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

[alsa-devel] [patch] ALSA: firewire-tascam: off by one in identify_model()
by Dan Carpenter 19 Oct '15

19 Oct '15

In the original code, we potentially put a NUL character in model[8] and it caused a static checker warning. We can put the NUL in model[7] instead. Fixes: 53b3ffee7885 ('ALSA: firewire-tascam: change device probing processing') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index c6747a4..6ef2bcc 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -51,7 +51,7 @@ static int identify_model(struct snd_tscm *tscm) } /* Pick up model name from certain addresses. */ - for (i = 0; i < 8; i++) { + for (i = 0; i < 7; i++) { c = config_rom[28 + i / 4] >> (24 - 8 * (i % 4)); if (c == '\0') break;

3 6

[alsa-devel] [PATCH 0/3] ALSA: oxfw: add support for TASCAM FireOne
by Takashi Sakamoto 19 Oct '15

19 Oct '15

Hi, This parchset adds support for TASCAM FireOne to ALSA OXFW driver. PCM and MIDI functionalities become available including physical controls. The first and second patches are preparation for it. This model supports some extra commands. So as the other ALSA drivers for sound and music units on IEEE 1394 bus, they should be implemented in userspace applications. For your information, I note them: (Jog/Shuttle display mode) ./firewire-request /dev/fw1 fcp 0x01ff0000022e46493110ffff response: 000: 0c ff 00 00 02 2e 46 49 31 10 06 * The last byte is: * 0x00: always off * 0x01: always on * 0x02: breathe * 0x03: metronome * 0x04: MIDI clock rotate * 0x05: MIDI clock flash * 0x06: jog slow rotate * 0x07: Jog track (Control surface mode) ./firewire-request /dev/fw1 fcp 0x01ff0000022e46493111ffff response: 000: 0c ff 00 00 02 2e 46 49 31 11 00 * The last bytes is: * 0x00: FireOne native * 0x01: Mackie Human User Interface emulation (Input monitor monaural) ./firewire-request /dev/fw1 fcp 0x01ff0000022e46493112ffff response: 000: 0c ff 00 00 02 2e 46 49 31 12 00 * The last byte is: * 0x00: off * 0x01: on (Firmware version, read-only) ./firewire-request /dev/fw1 fcp 0x01ff0000022e46493113ffff response: 000: 0c ff 00 00 02 2e 46 49 31 13 02 * The last byte is: * 0x02: version 2 I can see additional commands from Windows driver. They seem to be used to retrieve the name of data channels. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0000000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 00 00 12 0c ......HST`...... response: 010: 41 75 64 69 6f 20 4f 75 74 20 4c 00 Audio Out L. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0000010040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 00 01 12 0c ......HST`...... response: 010: 41 75 64 69 6f 20 4f 75 74 20 52 00 Audio Out R. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0000020040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 00 02 12 09 ......HST`...... response: 010: 4d 69 64 69 20 4f 75 74 20 Midi Out ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0001000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 01 00 12 0b ......HST`...... response: 010: 41 75 64 69 6f 20 49 6e 20 41 00 Audio In A. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0001010040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 01 01 12 0b ......HST`...... response: 010: 41 75 64 69 6f 20 49 6e 20 42 00 Audio In B. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0001020040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 01 02 12 08 ......HST`...... response: 010: 4d 69 64 69 20 49 6e 20 Midi In ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0000000040 response: 000: 0c ff 00 00 02 2e 48 53 54 70 ff 00 00 00 12 06 ......HSTp...... response: 010: 41 75 64 69 6f 20 Audio ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0000010040 response: 000: 0c ff 00 00 02 2e 48 53 54 70 ff 00 00 01 12 00 ......HSTp...... ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0000020040 response: 000: 0a ff 00 00 02 2e 48 53 54 70 ff 00 00 02 00 40 ......HSTp.....@ ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0001000040 response: 000: 0c ff 00 00 02 2e 48 53 54 70 ff 00 01 00 12 06 ......HSTp...... response: 010: 41 75 64 69 6f 20 Audio ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0001010040 response: 000: 0c ff 00 00 02 2e 48 53 54 70 ff 00 01 01 12 00 ......HSTp...... ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0001020040 response: 000: 0a ff 00 00 02 2e 48 53 54 70 ff 00 01 02 00 40 ......HSTp.....@ ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff8100000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 81 00 00 12 0c ......HST`...... response: 010: 4d 69 64 69 20 49 6e 20 4f 6e 65 00 Midi In One. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff8101000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 81 01 00 12 0d ......HST`...... response: 010: 4d 69 64 69 20 4f 75 74 20 4f 6e 65 00 Midi Out One. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff8200000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 82 00 00 12 0c ......HST`...... response: 010: 4d 69 64 69 20 49 6e 20 54 77 6f 00 Midi In Two. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff8201000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 82 01 00 12 0d ......HST`...... response: 010: 4d 69 64 69 20 4f 75 74 20 54 77 6f 00 Midi Out Two. Takashi Sakamoto (3): ALSA: oxfw: calculating MIDI ports in stream discover ALSA: oxfw: support more MIDI ports ALSA: oxfw: add an entry for TASCAM FireOne sound/firewire/oxfw/oxfw-midi.c | 24 +++--------------------- sound/firewire/oxfw/oxfw-stream.c | 37 +++++++++++++++++++++++++++++++++++-- sound/firewire/oxfw/oxfw.c | 17 +++++++++++++++++ 3 files changed, 55 insertions(+), 23 deletions(-) -- 2.1.4

2 5

[alsa-devel] [PATCH v3] drm/i915: Improve kernel-doc for i915_audio_component struct
by David Henningsson 19 Oct '15

19 Oct '15

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Now rebased against drm-intel git master. include/drm/i915_component.h | 69 ++++++++++++++++++++++++++++++++------------ 1 file changed, 51 insertions(+), 18 deletions(-) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index 30d89e0..fab1385 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -31,47 +31,80 @@ #define MAX_PORTS 5 /** - * struct i915_audio_component_ops - callbacks defined in gfx driver - * @owner: the module owner - * @get_power: get the POWER_DOMAIN_AUDIO power well - * @put_power: put the POWER_DOMAIN_AUDIO power well - * @codec_wake_override: Enable/Disable generating the codec wake signal - * @get_cdclk_freq: get the Core Display Clock in KHz - * @sync_audio_rate: set n/cts based on the sample rate + * struct i915_audio_component_ops - Ops implemented by i915 driver, called by hda driver */ struct i915_audio_component_ops { + /** + * @owner: i915 module + */ struct module *owner; + /** + * @get_power: get the POWER_DOMAIN_AUDIO power well + * + * Request the power well to be turned on. + */ void (*get_power)(struct device *); + /** + * @put_power: put the POWER_DOMAIN_AUDIO power well + * + * Allow the power well to be turned off. + */ void (*put_power)(struct device *); + /** + * @codec_wake_override: Enable/disable codec wake signal + */ void (*codec_wake_override)(struct device *, bool enable); + /** + * @get_cdclk_freq: Get the Core Display Clock in kHz + */ int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ int (*sync_audio_rate)(struct device *, int port, int rate); }; +/** + * struct i915_audio_component_audio_ops - Ops implemented by hda driver, called by i915 driver + */ struct i915_audio_component_audio_ops { + /** + * @audio_ptr: Pointer to be used in call to pin_eld_notify + */ void *audio_ptr; /** - * Call from i915 driver, notifying the HDA driver that - * pin sense and/or ELD information has changed. - * @audio_ptr: HDA driver object - * @port: Which port has changed (PORTA / PORTB / PORTC etc) + * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed + * + * Called when the i915 driver has set up audio pipeline or has just + * begun to tear it down. This allows the HDA driver to update its + * status accordingly (even when the HDA controller is in power save + * mode). */ void (*pin_eld_notify)(void *audio_ptr, int port); }; /** - * struct i915_audio_component - used for audio video interaction - * @dev: the device from gfx driver - * @aud_sample_rate: the array of audio sample rate per port - * @ops: callback for audio driver calling - * @audio_ops: Call from i915 driver + * struct i915_audio_component - Used for direct communication between i915 and hda drivers */ struct i915_audio_component { + /** + * @dev: i915 device, used as parameter for ops + */ struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ int aud_sample_rate[MAX_PORTS]; - + /** + * @ops: Ops implemented by i915 driver, called by hda driver + */ const struct i915_audio_component_ops *ops; - + /** + * @audio_ops: Ops implemented by hda driver, called by i915 driver + */ const struct i915_audio_component_audio_ops *audio_ops; }; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/5] ALSA: firewire: suppress sparse warnings due to endianness handling
by Takashi Sakamoto 19 Oct '15

19 Oct '15

Hi, This patchset fixes some issues related to sparse warnings about endianness handling for ALSA firewire stack. For writing this patchset, I use 'C=1 CF=-D__CHECK_ENDIAN__' options to 'make' command. If they're not enough, please inform to me. Takashi Sakamoto (5): ALSA: dice: correct variable types for __be32 data ALSA: dice: assign converted data to the same type of variable ALSA: fireworks: use u32 type for be32_to_cpup() macro ALSA: bebob: use correct type for __be32 data ALSA: firewire-digi00x/firewire-tascam: remove wrong conversion for Config ROM sound/firewire/bebob/bebob_maudio.c | 2 +- sound/firewire/dice/dice-stream.c | 12 ++++++------ sound/firewire/dice/dice.c | 3 ++- sound/firewire/digi00x/digi00x.c | 3 +-- sound/firewire/fireworks/fireworks_command.c | 2 +- sound/firewire/tascam/tascam.c | 3 +-- 6 files changed, 12 insertions(+), 13 deletions(-) -- 2.1.4

2 6

[alsa-devel] [PATCH] ALSA: firewire-lib: avoid NULL pointer dereference after closing MIDI port
by Takashi Sakamoto 19 Oct '15

19 Oct '15

When asynchronous MIDI port is closed before callbacked, the callback function causes NULL pointer dereference to missing MIDI substream. This commit fixes this bug. Fixes: e8a40d9bcb23('ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages') Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/lib.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c index edf1c8b..f80aafa 100644 --- a/sound/firewire/lib.c +++ b/sound/firewire/lib.c @@ -74,7 +74,11 @@ static void async_midi_port_callback(struct fw_card *card, int rcode, struct snd_fw_async_midi_port *port = callback_data; struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream); - if (rcode == RCODE_COMPLETE && substream != NULL) + /* This port is closed. */ + if (substream == NULL) + return; + + if (rcode == RCODE_COMPLETE) snd_rawmidi_transmit_ack(substream, port->consume_bytes); else if (!rcode_is_permanent_error(rcode)) /* To start next transaction immediately for recovery. */ -- 2.1.4

2 1

[alsa-devel] [PATCH v6 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 19 Oct '15

19 Oct '15

From: Libin Yang <libin.yang(a)intel.com> Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..8ad6f1b 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,13 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

5 20

[alsa-devel] audio-over-SPI driver for ASoC
by Brian Clarkson 18 Oct '15

18 Oct '15

Hi I'm about to embark on the development of an ASoC capture driver that uses SPI to stream the samples to the processor. I'm still spending a lot of my time going through the ALSA kernel code but as I do that may I have some suggestions as to which examples would be closest to my case? My development hardware is AM335x --> McSPI --> ADS8868. So far I think I have achieved a solid understanding of the "simple-card" case where you have dai-link that connects a processor peripheral driver (e.g. McASP ) with a codec driver and forms a snd_soc_card that can be registered. Is my best option to: - write a new CPU DAI (snd_soc_dai_driver) based on the existing SPI driver - write a new CODEC DAI (snd_soc_dai_driver) which will be mostly empty - write a new platform driver that populates and registers a new snd_soc_card struct with the above in a dai_link? Any advice would be much appreciated. Cheers, Brian -- Orthogonal Devices Tokyo, Japan www.orthogonaldevices.com +81 80 8700 2340

2 1

[alsa-devel] [PATCH 0/3] fix sending the same MIDI bytes in two transactions
by Takashi Sakamoto 18 Oct '15

18 Oct '15

Hi, In former commits, MIDI functionality is added to TASCAM driver, while it includes a bug to send the same MIDI bytes in two continuous transactions. This patchset is to fix this bug, including some code improvements. Takashi Sakamoto (3): ALSA: tascam: remove buffer initialization in driver side ALSA: tascam: improve loop condition to filling MIDI bytes in transaction ALSA: tascam: clear extra MIDI byte of asynchronous transaction sound/firewire/tascam/tascam-transaction.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) -- 2.1.4

2 7

[alsa-devel] Sildes of topology status for ELEC 2015
by Lin, Mengdong 18 Oct '15

18 Oct '15

Hi, The slides of topology status is attached. And Libin will update DP MST audio slides based on recent alignment. Thanks Mengdong

4 6

[alsa-devel] [PATCH] ALSA: USB-audio: Add support for Novation Nocturn MIDIcontrol surface
by Ricard Wanderlof 18 Oct '15

18 Oct '15

The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices. Tested that the Nocturn shows up in aconnect, and that it can be used as a control surface (using the xtor synthesizer patch editor). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- I've noticed that checkpatch complains about the space between the & and (, but in the interest of consistency I've not changed this. Is there some reason for the space, or would it be worth while cleaning up the file? sound/usb/quirks-table.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e475665..99de061 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2664,6 +2664,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE(0x1235, 0x000a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Novation", */ + /* .product_name = "Nocturn", */ + .ifnum = 0, + .type = QUIRK_MIDI_RAW_BYTES + } +}, +{ USB_DEVICE(0x1235, 0x000e), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "Novation", */ -- 2.1.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 4

[alsa-devel] [PATCH] ALSA: fireworks/bebob/oxfw/dice: enable to make as built-in
by Takashi Sakamoto 18 Oct '15

18 Oct '15

When committed to upstream, these four modules had wrong entries for Makefile. This forces them to be loadable modules even if they're set as built-in. This commit fixes this bug. Fixes: b5b04336015e('ALSA: fireworks: Add skelton for Fireworks based devices') Fixes: fd6f4b0dc167('ALSA: bebob: Add skelton for BeBoB based devices') Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory') Fixes: 14ff6a094815('ALSA: dice: Move file to its own directory') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/Makefile | 2 +- sound/firewire/dice/Makefile | 2 +- sound/firewire/fireworks/Makefile | 2 +- sound/firewire/oxfw/Makefile | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/Makefile b/sound/firewire/bebob/Makefile index 6cf470c..af7ed66 100644 --- a/sound/firewire/bebob/Makefile +++ b/sound/firewire/bebob/Makefile @@ -1,4 +1,4 @@ snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \ bebob_pcm.o bebob_hwdep.o bebob_terratec.o bebob_yamaha.o \ bebob_focusrite.o bebob_maudio.o bebob.o -obj-m += snd-bebob.o +obj-$(CONFIG_SND_BEBOB) += snd-bebob.o diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 9ef228e..55b4be9 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,3 +1,3 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o -obj-m += snd-dice.o +obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/fireworks/Makefile b/sound/firewire/fireworks/Makefile index 0c74408..15ef7f7 100644 --- a/sound/firewire/fireworks/Makefile +++ b/sound/firewire/fireworks/Makefile @@ -1,4 +1,4 @@ snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \ fireworks_stream.o fireworks_proc.o fireworks_midi.o \ fireworks_pcm.o fireworks_hwdep.o fireworks.o -obj-m += snd-fireworks.o +obj-$(CONFIG_SND_FIREWORKS) += snd-fireworks.o diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index a926850..06ff50f 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,3 +1,3 @@ snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \ oxfw-proc.o oxfw-midi.o oxfw-hwdep.o oxfw.o -obj-m += snd-oxfw.o +obj-$(CONFIG_SND_OXFW) += snd-oxfw.o -- 2.1.4

2 3

[alsa-devel] [PATCH 2/3] ALSA: hda - Remove leftover snd_hda_bus() prototype
by Takashi Iwai 17 Oct '15

17 Oct '15

It was forgotten to be removed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5bcfe7407d83..373fcad840ea 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -301,10 +301,6 @@ struct hda_codec { /* * constructors */ -int snd_hda_bus_new(struct snd_card *card, - const struct hdac_bus_ops *ops, - const struct hdac_io_ops *io_ops, - struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); -- 2.6.0

1 1

[alsa-devel] [PATCH 1/3] ALSA: hda - Fix bogus codec address check for mixer name assignment
by Takashi Iwai 17 Oct '15

17 Oct '15

The recent commit [7fbe824a0f0e: ALSA: hda - Update mixer name for the lower codec address] tried to improve the mixer chip name assignment in the order of codec address. However, this fix was utterly bogus; it checks the field set in each codec, thus this value is reset at each codec creation, of course. For really handling this priority, the assignment has to be remembered in the common place, namely in hda_bus, instead of hda_codec. Fixes: 7fbe824a0f0e ('ALSA: hda - Update mixer name for the lower codec address') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_bind.c | 4 ++-- sound/pci/hda/hda_codec.c | 1 - sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_controller.c | 1 + 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 152acdaa0a45..70671ad65d24 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -63,11 +63,11 @@ int snd_hda_codec_set_name(struct hda_codec *codec, const char *name) /* update the mixer name */ if (!*codec->card->mixername || - codec->mixer_assigned >= codec->core.addr) { + codec->bus->mixer_assigned >= codec->core.addr) { snprintf(codec->card->mixername, sizeof(codec->card->mixername), "%s %s", codec->core.vendor_name, codec->core.chip_name); - codec->mixer_assigned = codec->core.addr; + codec->bus->mixer_assigned = codec->core.addr; } return 0; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0e55c6a6cc7e..2eeaf5ea20f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -851,7 +851,6 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; codec->fixup_id = HDA_FIXUP_ID_NOT_SET; - codec->mixer_assigned = -1; #ifdef CONFIG_PM codec->power_jiffies = jiffies; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 055a80522282..5bcfe7407d83 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -70,6 +70,7 @@ struct hda_bus { unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ + unsigned int mixer_assigned; /* codec addr for mixer name */ }; /* from hdac_bus to hda_bus */ @@ -260,7 +261,6 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; #endif - unsigned int mixer_assigned; /* filter the requested power state per nid */ unsigned int (*power_filter)(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 944455997fdc..d6b93a20361b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1045,6 +1045,7 @@ int azx_bus_init(struct azx *chip, const char *model, mutex_init(&bus->prepare_mutex); bus->pci = chip->pci; bus->modelname = model; + bus->mixer_assigned = -1; bus->core.snoop = azx_snoop(chip); if (chip->get_position[0] != azx_get_pos_lpib || chip->get_position[1] != azx_get_pos_lpib) -- 2.6.0

1 0

[alsa-devel] [patch 2/2] ALSA: firewire-tascam: off by one in handle_midi_tx()
by Dan Carpenter 17 Oct '15

17 Oct '15

My static checker complains because tscm->spec->midi_capture_ports is either 2 or 4 but the tscm->tx_midi_substreams[] array has 4 elements so this is possibly off by one. I have looked at the code and I think it should be >= instead of > as well. Fixes: 107cc0129a68 ('ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 45c3ce3..d4f64ae 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -158,7 +158,7 @@ static void handle_midi_tx(struct fw_card *card, struct fw_request *request, port = b[0] >> 4; /* TODO: support virtual MIDI ports. */ - if (port > tscm->spec->midi_capture_ports) + if (port >= tscm->spec->midi_capture_ports) goto end; /* Assume the message length. */

2 1

[alsa-devel] [patch 1/2] ALSA: firewire-tascam: fix an LED bug
by Dan Carpenter 17 Oct '15

17 Oct '15

We recently tried to add some new code to support turning the LED on and off but the code in snd_tscm_transaction_reregister() is unreachable. Fixes: e65e2cb99e44 ('ALSA: firewire-tascam: Turn on/off FireWire LED') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 1c9a88b..45c3ce3 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -249,7 +249,7 @@ int snd_tscm_transaction_reregister(struct snd_tscm *tscm) /* Turn on messaging. */ reg = cpu_to_be32(0x00000001); - return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON, &reg, sizeof(reg), 0); if (err < 0)

2 1

[alsa-devel] [PATCH] ASoC: rt5645: Recheck the jack detect status after resuming from S3
by Oder Chiou 16 Oct '15

