmidi_status sound: core: Avoid using timespec for struct snd_timer_status uapi: sound: Avoid using timespec for struct snd_ctl_elem_value sound: core: Avoid using timespec for struct snd_pcm_sync_ptr sound: core: Avoid using timespec for struct snd_timer_tread include/sound/pcm.h | 126 +++++++++- include/sound/timer.h | 4 +- include/uapi/sound/asound.h | 19 +- sound/core/pcm.c | 20 +- sound/core/pcm_compat.c | 469 ++++++++++++++++++++++++++++--------- sound/core/pcm_lib.c | 33 ++- sound/core/pcm_native.c | 220 +++++++++++++---- sound/core/rawmidi.c | 81 ++++++- sound/core/rawmidi_compat.c | 121 +++++++--- sound/core/timer.c | 253 ++++++++++++++++---- sound/core/timer_compat.c | 67 ++---- sound/pci/hda/hda_controller.c | 10 +- sound/soc/intel/skylake/skl-pcm.c | 4 +- 13 files changed, 1090 insertions(+), 337 deletions(-) -- 1.7.9.5

4 19

[alsa-devel] [PATCH v3] ASoC: rt5663: Delay and retry reading rt5663 ID register
by Oder Chiou 09 Nov '17

09 Nov '17

In the probe, the codec may not be ready for I2C reading or there are some glitches on the i2c line. So if the i2c reading value is incorrect, it will read again after delay. This issue is similar the patch https://patchwork.kernel.org/patch/9681421/. In current project, these 2 devices were connected to the same i2c line, and they met the same problem. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index fcd1643..18dc1ec 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3429,7 +3429,16 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret); return ret; } - regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + + ret = regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + if (ret || (val != RT5663_DEVICE_ID_2 && val != RT5663_DEVICE_ID_1)) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5663, retry one time.\n", + val); + msleep(100); + regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + } + switch (val) { case RT5663_DEVICE_ID_2: rt5663->regmap = devm_regmap_init_i2c(i2c, &rt5663_v2_regmap); -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: sun8i-codec: Set the BCLK divider
by Maxime Ripard 09 Nov '17

09 Nov '17

While the current code was reporting to be able to work in master mode, it failed to do so because the BCLK divider wasn't programmed, meaning that the BCLK would run at the PLL's frequency no matter the sample rate. It was obviously a bit too fast. Add support to retrieve the divider to use, and set it. Since our PLL is not always able to generate a perfect multiple of the sample rate, we'll have to choose the closest divider that matches our setup. Fixes: 36c684936fae ("ASoC: Add sun8i digital audio codec") Cc: <stable(a)vger.kernel.org> Reviewed-by: Chen-Yu Tsai <wens(a)csie.org> Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- Changes from v1: - Hardcoded the bit width to remain consistent with the rest of the driver --- sound/soc/sunxi/sun8i-codec.c | 51 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index c2ceca485d6a..7a312168f864 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -73,6 +73,7 @@ #define SUN8I_SYS_SR_CTRL_AIF2_FS_MASK GENMASK(11, 8) #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_MASK GENMASK(5, 4) #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK GENMASK(8, 6) +#define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK GENMASK(12, 9) struct sun8i_codec { struct device *dev; @@ -226,12 +227,57 @@ static int sun8i_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +struct sun8i_codec_clk_div { + u8 div; + u8 val; +}; + +static const struct sun8i_codec_clk_div sun8i_codec_bclk_div[] = { + { .div = 1, .val = 0 }, + { .div = 2, .val = 1 }, + { .div = 4, .val = 2 }, + { .div = 6, .val = 3 }, + { .div = 8, .val = 4 }, + { .div = 12, .val = 5 }, + { .div = 16, .val = 6 }, + { .div = 24, .val = 7 }, + { .div = 32, .val = 8 }, + { .div = 48, .val = 9 }, + { .div = 64, .val = 10 }, + { .div = 96, .val = 11 }, + { .div = 128, .val = 12 }, + { .div = 192, .val = 13 }, +}; + +static u8 sun8i_codec_get_bclk_div(struct sun8i_codec *scodec, + unsigned int rate, + unsigned int word_size) +{ + unsigned long clk_rate = clk_get_rate(scodec->clk_module); + unsigned int div = clk_rate / rate / word_size / 2; + unsigned int best_val = 0, best_diff = ~0; + int i; + + for (i = 0; i < ARRAY_SIZE(sun8i_codec_bclk_div); i++) { + const struct sun8i_codec_clk_div *bdiv = &sun8i_codec_bclk_div[i]; + unsigned int diff = abs(bdiv->div - div); + + if (diff < best_diff) { + best_diff = diff; + best_val = bdiv->val; + } + } + + return best_val; +} + static int sun8i_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct sun8i_codec *scodec = snd_soc_codec_get_drvdata(dai->codec); int sample_rate; + u8 bclk_div; /* * The CPU DAI handles only a sample of 16 bits. Configure the @@ -241,6 +287,11 @@ static int sun8i_codec_hw_params(struct snd_pcm_substream *substream, SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_MASK, SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_16); + bclk_div = sun8i_codec_get_bclk_div(scodec, params_rate(params), 16); + regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, + SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK, + bclk_div << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV); + regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK, SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16); -- 2.14.3