16 Oct '15

The patch rechecks the jack detect status after resuming from S3. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 89ba4ff..4d54999 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2745,6 +2745,10 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_FV1 | RT5645_PWR_FV2, RT5645_PWR_FV1 | RT5645_PWR_FV2); + if (rt5645->en_button_func && + snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, msecs_to_jiffies(0)); break; case SND_SOC_BIAS_OFF: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH RESEND] ASoC: rt298: correct index default value
by Bard Liao 16 Oct '15

16 Oct '15

Some of the default value on rt298_index_def are incorrect. Change them to the correct value. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This patch is base on topic/rt298 branch. --- sound/soc/codecs/rt298.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index ff126a7..8c5cb7e 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -50,24 +50,24 @@ struct rt298_priv { }; static struct reg_default rt298_index_def[] = { - { 0x01, 0xaaaa }, - { 0x02, 0x8aaa }, + { 0x01, 0xa5a8 }, + { 0x02, 0x8e95 }, { 0x03, 0x0002 }, - { 0x04, 0xaf01 }, - { 0x08, 0x000d }, - { 0x09, 0xd810 }, - { 0x0a, 0x0120 }, + { 0x04, 0xaf67 }, + { 0x08, 0x200f }, + { 0x09, 0xd010 }, + { 0x0a, 0x0100 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, - { 0x0f, 0x0000 }, - { 0x19, 0x0a17 }, + { 0x0f, 0x0022 }, + { 0x19, 0x0217 }, { 0x20, 0x0020 }, { 0x33, 0x0208 }, { 0x46, 0x0300 }, - { 0x49, 0x0004 }, - { 0x4f, 0x50e9 }, - { 0x50, 0x2000 }, - { 0x63, 0x2902 }, + { 0x49, 0x4004 }, + { 0x4f, 0x50c9 }, + { 0x50, 0x3000 }, + { 0x63, 0x1b02 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, { 0x69, 0x273f }, -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] Remove Renesas sound SRU for Gen1 support
by Kuninori Morimoto 16 Oct '15

16 Oct '15

Hi ALSA, and SH ML I would like to remove Renesas sound SRU support of Gen1 in the future. It was created for preparation of Gen2 SRC support, and it converts sampling rate. But no-one is using this feature in Gen1. board-bockw had used SRU before, but it was pass through mode. This means it is same as SSI. 2nd reason is that board-bockw was removed from upstream code. BockW is now supported by DT, but it doesn't use SRU. More detail, r8a7778.dtsi has rcar_sound,src entry, but no-one is using this feature. SRU proving has no relation to this removing feature. This means there is no effect for DT compatibility. Gen2 SRC was created from Gen1 SRU, these are similar but not same IP. Keeping Gen1 SRU in current driver is a little bit difficult, and no-one is using it today. Gen1 sound is still supported via SSI. If there is no objection, I would like to remove it from upstream step1. add warning message in v4.x step2. remove it in v4.x+1 Best regards --- Kuninori Morimoto

2 2

[alsa-devel] [PATCH] ASoC: rsnd: Gen1 probe is not error
by Kuninori Morimoto 16 Oct '15

16 Oct '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Probing from Gen1 is not error. This patch fixup it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ctu.c | 6 ++---- sound/soc/sh/rcar/dvc.c | 6 ++---- sound/soc/sh/rcar/mix.c | 6 ++---- 3 files changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 6e34909c..27d56a0 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -125,10 +125,8 @@ int rsnd_ctu_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "CTU is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_ctu(pdev, of_data, priv); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 08e58d0..63e8e52 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -338,10 +338,8 @@ int rsnd_dvc_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "CMD is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_dvc(pdev, of_data, priv); diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 2311d49..a5485b2 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -158,10 +158,8 @@ int rsnd_mix_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "MIX is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_mix(pdev, of_data, priv); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Add info callback for SX_TLV controls
by Charles Keepax 16 Oct '15

16 Oct '15

SX_TLV controls are intended for situations where the register behind the control has some non-zero value indicating the minimum gain and then gains increasing from there and eventually overflowing through zero. Currently every CODEC implementing these controls specifies the minimum as the non-zero value for the minimum and the maximum as the number of gain settings available. This means when the info callback subtracts the minimum value from the maximum value to calculate the number of gain levels available it is actually under reporting the available levels. This patch fixes this issue by adding a new snd_soc_info_volsw_sx callback that does not subtract the minimum value. Fixes: 1d99f2436d0d ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 6 ++++-- sound/soc/soc-ops.c | 28 ++++++++++++++++++++++++++++ 2 files changed, 32 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 98e080b..bd85564 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -86,7 +86,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array),\ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw_sx, \ .get = snd_soc_get_volsw_sx,\ .put = snd_soc_put_volsw_sx, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ @@ -156,7 +156,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw_sx, \ .get = snd_soc_get_volsw_sx, \ .put = snd_soc_put_volsw_sx, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ @@ -593,6 +593,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); #define snd_soc_info_bool_ext snd_ctl_boolean_mono_info int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 100d92b..05977ae 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -207,6 +207,34 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_info_volsw); /** + * snd_soc_info_volsw_sx - Mixer info callback for SX TLV controls + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a single mixer control, or a double + * mixer control that spans 2 registers of the SX TLV type. SX TLV controls + * have a range that represents both positive and negative values either side + * of zero but without a sign bit. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + snd_soc_info_volsw(kcontrol, uinfo); + /* Max represents the number of levels in an SX control not the + * maximum value, so add the minimum value back on + */ + uinfo->value.integer.max += mc->min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_sx); + +/** * snd_soc_get_volsw - single mixer get callback * @kcontrol: mixer control * @ucontrol: control element information -- 1.7.2.5

3 3

[alsa-devel] race condition between dmaengine_pcm_dma_complete and snd_pcm_release
by Matt Campbell 16 Oct '15

16 Oct '15

Hi All, I've been trying to track down a kernel crash I've been getting when closing ALSA devices. There is a race condition between the bottom half interrupt handler for the DMA system (dmaengine_pcm_dma_complete in pcm_dmaengine.c) and the releasing of the sub-stream resources it receives as an argument (when snd_pcm_release is called). An older thread from 2013 discussed this to a good extent so I wont go into the details here. I've been unable to track down either a patch to fix this or even a good solution that I could implement. I've spent a couple days trying to think of an elegant solution and come up with nothing so far. Any input would be appreciated. Link to original thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-October/067111.ht… It's worth nothing that the original thread points out how this can arise on multi-core systems. In my case I'm on a single core system, but with the real-time patch enabled and the userspace ALSA thread running at a higher priority than the kernel's tasklet thread which can reproduce this almost 100% of the time. ~Matt

2 5

[alsa-devel] [PATCH] ALSA: timer: add config item to export PCM timer disabling for expert
by Jie Yang 16 Oct '15

16 Oct '15

PCM timer is not always used. For embedded device, we need an interface to disable it when it is not needed, to shrink the kernel size and memory footprint, here add CONFIG_SND_PCM_TIMER for it. When both CONFIG_SND_PCM_TIMER and CONFIG_SND_TIMER is unselected, about 25KB saving bonus we can get. Please be noted that when disabled, those stubs who using pcm timer (e.g. dmix, dsnoop & co) may work incorrectlly. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/pcm.h | 8 +++++++- sound/core/Kconfig | 13 ++++++++++++- sound/core/Makefile | 3 ++- sound/core/pcm_lib.c | 2 ++ sound/core/pcm_native.c | 36 ++++++++++++++++-------------------- 5 files changed, 39 insertions(+), 23 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee..79cf943 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1117,10 +1117,16 @@ static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substrea * Timer interface */ +#ifdef CONFIG_SND_PCM_TIMER void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream); void snd_pcm_timer_init(struct snd_pcm_substream *substream); void snd_pcm_timer_done(struct snd_pcm_substream *substream); - +#else +static inline void +snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} +#endif /** * snd_pcm_gettime - Fill the timespec depending on the timestamp mode * @runtime: PCM runtime instance diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6c96fee..e3e9491 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -4,7 +4,7 @@ config SND_TIMER config SND_PCM tristate - select SND_TIMER + select SND_TIMER if SND_PCM_TIMER config SND_PCM_ELD bool @@ -93,6 +93,17 @@ config SND_PCM_OSS_PLUGINS support conversion of channels, formats and rates. It will behave like most of new OSS/Free drivers in 2.4/2.6 kernels. +config SND_PCM_TIMER + bool "PCM timer interface" if EXPERT + default y + help + If you disable this option, pcm timer will be inavailable, so + those stubs used pcm timer (e.g. dmix, dsnoop & co) may work + incorrectlly. + + For some embedded device, we may disable it to reduce memory + footprint, about 20KB on x86_64 platform. + config SND_SEQUENCER_OSS bool "OSS Sequencer API" depends on SND_SEQUENCER diff --git a/sound/core/Makefile b/sound/core/Makefile index 3354f91..48ab4b8 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -13,8 +13,9 @@ snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o -snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ +snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o +snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d45645..6dc4277 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1883,8 +1883,10 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; +#ifdef CONFIG_SND_PCM_TIMER if (substream->timer_running) snd_timer_interrupt(substream->timer, 1); +#endif _end: snd_pcm_stream_unlock_irqrestore(substream, flags); if (runtime->transfer_ack_end) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 75888dd..d5672ed 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -486,6 +486,16 @@ static void snd_pcm_set_state(struct snd_pcm_substream *substream, int state) snd_pcm_stream_unlock_irq(substream); } +static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, + int event) +{ +#ifdef CONFIG_SND_PCM_TIMER + if (substream->timer) + snd_timer_notify(substream->timer, event, + &substream->runtime->trigger_tstamp); +#endif +} + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -1042,9 +1052,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } static struct action_ops snd_pcm_action_start = { @@ -1092,9 +1100,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) if (runtime->status->state != state) { snd_pcm_trigger_tstamp(substream); runtime->status->state = state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP); } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); @@ -1208,18 +1214,12 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) snd_pcm_trigger_tstamp(substream); if (push) { runtime->status->state = SNDRV_PCM_STATE_PAUSED; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MPAUSE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MPAUSE); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MCONTINUE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MCONTINUE); } } @@ -1267,9 +1267,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) snd_pcm_trigger_tstamp(substream); runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSUSPEND); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } @@ -1373,9 +1371,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->status->state = runtime->status->suspended_state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME); } static struct action_ops snd_pcm_action_resume = { -- 2.1.4

2 1

[alsa-devel] [PATCH V3 0/4] BAT: some fixes
by han.lu＠intel.com 16 Oct '15

16 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> some fixes and adjusts. Lu, Han (4): BAT: Merge message strings BAT: Use colon instead of comma for separation BAT: Adjust parameters and strings BAT: Use dynamic temp file bat/bat.c | 77 +++++++++++++++++++++++++++++++++--------------------------- bat/common.h | 2 +- 2 files changed, 43 insertions(+), 36 deletions(-) -- 1.9.1

3 6

[alsa-devel] [RFC PATCH] ASoC: Modify check condition of multiple bindings of components
by Koro Chen 16 Oct '15

16 Oct '15

The patch "ASoC: Prevent components from being bound to multiple cards" adds check to prevent multiple cards from using the same component. However, snd_soc_register_platform() or snd_soc_register_codec() will also create components, and sharing the same platform by multiple cards is then refused. This happens with a platform having multiple independent DAIs that share the same DMA controller. Relax the condition by checking component->registered_as_component, which is only true in case of snd_soc_register_component() and will be false for components created by snd_soc_register_platform() or snd_soc_register_codec(). Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 42575b0..eca169a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1106,7 +1106,8 @@ static int soc_probe_component(struct snd_soc_card *card, return 0; if (component->card) { - if (component->card != card) { + if (component->card != card && + component->registered_as_component) { dev_err(component->dev, "Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n", card->name, component->card->name); -- 1.7.9.5

3 9

[alsa-devel] [PATCH MC Next Gen 0/2] Add mixer and control interface media entities
by Shuah Khan 15 Oct '15

15 Oct '15

This patch series adds the following: Add support for creating MEDIA_ENT_F_AUDIO_MIXER entity for each mixer and a MEDIA_INTF_T_ALSA_CONTROL control interface entity that links to mixer entities. MEDIA_INTF_T_ALSA_CONTROL entity corresponds to the control device for the card. Change au0828_create_media_graph() to create pad link between MEDIA_ENT_F_AUDIO_MIXER entity and decoder's AU8522_PAD_AUDIO_OUT. This patch series is dependent on an earlier patch series: Update ALSA, and au0828 drivers to use Managed Media Controller API: https://www.mail-archive.com/linux-media@vger.kernel.org/msg92752.html Please find the media graph with mixer and control interface nodes: https://drive.google.com/folderview?id=0B0NIL0BQg-Alb3JFb2diMXRoQlU&usp=sha… New file: graph_mixer.png Shuah Khan (2): sound/usb: Create media mixer function and control interface entities media: au0828 create link between ALSA Mixer and decoder drivers/media/usb/au0828/au0828-core.c | 12 +++++ drivers/media/usb/au0828/au0828.h | 1 + sound/usb/card.c | 5 ++ sound/usb/media.c | 85 ++++++++++++++++++++++++++++++++++ sound/usb/media.h | 4 ++ sound/usb/mixer.h | 1 + sound/usb/usbaudio.h | 1 + 7 files changed, 109 insertions(+) -- 2.1.4

1 2

[alsa-devel] DSD: Playing silence before and after actual audio data
by Martin Pietryka 15 Oct '15

15 Oct '15

Hi guys, I'm trying to figure out the best way to insert ~40ms worth of silence pattern before and after playing back DSD data. There seems to be support inside ALSA for a DSD silence pattern, but as far as I have seen it's only used when an underrun might occur (silence_threshold, silence_size). Is there some possibility inside ALSA to basically insert silence before and after playback, or how hard would it be to implement something like this? I have looked a bit for myself and for me it seems that the modifications needed inside ALSA to achieve this feature are substantial, but of course I also might have missed something. Martin

2 1

[alsa-devel] Audio Mini Summit 2015, 8th October, Dublin
by Mark Brown 15 Oct '15

15 Oct '15

[This message is BCCed to everyone currently on the signup list] As previously announce the audio mini-summit for this year will be this Thursday (8th August) at CCD in Dublin. Topics we currently have on the list for discussion are: - Power saving and general system design decisions (is what we're doing lining up with modern hardware with things like multiple output streams?). - What to do with HDMI and its integration with graphics. - Test tools (Intel BAT and others). - BATCH flag. - ACPI simple-card and general generic card support. - Topology & media controller integration. - Kernel tinification. - Android <-> alsa-lib configuration translation. - HDA restructuring. - PCM core simplification - the code is currently quite complex! - Soundwire driver support plans. If you're planning to attend and haven't done so already please sign up at: https://docs.google.com/spreadsheets/d/1bi-8GHsqlzt41FR20WiMuWILe_cZowFireV… Once I have details on the room & so on I will follow up to this mail with both. We'll also try and sort out some dinner on Thursday night, if there are any special requirements for that please say (I'm assuming that we will have at least some vegetarians).

9 25

[alsa-devel] [PATCH] ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers
by Luis de Bethencourt 15 Oct '15

15 Oct '15

These platform drivers have a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/bcm/bcm2835-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 03fa1cb..8c435be 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -862,6 +862,8 @@ static const struct of_device_id bcm2835_i2s_of_match[] = { {}, }; +MODULE_DEVICE_TABLE(of, bcm2835_i2s_of_match); + static struct platform_driver bcm2835_i2s_driver = { .probe = bcm2835_i2s_probe, .driver = { -- 2.4.6

2 2

Re: [alsa-devel] Sound not working after new kernel installed
by James Cameron 15 Oct '15

15 Oct '15

On Thu, Oct 15, 2015 at 01:17:15PM +0200, Simon Wolk wrote: > The Kernel changed from 3.2.0 to 3.19.0 for me. By installing the > 3.19. the sound stopped working. Yes, Ubuntu kernels change like that, but you can easily run more recent kernel versions, see Ubuntu Wiki page: https://wiki.ubuntu.com/Kernel/MainlineBuilds While you wait for help, you can install latest v4.2.3 built 3rd October 2015. You can also test kernels between 3.2 and 3.19, if you don't get any other help. -- James Cameron http://quozl.linux.org.au/

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - consolidate chip rename functions
by Takashi Iwai 15 Oct '15

15 Oct '15

A few multiple codec drivers do renaming the chip_name string but all these are open-coded and some of them have even no error check. Let's make common helpers to do it properly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 1 + sound/hda/hdac_device.c | 22 ++++++++++++++++++++++ sound/pci/hda/hda_bind.c | 35 +++++++++++++++++++++++------------ sound/pci/hda/hda_codec.h | 2 ++ sound/pci/hda/hda_sysfs.c | 3 +-- sound/pci/hda/patch_realtek.c | 12 +----------- sound/pci/hda/patch_via.c | 18 ++++-------------- 7 files changed, 54 insertions(+), 39 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 26e956f4b7c6..49df61c7afdc 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -117,6 +117,7 @@ int snd_hdac_device_init(struct hdac_device *dev, struct hdac_bus *bus, void snd_hdac_device_exit(struct hdac_device *dev); int snd_hdac_device_register(struct hdac_device *codec); void snd_hdac_device_unregister(struct hdac_device *codec); +int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name); int snd_hdac_refresh_widgets(struct hdac_device *codec); int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b3b0ad289df1..4b06b26cee06 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -164,6 +164,28 @@ void snd_hdac_device_unregister(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_device_unregister); /** + * snd_hdac_device_set_chip_name - set/update the codec name + * @codec: the HDAC device + * @name: name string to set + * + * Returns 0 if the name is set or updated, or a negative error code. + */ +int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name) +{ + char *newname; + + if (!name) + return 0; + newname = kstrdup(name, GFP_KERNEL); + if (!newname) + return -ENOMEM; + kfree(codec->chip_name); + codec->chip_name = newname; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name); + +/** * snd_hdac_make_cmd - compose a 32bit command word to be sent to the * HD-audio controller * @codec: the codec object diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index d5ac25cc7fee..ef6b8f836a87 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -45,15 +45,31 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) codec->patch_ops.unsol_event(codec, ev); } -/* reset the codec name from the preset */ -static int codec_refresh_name(struct hda_codec *codec, const char *name) +/** + * snd_hda_codec_set_name - set the codec name + * @codec: the HDA codec + * @name: name string to set + */ +int snd_hda_codec_set_name(struct hda_codec *codec, const char *name) { - if (name) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup(name, GFP_KERNEL); + int err; + + if (!name) + return 0; + err = snd_hdac_device_set_chip_name(&codec->core, name); + if (err < 0) + return err; + + /* update the mixer name */ + if (!*codec->card->mixername) { + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", + codec->core.vendor_name, codec->core.chip_name); } - return codec->core.chip_name ? 0 : -ENOMEM; + + return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_set_name); static int hda_codec_driver_probe(struct device *dev) { @@ -64,7 +80,7 @@ static int hda_codec_driver_probe(struct device *dev) if (WARN_ON(!codec->preset)) return -EINVAL; - err = codec_refresh_name(codec, codec->preset->name); + err = snd_hda_codec_set_name(codec, codec->preset->name); if (err < 0) goto error; err = snd_hdac_regmap_init(&codec->core); @@ -251,11 +267,6 @@ int snd_hda_codec_configure(struct hda_codec *codec) } } - /* audio codec should override the mixer name */ - if (codec->core.afg || !*codec->card->mixername) - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), "%s %s", - codec->core.vendor_name, codec->core.chip_name); return 0; error: diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 95991e463abb..b6d937784afa 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -463,6 +463,8 @@ void snd_hda_unlock_devices(struct hda_bus *bus); void snd_hda_bus_reset(struct hda_bus *bus); void snd_hda_bus_reset_codecs(struct hda_bus *bus); +int snd_hda_codec_set_name(struct hda_codec *codec, const char *name); + /* * power management */ diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index a6e3d9b511ab..64e0d1d81ca5 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -595,8 +595,7 @@ static void parse_model_mode(char *buf, struct hda_bus *bus, static void parse_chip_name_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - kfree((*codecp)->core.chip_name); - (*codecp)->core.chip_name = kstrdup(buf, GFP_KERNEL); + snd_hda_codec_set_name(*codecp, buf); } #define DEFINE_PARSE_ID_MODE(name) \ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 16b8dcba5c12..e1ffb0202ebc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -822,17 +822,7 @@ static const struct hda_codec_ops alc_patch_ops = { }; -/* replace the codec chip_name with the given string */ -static int alc_codec_rename(struct hda_codec *codec, const char *name) -{ - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup(name, GFP_KERNEL); - if (!codec->core.chip_name) { - alc_free(codec); - return -ENOMEM; - } - return 0; -} +#define alc_codec_rename(codec, name) snd_hda_codec_set_name(codec, name) /* * Rename codecs appropriately from COEF value or subvendor id diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index da5366405eda..d714a57e9460 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -785,21 +785,11 @@ static int patch_vt1708S(struct hda_codec *codec) override_mic_boost(codec, 0x1e, 0, 3, 40); /* correct names for VT1708BCE */ - if (get_codec_type(codec) == VT1708BCE) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup("VT1708BCE", GFP_KERNEL); - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), - "%s %s", codec->core.vendor_name, codec->core.chip_name); - } + if (get_codec_type(codec) == VT1708BCE) + snd_hda_codec_set_name(codec, "VT1708BCE"); /* correct names for VT1705 */ - if (codec->core.vendor_id == 0x11064397) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup("VT1705", GFP_KERNEL); - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), - "%s %s", codec->core.vendor_name, codec->core.chip_name); - } + if (codec->core.vendor_id == 0x11064397) + snd_hda_codec_set_name(codec, "VT1705"); /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); -- 2.6.1

1 1

[alsa-devel] [PATCH v6 0/10] Fix broken DMAFLUSHP on Rockchips platform
by Shawn Lin 15 Oct '15

15 Oct '15

The purpose of the DMAFLUSHP instruction: - Tell the peripheral to clear its status and control registers. - Send a message to the peripheral to resend its level status. There are 3 timings described in PL330 Technical Reference Manual: - Timing 1: Burst request, can work well without DMAFLUSHP. - Timing 2: Single and burst request, DMAC will ignore the single transfer request. This timing happens if there are single and burst request. - Timing 3: Single transfers for a burst request, DMAC should signals datype to request the peripheral to flush the contents of any control registers. This timing happens if there is not enough MFIFO to places the burst data. A peripheral may signal a DMA request during the execution of DMAFLUSHP instruction, that cause DMA request being ignored by DMAC. But DMAC and all peripherals on RK3X SoCs DO NOT support DMAFLUSHP. It can't send a message to the peripheral to resend DMA request, and the peripheral can't acknowledge a flush request from DMAC. So all DMA requests should NOT be ignored by DMAC, and DMAC will not notify the peripheral to flush. To fix this problem, we need: - Do NOT execute DMAFLUSHP instruction. - Timing 2 and timing 3 should not happen. Because on RK3X SoCs, there are 6 or below channels and 32 MFIFO depth for DMAC_BUS, and 8 channels and 64 MFIFO depth for DMAC_PERI, it is impossible to hit the timing 3 if burst length is equal or less than 4. Since the request type signal by the peripheral can only be set by software. We can set Rockchip Soc's GRF_PERIDMAC_CON0[2:1] to select single or burst request, if it is set b01, all of the peripharals will signal a brust request. So the timing 2 will not happen, too. So DMAC on RK3X can support single or burst transfer, but can't support mixed transfer. Because burst transfer is more efficient than single transfer, this is confirmed by our ASIC team, who strongly suggest to use burst transfer. And this is confirmed by Addy's test on RK3288-Pink2 board, the speed of spi flash burst transfer will increase about two times than single transfer. Also, I have tested dw_mmc with pl330 on RK3188 platform to double confirm the result. That means burst transfer is reansonable. So we need a quirk not to execute DMAFLUSHP instruction and to use burst transfer. Note: - The Rockchip Soc default value of GRF_PERIDMAC_CON0[2:1] is b01. To support brust transfer, these bits should not be changed in bootloader. Changes in v6: - remove expose quirk and add dma max_burst caps for clients - remove expose quirks and add max_burst for dmaengine - remove get quirks and add get slave caps - remove quirks and get dma caps in order to limit burst Changes in v5: - add Mark's tag for spi changes - remove unnecessary whitespace change - use switch statement for i2s quirk - use switch statement for dma_quirk's manipulation Changes in v4: - remove spi & i2s dts changes and query quirk from dmaengine API suggeseted by Mark. - fix typo - Add dmaengine_get_quirk hook and implement it for pl330 Changes in v3: - add Sunny's tag - add more rockchip drivers' changes in this patchset - add Reviewed-by: Sonny Rao <sonnyrao(a)chromium.org> Changes in v2: - amend the author - reorder the patches suggested by Doug - add Reviewed-by: Doug Anderson <dianders(a)chromium.org> for rk3288.dtsi patch and arm-pl330.txt patch - fix Olof's mail address Changes in v1: - add From original author. - remove Sunny's tag - rename broken-no-flushp to "arm,pl330-broken-no-flushp" suggested by Krzysztof. Addy Ke (3): DMA: pl330: add quirk for broken no flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3288 platform spi: rockchip: modify DMA max burst to 1 Boojin Kim (1): DMA: pl330: support burst mode for dev-to-mem and mem-to-dev transmit Shawn Lin (5): Documentation: arm-pl330: add description of arm,pl330-broken-no-flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3xxx platform DMA: dmaengine: expose max burst capability to clients DMA: pl330: add max burst for dmaengine snd: dmaengine-pcm: add snd_dmaengine_pcm_get_caps interface Yiwei Cai (1): ASoC: rockchip_i2s: modify DMA max burst to 1 .../devicetree/bindings/dma/arm-pl330.txt | 1 + arch/arm/boot/dts/rk3288.dtsi | 3 + arch/arm/boot/dts/rk3xxx.dtsi | 3 + drivers/dma/dmaengine.c | 1 + drivers/dma/pl330.c | 106 +++++++++++++++------ drivers/spi/spi-rockchip.c | 12 ++- include/linux/dmaengine.h | 4 + sound/soc/rockchip/rockchip_i2s.c | 18 +++- sound/soc/soc-generic-dmaengine-pcm.c | 24 +++++ 9 files changed, 138 insertions(+), 34 deletions(-) -- 2.3.7

4 16

[alsa-devel] [PATCH] ALSA: hda - Enable widget power saving for Cirrus codecs
by Takashi Iwai 15 Oct '15

15 Oct '15

Cirrus codecs have also fine power controls on each widget, thus it gets benefit from the recent widget power-saving feature. As we haven't seen any obvious regressions with tests on some MacBooks, let's try to enable it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 85813de26da8..01c464a468f6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -570,6 +570,7 @@ static struct cs_spec *cs_alloc_spec(struct hda_codec *codec, int vendor_nid) return NULL; codec->spec = spec; spec->vendor_nid = vendor_nid; + codec->power_save_node = 1; snd_hda_gen_spec_init(&spec->gen); return spec; -- 2.6.1

1 0

[alsa-devel] Sound not working after new kernel installed
by Simon Wolk 15 Oct '15

15 Oct '15

Hello everybody, I don't know if my question is right here, but either way I am asking: I am using an old Benq notebook (Joybook S73) mit Xubuntu 15.04. When I installed a new kernel (then 3.19.0-16, but now I am using 3.19.0.28) the sound on the speakers stoped working and there were no more audible sounds. With headphones plugged in and a the highest possible volume I could hear very silent the music played on the PC. Up to Kernel 3.2.0-70 the sound worked fine. System update and upgrade unfortunately did not help. Do you have any idea what may be the problem? I did the following to show the chip used: simon@simon-Joybook-S73:~$ lspci -nnk | grep -i audio -A2 00:1b.0 Audio device [0403]: Intel Corporation NM10/ICH7 Family High Definition Audio Controller [8086:27d8] (rev 02) Subsystem: Benq Corporation Device [17ff:0600] Kernel driver in use: snd_hda_intel simon@simon-Joybook-S73:~$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x6c400000 irq 29 Thank you for your help! Simon

3 2

Re: [alsa-devel] ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages
by Dan Carpenter 15 Oct '15

15 Oct '15

Hello Takashi Sakamoto, This is a semi-automatic email about new static checker warnings. The patch e8a40d9bcb23: "ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages" from Oct 9, 2015, leads to the following Smatch complaint: sound/firewire/lib.c:88 async_midi_port_callback() error: we previously assumed 'substream' could be null (see line 77) sound/firewire/lib.c 76 77 if (rcode == RCODE_COMPLETE && substream != NULL) ^^^^^^^^^^^^^^^^^ Existing code assumes substream can be NULL. 78 snd_rawmidi_transmit_ack(substream, port->consume_bytes); 79 else if (!rcode_is_permanent_error(rcode)) 80 /* To start next transaction immediately for recovery. */ 81 port->next_ktime = ktime_set(0, 0); 82 else 83 /* Don't continue processing. */ 84 port->error = true; 85 86 port->idling = true; 87 88 if (!snd_rawmidi_transmit_empty(substream)) ^^^^^^^^^ Patch adds an unchecked dereference (inside the function call). 89 schedule_work(&port->work); 90 } regards, dan carpenter

2 1

[alsa-devel] [PATCH v4 28/79] hdspm.h: use __u8, __u32 and __u64 from linux/types.h instead of stdint.h
by Mikko Rapeli 15 Oct '15

15 Oct '15

Kernel headers should use linux/types.h based definitions. Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/hdspm.h | 40 ++++++++++++++++++---------------------- 1 file changed, 18 insertions(+), 22 deletions(-) diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index 5737332..c4db6f5 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,11 +20,7 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ -#ifdef __KERNEL__ #include <linux/types.h> -#else -#include <stdint.h> -#endif /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 @@ -46,15 +42,15 @@ enum hdspm_speed { /* -------------------- IOCTL Peak/RMS Meters -------------------- */ struct hdspm_peak_rms { - uint32_t input_peaks[64]; - uint32_t playback_peaks[64]; - uint32_t output_peaks[64]; + __u32 input_peaks[64]; + __u32 playback_peaks[64]; + __u32 output_peaks[64]; - uint64_t input_rms[64]; - uint64_t playback_rms[64]; - uint64_t output_rms[64]; + __u64 input_rms[64]; + __u64 playback_rms[64]; + __u64 output_rms[64]; - uint8_t speed; /* enum {ss, ds, qs} */ + __u8 speed; /* enum {ss, ds, qs} */ int status2; }; @@ -155,21 +151,21 @@ enum hdspm_syncsource { }; struct hdspm_status { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ enum hdspm_syncsource autosync_source; - uint64_t card_clock; - uint32_t master_period; + __u64 card_clock; + __u32 master_period; union { struct { - uint8_t sync_wc; /* enum hdspm_sync */ - uint8_t sync_madi; /* enum hdspm_sync */ - uint8_t sync_tco; /* enum hdspm_sync */ - uint8_t sync_in; /* enum hdspm_sync */ - uint8_t madi_input; /* enum hdspm_madi_input */ - uint8_t channel_format; /* enum hdspm_madi_channel_format */ - uint8_t frame_format; /* enum hdspm_madi_frame_format */ + __u8 sync_wc; /* enum hdspm_sync */ + __u8 sync_madi; /* enum hdspm_sync */ + __u8 sync_tco; /* enum hdspm_sync */ + __u8 sync_in; /* enum hdspm_sync */ + __u8 madi_input; /* enum hdspm_madi_input */ + __u8 channel_format; /* enum hdspm_madi_channel_format */ + __u8 frame_format; /* enum hdspm_madi_frame_format */ } madi; } card_specific; }; @@ -184,7 +180,7 @@ struct hdspm_status { #define HDSPM_ADDON_TCO 1 struct hdspm_version { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ char cardname[20]; unsigned int serial; unsigned short firmware_rev; -- 2.5.0

2 1

[alsa-devel] [PATCH v4 17/79] include/uapi/sound/emu10k1.h: added EMU10K1 version of DECLARE_BITMAP macro
by Mikko Rapeli 15 Oct '15

15 Oct '15

Fixes userspace compilation error: error: expected specifier-qualifier-list before ‘DECLARE_BITMAP’ DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ DECLARE_BITMAP macro is not meant for userspace headers and thus added here as private copy for emu10k.h. Fix was suggested by Arnd Bergmann <arnd(a)arndb.de> in message <2168807.4Yxh5gl11Q@wuerfel> and Takashi Iwai <tiwai(a)suse.de> in message <s5h1thx88tk.wl-tiwai(a)suse.de> on lkml. Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/emu10k1.h | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index ec1535b..5175e16 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -34,6 +34,14 @@ #define EMU10K1_FX8010_PCM_COUNT 8 +/* + * Following definition is copied from linux/types.h to support compiling + * this header file in userspace since they are not generally available for + * uapi headers. + */ +#define __EMU10K1_DECLARE_BITMAP(name,bits) \ + unsigned long name[(bits) / (sizeof(unsigned long) * 8)] + /* instruction set */ #define iMAC0 0x00 /* R = A + (X * Y >> 31) ; saturation */ #define iMAC1 0x01 /* R = A + (-X * Y >> 31) ; saturation */ @@ -300,7 +308,7 @@ struct snd_emu10k1_fx8010_control_old_gpr { struct snd_emu10k1_fx8010_code { char name[128]; - DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ __u32 __user *gpr_map; /* initializers */ unsigned int gpr_add_control_count; /* count of GPR controls to add/replace */ @@ -313,11 +321,11 @@ struct snd_emu10k1_fx8010_code { unsigned int gpr_list_control_total; /* total count of GPR controls */ struct snd_emu10k1_fx8010_control_gpr __user *gpr_list_controls; /* listed GPR controls */ - DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ __u32 __user *tram_data_map; /* data initializers */ __u32 __user *tram_addr_map; /* map initializers */ - DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ + __EMU10K1_DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ __u32 __user *code; /* one instruction - 64 bits */ }; -- 2.5.0

2 1

[alsa-devel] [patch] ALSA: oss: underflow in snd_mixer_oss_proc_write()
by Dan Carpenter 15 Oct '15

15 Oct '15

We cap the upper bound of "idx" but not the negative side. Let's make it unsigned to fix this. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index a99f720..7a8c79d 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1177,7 +1177,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, struct snd_mixer_oss *mixer = entry->private_data; char line[128], str[32], idxstr[16]; const char *cptr; - int ch, idx; + unsigned int idx; + int ch; struct snd_mixer_oss_assign_table *tbl; struct slot *slot;

2 1

[alsa-devel] [PATCH v5 0/10] Fix broken DMAFLUSHP on Rockchips platform
by Shawn Lin 14 Oct '15

14 Oct '15

The purpose of the DMAFLUSHP instruction: - Tell the peripheral to clear its status and control registers. - Send a message to the peripheral to resend its level status. There are 3 timings described in PL330 Technical Reference Manual: - Timing 1: Burst request, can work well without DMAFLUSHP. - Timing 2: Single and burst request, DMAC will ignore the single transfer request. This timing happens if there are single and burst request. - Timing 3: Single transfers for a burst request, DMAC should signals datype to request the peripheral to flush the contents of any control registers. This timing happens if there is not enough MFIFO to places the burst data. A peripheral may signal a DMA request during the execution of DMAFLUSHP instruction, that cause DMA request being ignored by DMAC. But DMAC and all peripherals on RK3X SoCs DO NOT support DMAFLUSHP. It can't send a message to the peripheral to resend DMA request, and the peripheral can't acknowledge a flush request from DMAC. So all DMA requests should NOT be ignored by DMAC, and DMAC will not notify the peripheral to flush. To fix this problem, we need: - Do NOT execute DMAFLUSHP instruction. - Timing 2 and timing 3 should not happen. Because on RK3X SoCs, there are 6 or below channels and 32 MFIFO depth for DMAC_BUS, and 8 channels and 64 MFIFO depth for DMAC_PERI, it is impossible to hit the timing 3 if burst length is equal or less than 4. Since the request type signal by the peripheral can only be set by software. We can set Rockchip Soc's GRF_PERIDMAC_CON0[2:1] to select single or burst request, if it is set b01, all of the peripharals will signal a brust request. So the timing 2 will not happen, too. So DMAC on RK3X can support single or burst transfer, but can't support mixed transfer. Because burst transfer is more efficient than single transfer, this is confirmed by our ASIC team, who strongly suggest to use burst transfer. And this is confirmed by Addy's test on RK3288-Pink2 board, the speed of spi flash burst transfer will increase about two times than single transfer. Also, I have tested dw_mmc with pl330 on RK3188 platform to double confirm the result. That means burst transfer is reansonable. So we need a quirk not to execute DMAFLUSHP instruction and to use burst transfer. Note: - The Rockchip Soc default value of GRF_PERIDMAC_CON0[2:1] is b01. To support brust transfer, these bits should not be changed in bootloader. Changes in v5: - add Mark's tag for spi changes - remove unnecessary whitespace change - use switch statement for I2S dma_quirk's manipulation Changes in v4: - remove spi & i2s dts changes and query quirk from dmaengine API suggeseted by Mark. - fix typo - Add dmaengine_get_quirk hook and implement it for pl330 Changes in v3: - add Sunny's tag - add more rockchip drivers' changes in this patchset - add Reviewed-by: Sonny Rao <sonnyrao(a)chromium.org> Changes in v2: - amend the author - reorder the patches suggested by Doug - add Reviewed-by: Doug Anderson <dianders(a)chromium.org> for rk3288.dtsi patch and arm-pl330.txt patch - amend Olof's mail address Changes in v1: - rename broken-no-flushp to "arm,pl330-broken-no-flushp" suggested by Krzysztof. - add From original author. - remove Sunny's tag Addy Ke (3): DMA: pl330: add quirk for broken no flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3288 platform spi: rockchip: modify DMA max burst to 1 Boojin Kim (1): DMA: pl330: support burst mode for dev-to-mem and mem-to-dev transmit Shawn Lin (5): Documentation: arm-pl330: add description of arm,pl330-broken-no-flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3xxx platform dmaengine: add API for getting dma controller's quirk DMA: pl330: implement dmaengine_get_quirks hook snd: dmaengine-pcm: add snd_dmaengine_pcm_get_quirks interface Yiwei Cai (1): ASoC: rockchip_i2s: modify DMA max burst to 1 .../devicetree/bindings/dma/arm-pl330.txt | 1 + arch/arm/boot/dts/rk3288.dtsi | 3 + arch/arm/boot/dts/rk3xxx.dtsi | 3 + drivers/dma/pl330.c | 110 +++++++++++++++------ drivers/spi/spi-rockchip.c | 13 ++- include/linux/dmaengine.h | 9 ++ sound/soc/rockchip/rockchip_i2s.c | 21 ++++ sound/soc/soc-generic-dmaengine-pcm.c | 24 +++++ 8 files changed, 153 insertions(+), 31 deletions(-) -- 2.3.7

4 21

[alsa-devel] snd_pcm_hw_params_set_buffer_size_near yields EINVAL for the value 1681
by Andrew Kelley 14 Oct '15

14 Oct '15

When I call snd_pcm_hw_params_set_buffer_size_near with the value 1681, I get an EINVAL error. These other values all work fine: 1 1679 1680 1682 1683 1684 99999 What in the world is going on here?