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[alsa-devel] [PATCH] ASoC: sun8i-codec: Add a comment on the LRCK inversion
by Maxime Ripard 09 Nov '17

09 Nov '17

The current code might be a bit intriguing without having experienced the issue before, and might come up as a mistake. Make explicit what's going on by adding a comment. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/soc/sunxi/sun8i-codec.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 7a312168f864..3dd183be08a4 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -198,6 +198,17 @@ static int sun8i_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, BIT(SUN8I_AIF1CLK_CTRL_AIF1_BCLK_INV), value << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_INV); + + /* + * It appears that the DAI and the codec don't share the same + * polarity for the LRCK signal when they mean 'normal' and + * 'inverted' in the datasheet. + * + * Since the DAI here is our regular i2s driver that have been + * tested with way more codecs than just this one, it means + * that the codec probably gets it backward, and we have to + * invert the value here. + */ regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL, BIT(SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV), !value << SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV); -- 2.14.3

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[alsa-devel] [PATCH] ASoC: da7213: add support for DSP modes
by Pierre-Louis Bossart 09 Nov '17

09 Nov '17

DSP modes are documented in the data sheet but not enabled in the driver. The work-around already implemented for DA7218/9 is also required to make sure the bit clock handling in DSP modes follows ASoC conventions. Tested with ARD-AUDIO-DA7212 and Minnowmax Turbot boards Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/da7213.c | 58 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/da7213.h | 1 + 2 files changed, 49 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index cc0b2d2eaf15..41d9b1da27c2 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1220,6 +1220,7 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct snd_soc_codec *codec = codec_dai->codec; struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); u8 dai_clk_mode = 0, dai_ctrl = 0; + u8 dai_offset = 0; /* Set master/slave mode */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1234,17 +1235,46 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } /* Set clock normal/inverted */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - dai_clk_mode |= DA7213_DAI_WCLK_POL_INV; - break; - case SND_SOC_DAIFMT_IB_NF: - dai_clk_mode |= DA7213_DAI_CLK_POL_INV; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7213_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV | + DA7213_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } break; - case SND_SOC_DAIFMT_IB_IF: - dai_clk_mode |= DA7213_DAI_WCLK_POL_INV | DA7213_DAI_CLK_POL_INV; + case SND_SOC_DAI_FORMAT_DSP_A: + case SND_SOC_DAI_FORMAT_DSP_B: + /* The bclk is inverted wrt ASoC conventions */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + dai_clk_mode |= DA7213_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV | + DA7213_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV; + break; + default: + return -EINVAL; + } break; default: return -EINVAL; @@ -1261,6 +1291,13 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_RIGHT_J: dai_ctrl |= DA7213_DAI_FORMAT_RIGHT_J; break; + case SND_SOC_DAI_FORMAT_DSP_A: /* L data MSB after FRM LRC */ + dai_ctrl |= DA7213_DAI_FORMAT_DSP; + dai_offset = 1; + break; + case SND_SOC_DAI_FORMAT_DSP_B: /* L data MSB during FRM LRC */ + dai_ctrl |= DA7213_DAI_FORMAT_DSP; + break; default: return -EINVAL; } @@ -1271,6 +1308,7 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) snd_soc_write(codec, DA7213_DAI_CLK_MODE, dai_clk_mode); snd_soc_update_bits(codec, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK, dai_ctrl); + snd_soc_write(codec, DA7213_DAI_OFFSET, dai_offset); return 0; } diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 16ef56f77cd4..5a78dba1dcb5 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -188,6 +188,7 @@ #define DA7213_DAI_FORMAT_I2S_MODE (0x0 << 0) #define DA7213_DAI_FORMAT_LEFT_J (0x1 << 0) #define DA7213_DAI_FORMAT_RIGHT_J (0x2 << 0) +#define DA7213_DAI_FORMAT_DSP (0x3 << 0) #define DA7213_DAI_FORMAT_MASK (0x3 << 0) #define DA7213_DAI_WORD_LENGTH_S16_LE (0x0 << 2) #define DA7213_DAI_WORD_LENGTH_S20_LE (0x1 << 2) -- 2.11.0