2 1

[alsa-devel] [PATCH v4 0/8] Add SPDIF support for rockchip
by Sjoerd Simons 13 Oct '15

13 Oct '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board and Radxa Rock 2 Square board. Quite a few changes since v3 (mostly based on review comment). One important addition is proper RK3288 SoC support; While according to the documentation it has 2 SPDIF "solutions" with the default being the old one, this variation doesn't actually work.. So support has been added to switch to the new SPDIF solution on RK3288. Changes in v4: - Require rockchip,grf on RK3288 as the 8 channel solution has to be selected on that SoC - Make the compatibility string one of a known list rather then requiring a precise list of options. - Change the clock names to hclk and mclk instead of spdif_hclk and spdif_mclk to better match the implementation and data sheets. - Use hclk and mclk as the clock names matching the update devicetree documentation - Drop something useless to_info function - Open-code tx control in rk_spdif_trigge - Use devm_regmap_init_mmio_clk to allow the regmap subsystem to manage the hclk as needed - Use a more conventionale pm setup (turn everything on, mark the device as active, enable pm and request the device to go idle) - re-add the platfrorm:rockchip-spdif module alias - Also manage the hclk in the PM callbacks - Check the return code of clk_set_rate - Switch to the 8 channel spdif solutions on RK3288 - Add support for 192khz audio rate - Adjust clock names to match the new names (hclk/mclk) Changes in v3: - Fix typos in commit message - Resent patches to a more complete list of maintainers. Changes in v2: - Remove platform: module alias as it was unused - Call MODULE_DEVICE_TABLE(of, ) right after the of match table - use rk_spdif as a prefix consistenly throughout the driver - Check return code of regmap_update and bubble it up - Sort the spdif node properties - Drop the 0x prefix from the node name - Rename the spdif@ node to sound@ Sjoerd Simons (8): ASoC: dt-bindings: add rockchip tranceiver bindings ASoC: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock clk: rockchip: Allow the RK3288 SPDIF clocks to change their parent ARM: dts: rockchip: Add SPDIF transceiver for RK3288 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock2 Square ARM: multi_v7_defconfig: Add rockchip audio support .../devicetree/bindings/sound/rockchip-spdif.txt | 44 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 + arch/arm/boot/dts/rk3188.dtsi | 22 ++ arch/arm/boot/dts/rk3288-rock2-square.dts | 18 + arch/arm/boot/dts/rk3288.dtsi | 23 ++ arch/arm/configs/multi_v7_defconfig | 5 + drivers/clk/rockchip/clk-rk3288.c | 16 +- sound/soc/rockchip/Kconfig | 8 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_spdif.c | 409 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 11 files changed, 621 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.6.1

7 17

[alsa-devel] [PATCH v3 0/2] make it possible to not compile compress APIs
by Jie Yang 13 Oct '15

13 Oct '15

Compress APIs are not always needed, but they are always built currentlly, in both ALSA core and ASoC core. This series make it possible to not compile it when unneeded. Hi, Takashi & all, I am not sure you will like the patch 2/2, it split soc-compress to a module, but introduced several extra 'EXPORT_SYMBOL_GPL' there. If don't like it, you can just apply the first one(1/2) and skip 2/2. thanks, ~Keyon Changes in this version: 1. change to use a callback for compress_new, and remove compress_dai bool. 2. remove the empty module_init/exit for soc-compress module. Jie Yang (2): ASoC: soc-compress: add a config item for soc-compress ASoC: soc-compress: split soc-compress to a module include/sound/soc-dai.h | 2 +- include/sound/soc.h | 2 ++ sound/soc/Kconfig | 6 +++++- sound/soc/Makefile | 5 ++++- sound/soc/intel/Kconfig | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/soc-compress.c | 10 ++++++++++ sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 1 + sound/soc/soc-pcm.c | 13 +++++++++++++ 10 files changed, 40 insertions(+), 6 deletions(-) -- 1.9.1

5 12

[alsa-devel] [PATCH v2] drm/i915: Improve kernel-doc for i915_audio_component struct
by David Henningsson 13 Oct '15

13 Oct '15

To make kernel-doc happy, the i915_audio_component_audio_ops struct cannot be nested. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/DocBook/drm.tmpl | 1 + include/drm/i915_component.h | 92 ++++++++++++++++++++++++++++++------------ 2 files changed, 67 insertions(+), 26 deletions(-) diff --git a/Documentation/DocBook/drm.tmpl b/Documentation/DocBook/drm.tmpl index 9ddf8c6..f16e4e8 100644 --- a/Documentation/DocBook/drm.tmpl +++ b/Documentation/DocBook/drm.tmpl @@ -4051,6 +4051,7 @@ int num_ioctls;</synopsis> <title>High Definition Audio</title> !Pdrivers/gpu/drm/i915/intel_audio.c High Definition Audio over HDMI and Display Port !Idrivers/gpu/drm/i915/intel_audio.c +!Iinclude/drm/i915_component.h </sect2> <sect2> <title>Panel Self Refresh PSR (PSR/SRD)</title> diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index 89dc7d6..76c10c8 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -30,38 +30,78 @@ */ #define MAX_PORTS 5 +/** + * struct i915_audio_component_ops - Ops implemented by i915 driver, called by hda driver + */ +struct i915_audio_component_ops { + /** + * @owner: i915 module + */ + struct module *owner; + /** + * @get_power: Request that power well is to be turned on + */ + void (*get_power)(struct device *); + /** + * @put_power: Allow the power well to be turned off + */ + void (*put_power)(struct device *); + /** + * @codec_wake_override: Force the audio codec to stay awake + */ + void (*codec_wake_override)(struct device *, bool enable); + /** + * @get_cdclk_freq: Query the i915 driver about the current cdclk frequency + */ + int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ + int (*sync_audio_rate)(struct device *, int port, int rate); +}; + +/** + * struct i915_audio_component_audio_ops - Ops implemented by hda driver, called by i915 driver + */ +struct i915_audio_component_audio_ops { + /** + * @audio_ptr: Pointer to be used in call to pin_eld_notify + */ + void *audio_ptr; + /** + * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed + * + * Called when the i915 driver has set up audio pipeline or has just + * begun to tear it down. This allows the HDA driver to update its + * status accordingly (even when the HDA controller is in power save + * mode). + */ + void (*pin_eld_notify)(void *audio_ptr, int port); +}; + +/** + * struct i915_audio_component - Used for direct communication between i915 and hda drivers + */ struct i915_audio_component { + /** + * @dev: i915 device, used as parameter for ops + */ struct device *dev; /** * @aud_sample_rate: the array of audio sample rate per port */ int aud_sample_rate[MAX_PORTS]; - - const struct i915_audio_component_ops { - struct module *owner; - void (*get_power)(struct device *); - void (*put_power)(struct device *); - void (*codec_wake_override)(struct device *, bool enable); - int (*get_cdclk_freq)(struct device *); - /** - * @sync_audio_rate: set n/cts based on the sample rate - * - * Called from audio driver. After audio driver sets the - * sample rate, it will call this function to set n/cts - */ - int (*sync_audio_rate)(struct device *, int port, int rate); - } *ops; - - const struct i915_audio_component_audio_ops { - void *audio_ptr; - /** - * Call from i915 driver, notifying the HDA driver that - * pin sense and/or ELD information has changed. - * @audio_ptr: HDA driver object - * @port: Which port has changed (PORTA / PORTB / PORTC etc) - */ - void (*pin_eld_notify)(void *audio_ptr, int port); - } *audio_ops; + /** + * @ops: Ops implemented by i915 driver, called by hda driver + */ + const struct i915_audio_component_ops *ops; + /** + * @audio_ops: Ops implemented by hda driver, called by i915 driver + */ + const struct i915_audio_component_audio_ops *audio_ops; }; #endif /* _I915_COMPONENT_H_ */ -- 1.9.1

3 4

[alsa-devel] Feasibility of adding alternative audio transport besides I2S/PWM/SPDIF, etc
by Caleb Crome 13 Oct '15

13 Oct '15

Hi All, I'm in a constant struggle to bring up many channel audio on each separate SoC. I can easily put a microcontroller in place that will collect and distrubute all the TDM channels to the codecs, and connect the hardware via an SPI interface to the SoC. So, instead of: CODECS <---TDM---> SoC It would be CODECS <---TDM---> uC <---SPI---> SoC So, my questions are: * I suspect the SPI interface could be used more universally than each individual I2S/TDM interface (like FSL SSI vs. Ti McBSP vs. Ti McASP, etc). and the SPI port would provide a very common API regardless of SoC. Is that true? * Is it feasible to integrate an SPI transport into the SoC/alsa layer? What are the difficulties involved with that? Cheers, -Caleb

4 7

[alsa-devel] [PATCH v2] drm/i915: set proper N/CTS in modeset
by libin.yang＠intel.com 13 Oct '15

13 Oct '15

From: Libin Yang <libin.yang(a)intel.com> When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- drivers/gpu/drm/i915/intel_audio.c | 57 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 10 +++++++ 2 files changed, 58 insertions(+), 9 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index e3c32ce..642dfbd 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -128,6 +128,20 @@ static int audio_config_get_n(const struct drm_display_mode *mode, int rate) return 0; } +static uint32_t audio_config_setup_n_reg(int n, uint32_t val) +{ + int n_low, n_up; + uint32_t tmp = val; + + n_low = n & 0xfff; + n_up = (n >> 12) & 0xff; + tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); + tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | + (n_low << AUD_CONFIG_LOWER_N_SHIFT) | + AUD_CONFIG_N_PROG_ENABLE); + return tmp; +} + /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, struct drm_display_mode *mode) @@ -262,9 +276,14 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, struct drm_i915_private *dev_priv = connector->dev->dev_private; struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc); enum pipe pipe = intel_crtc->pipe; + struct i915_audio_component *acomp = dev_priv->audio_component; const uint8_t *eld = connector->eld; + struct intel_digital_port *intel_dig_port = + enc_to_dig_port(&encoder->base); + enum port port = intel_dig_port->port; uint32_t tmp; int len, i; + int n, rate; DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n", pipe_name(pipe), drm_eld_size(eld)); @@ -302,12 +321,29 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, /* Enable timestamps */ tmp = I915_READ(HSW_AUD_CFG(pipe)); tmp &= ~AUD_CONFIG_N_VALUE_INDEX; - tmp &= ~AUD_CONFIG_N_PROG_ENABLE; tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) tmp |= AUD_CONFIG_N_VALUE_INDEX; else tmp |= audio_config_hdmi_pixel_clock(mode); + + tmp &= ~AUD_CONFIG_N_PROG_ENABLE; + if (audio_rate_need_prog(intel_crtc, mode)) { + if (!acomp) + rate = 0; + else if (port >= PORT_A && port <= PORT_E) + rate = acomp->aud_sample_rate[port]; + else { + DRM_ERROR("invalid port: %d\n", port); + rate = 0; + } + n = audio_config_get_n(mode, rate); + if (n != 0) + tmp = audio_config_setup_n_reg(n, tmp); + else + DRM_DEBUG_KMS("no suitable N value is found\n"); + } + I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -594,9 +630,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, struct intel_digital_port *intel_dig_port; struct intel_crtc *crtc; struct drm_display_mode *mode; + struct i915_audio_component *acomp = dev_priv->audio_component; enum pipe pipe = -1; u32 tmp; - int n_low, n_up, n; + int n; /* HSW, BDW SKL need this fix */ if (!IS_SKYLAKE(dev_priv) && @@ -630,6 +667,9 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; + /* port must be valid now, otherwise the pipe will be invalid */ + acomp->aud_sample_rate[port] = rate; + /* 2. check whether to set the N/CTS/M manually or not */ if (!audio_rate_need_prog(crtc, mode)) { tmp = I915_READ(HSW_AUD_CFG(pipe)); @@ -649,15 +689,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, mutex_unlock(&dev_priv->av_mutex); return 0; } - n_low = n & 0xfff; - n_up = (n >> 12) & 0xff; - /* 4. set the N/CTS/M */ + /* 3. set the N/CTS/M */ tmp = I915_READ(HSW_AUD_CFG(pipe)); - tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); - tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | - (n_low << AUD_CONFIG_LOWER_N_SHIFT) | - AUD_CONFIG_N_PROG_ENABLE); + tmp = audio_config_setup_n_reg(n, tmp); I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -678,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev, { struct i915_audio_component *acomp = data; struct drm_i915_private *dev_priv = dev_to_i915(i915_dev); + int i; if (WARN_ON(acomp->ops || acomp->dev)) return -EEXIST; @@ -685,6 +721,9 @@ static int i915_audio_component_bind(struct device *i915_dev, drm_modeset_lock_all(dev_priv->dev); acomp->ops = &i915_audio_component_ops; acomp->dev = i915_dev; + BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS); + for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++) + acomp->aud_sample_rate[i] = 0; dev_priv->audio_component = acomp; drm_modeset_unlock_all(dev_priv->dev); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7..89dc7d6 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,18 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- 1.9.1

7 7

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake - Add Skylake RT286 I2S machine driver
by Vinod Koul 13 Oct '15

13 Oct '15

From: Omair M Abdullah <omair.m.abdullah(a)intel.com> Add the SKL I2S machine driver using Realtek ALC286S codec in I2S mode. Signed-off-by: Omair M Abdullah <omair.m.abdullah(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v2: added depends on X86 and ACPI based on LKP reports sound/soc/intel/Kconfig | 13 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_rt286.c | 268 +++++++++++++++++++++++++++++++++++++ 3 files changed, 283 insertions(+) create mode 100644 sound/soc/intel/boards/skl_rt286.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 664df1f44530..6d67357f2d74 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -140,3 +140,16 @@ config SND_SOC_INTEL_SKYLAKE select SND_HDA_EXT_CORE select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + +config SND_SOC_INTEL_SKL_RT286_MACH + tristate "ASoC Audio driver for SKL with RT286 I2S mode" + depends on X86 && ACPI + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT286 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Skylake platforms + with RT286 I2S audio codec. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index cb94895c9edb..371c4565cad8 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-skl_rt286-objs := skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -15,3 +16,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c new file mode 100644 index 000000000000..71907b2c1611 --- /dev/null +++ b/sound/soc/intel/boards/skl_rt286.c @@ -0,0 +1,268 @@ +/* + * Intel Skylake I2S Machine Driver + * + * Copyright (C) 2014-2015, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * 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. + * + * This program is distributed in the hope that 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. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/rt286.h" + +static struct snd_soc_jack skylake_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin skylake_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new skylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget skylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route skylake_rt286_map[] = { + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC2"}, + {"DMIC AIF", NULL, "SoC DMIC"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + { "ssp0 Tx", NULL, "codec1_out"}, + + { "codec0_in", NULL, "ssp0 Rx" }, + { "codec1_in", NULL, "ssp0 Rx" }, + { "ssp0 Rx", NULL, "AIF1 Capture" }, + + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + + { "hif1", NULL, "iDisp Tx"}, + { "iDisp Tx", NULL, "iDisp_out"}, + +}; + +static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &skylake_headset, + skylake_headset_pins, ARRAY_SIZE(skylake_headset_pins)); + + if (ret) + return ret; + + rt286_mic_detect(codec, &skylake_headset); + + return 0; +} + + +static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The output is 48KHz, stereo, 16bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int skylake_rt286_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 *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops skylake_rt286_ops = { + .hw_params = skylake_rt286_hw_params, +}; + +/* skylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link skylake_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "Skl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + }, + { + .name = "Skl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_capture = 1, + }, + { + .name = "Skl Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .ignore_suspend = 1, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .be_id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .init = skylake_rt286_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp0_fixup, + .ops = &skylake_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .be_id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +/* skylake audio machine driver for SPT + RT286S */ +static struct snd_soc_card skylake_rt286 = { + .name = "skylake-rt286", + .owner = THIS_MODULE, + .dai_link = skylake_rt286_dais, + .num_links = ARRAY_SIZE(skylake_rt286_dais), + .controls = skylake_controls, + .num_controls = ARRAY_SIZE(skylake_controls), + .dapm_widgets = skylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), + .dapm_routes = skylake_rt286_map, + .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), + .fully_routed = true, +}; + +static int skylake_audio_probe(struct platform_device *pdev) +{ + skylake_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&skylake_rt286); +} + +static int skylake_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&skylake_rt286); + return 0; +} + +static struct platform_driver skylake_audio = { + .probe = skylake_audio_probe, + .remove = skylake_audio_remove, + .driver = { + .name = "skl_alc286s_i2s", + }, +}; + +module_platform_driver(skylake_audio) + +/* Module information */ +MODULE_AUTHOR("Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com>"); +MODULE_DESCRIPTION("Intel SST Audio for Skylake"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_alc286s_i2s"); -- 2.4.3

2 1

[alsa-devel] [patch] ALSA: pcm: harmless underflow issue in snd_pcm_sw_params()
by Dan Carpenter 13 Oct '15

13 Oct '15

"params->tstamp_mode" is a user supplied int. We cap the maximum but allow negative values by mistake. It's harmless but static checkers complain. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 75888dd..4863af5 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -650,7 +650,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, } snd_pcm_stream_unlock_irq(substream); - if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) + if (params->tstamp_mode < 0 || + params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) return -EINVAL; if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) && params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST)

2 2

[alsa-devel] [PATCH V2 0/7] BAT: Add Basic Audio Tester command line tool
by han.lu＠intel.com 13 Oct '15

13 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> BAT (Basic Audio Tester) is a simple command line utility intended to automate audio driver and sound server QA testing with no human interaction. BAT works by playing an audio stream and capturing the same stream in either a digital or analog loop back. It then compares the captured stream to the original to determine if the test case passes or fails. The main idea of frequency detecting is: The analysis function reads data from WAV file, run FFT against the data to get magnitude of frequency vectors, and then calculates the average value and standard deviation of frequency vectors. After that, we define a threshold: threshold = 3 * standard_deviation + average_value Frequencies with amplitude larger than threshold will be recognized as a peak, and the frequency with largest peak value will be recognized as a detected frequency. BAT then compares the detected frequency to target frequency, to decide if the detecting passes or fails. BAT supports 4 working modes: 1. Single line playback; 2. Single line capture and analysis; 3. Playback and capture in loop, and analyze captured data; 4. Local analyze without actual playback or capture. BAT will check devices input by user to decide which mode to use. BAT will create threads for playback and record, and will run spectrum analysis for mode 2, 3 or 4. Changes on V2: 1. Adopt Caleb Crome's method to generate sine signal to avoids unwanted distortion on waveform. Lu, Han (7): BAT: Add initial functions BAT: Add common definitions and functions BAT: Add playback and record functions BAT: Add signal generator BAT: Add converting functions BAT: Add spectrum analysis functions BAT: Add Makefile and configures Makefile.am | 3 + bat/Makefile.am | 14 ++ bat/alsa.c | 594 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/alsa.h | 20 ++ bat/analyze.c | 314 +++++++++++++++++++++++++++++ bat/analyze.h | 16 ++ bat/bat.c | 603 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/common.c | 198 +++++++++++++++++++ bat/common.h | 176 +++++++++++++++++ bat/convert.c | 113 +++++++++++ bat/convert.h | 23 +++ bat/signal.c | 182 +++++++++++++++++ bat/signal.h | 30 +++ configure.ac | 19 +- 14 files changed, 2304 insertions(+), 1 deletion(-) create mode 100644 bat/Makefile.am create mode 100644 bat/alsa.c create mode 100644 bat/alsa.h create mode 100644 bat/analyze.c create mode 100644 bat/analyze.h create mode 100644 bat/bat.c create mode 100644 bat/common.c create mode 100644 bat/common.h create mode 100644 bat/convert.c create mode 100644 bat/convert.h create mode 100644 bat/signal.c create mode 100644 bat/signal.h -- 1.9.1

4 12

[alsa-devel] [PATCH v4] ALSA: usb-audio: Fix max packet size calculation for USB audio
by Ricard Wanderlof 13 Oct '15

13 Oct '15

Rounding must take place before multiplication with the frame size, since each packet contains a whole number of frames. We must also properly consider the data interval, as a larger data interval will result in larger packets, which, depending on the sampling frequency, can result in packet sizes that are less than integral multiples of the packet size for a lower data interval. Detailed explanation and rationale: The code before this commit had the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of frames per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor (which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively)). For this discussion we will initially assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is twofold. First, the rounding operation consists of the addition of 0x0.ffff and subsequent conversion to integer, but as the expression stands, the conversion to integer is done after multiplication with the frame size, rather than before. This results in the resulting maxsize becoming too large. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000. The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). This is fixed by performing the complete rounding operation prior to multiplication with the frame rate. The second problem is that when considering the ep->datainterval, this must be done before rounding, in order to take the advantage of the fact that if the number of bytes per packet is not an integer, the resulting rounded-up integer is not necessarily a factor of two when the data interval is increased by the same factor. For instance, assuming a freqency of 41 kHz, the resulting bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or 0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0), this means that 6 frames per packet are needed, whereas with a data interval of 2 we need 10.25, i.e. 11 frames needed. Rephrasing the maxsize expression to: maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * (frame_bits >> 3); for the above 96 kHz example we instead get ((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)): Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56 New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 104, with ep->maxpacksize then having the same value. As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to (104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111 (with decimals 111.99988). Clearly, we should get back the 104 here, which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 . (The error has not been a problem because it only results in maxsize being a bit too big which just wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long.) Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V2: Added tested device to commit message. No other changes. V3: Updated after input from Clemens. Since the expression is now getting rather convoluted, added a description in the comments in the code. New commit message to reflect the new version. Updated rationale below. V4: Moved most of the rationale (slightly rephrased) to the commit message as suggested by Takashi. Although the error is tolerable because it just wastes a couple of bytes in each packet, however, I'm working on a quirk for a device which stuffs an extra four (non sample) bytes into each isochronous packet, and that means that the maxsize calculation must take this into account. During this work it became obvious that there was something wrong with the maxsize calculation, as the resulting packet size when wMaxPacketSize was hit did not correspond to the packet size which triggered the if statement to start with. Since this seems to me an obvious (after looking at it for a bit...) bug I'm submitting this as a separate patch rather than part of an upcoming patch series. sound/usb/endpoint.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6f7189..a77d9c8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -610,8 +610,23 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, /* assume max. frequency is 25% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); /* but wMaxPacketSize might reduce this */ if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] ALSA: hdac: Explicitly add io.h
by Vinod Koul 13 Oct '15

13 Oct '15

Compiling the hdac extended core on arm fails with below error: sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_writel': >> sound/hda/ext/hdac_ext_bus.c:29:2: error: implicit declaration of >> function +'writel' [-Werror=implicit-function-declaration] writel(value, addr); ^ sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_readl': >> sound/hda/ext/hdac_ext_bus.c:34:2: error: implicit declaration of >> function +'readl' [-Werror=implicit-function-declaration] return readl(addr); This is fixed by explicitly including io.h Fixes: 99463b3a3994 - ('ALSA: hda: provide default bus io ops extended hdac') Reported-by: kbuild test robot <lkp(a)intel.com> Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 4449d1a99089..2433f7c81472 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/slab.h> +#include <linux/io.h> #include <sound/hdaudio_ext.h> MODULE_DESCRIPTION("HDA extended core"); -- 2.4.3

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix inverted internal mic on Lenovo G50-80
by David Henningsson 13 Oct '15

13 Oct '15

Add the appropriate quirk to indicate the Lenovo G50-80 has a stereo mic input where one channel has reverse polarity. Alsa-info available at: https://launchpadlibrarian.net/220846272/AlsaInfo.txt Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1504778 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ca03c40..2f0ec7c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -819,6 +819,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), + SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), -- 1.9.1

2 1

[alsa-devel] How is /proc/asound/cardX/eld#Y.Z generated?
by VDR User 12 Oct '15

12 Oct '15

Hi. I was wondering if someone can tell me how exactly the eld files are generated. Additionally, if they can be overridden somehow. The reason is if I start x while my tv is turned on, the eld reports only stereo audio capability for hdmi even though the tv is plugged into my surround receiver. I don't care what the tv reports, my surround receiver is 7.1 so I always want the eld to reflect my -actual- audio channel setup. Thanks.

2 2

[alsa-devel] [PATCH] ASoC: rt298: set register non-volatile by default
by Bard Liao 12 Oct '15

12 Oct '15

It is not necessary to set registers volatile. So, return false for default case of rt298_volatile_register. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 5a9aadb..56e5e1a 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -128,7 +128,7 @@ static bool rt298_volatile_register(struct device *dev, unsigned int reg) case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_HP_OUT, 0): return true; default: - return true; + return false; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt298: correct index default value
by Bard Liao 12 Oct '15

12 Oct '15

Some of the default value on rt298_index_def are incorrect. Change them to the correct value. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 83bbc62..5a9aadb 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -49,24 +49,24 @@ struct rt298_priv { }; static const struct reg_default rt298_index_def[] = { - { 0x01, 0xaaaa }, - { 0x02, 0x8aaa }, + { 0x01, 0xa5a8 }, + { 0x02, 0x8e95 }, { 0x03, 0x0002 }, - { 0x04, 0xaf01 }, - { 0x08, 0x000d }, - { 0x09, 0xd810 }, - { 0x0a, 0x0120 }, + { 0x04, 0xaf67 }, + { 0x08, 0x200f }, + { 0x09, 0xd010 }, + { 0x0a, 0x0100 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, - { 0x0f, 0x0000 }, - { 0x19, 0x0a17 }, + { 0x0f, 0x0022 }, + { 0x19, 0x0217 }, { 0x20, 0x0020 }, { 0x33, 0x0208 }, { 0x46, 0x0300 }, - { 0x49, 0x0004 }, - { 0x4f, 0x50e9 }, - { 0x50, 0x2000 }, - { 0x63, 0x2902 }, + { 0x49, 0x4004 }, + { 0x4f, 0x50c9 }, + { 0x50, 0x3000 }, + { 0x63, 0x1b02 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, { 0x69, 0x273f }, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3] ASoC: rt5645: Add the HW EQ for the customized speaker output of Google Celes
by Oder Chiou 12 Oct '15

12 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5645.h | 1 + sound/soc/codecs/rt5645.c | 86 ++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 86 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf615..42f0842 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,7 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Invert JD when jack insert */ bool jd_invert; + bool hweq; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c4fe6b..d9532bb 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -22,6 +22,7 @@ #include <linux/acpi.h> #include <linux/dmi.h> #include <linux/regulator/consumer.h> +#include <linux/firmware.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -224,6 +225,16 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5645_eq_param_s { + unsigned short reg; + unsigned short val; +}; + +struct rt5645_hweq_s { + unsigned short num; + struct rt5645_eq_param_s *param; +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -240,6 +251,7 @@ struct rt5645_priv { struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; + struct rt5645_hweq_s hweq; int codec_type; int sysclk; @@ -631,6 +643,59 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_read_hweq_param(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + const struct firmware *fw; + int i; + + if (!request_firmware(&fw, "rt5645_hweq.bin", codec->dev)) { + rt5645->hweq.num = fw->size / sizeof(struct rt5645_eq_param_s); + rt5645->hweq.param = devm_kmemdup(codec->dev, fw->data, + fw->size, GFP_KERNEL); + if (rt5645->hweq.param) { + for (i = 0; i < rt5645->hweq.num; i++) { + rt5645->hweq.param[i].reg = + be16_to_cpu(rt5645->hweq.param[i].reg); + rt5645->hweq.param[i].val = + be16_to_cpu(rt5645->hweq.param[i].val); + } + } + release_firmware(fw); + } + + return 0; +} + +static bool rt5645_validate_hweq(unsigned short reg) +{ + if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) | + (reg == RT5645_EQ_CTRL2)) + return true; + + return false; +} + +static int rt5645_enable_hweq(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int i; + + if (!rt5645->hweq.param) + rt5645_read_hweq_param(codec); + + if (rt5645->hweq.param) { + for (i = 0; i < rt5645->hweq.num; i++) { + if (rt5645_validate_hweq(rt5645->hweq.param[i].reg)) + regmap_write(rt5645->regmap, + rt5645->hweq.param[i].reg, + rt5645->hweq.param[i].val); + } + } + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1532,9 +1597,12 @@ static int rt5645_spk_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 rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: + if (rt5645->pdata.hweq) + rt5645_enable_hweq(codec); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1611,8 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + if (rt5645->pdata.hweq) + snd_soc_write(codec, RT5645_EQ_CTRL2, 0); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3205,6 +3275,20 @@ static int strago_quirk_cb(const struct dmi_system_id *id) return 1; } +static struct rt5645_platform_data celes_platform_data = { + .dmic1_data_pin = RT5645_DMIC1_DISABLE, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, + .hweq = true, +}; + +static int celes_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &celes_platform_data; + + return 1; +} + static const struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", @@ -3215,7 +3299,7 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { }, { .ident = "Google Celes", - .callback = strago_quirk_cb, + .callback = celes_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] [PATCH] ASoC: Intel: Skylake - Add Skylake RT286 I2S machine driver
by Koul, Vinod 12 Oct '15

12 Oct '15

On Sat, 2015-10-10 at 16:42 +0800, kbuild test robot wrote: > Hi Omair, > > [auto build test ERROR on next-20151009 -- if it's inappropriate base, please > ignore] > > config: arm-allmodconfig (attached as .config) > reproduce: That's nice, thanks. We should add a depends on X86 for this to not compile on non X86 code Will send updated patch -- ~Vinod

3 3

[alsa-devel] Applied "ASoc: sh: Fit typo in Kconfig" to the asoc tree
by Mark Brown 12 Oct '15

12 Oct '15

The patch ASoc: sh: Fit typo in Kconfig has been applied to the asoc tree at git://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 d795eb3f5c10d36ed51ca7cd5ce411677f6f7aab Mon Sep 17 00:00:00 2001 From: Masanari Iida <standby24x7(a)gmail.com> Date: Sat, 10 Oct 2015 14:17:58 +0900 Subject: [PATCH] ASoc: sh: Fit typo in Kconfig s/SUR/SRU/g Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 07114b0..de668b2 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_RCAR select SND_SIMPLE_CARD select REGMAP_MMIO help - This option enables R-Car SUR/SCU/SSIU/SSI sound support + This option enables R-Car SRU/SCU/SSIU/SSI sound support config SND_SOC_RSRC_CARD tristate "Renesas Sampling Rate Convert Sound Card" -- 2.5.0

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Fix max packet size calculation for USB audio
by Ricard Wanderlof 12 Oct '15

12 Oct '15

The addition of 0xffff ("almost 1" in Q16.16 format), resulting in the rounding up of the resulting value, was in the wrong place, as it is the resulting packet size that needs to be rounded up to the nearest integer (e.g. if the resulting packet size is calculated as 55.125 bytes, we need a packet of 56 bytes), rather than the rate value (ep->freqmax). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- Since this clearly is a sensitive part of the code, here follows a lengthy rationale with examples to illustrate the problem. The code as it stands has the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of samples per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor, which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively). For this discussion we will assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is that the addition of 0xffff is in the wrong place. We don't want the rounding up to take place on the frequency, but on the resulting packet size. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 786432 (see usb_high_speed_rate()). Add 25% (line 612) and we get 983040. The calculated maxsize is then ((983040 + 65535) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get (96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). Rephrasing the maxsize expression to: maxsize = (ep->freqmax * (frame_bits >> 3) + 0xffff) >> (16 - ep->datainterval); we instead get (983040 * 8 + 65535) >> 16 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 361267, and resulting ep->freqmax 361267 * 1.25 = 451583 (rounded down). Original maxsize = ((451583 + 65535) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = (44100 * 1.25 / 8000) * 8 = 55.125, i.e. 56 bytes required New maxsize = (451583 * 8 + 65535) >> 16 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 108, with ep->maxpacksize then having the same value. As 108 < 127, we get maxsize = 108. ep->freqmax is then recalculated to (108 / 8) << 16 = 884736 . Putting that rate into the original maxsize calculation yields a maxsize of ((884736 + 65535) * 8) >> 16 = 115 (with decimals 115.99988). Clearly, we should get back the 108 here, which we would with the new expression: (884736 * 8 + 65535) >> 16 = 108 . As it currently stands, the error is tolerable because it only results in maxsize being a bit too big which wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long. However, I'm working on a quirk for a device which stuffs an extra four (non sample) bytes into each isochronous packet, and that means that the maxsize calculation must take this into account. During this work it became obvious that there was something wrong with the maxsize calculation, as the resulting packet size when wMaxPacketSize was hit did not correspond to the packet size which triggered the if statement to start with. Since this seems to me an obvious (after looking at it for a bit...) bug I'm submitting this as a separate patch rather than part of the upcoming patch series. sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6f7189..be8a972 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -610,7 +610,7 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, /* assume max. frequency is 25% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + maxsize = (ep->freqmax * (frame_bits >> 3) + 0xffff) >> (16 - ep->datainterval); /* but wMaxPacketSize might reduce this */ if (ep->maxpacksize && ep->maxpacksize < maxsize) { -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 2

[alsa-devel] [PATCH alsa-lib] fix _GNU_SOURCE handling & header inclusion
by Mike Frysinger 12 Oct '15

12 Oct '15

From: Mike Frysinger <vapier(a)chromium.org> The configure script blindly adds -D_GNU_SOURCE to all build settings, even on non-GNU systems. This isn't too much of a big deal (even if it uses the wrong variable -- CFLAGS instead of CPPFLAGS), except that the alsa-lib source itself determines whether to use GNU features when this is defined (such as versionsort). So when we build on non-glibc systems, we get build failures like: src/ucm/parser.c:1268:18: error: 'versionsort' undeclared (first use in this function) #define SORTFUNC versionsort ^ src/ucm/parser.c:1272:54: note: in expansion of macro 'SORTFUNC' err = scandir(filename, &namelist, filename_filter, SORTFUNC); ^ The correct way to add these flags is to use the autoconf helper AC_USE_SYSTEM_EXTENSIONS. Unfortunately, that triggers some more bugs in the alsa build. This macro adds defines to config.h and not directly to CPPFLAGS, so it relies on files correctly including config.h before anything else. A number of alsa files do not do this leading to build failures. The fix there is to shuffle the includes around so that the local ones come first. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- configure.ac | 4 +--- src/conf.c | 2 +- src/timer/timer.c | 9 ++------- src/timer/timer_hw.c | 6 ------ src/timer/timer_local.h | 5 ++--- src/timer/timer_query.c | 6 ------ src/timer/timer_query_hw.c | 6 ------ src/ucm/ucm_local.h | 2 +- 8 files changed, 7 insertions(+), 33 deletions(-) diff --git a/configure.ac b/configure.ac index a14e52d..9cb8614 100644 --- a/configure.ac +++ b/configure.ac @@ -27,11 +27,9 @@ AC_PREFIX_DEFAULT(/usr) dnl Checks for programs. -CFLAGS="$CFLAGS -D_GNU_SOURCE" - - AC_PROG_CC AC_PROG_CPP +AC_USE_SYSTEM_EXTENSIONS AC_PROG_INSTALL AC_PROG_LN_S AC_DISABLE_STATIC diff --git a/src/conf.c b/src/conf.c index c6a83ee..91fca25 100644 --- a/src/conf.c +++ b/src/conf.c @@ -414,12 +414,12 @@ beginning:</P> */ +#include "local.h" #include <stdarg.h> #include <limits.h> #include <sys/stat.h> #include <dirent.h> #include <locale.h> -#include "local.h" #ifdef HAVE_LIBPTHREAD #include <pthread.h> #endif diff --git a/src/timer/timer.c b/src/timer/timer.c index b71a9f8..a25e4f7 100644 --- a/src/timer/timer.c +++ b/src/timer/timer.c @@ -67,15 +67,10 @@ This example shows opening a timer device and reading of timer events. * \anchor example_test_timer */ -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <fcntl.h> -#include <signal.h> -#include <sys/ioctl.h> #include "timer_local.h" +#include <signal.h> + static int snd_timer_open_conf(snd_timer_t **timer, const char *name, snd_config_t *timer_root, snd_config_t *timer_conf, int mode) diff --git a/src/timer/timer_hw.c b/src/timer/timer_hw.c index e833fc8..e61b994 100644 --- a/src/timer/timer_hw.c +++ b/src/timer/timer_hw.c @@ -19,12 +19,6 @@ * */ -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <fcntl.h> -#include <sys/ioctl.h> #include "timer_local.h" #ifndef PIC diff --git a/src/timer/timer_local.h b/src/timer/timer_local.h index 8040b05..eef3b06 100644 --- a/src/timer/timer_local.h +++ b/src/timer/timer_local.h @@ -19,10 +19,9 @@ * */ -#include <stdio.h> -#include <stdlib.h> -#include <limits.h> #include "local.h" +#include <limits.h> +#include <sys/ioctl.h> #ifndef DOC_HIDDEN typedef struct { diff --git a/src/timer/timer_query.c b/src/timer/timer_query.c index 50b098a..93d2455 100644 --- a/src/timer/timer_query.c +++ b/src/timer/timer_query.c @@ -26,12 +26,6 @@ * */ -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <fcntl.h> -#include <sys/ioctl.h> #include "timer_local.h" static int snd_timer_query_open_conf(snd_timer_query_t **timer, diff --git a/src/timer/timer_query_hw.c b/src/timer/timer_query_hw.c index 289ca52..d0c4391 100644 --- a/src/timer/timer_query_hw.c +++ b/src/timer/timer_query_hw.c @@ -19,12 +19,6 @@ * */ -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <fcntl.h> -#include <sys/ioctl.h> #include "timer_local.h" #ifndef PIC diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index c1655c7..3a5d2c2 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -36,8 +36,8 @@ #define UC_MGR_DEBUG #endif -#include <pthread.h> #include "local.h" +#include <pthread.h> #include "use-case.h" #define MAX_FILE 256 -- 2.5.2

2 1

[alsa-devel] [RFC] i2c: Revert back to old device naming for ACPI enumerated I2C slaves
by Jarkko Nikula 12 Oct '15