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[alsa-devel] Applied "ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+" to the asoc tree
by Mark Brown 09 Nov '17

09 Nov '17

The patch ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+ has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8a0f95ce48304364979066600eba7cc5c939ba83 Mon Sep 17 00:00:00 2001 From: Andrea Bondavalli <andrea.bondavalli74(a)gmail.com> Date: Tue, 31 Oct 2017 12:58:18 +0100 Subject: [PATCH] ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+ 32bit and 24bit audio capture formats for H3/H2+ are broken because the RX_SAMPLE_BITS and the RX_FIFO_MODE bits of AC_ADC_FIFOC register of the audio codec are not set to operate in 24bit mode but in 16bit mode only. The following patch sets the H3 audio codec registers and the DMA bus width properly when a 24/32bit capture is requested. Signed-off-by: Andrea Bondavalli <andrea.bondavalli74(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index baa9007464ed..5da4efe7a550 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -346,11 +346,6 @@ static int sun4i_codec_prepare_capture(struct snd_pcm_substream *substream, 0x3 << 8, 0x1 << 8); - /* Fill most significant bits with valid data MSB */ - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), - BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); - return 0; } @@ -490,6 +485,30 @@ static int sun4i_codec_hw_params_capture(struct sun4i_codec *scodec, BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), 0); + /* Set the number of sample bits to either 16 or 24 bits */ + if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) { + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS), + BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS)); + + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), + 0); + + scodec->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + } else { + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS), + 0); + + /* Fill most significant bits with valid data MSB */ + regmap_field_update_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); + + scodec->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + } + return 0; } -- 2.15.0

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[alsa-devel] [PATCH v3] ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2
by Alex Deucher 09 Nov '17

09 Nov '17

From: Akshu Agrawal <akshu.agrawal(a)amd.com> Minimum time required between power On of codec and read of RT5645_VENDOR_ID2 is 400msec. We should wait that long before reading the value. TEST=Cold boot the device and check for sound device. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- v2: rework the patch based on mailing list discussion. Just wait before reading the register. v3: Drop the extra read sound/soc/codecs/rt5645.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 23cc2cb8393f..5f24df4fae8e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -55,6 +55,8 @@ MODULE_PARM_DESC(quirk, "RT5645 pdata quirk override"); #define RT5645_HWEQ_NUM 57 +#define TIME_TO_POWER_MS 400 + static const struct regmap_range_cfg rt5645_ranges[] = { { .name = "PR", @@ -3784,6 +3786,12 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret); return ret; } + + /* + * Read after 400msec, as it is the interval required between + * read and power On. + */ + msleep(TIME_TO_POWER_MS); regmap_read(regmap, RT5645_VENDOR_ID2, &val); switch (val) { -- 2.13.6

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