12 Oct '15

Commit 70762abb9f89 ("i2c: Use stable dev_name for ACPI enumerated I2C slaves") broke the lm-sensors which relies on I2C hwmon slave devices under /sys/bus/i2c/devices/ to be named as "x-00yz". However if those hwmon devices are ACPI 5 enumerated their name became "i2c-INTABCD:ij" and sysfs code in lm-sensors does not find them anymore: lib/sysfs.c:665: if ((!subsys || !strcmp(subsys, "i2c")) && sscanf(dev_name, "%hd-%x", &entry.chip.bus.nr, &entry.chip.addr) == 2) { This patch fixes this by reverting back the old device naming in i2c-core but at the same avoids regression to ALSA SoC drivers that depend on stable device binding. Reverted I2C slave device naming is handled in ALSA SoC core by using the ACPI name as a component name if device is ACPI probed. This keeps the component binding in ALSA SoC stable and requires only minimal changes to affected machine drivers. Fixes: 70762abb9f89 i2c: Use stable dev_name for ACPI enumerated I2C slaves Reported-by: Dustin Byford <dustin(a)cumulusnetworks.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- This is on top of 4.2.0-rc8. I didn't check exact kernel versions where this still applies but I know we would need a few different versions for older stable versions. Mainly because of added and moved drivers in sound/soc/intel/. This is for discussion so I didn't cc stable(a)vger.kernel.org yet. I was thinking would it work if we'd keep the stable name but have an another symlink in /sys/bus/i2c/devices/ that uses "x-00yz" name. However this feels ill-use of devices directory and probably causes more troubles elsewhere. I don't know how common ACPI enumerated I2C hwmon devices are but I guess they exists since Dustin notices this issue and wrote "With that change, /sys/bus/i2c/devices/i2c-0-004a, for example, became /sys/bus/i2c/devices/i2c-PNPXXXX:xx". It could be possible there is use for the new device naming. I found a comment from Documentation/hid/hid-sensor.txt introduced by commit b2eafd7282fd ("HID: sensor: Update document for custom sensor") that seems to indicate current i2c-INTABCD:xy naming may be used from userspace too but I didn't find examples of that. --- drivers/i2c/i2c-core.c | 21 ++++----------------- sound/soc/intel/boards/broadwell.c | 6 +++--- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +++--- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/soc-core.c | 10 ++++++++++ 10 files changed, 28 insertions(+), 31 deletions(-) diff --git a/drivers/i2c/i2c-core.c b/drivers/i2c/i2c-core.c index c83e4d13cfc5..e9c227b9dc92 100644 --- a/drivers/i2c/i2c-core.c +++ b/drivers/i2c/i2c-core.c @@ -913,22 +913,6 @@ void i2c_unlock_adapter(struct i2c_adapter *adapter) } EXPORT_SYMBOL_GPL(i2c_unlock_adapter); -static void i2c_dev_set_name(struct i2c_adapter *adap, - struct i2c_client *client) -{ - struct acpi_device *adev = ACPI_COMPANION(&client->dev); - - if (adev) { - dev_set_name(&client->dev, "i2c-%s", acpi_dev_name(adev)); - return; - } - - /* For 10-bit clients, add an arbitrary offset to avoid collisions */ - dev_set_name(&client->dev, "%d-%04x", i2c_adapter_id(adap), - client->addr | ((client->flags & I2C_CLIENT_TEN) - ? 0xa000 : 0)); -} - /** * i2c_new_device - instantiate an i2c device * @adap: the adapter managing the device @@ -987,7 +971,10 @@ i2c_new_device(struct i2c_adapter *adap, struct i2c_board_info const *info) client->dev.of_node = info->of_node; client->dev.fwnode = info->fwnode; - i2c_dev_set_name(adap, client); + /* For 10-bit clients, add an arbitrary offset to avoid collisions */ + dev_set_name(&client->dev, "%d-%04x", i2c_adapter_id(adap), + client->addr | ((client->flags & I2C_CLIENT_TEN) + ? 0xa000 : 0)); status = device_register(&client->dev); if (status) goto out_err; diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 8bafaf6ceab1..3abcf0d7682e 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -205,7 +205,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, - .codec_name = "i2c-INT343A:00", + .codec_name = "INT343A:00", .codec_dai_name = "rt286-aif1", .init = broadwell_rt286_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | @@ -223,7 +223,7 @@ static int broadwell_suspend(struct snd_soc_card *card){ struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + if (!strcmp(codec->component.name, "INT343A:00")) { dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); rt286_mic_detect(codec, NULL); break; @@ -236,7 +236,7 @@ static int broadwell_resume(struct snd_soc_card *card){ struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + if (!strcmp(codec->component.name, "INT343A:00")) { dev_dbg(codec->dev, "enabling jack detect for resume.\n"); rt286_mic_detect(codec, &broadwell_headset); break; diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 7ab8cc9fbfd5..ae336b8522a8 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -116,7 +116,7 @@ static struct snd_soc_dai_link byt_max98090_dais[] = { .stream_name = "Audio", .cpu_dai_name = "baytrail-pcm-audio", .codec_dai_name = "HiFi", - .codec_name = "i2c-193C9890:00", + .codec_name = "193C9890:00", .platform_name = "baytrail-pcm-audio", .init = byt_max98090_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c index ae89b9b966d9..275b056c495c 100644 --- a/sound/soc/intel/boards/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -186,7 +186,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .stream_name = "Audio", .cpu_dai_name = "baytrail-pcm-audio", .codec_dai_name = "rt5640-aif1", - .codec_name = "i2c-10EC5640:00", + .codec_name = "10EC5640:00", .platform_name = "baytrail-pcm-audio", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7f55d59024a8..973f07548b08 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -171,7 +171,7 @@ static struct snd_soc_dai_link byt_dailink[] = { .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "rt5640-aif1", - .codec_name = "i2c-10EC5640:00", + .codec_name = "10EC5640:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .be_hw_params_fixup = byt_codec_fixup, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 70f832114a5a..1a5348771e03 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -232,7 +232,7 @@ static struct snd_soc_ops cht_be_ssp2_ops = { static struct snd_soc_aux_dev cht_max98090_headset_dev = { .name = "Headset Chip", .init = cht_max98090_headset_init, - .codec_name = "i2c-104C227E:00", + .codec_name = "104C227E:00", }; static struct snd_soc_dai_link cht_dailink[] = { @@ -265,7 +265,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "HiFi", - .codec_name = "i2c-193C9890:00", + .codec_name = "193C9890:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf467842a..c05841264dc7 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -290,7 +290,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "rt5645-aif1", - .codec_name = "i2c-10EC5645:00", + .codec_name = "10EC5645:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, @@ -365,7 +365,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } } card->dev = &pdev->dev; - sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + sprintf(codec_name, "%s:00", drv->acpi_card->codec_id); /* set correct codec name */ strcpy((char *)card->dai_link[2].codec_name, codec_name); snd_soc_card_set_drvdata(card, drv); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2c9cc5be439e..632926d0c595 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -280,7 +280,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .no_pcm = 1, .nonatomic = true, .codec_dai_name = "rt5670-aif1", - .codec_name = "i2c-10EC5670:00", + .codec_name = "10EC5670:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, @@ -296,7 +296,7 @@ static int cht_suspend_pre(struct snd_soc_card *card) struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + if (!strcmp(codec->component.name, "10EC5670:00")) { dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); rt5670_jack_suspend(codec); break; @@ -310,7 +310,7 @@ static int cht_resume_post(struct snd_soc_card *card) struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + if (!strcmp(codec->component.name, "10EC5670:00")) { dev_dbg(codec->dev, "enabling jack detect for resume.\n"); rt5670_jack_resume(codec); break; diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 22558572cb9c..6fd6491e368d 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -160,7 +160,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, - .codec_name = "i2c-INT33CA:00", + .codec_name = "INT33CA:00", .codec_dai_name = "rt5640-aif1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0e1e69c7abd5..811d63e31a7a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -34,6 +34,7 @@ #include <linux/ctype.h> #include <linux/slab.h> #include <linux/of.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -2476,10 +2477,18 @@ static char *fmt_single_name(struct device *dev, int *id) { char *found, name[NAME_SIZE]; int id1, id2; + struct acpi_device *adev; if (dev_name(dev) == NULL) return NULL; + if (ACPI_HANDLE(dev) && + !acpi_bus_get_device(ACPI_HANDLE(dev), &adev)) { + /* use ACPI name as a component name for ACPI probed devices */ + strlcpy(name, acpi_dev_name(adev), NAME_SIZE); + goto out; + } + strlcpy(name, dev_name(dev), NAME_SIZE); /* are we a "%s.%d" name (platform and SPI components) */ @@ -2508,6 +2517,7 @@ static char *fmt_single_name(struct device *dev, int *id) *id = 0; } +out: return kstrdup(name, GFP_KERNEL); } -- 2.5.0

6 16

[alsa-devel] [PATCH 0/6 v3] ALSA: firewire-digi00x: support MIDI functionality
by Takashi Sakamoto 12 Oct '15

12 Oct '15

Hi, This commit updates my former post (patch 13, 15, 16): [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html Digi 00x driver may be newly available in Linux 4.4. This patchset adds ALSA MIDI ports to support physical MIDI ports and physical controls. I tested physical MIDI ports, while never tested physical controls because I'm not an owner of 'console' models. The functionality for physical controls is written according to my assumption. When I investigated Digi 002 rack, Windows driver registers a certain address to the device, and adds virtual MIDI ports to the system. These ports are named as 'control'. When MIDI application sends MIDI messages to the port, asynchronous transactions start on IEEE 1394 bus. On the other hand, I have no way to confirm asynchronous transaction in opposite direction because the rack model doesn't have the functionality. In this time, I apply the same format of outgoing asynchronous transaction to incoming asynchronous transaction. Takashi Sakamoto (6): firewire-digi00x: use in-kernel representation for the type of 8 bits firewire-digi00x: handle MIDI messages in isochronous packets firewire-digi00x: add support for MIDI ports corresponding to isochronous packet streaming firewire-digi00x: add support of asynchronous transaction for incoming MIDI messages from physical controls firewire-digi00x: add support of asynchronous transaction for outgoing MIDI messages to physical controls firewire-digi00x: add support for MIDI ports for physical controls sound/firewire/digi00x/Makefile | 2 +- sound/firewire/digi00x/amdtp-dot.c | 150 +++++++++++++++++++++---- sound/firewire/digi00x/digi00x-midi.c | 160 +++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-stream.c | 6 +- sound/firewire/digi00x/digi00x-transaction.c | 70 ++++++++++-- sound/firewire/digi00x/digi00x.c | 4 + sound/firewire/digi00x/digi00x.h | 15 ++- 7 files changed, 376 insertions(+), 31 deletions(-) create mode 100644 sound/firewire/digi00x/digi00x-midi.c -- 2.1.4

2 8

[alsa-devel] [RESEND PATCH 1/1] Allow any MIDI endpoint to drive use of interrupt transfer on newer Roland devices
by maillist＠superlative.org 11 Oct '15

11 Oct '15

From: "Keith A. Milner" <maillist(a)superlative.org> This patch enables interrupt transfer mode for MIDI ports on newer Boss/Roland devices such as the GT-100/001 which support interrupt transfer on both IN and OUT MIDI endpoints. Previously this wasn't being enabled for these devices as the code was specifically looking for the scenario where the IN endpoint supported interrupt transfer and the OUT endpoint was bulk transfer. Newer devices support interrupt transfer for both endpoints. This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland VS-20. It would benefit from some regresison testing with other devices if possible. Signed-off-by: Keith A. Milner <maillist(a)superlative.org> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b901f46..d50c649 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1860,9 +1860,14 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) hostif = &intf->altsetting[1]; intfd = get_iface_desc(hostif); + /* If either or both of the endpoints support interrupt transfer, + * then use the alternate setting + */ if (intfd->bNumEndpoints != 2 || - (get_endpoint(hostif, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK || - (get_endpoint(hostif, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT) + !((get_endpoint(hostif, 0)->bmAttributes + & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT || + (get_endpoint(hostif, 1)->bmAttributes + & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)) return; snd_printdd(KERN_INFO "switching to altsetting %d with int ep\n", -- 1.9.1

3 5

[alsa-devel] [PATCH v3] ALSA: usb-audio: Fix max packet size calculation for USB audio
by Ricard Wanderlof 11 Oct '15

11 Oct '15

Rounding must take place before multiplication with the frame size, since each packet contains a whole number of frames. We must also properly consider the data interval, as a larger data interval will result in larger packets, which, depending on the sampling frequency, can result in packet sizes that are less than integral multiples of the packet size for a lower data interval. Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V2: Added tested device to commit message. No other changes. V3: Updated after input from Clemens. Since the expression is now getting rather convoluted, added a description in the comments in the code. New commit message to reflect the new version. Updated rationale below. The code as it stands has the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of frames per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor (which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively)). For this discussion we will initially assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is twofold. First, the rounding operation consists of the addition of 0x0.ffff and subsequent conversion to integer, but as the expression stands, the conversion to integer is done after multiplication with the frame size, rather than before. This results in the resulting maxsize becoming too large. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000. The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). This is fixed by performing the complete rounding operation prior to multiplication with the frame rate. The second problem is that when considering the ep->datainterval, this must be done before rounding, in order to take the advantage of the fact that the number of bytes per packet is not an integer, the resulting rounded-up integer is not necessarily a factor of two when the data interval is increased by the same factor. For instance, assuming a freqency of 41 kHz, the resulting bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or 0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0), this 6 frames per packet is required, whereas with a data interval of 2 we need 10.25, i.e. 11 frames needed. Rephrasing the maxsize expression to: maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * (frame_bits >> 3); for the above 96 kHz example we instead get ((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)): Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56 New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 104, with ep->maxpacksize then having the same value. As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to (104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111 (with decimals 111.99988). Clearly, we should get back the 104 here, which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 . As it currently stands, the error is tolerable because it only results in maxsize being a bit too big which wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long. However, I'm working on a quirk for a device which stuffs an extra four (non sample) bytes into each isochronous packet, and that means that the maxsize calculation must take this into account. During this work it became obvious that there was something wrong with the maxsize calculation, as the resulting packet size when wMaxPacketSize was hit did not correspond to the packet size which triggered the if statement to start with. Since this seems to me an obvious (after looking at it for a bit...) bug I'm submitting this as a separate patch rather than part of an upcoming patch series. sound/usb/endpoint.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6f7189..a77d9c8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -610,8 +610,23 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, /* assume max. frequency is 25% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); /* but wMaxPacketSize might reduce this */ if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 3

[alsa-devel] [PATCH] ALSA: firewire-lib: continue packet processing at detecting wrong CIP headers
by Takashi Sakamoto 11 Oct '15

11 Oct '15

In firewire-lib, isochronous packet streaming is stopped when detecting wrong value for FMT field of CIP headers. Although this is appropriate to IEC 61883-1 and 6, some BeBoB based devices with vendors' customization use invalid value to FMT field of CIP headers in the beginning of streaming. $ journalctl snd-bebob fw1.0: Detect unexpected protocol: 01000000 8000ffff I got this log with M-Audio FireWire 1814. In this line, the value of FMT field is 0x00, while it should be 0x10 in usual AMDTP. Except for the beginning, these devices continue to transfer packets with valid value for FMT field, except for the beginning. Therefore, in this case, firewire-lib should continue to process packets. The former implementation of firewire-lib performs it. This commit loosens the handling of wrong value, to continue packet processing in the case. Fixes: 414ba022a528('ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index fa10b58..ed29026 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -468,10 +468,12 @@ static int handle_in_packet(struct amdtp_stream *s, /* Check valid protocol or not. */ fmt = (cip_header[1] & CIP_FMT_MASK) >> CIP_FMT_SHIFT; if (fmt != s->fmt) { - dev_err(&s->unit->device, - "Detect unexpected protocol: %08x %08x\n", - cip_header[0], cip_header[1]); - return -EIO; + dev_info_ratelimited(&s->unit->device, + "Detect unexpected protocol: %08x %08x\n", + cip_header[0], cip_header[1]); + *data_blocks = 0; + pcm_frames = 0; + goto end; } /* Calculate data blocks */ -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: bebob: constify snd_bebob_rate_spec structures
by Julia Lawall 11 Oct '15

11 Oct '15

The snd_bebob_rate_spec structure type contains only function pointers. In all instances of this structure type, the functions are provided statically and are never changed subsequently. Add a const annotation on all initializations and uses. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/firewire/bebob/bebob.c | 2 +- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_focusrite.c | 4 ++-- sound/firewire/bebob/bebob_maudio.c | 4 ++-- sound/firewire/bebob/bebob_pcm.c | 2 +- sound/firewire/bebob/bebob_proc.c | 2 +- sound/firewire/bebob/bebob_stream.c | 2 +- sound/firewire/bebob/bebob_terratec.c | 2 +- sound/firewire/bebob/bebob_yamaha.c | 2 +- 9 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/firewire/bebob/bebob_yamaha.c b/sound/firewire/bebob/bebob_yamaha.c index 5810170..2a9ffdb 100644 --- a/sound/firewire/bebob/bebob_yamaha.c +++ b/sound/firewire/bebob/bebob_yamaha.c @@ -51,7 +51,7 @@ static struct snd_bebob_clock_spec clock_spec = { .types = clk_src_types, .get = &clk_src_get, }; -static struct snd_bebob_rate_spec rate_spec = { +static const struct snd_bebob_rate_spec rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c index a1a3949..e62923e 100644 --- a/sound/firewire/bebob/bebob_focusrite.c +++ b/sound/firewire/bebob/bebob_focusrite.c @@ -260,7 +260,7 @@ saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size) return err; } -static struct snd_bebob_rate_spec saffirepro_both_rate_spec = { +static const struct snd_bebob_rate_spec saffirepro_both_rate_spec = { .get = &saffirepro_both_clk_freq_get, .set = &saffirepro_both_clk_freq_set, }; @@ -287,7 +287,7 @@ struct snd_bebob_spec saffirepro_10_spec = { .meter = NULL }; -static struct snd_bebob_rate_spec saffire_both_rate_spec = { +static const struct snd_bebob_rate_spec saffire_both_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 0ed5e5f..091290d 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -335,7 +335,7 @@ static void bebob_remove(struct fw_unit *unit) snd_card_free_when_closed(bebob->card); } -static struct snd_bebob_rate_spec normal_rate_spec = { +static const struct snd_bebob_rate_spec normal_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate }; diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index d3c9d8d..aff9da3 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -71,7 +71,7 @@ struct snd_bebob_meter_spec { }; struct snd_bebob_spec { struct snd_bebob_clock_spec *clock; - struct snd_bebob_rate_spec *rate; + const struct snd_bebob_rate_spec *rate; struct snd_bebob_meter_spec *meter; }; diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index 9242e33..470d526 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -55,7 +55,7 @@ phase24_series_clk_src_get(struct snd_bebob *bebob, unsigned int *id) return 0; } -static struct snd_bebob_rate_spec phase_series_rate_spec = { +static const struct snd_bebob_rate_spec phase_series_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 2fdc1f1..ef224d6 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -155,7 +155,7 @@ static int pcm_open(struct snd_pcm_substream *substream) { struct snd_bebob *bebob = substream->private_data; - struct snd_bebob_rate_spec *spec = bebob->spec->rate; + const struct snd_bebob_rate_spec *spec = bebob->spec->rate; unsigned int sampling_rate; enum snd_bebob_clock_type src; int err; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index a2baa47..fc7a66d 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -580,7 +580,7 @@ end: int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) { - struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; + const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; struct amdtp_stream *master, *slave; enum cip_flags sync_mode; unsigned int curr_rate; diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index 057495d..dc9cd0c 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -712,7 +712,7 @@ end: } /* for special customized devices */ -static struct snd_bebob_rate_spec special_rate_spec = { +static const struct snd_bebob_rate_spec special_rate_spec = { .get = &special_get_rate, .set = &special_set_rate, }; @@ -733,7 +733,7 @@ struct snd_bebob_spec maudio_special_spec = { }; /* Firewire 410 specification */ -static struct snd_bebob_rate_spec usual_rate_spec = { +static const struct snd_bebob_rate_spec usual_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c index 301cc6a..0f3b2ca 100644 --- a/sound/firewire/bebob/bebob_proc.c +++ b/sound/firewire/bebob/bebob_proc.c @@ -138,7 +138,7 @@ proc_read_clock(struct snd_info_entry *entry, "SYT-Match", }; struct snd_bebob *bebob = entry->private_data; - struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; + const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; enum snd_bebob_clock_type src; unsigned int rate;

3 8

[alsa-devel] [PATCH 1/1] Allow any MIDI endpoint to drive use of interrupt transfer on Roland devices
by maillist＠superlative.org 11 Oct '15

11 Oct '15

From: "Keith A. Milner" <kamilner(a)superlative.org> Signed-off-by: Keith A. Milner <maillist(a)superlative.org> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b901f46..52b8c72 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1860,9 +1860,14 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) hostif = &intf->altsetting[1]; intfd = get_iface_desc(hostif); + /* If either or both of the endpoints support interrupt transfer, + * then use the alternate setting + */ if (intfd->bNumEndpoints != 2 || - (get_endpoint(hostif, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK || - (get_endpoint(hostif, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT) + !((get_endpoint(hostif, 0)->bmAttributes + & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT || + (get_endpoint(hostif, 1)->bmAttributes + & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)) return; snd_printdd(KERN_INFO "switching to altsetting %d with int ep\n", -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ALSA: usb-audio: Fix max packet size calculation for USB audio
by Ricard Wanderlof 10 Oct '15

10 Oct '15

The addition of 0xffff ("almost 1" in Q16.16 format), resulting in the rounding up of the resulting value, was in the wrong place, as it is the resulting packet size that needs to be rounded up to the nearest integer (e.g. if the resulting packet size is calculated as 55.125 bytes, we need a packet of 56 bytes), rather than the rate value (ep->freqmax). Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V2: Added tested device to commit message. No other changes. Since this clearly is a sensitive part of the code, here follows a lengthy rationale with examples to illustrate the problem. The code as it stands has the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of samples per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor, which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively). For this discussion we will assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is that the addition of 0xffff is in the wrong place. We don't want the rounding up to take place on the frequency, but on the resulting packet size. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 786432 (see usb_high_speed_rate()). Add 25% (line 612) and we get 983040. The calculated maxsize is then ((983040 + 65535) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get (96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). Rephrasing the maxsize expression to: maxsize = (ep->freqmax * (frame_bits >> 3) + 0xffff) >> (16 - ep->datainterval); we instead get (983040 * 8 + 65535) >> 16 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 361267, and resulting ep->freqmax 361267 * 1.25 = 451583 (rounded down). Original maxsize = ((451583 + 65535) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = (44100 * 1.25 / 8000) * 8 = 55.125, i.e. 56 bytes required New maxsize = (451583 * 8 + 65535) >> 16 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 108, with ep->maxpacksize then having the same value. As 108 < 127, we get maxsize = 108. ep->freqmax is then recalculated to (108 / 8) << 16 = 884736 . Putting that rate into the original maxsize calculation yields a maxsize of ((884736 + 65535) * 8) >> 16 = 115 (with decimals 115.99988). Clearly, we should get back the 108 here, which we would with the new expression: (884736 * 8 + 65535) >> 16 = 108 . As it currently stands, the error is tolerable because it only results in maxsize being a bit too big which wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long. However, I'm working on a quirk for a device which stuffs an extra four (non sample) bytes into each isochronous packet, and that means that the maxsize calculation must take this into account. During this work it became obvious that there was something wrong with the maxsize calculation, as the resulting packet size when wMaxPacketSize was hit did not correspond to the packet size which triggered the if statement to start with. Since this seems to me an obvious (after looking at it for a bit...) bug I'm submitting this as a separate patch rather than part of an upcoming patch series. sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6f7189..be8a972 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -610,7 +610,7 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, /* assume max. frequency is 25% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + maxsize = (ep->freqmax * (frame_bits >> 3) + 0xffff) >> (16 - ep->datainterval); /* but wMaxPacketSize might reduce this */ if (ep->maxpacksize && ep->maxpacksize < maxsize) { -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 4

[alsa-devel] How to download and install patch for ICS-43432 codec driver
by Ankhit Vivekananda 10 Oct '15

10 Oct '15

Hello, I am new to developing with ALSA. I have a situation where I need to use ICS-43432 MEMS microphone with a TMS320DM365 processor. I wanted to know if I could download the patch for ICS-43432 codec driver: http://permalink.gmane.org/gmane.linux.alsa.devel/143442. Appreciate any help. Thank you, Ankhit

2 1

[alsa-devel] Boss GT-001
by maillist＠superlative.org 10 Oct '15

10 Oct '15

I recently got access to a Boss GT-001. This is a small USB audio Interface for guitar with onboard FX which doesn't work with ALSA under Linux Mint 17 (running kernel 3.16.0-38-generic). I've not tried it with the latest ALSA development branch, but scanning the recent ALSA Dev archives, I can't see anything which might fix this. # lsusb ... Bus 001 Device 011: ID 0582:0187 Roland Corp (detailed lsusb output is towards the bottom of this email) The unit is moderately quirky compared to most audio devices. This unit includes on-board Boss guitar amp emulations and effects and the USB connection allows you to simultaneously record both the "wet" (with effects) and "dry" (without effects) signal. Thus the USB carries two separate stereo pairs to the computer. It also allows "re-amping" which allows you to apply effects to the output signal. The way this seems to do this is by having two separate stereo pair outputs from the computer one of which is processed via the onboard effects, and one of which is not. This is all just background, but the upshot of this is that the unit is a 2x stereo pair in, and 2x stereo pair out. However, this not only does not show up properly under Linux, but it will not play or record anything. ALSA sees it thusly: # aplay -l ... card 4: GT001 [GT-001], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 PLaying a CD-format file: # aplay hw:4 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1233: Sample format not available Available formats: - S32_LE # aplay hw:4 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo No audio is output For input: # arecord -l card 4: GT001 [GT-001], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 The unit also supports MIDI which is used for both control of the unit itself via a PC editing application (editing settings, changing patches, etc.) and for simple DAW control. There are three USB MIDI ports for this which, on Windows, are labelled "GT-001", "GT-001 DAW CTRL" and "GT-001 CTRL" On Linux these show up as follows: # amidi -l IO hw:4,0,0 GT-001 MIDI 1 IO hw:4,0,1 GT-001 MIDI 2 IO hw:4,0,2 GT-001 MIDI 3 Any advice and pointers to get this working would be appreciated. Dmesg output: [17530.759008] usb 1-1: new high-speed USB device number 12 using ehci-pci [17530.893299] usb 1-1: New USB device found, idVendor=0582, idProduct=0187 [17530.893303] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [17530.893305] usb 1-1: Product: GT-001 [17530.893307] usb 1-1: Manufacturer: BOSS [17530.894412] snd-usb-audio: probe of 1-1:1.0 failed with error -5 [17530.927591] usb 1-1: Unable to change format on ep #8e: already in use [17530.927614] usb 1-1: Unable to change format on ep #8e: already in use [17530.927764] usb 1-1: Unable to change format on ep #8e: already in use [17530.927945] usb 1-1: Unable to change format on ep #8e: already in use [17530.928161] usb 1-1: Unable to change format on ep #8e: already in use [17530.928600] usb 1-1: Unable to change format on ep #8e: already in use [17530.928620] usb 1-1: Unable to change format on ep #8e: already in use [17530.928693] usb 1-1: Unable to change format on ep #8e: already in use [17530.928832] usb 1-1: Unable to change format on ep #8e: already in use [17530.928958] usb 1-1: Unable to change format on ep #8e: already in use [17530.929350] usb 1-1: Unable to change format on ep #8e: already in use [17530.929377] usb 1-1: Unable to change format on ep #8e: already in use [17530.929500] usb 1-1: Unable to change format on ep #8e: already in use [17530.929743] usb 1-1: Unable to change format on ep #8e: already in use [17530.929988] usb 1-1: Unable to change format on ep #8e: already in use [17530.930577] usb 1-1: Unable to change format on ep #8e: already in use [17530.930617] usb 1-1: Unable to change format on ep #8e: already in use [17530.930808] usb 1-1: Unable to change format on ep #8e: already in use [17530.931064] usb 1-1: Unable to change format on ep #8e: already in use [17530.931213] usb 1-1: Unable to change format on ep #8e: already in use [17531.330778] usb 1-1: Unable to change format on ep #8e: already in use [17531.334500] usb 1-1: Unable to change format on ep #8e: already in use lsusb output: lsusb -vv -d 0582:0187 Bus 001 Device 011: ID 0582:0187 Roland Corp. Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x0187 bcdDevice 0.00 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 176 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 04 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x0d EP 13 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 04 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8e EP 14 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 03 03 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Cheers, Keith

4 19

[alsa-devel] [asoc:topic/rockchip 4/4] sound/soc/rockchip/rockchip_spdif.c:67:12: warning: 'rk_spdif_runtime_resume' defined but not used
by kbuild test robot 10 Oct '15

10 Oct '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rockchip head: f874b80e1571118fcf4554878633556f06f998e6 commit: f874b80e1571118fcf4554878633556f06f998e6 [4/4] ASoC: rockchip: Add rockchip SPDIF transceiver driver config: microblaze-allmodconfig (attached as .config) 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 f874b80e1571118fcf4554878633556f06f998e6 # save the attached .config to linux build tree make.cross ARCH=microblaze All warnings (new ones prefixed by >>): >> sound/soc/rockchip/rockchip_spdif.c:67:12: warning: 'rk_spdif_runtime_resume' defined but not used [-Wunused-function] static int rk_spdif_runtime_resume(struct device *dev) ^ vim +/rk_spdif_runtime_resume +67 sound/soc/rockchip/rockchip_spdif.c 51 { .compatible = "rockchip,rk3288-spdif", 52 .data = (void *) RK_SPDIF_RK3288 }, 53 {}, 54 }; 55 MODULE_DEVICE_TABLE(of, rk_spdif_match); 56 57 static int rk_spdif_runtime_suspend(struct device *dev) 58 { 59 struct rk_spdif_dev *spdif = dev_get_drvdata(dev); 60 61 clk_disable_unprepare(spdif->mclk); 62 clk_disable_unprepare(spdif->hclk); 63 64 return 0; 65 } 66 > 67 static int rk_spdif_runtime_resume(struct device *dev) 68 { 69 struct rk_spdif_dev *spdif = dev_get_drvdata(dev); 70 int ret; 71 72 ret = clk_prepare_enable(spdif->mclk); 73 if (ret) { 74 dev_err(spdif->dev, "mclk clock enable failed %d\n", ret); 75 return ret; --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] Dummy module and sample rates
by Uli Franke 10 Oct '15

10 Oct '15

Hi Clemens, I fiddled around with the Dummy module and it seems to be the thing we're looking for. I observed a rather strange behavior though: I set the hw params sample rate to 44k1 without error, but a subsequent call of snd_pcm_start returns EPIPE. When setting hw params sample rate to 4k this does not happen. Did I miss anything? Regards Uli ____ uli@ankerklause:~/.../dlna/wdmr$ modinfo snd-dummy filename: /lib/modules/3.16.0-46-generic/kernel/sound/drivers/snd-dummy.ko license: GPL description: Dummy soundcard (/dev/null) author: Jaroslav Kysela <perex(a)perex.cz> srcversion: BB2C5716F580F3D7D0F53D1 depends: snd-pcm,snd intree: Y vermagic: 3.16.0-46-generic SMP mod_unload modversions 686 signer: Magrathea: Glacier signing key sig_key: 6C:3B:5C:5E:34:A0:6D:7D:5F:7D:A3:D0:50:98:75:D9:6C:6D:E6:C4 sig_hashalgo: sha512 parm: index:Index value for dummy soundcard. (array of int) parm: id:ID string for dummy soundcard. (array of charp) parm: enable:Enable this dummy soundcard. (array of bool) parm: model:Soundcard model. (array of charp) parm: pcm_devs:PCM devices # (0-4) for dummy driver. (array of int) parm: pcm_substreams:PCM substreams # (1-128) for dummy driver. (array of int) parm: fake_buffer:Fake buffer allocations. (bool) parm: hrtimer:Use hrtimer as the timer source. (bool)

3 4

[alsa-devel] [PATCH] ASoC: Intel: Skylake - Add Skylake RT286 I2S machine driver
by Vinod Koul 09 Oct '15

09 Oct '15

From: Omair M Abdullah <omair.m.abdullah(a)intel.com> Add the SKL I2S machine driver using Realtek ALC286S codec in I2S mode. Signed-off-by: Omair M Abdullah <omair.m.abdullah(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 12 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_rt286.c | 268 +++++++++++++++++++++++++++++++++++++ 3 files changed, 282 insertions(+) create mode 100644 sound/soc/intel/boards/skl_rt286.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 664df1f44530..ea3221a8b615 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -140,3 +140,15 @@ config SND_SOC_INTEL_SKYLAKE select SND_HDA_EXT_CORE select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + +config SND_SOC_INTEL_SKL_RT286_MACH + tristate "ASoC Audio driver for SKL with RT286 I2S mode" + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT286 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Skylake platforms + with RT286 I2S audio codec. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index cb94895c9edb..371c4565cad8 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-skl_rt286-objs := skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -15,3 +16,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c new file mode 100644 index 000000000000..71907b2c1611 --- /dev/null +++ b/sound/soc/intel/boards/skl_rt286.c @@ -0,0 +1,268 @@ +/* + * Intel Skylake I2S Machine Driver + * + * Copyright (C) 2014-2015, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * 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. + * + * This program is distributed in the hope that 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. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/rt286.h" + +static struct snd_soc_jack skylake_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin skylake_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new skylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget skylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route skylake_rt286_map[] = { + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC2"}, + {"DMIC AIF", NULL, "SoC DMIC"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + { "ssp0 Tx", NULL, "codec1_out"}, + + { "codec0_in", NULL, "ssp0 Rx" }, + { "codec1_in", NULL, "ssp0 Rx" }, + { "ssp0 Rx", NULL, "AIF1 Capture" }, + + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + + { "hif1", NULL, "iDisp Tx"}, + { "iDisp Tx", NULL, "iDisp_out"}, + +}; + +static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &skylake_headset, + skylake_headset_pins, ARRAY_SIZE(skylake_headset_pins)); + + if (ret) + return ret; + + rt286_mic_detect(codec, &skylake_headset); + + return 0; +} + + +static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The output is 48KHz, stereo, 16bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int skylake_rt286_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 *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops skylake_rt286_ops = { + .hw_params = skylake_rt286_hw_params, +}; + +/* skylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link skylake_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "Skl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + }, + { + .name = "Skl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_capture = 1, + }, + { + .name = "Skl Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .ignore_suspend = 1, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .be_id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .init = skylake_rt286_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp0_fixup, + .ops = &skylake_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .be_id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +/* skylake audio machine driver for SPT + RT286S */ +static struct snd_soc_card skylake_rt286 = { + .name = "skylake-rt286", + .owner = THIS_MODULE, + .dai_link = skylake_rt286_dais, + .num_links = ARRAY_SIZE(skylake_rt286_dais), + .controls = skylake_controls, + .num_controls = ARRAY_SIZE(skylake_controls), + .dapm_widgets = skylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), + .dapm_routes = skylake_rt286_map, + .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), + .fully_routed = true, +}; + +static int skylake_audio_probe(struct platform_device *pdev) +{ + skylake_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&skylake_rt286); +} + +static int skylake_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&skylake_rt286); + return 0; +} + +static struct platform_driver skylake_audio = { + .probe = skylake_audio_probe, + .remove = skylake_audio_remove, + .driver = { + .name = "skl_alc286s_i2s", + }, +}; + +module_platform_driver(skylake_audio) + +/* Module information */ +MODULE_AUTHOR("Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com>"); +MODULE_DESCRIPTION("Intel SST Audio for Skylake"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_alc286s_i2s"); -- 2.4.3

2 1

[alsa-devel] [asoc:topic/rockchip 4/4] sound/soc/rockchip/rockchip_spdif.c:283:6: warning: cast from pointer to integer of different size
by kbuild test robot 09 Oct '15

09 Oct '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rockchip head: f874b80e1571118fcf4554878633556f06f998e6 commit: f874b80e1571118fcf4554878633556f06f998e6 [4/4] ASoC: rockchip: Add rockchip SPDIF transceiver driver config: x86_64-allmodconfig (attached as .config) reproduce: git checkout f874b80e1571118fcf4554878633556f06f998e6 # save the attached .config to linux build tree make ARCH=x86_64 All warnings (new ones prefixed by >>): sound/soc/rockchip/rockchip_spdif.c: In function 'rk_spdif_probe': >> sound/soc/rockchip/rockchip_spdif.c:283:6: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] if ((int) match->data == RK_SPDIF_RK3288) { ^ vim +283 sound/soc/rockchip/rockchip_spdif.c 267 .writeable_reg = rk_spdif_wr_reg, 268 .readable_reg = rk_spdif_rd_reg, 269 .volatile_reg = rk_spdif_volatile_reg, 270 .cache_type = REGCACHE_FLAT, 271 }; 272 273 static int rk_spdif_probe(struct platform_device *pdev) 274 { 275 struct device_node *np = pdev->dev.of_node; 276 struct rk_spdif_dev *spdif; 277 const struct of_device_id *match; 278 struct resource *res; 279 void __iomem *regs; 280 int ret; 281 282 match = of_match_node(rk_spdif_match, np); > 283 if ((int) match->data == RK_SPDIF_RK3288) { 284 struct regmap *grf; 285 286 grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf"); 287 if (IS_ERR(grf)) { 288 dev_err(&pdev->dev, 289 "rockchip_spdif missing 'rockchip,grf' \n"); 290 return PTR_ERR(grf); 291 } --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Modify the log level
by Vinod Koul 09 Oct '15

09 Oct '15

dev_info is too noisy for tplg wiget loading, so move it to debug level Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index b1bffa52cbcd..a5d8420ce69a 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1178,7 +1178,7 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, bind_event: if (tplg_w->event_type == 0) { - dev_info(bus->dev, "ASoC: No event handler required\n"); + dev_dbg(bus->dev, "ASoC: No event handler required\n"); return 0; } -- 2.4.3

2 1

[alsa-devel] [PATCH 1/1] ASoC: da7213: Correct units description of dmic-clkrate
by Adam Thomson 09 Oct '15

09 Oct '15

Description previously stated MHz whereas the resolution is in Hz. This has now has been updated to align with reality. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- This patch is based on changes introduced under patch: 'ASoC: da7213: Add bindings documentation for codec driver' (Commit a97d4e93a9bc556d0b3c2efb7695eb4c79938de7, kernel/git/broonie/sound.git) Documentation/devicetree/bindings/sound/da7213.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt index 7280e82..5890280 100644 --- a/Documentation/devicetree/bindings/sound/da7213.txt +++ b/Documentation/devicetree/bindings/sound/da7213.txt @@ -18,7 +18,7 @@ Optional properties: ["lrise_rfall", "lfall_rrise"] - dlg,dmic-samplephase : When to sample audio from DMIC. ["on_clkedge", "between_clkedge"] -- dlg,dmic-clkrate : DMIC clock frequency (MHz). +- dlg,dmic-clkrate : DMIC clock frequency (Hz). [<1500000>, <3000000>] ====== -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: nau8825: add acpi match ID
by yang.a.fang＠intel.com 09 Oct '15

09 Oct '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> This patch adds the acpi match ID for nau8825 codec Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/nau8825.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c44a7a1..5c1badf 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -16,6 +16,7 @@ #include <linux/regmap.h> #include <linux/slab.h> #include <linux/clk.h> +#include <linux/acpi.h> #include <sound/initval.h> #include <sound/tlv.h> @@ -1090,10 +1091,19 @@ static const struct of_device_id nau8825_of_ids[] = { MODULE_DEVICE_TABLE(of, nau8825_of_ids); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8825_acpi_match[] = { + { "10508825", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8825_acpi_match); +#endif + static struct i2c_driver nau8825_driver = { .driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), + .acpi_match_table = ACPI_PTR(nau8825_acpi_match), }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, -- 1.7.9.5

2 1

[alsa-devel] Applied "ASoC: rockchip: Add rockchip SPDIF transceiver driver" to the asoc tree
by Mark Brown 09 Oct '15

09 Oct '15

The patch ASoC: rockchip: Add rockchip SPDIF transceiver driver has been applied to the asoc tree at git://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 f874b80e1571118fcf4554878633556f06f998e6 Mon Sep 17 00:00:00 2001 From: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> Date: Thu, 8 Oct 2015 15:31:13 +0200 Subject: [PATCH] ASoC: rockchip: Add rockchip SPDIF transceiver driver Add a driver for the SPDIF transceiver available on RK3066, RK3188 and RK3288. Heavily based on the rockchip i2s driver. Signed-off-by: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/rockchip/Kconfig | 8 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_spdif.c | 409 ++++++++++++++++++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++++ 4 files changed, 482 insertions(+) create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 5709057..f1e0c70 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -15,6 +15,14 @@ config SND_SOC_ROCKCHIP_I2S Rockchip I2S device. The device supports upto maximum of 8 channels each for play and record. +config SND_SOC_ROCKCHIP_SPDIF + tristate "Rockchip SPDIF Device Driver" + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for SPDIF driver for + Rockchip SPDIF transceiver device. + config SND_SOC_ROCKCHIP_MAX98090 tristate "ASoC support for Rockchip boards using a MAX98090 codec" depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index e9ba558..c0bf560 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,7 +1,9 @@ # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-spdif-objs := rockchip_spdif.o obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o +obj-$(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 diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c new file mode 100644 index 0000000..9d5c470 --- /dev/null +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -0,0 +1,409 @@ +/* sound/soc/rockchip/rk_spdif.c + * + * ALSA SoC Audio Layer - Rockchip I2S Controller driver + * + * Copyright (c) 2014 Rockchip Electronics Co. Ltd. + * Author: Jianqun <jay.xu(a)rock-chips.com> + * Copyright (c) 2015 Collabora Ltd. + * Author: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> + * + * 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/delay.h> +#include <linux/of_gpio.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> +#include <linux/mfd/syscon.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> + +#include "rockchip_spdif.h" + +enum rk_spdif_type { + RK_SPDIF_RK3066, + RK_SPDIF_RK3188, + RK_SPDIF_RK3288, +}; + +#define RK3288_GRF_SOC_CON2 0x24c + +struct rk_spdif_dev { + struct device *dev; + + struct clk *mclk; + struct clk *hclk; + + struct snd_dmaengine_dai_dma_data playback_dma_data; + + struct regmap *regmap; +}; + +static const struct of_device_id rk_spdif_match[] = { + { .compatible = "rockchip,rk3066-spdif", + .data = (void *) RK_SPDIF_RK3066 }, + { .compatible = "rockchip,rk3188-spdif", + .data = (void *) RK_SPDIF_RK3188 }, + { .compatible = "rockchip,rk3288-spdif", + .data = (void *) RK_SPDIF_RK3288 }, + {}, +}; +MODULE_DEVICE_TABLE(of, rk_spdif_match); + +static int rk_spdif_runtime_suspend(struct device *dev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + + clk_disable_unprepare(spdif->mclk); + clk_disable_unprepare(spdif->hclk); + + return 0; +} + +static int rk_spdif_runtime_resume(struct device *dev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(spdif->mclk); + if (ret) { + dev_err(spdif->dev, "mclk clock enable failed %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(spdif->hclk); + if (ret) { + dev_err(spdif->dev, "hclk clock enable failed %d\n", ret); + return ret; + } + + return 0; +} + +static int rk_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai); + unsigned int val = SPDIF_CFGR_HALFWORD_ENABLE; + int srate, mclk; + int ret; + + srate = params_rate(params); + switch (srate) { + case 32000: + case 48000: + case 96000: + mclk = 96000 * 128; /* 12288000 hz */ + break; + case 44100: + mclk = 44100 * 256; /* 11289600 hz */ + break; + case 192000: + mclk = 192000 * 128; /* 24576000 hz */ + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val |= SPDIF_CFGR_VDW_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= SPDIF_CFGR_VDW_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= SPDIF_CFGR_VDW_24; + break; + default: + return -EINVAL; + } + + /* Set clock and calculate divider */ + ret = clk_set_rate(spdif->mclk, mclk); + if (ret != 0) { + dev_err(spdif->dev, "Failed to set module clock rate: %d\n", + ret); + return ret; + } + + val |= SPDIF_CFGR_CLK_DIV(mclk/(srate * 256)); + ret = regmap_update_bits(spdif->regmap, SPDIF_CFGR, + SPDIF_CFGR_CLK_DIV_MASK | SPDIF_CFGR_HALFWORD_ENABLE | + SDPIF_CFGR_VDW_MASK, + val); + + return ret; +} + +static int rk_spdif_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai); + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR, + SPDIF_DMACR_TDE_ENABLE, + SPDIF_DMACR_TDE_ENABLE); + + if (ret != 0) + return ret; + + ret = regmap_update_bits(spdif->regmap, SPDIF_XFER, + SPDIF_XFER_TXS_START, + SPDIF_XFER_TXS_START); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR, + SPDIF_DMACR_TDE_ENABLE, + SPDIF_DMACR_TDE_DISABLE); + + if (ret != 0) + return ret; + + ret = regmap_update_bits(spdif->regmap, SPDIF_XFER, + SPDIF_XFER_TXS_START, + SPDIF_XFER_TXS_STOP); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int rk_spdif_dai_probe(struct snd_soc_dai *dai) +{ + struct rk_spdif_dev *spdif = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &spdif->playback_dma_data; + + return 0; +} + +static const struct snd_soc_dai_ops rk_spdif_dai_ops = { + .hw_params = rk_spdif_hw_params, + .trigger = rk_spdif_trigger, +}; + +static struct snd_soc_dai_driver rk_spdif_dai = { + .probe = rk_spdif_dai_probe, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &rk_spdif_dai_ops, +}; + +static const struct snd_soc_component_driver rk_spdif_component = { + .name = "rockchip-spdif", +}; + +static bool rk_spdif_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SPDIF_CFGR: + case SPDIF_DMACR: + case SPDIF_INTCR: + case SPDIF_XFER: + case SPDIF_SMPDR: + return true; + default: + return false; + } +} + +static bool rk_spdif_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SPDIF_CFGR: + case SPDIF_SDBLR: + case SPDIF_INTCR: + case SPDIF_INTSR: + case SPDIF_XFER: + return true; + default: + return false; + } +} + +static bool rk_spdif_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SPDIF_INTSR: + case SPDIF_SDBLR: + return true; + default: + return false; + } +} + +static const struct regmap_config rk_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SPDIF_SMPDR, + .writeable_reg = rk_spdif_wr_reg, + .readable_reg = rk_spdif_rd_reg, + .volatile_reg = rk_spdif_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int rk_spdif_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct rk_spdif_dev *spdif; + const struct of_device_id *match; + struct resource *res; + void __iomem *regs; + int ret; + + match = of_match_node(rk_spdif_match, np); + if ((int) match->data == RK_SPDIF_RK3288) { + struct regmap *grf; + + grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf"); + if (IS_ERR(grf)) { + dev_err(&pdev->dev, + "rockchip_spdif missing 'rockchip,grf' \n"); + return PTR_ERR(grf); + } + + /* Select the 8 channel SPDIF solution on RK3288 as + * the 2 channel one does not appear to work + */ + regmap_write(grf, RK3288_GRF_SOC_CON2, BIT(1) << 16); + } + + spdif = devm_kzalloc(&pdev->dev, sizeof(*spdif), GFP_KERNEL); + if (!spdif) + return -ENOMEM; + + spdif->hclk = devm_clk_get(&pdev->dev, "hclk"); + if (IS_ERR(spdif->hclk)) { + dev_err(&pdev->dev, "Can't retrieve rk_spdif bus clock\n"); + return PTR_ERR(spdif->hclk); + } + ret = clk_prepare_enable(spdif->hclk); + if (ret) { + dev_err(spdif->dev, "hclock enable failed %d\n", ret); + return ret; + } + + spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(spdif->mclk)) { + dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); + return PTR_ERR(spdif->mclk); + } + + ret = clk_prepare_enable(spdif->mclk); + if (ret) { + dev_err(spdif->dev, "clock enable failed %d\n", ret); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, + &rk_spdif_regmap_config); + if (IS_ERR(spdif->regmap)) { + dev_err(&pdev->dev, + "Failed to initialise managed register map\n"); + return PTR_ERR(spdif->regmap); + } + + spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; + spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + spdif->playback_dma_data.maxburst = 4; + + spdif->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, spdif); + + pm_runtime_set_active(&pdev->dev); + pm_runtime_enable(&pdev->dev); + pm_request_idle(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, + &rk_spdif_component, + &rk_spdif_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI\n"); + goto err_pm_runtime; + } + + ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM\n"); + goto err_pcm_register; + } + + return 0; + +err_pcm_register: + snd_dmaengine_pcm_unregister(&pdev->dev); +err_pm_runtime: + pm_runtime_disable(&pdev->dev); + + return ret; +} + +static int rk_spdif_remove(struct platform_device *pdev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + rk_spdif_runtime_suspend(&pdev->dev); + + clk_disable_unprepare(spdif->mclk); + clk_disable_unprepare(spdif->hclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops rk_spdif_pm_ops = { + SET_RUNTIME_PM_OPS(rk_spdif_runtime_suspend, rk_spdif_runtime_resume, + NULL) +}; + +static struct platform_driver rk_spdif_driver = { + .probe = rk_spdif_probe, + .remove = rk_spdif_remove, + .driver = { + .name = "rockchip-spdif", + .of_match_table = of_match_ptr(rk_spdif_match), + .pm = &rk_spdif_pm_ops, + }, +}; +module_platform_driver(rk_spdif_driver); + +MODULE_ALIAS("platform:rockchip-spdif"); +MODULE_DESCRIPTION("ROCKCHIP SPDIF transceiver Interface"); +MODULE_AUTHOR("Sjoerd Simons <sjoerd.simons(a)collabora.co.uk>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h new file mode 100644 index 0000000..07f86a2 --- /dev/null +++ b/sound/soc/rockchip/rockchip_spdif.h @@ -0,0 +1,63 @@ +/* + * ALSA SoC Audio Layer - Rockchip SPDIF transceiver driver + * + * Copyright (c) 2015 Collabora Ltd. + * Author: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> + * + * 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 _ROCKCHIP_SPDIF_H +#define _ROCKCHIP_SPDIF_H + +/* + * CFGR + * transfer configuration register +*/ +#define SPDIF_CFGR_CLK_DIV_SHIFT (16) +#define SPDIF_CFGR_CLK_DIV_MASK (0xff << SPDIF_CFGR_CLK_DIV_SHIFT) +#define SPDIF_CFGR_CLK_DIV(x) (x << SPDIF_CFGR_CLK_DIV_SHIFT) + +#define SPDIF_CFGR_HALFWORD_SHIFT 2 +#define SPDIF_CFGR_HALFWORD_DISABLE (0 << SPDIF_CFGR_HALFWORD_SHIFT) +#define SPDIF_CFGR_HALFWORD_ENABLE (1 << SPDIF_CFGR_HALFWORD_SHIFT) + +#define SPDIF_CFGR_VDW_SHIFT 0 +#define SPDIF_CFGR_VDW(x) (x << SPDIF_CFGR_VDW_SHIFT) +#define SDPIF_CFGR_VDW_MASK (0xf << SPDIF_CFGR_VDW_SHIFT) + +#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x00) +#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x01) +#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x10) + +/* + * DMACR + * DMA control register +*/ +#define SPDIF_DMACR_TDE_SHIFT 5 +#define SPDIF_DMACR_TDE_DISABLE (0 << SPDIF_DMACR_TDE_SHIFT) +#define SPDIF_DMACR_TDE_ENABLE (1 << SPDIF_DMACR_TDE_SHIFT) + +#define SPDIF_DMACR_TDL_SHIFT 0 +#define SPDIF_DMACR_TDL(x) ((x) << SPDIF_DMACR_TDL_SHIFT) +#define SPDIF_DMACR_TDL_MASK (0x1f << SDPIF_DMACR_TDL_SHIFT) + +/* + * XFER + * Transfer control register +*/ +#define SPDIF_XFER_TXS_SHIFT 0 +#define SPDIF_XFER_TXS_STOP (0 << SPDIF_XFER_TXS_SHIFT) +#define SPDIF_XFER_TXS_START (1 << SPDIF_XFER_TXS_SHIFT) + +#define SPDIF_CFGR (0x0000) +#define SPDIF_SDBLR (0x0004) +#define SPDIF_DMACR (0x0008) +#define SPDIF_INTCR (0x000c) +#define SPDIF_INTSR (0x0010) +#define SPDIF_XFER (0x0018) +#define SPDIF_SMPDR (0x0020) + +#endif /* _ROCKCHIP_SPDIF_H */ -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: rt5645: fix build warning
by Sudip Mukherjee 09 Oct '15

09 Oct '15

We were getting build warning about "Section mismatch". dmi_platform_intel_broadwell is being referenced from the probe function rt5645_i2c_probe(), but dmi_platform_intel_broadwell was marked with __initdata. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- 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 4c4fe6b..89ba4ff 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3244,7 +3244,7 @@ static int buddy_quirk_cb(const struct dmi_system_id *id) return 1; } -static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { +static struct dmi_system_id dmi_platform_intel_broadwell[] = { { .ident = "Chrome Buddy", .callback = buddy_quirk_cb, -- 1.9.1

3 4

[alsa-devel] [PATCH 0/4] ASoC: Intel: Skylake: Enable runtime pm
by Vinod Koul 09 Oct '15

09 Oct '15

Although driver already has runtime pm handlers, some of things were missing to have it properly functional, so add the fixes and correct handling Jeeja KP (4): ASoC: Intel: Skylake: Check CPA bit in DSP core power down ASoC: Intel: Skylake: Verify the status bit before handling interrupt ASoC: Intel: Skylake: Correct the runtime handler behaviour ASoC: Intel: Skylake: power down all link in suspend sound/soc/intel/common/sst-dsp-priv.h | 1 + sound/soc/intel/skylake/skl-sst-dsp.c | 7 ++++- sound/soc/intel/skylake/skl-sst-ipc.c | 12 ++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 1 + sound/soc/intel/skylake/skl-sst.c | 56 +++++++++++++++++++---------------- sound/soc/intel/skylake/skl.c | 3 ++ 6 files changed, 53 insertions(+), 27 deletions(-) -- 2.4.3

2 8

[alsa-devel] [PATCH 0/5 v4] ALSA: add helper functions for MIDI message transmission on IEEE 1394 asynchronous transaction
by Takashi Sakamoto 09 Oct '15

09 Oct '15

Hi, This patchset updates a part of my previous post: [alsa-devel] [PATCH 0/5] ALSA: add helper functions for MIDI message transmission on IEEE 1394 asynchronous transaction http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/098536.ht… (Besides, the patchset should have been version 3, thus this patchset is v4.) This patchset adds some helper functions for MIDI transmission via IEEE 1394 asynchronous transaction. Changes: * fix compile error due to missing member of structure. * improve comments Takashi Sakamoto (5): ALSA: firewire-lib: add helper functions for asynchronous transactions to transfer MIDI messages ALSA: firewire-lib: add a restriction for a transaction at once ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error sound/firewire/lib.c | 138 +++++++++++++++++++++++++++++++++++++++++++++++++++ sound/firewire/lib.h | 56 +++++++++++++++++++++ 2 files changed, 194 insertions(+) -- 2.1.4

2 6

Re: [alsa-devel] [PATCH] ALSA: seq_oss: fix waitqueue_active without memory barrier in snd-seq-oss
by Takashi Iwai 09 Oct '15

09 Oct '15

On Fri, 09 Oct 2015 02:35:54 +0200, Kosuke Tatsukawa wrote: > > snd_seq_oss_readq_put_event() seems to be missing a memory barrier which > might cause the waker to not notice the waiter and miss sending a > wake_up as in the following figure. > > snd_seq_oss_readq_put_event snd_seq_oss_readq_wait > ------------------------------------------------------------------------ > /* wait_event_interruptible_timeout */ > /* __wait_event_interruptible_timeout */ > /* ___wait_event */ > for (;;) { prepare_to_wait_event(&wq, &__wait, > state); > spin_lock_irqsave(&q->lock, flags); > if (waitqueue_active(&q->midi_sleep)) > /* The CPU might reorder the test for > the waitqueue up here, before > prior writes complete */ > if ((q->qlen>0 || q->head==q->tail) > ... > __ret = schedule_timeout(__ret) > if (q->qlen >= q->maxlen - 1) { > memcpy(&q->q[q->tail], ev, sizeof(*ev)); > q->tail = (q->tail + 1) % q->maxlen; > q->qlen++; > ------------------------------------------------------------------------ > > There are two other place in sound/core/seq/oss/ which have similar > code. The attached patch removes the call to waitqueue_active() leaving > just wake_up() behind. This fixes the problem because the call to > spin_lock_irqsave() in wake_up() will be an ACQUIRE operation. > > I found this issue when I was looking through the linux source code > for places calling waitqueue_active() before wake_up*(), but without > preceding memory barriers, after sending a patch to fix a similar > issue in drivers/tty/n_tty.c (Details about the original issue can be > found here: https://lkml.org/lkml/2015/9/28/849). > > Signed-off-by: Kosuke Tatsukawa <tatsu(a)ab.jp.nec.com> Thanks, applied now. Takashi > --- > sound/core/seq/oss/seq_oss_readq.c | 6 ++---- > sound/core/seq/oss/seq_oss_writeq.c | 4 +--- > 2 files changed, 3 insertions(+), 7 deletions(-) > > diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c > index ccd8935..046cb58 100644 > --- a/sound/core/seq/oss/seq_oss_readq.c > +++ b/sound/core/seq/oss/seq_oss_readq.c > @@ -91,8 +91,7 @@ snd_seq_oss_readq_clear(struct seq_oss_readq *q) > q->head = q->tail = 0; > } > /* if someone sleeping, wake'em up */ > - if (waitqueue_active(&q->midi_sleep)) > - wake_up(&q->midi_sleep); > + wake_up(&q->midi_sleep); > q->input_time = (unsigned long)-1; > } > > @@ -138,8 +137,7 @@ snd_seq_oss_readq_put_event(struct seq_oss_readq *q, union evrec *ev) > q->qlen++; > > /* wake up sleeper */ > - if (waitqueue_active(&q->midi_sleep)) > - wake_up(&q->midi_sleep); > + wake_up(&q->midi_sleep); > > spin_unlock_irqrestore(&q->lock, flags); > > diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c > index d50338b..1f6788a 100644 > --- a/sound/core/seq/oss/seq_oss_writeq.c > +++ b/sound/core/seq/oss/seq_oss_writeq.c > @@ -138,9 +138,7 @@ snd_seq_oss_writeq_wakeup(struct seq_oss_writeq *q, abstime_t time) > spin_lock_irqsave(&q->sync_lock, flags); > q->sync_time = time; > q->sync_event_put = 0; > - if (waitqueue_active(&q->sync_sleep)) { > - wake_up(&q->sync_sleep); > - } > + wake_up(&q->sync_sleep); > spin_unlock_irqrestore(&q->sync_lock, flags); > } > > -- > 1.7.1 >

1 0

[alsa-devel] Release of the tool for benchmarking sound server power efficiency
by Alexander E. Patrakov 09 Oct '15

09 Oct '15

Earlier today (sorry - yesterday!) at the audio mini-summit I promised to release a tool that I used for evaluating sound server energy efficiency. The tool is a shell script, which is attached, together with an example report. It needs to be run as root (in a throwaway VM, if you need), with one argument: the name of a directory to be created and populated with various files. The script needs the "debootstrap" program to be installed. The script will populate the directory with a Debian Jessie amd64 installation, compile CRAS there, and pack everything into an initramfs file that can be used for evaluation. It will produce two files as the main result: vmlinuz and initramfs.img, near the directory that it creates. The initramfs will play a sound file in a loop (please use headphones, so that your laptop does not annoy anyone) and log the laptop battery energy or charge. If it drops below 75% of the initial value, the experiment is terminated, the log is saved to the disk, and the laptop is powered off. You can add this to your boot loader. Example for extlinux: LABEL battery-pulse7 MENU LABEL Battery Benchmark (pulseaudio 7) LINUX /boot/battery-benchmark/vmlinuz INITRD /boot/battery-benchmark/initramfs.img APPEND quiet root=ram save=/dev/md126p1 test_case=pulse pa_version=7 latency_msec=28 dobenchmark Here is the meaning of the arguments: quiet: don't print verbose messages to the console root=ram: use ramfs (not tmpfs) for the extracted initramfs save=/dev/...: save the log to this block device at the end. The /battery-reports directory will be created, and the log will be placed there. FAT32, NTFS and EXT4 filesystems are supported. test_case=pulse|alsa|cras: what we want to test. alsa means raw hw device. pa_version=5|6|7: version of PulseAudio to test latency_msec: buffer size or its equivalent, in milliseconds cras_output: the correct output node for CRAS. Usually 5:0 (the default) or 6:0, please see cras_test_client --dump_server_info resampler: force a given resampler for PulseAudio (if not given, no resampler is used) dobenchmark: dpms-blank the screen and perform the measurement automatically To replicate the results in my presentation, it is sufficient to obtain logs for all three versions of PulseAudio, as well as ALSA and CRAS, for the following latency_msec values: 7, 28, 448. Please send any reports created by this tool to me directly, and mention the laptop model. A big "thanks" goes to Felipe Sateler for providing Debian packages of PulseAudio that are mostly compatible with Debian Jessie. -- Alexander E. Patrakov

1 0

[alsa-devel] [PATCH alsa-utils] build: Do not try to detect cross-compiler
by Mike Frysinger 08 Oct '15

08 Oct '15

From: Mike Frysinger <vapier(a)chromium.org> cross compilers are passed via path may not be a gcc based cross compiler in such cases this check fails and try's to force gcc based cross compiler detection, This code is a convenience that limits the build system. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- Note: this is the same fix that landed in alsa-lib. configure.ac | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/configure.ac b/configure.ac index 8c2d1a5..79a9aaf 100644 --- a/configure.ac +++ b/configure.ac @@ -12,20 +12,6 @@ AM_GNU_GETTEXT_VERSION([0.15]) dnl Checks for programs. -dnl try to gues cross-compiler if not set -if test "x$target" != "x$host" -a -z "`echo $CC | grep -e '-gcc'`"; -then - AC_MSG_CHECKING(for cross-compiler) - - which ${program_prefix}gcc >/dev/null 2>&1 && CC=${program_prefix}gcc - which ${target_cpu}-${target_os}-gcc >/dev/null 2>&1 \ - && CC=${target_cpu}-${target-os}-gcc - which ${target_cpu}-${target_vendor}-${target_os}-gcc >/dev/null 2>&1 \ - && CC=${target_cpu}-${target_vendor}-${target_os}-gcc - - AC_MSG_RESULT($CC) -fi - AC_PROG_CC dnl AC_PROG_CXX AC_PROG_INSTALL -- 2.5.2

2 1

[alsa-devel] [PATCH 1/2] ALSA: hdac: Copy codec helpers to core
by Vinod Koul 08 Oct '15

08 Oct '15

From: "Subhransu S. Prusty" <subhransu.s.prusty(a)intel.com> The current codec helpers are local to hda code and needs to be moved to core so that other users can use it. The helpers to read/write the codec and to check the power state of widgets is copied Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/hdaudio.h | 6 ++++ sound/hda/hdac_device.c | 81 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 87 insertions(+) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 49bc836fcd84..26e956f4b7c6 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -147,6 +147,12 @@ int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid, unsigned int format); +int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm); +int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm); +bool snd_hdac_check_power_state(struct hdac_device *hdac, + hda_nid_t nid, unsigned int target_state); /** * snd_hdac_read_parm - read a codec parameter * @codec: the codec object diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index db96042a497f..b3b0ad289df1 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -952,3 +952,84 @@ bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid, return true; } EXPORT_SYMBOL_GPL(snd_hdac_is_supported_format); + +static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); + unsigned int res; + + if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) + return -1; + + return res; +} + +static int codec_write(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); + + return snd_hdac_exec_verb(hdac, cmd, flags, NULL); +} + +/** + * snd_hdac_codec_read - send a command and get the response + * @hdac: the HDAC device + * @nid: NID to send the command + * @flags: optional bit flags + * @verb: the verb to send + * @parm: the parameter for the verb + * + * Send a single command and read the corresponding response. + * + * Returns the obtained response value, or -1 for an error. + */ +int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + return codec_read(hdac, nid, flags, verb, parm); +} +EXPORT_SYMBOL_GPL(snd_hdac_codec_read); + +/** + * snd_hdac_codec_write - send a single command without waiting for response + * @hdac: the HDAC device + * @nid: NID to send the command + * @flags: optional bit flags + * @verb: the verb to send + * @parm: the parameter for the verb + * + * Send a single command without waiting for response. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + return codec_write(hdac, nid, flags, verb, parm); +} +EXPORT_SYMBOL_GPL(snd_hdac_codec_write); + +/* + * snd_hdac_check_power_state: check whether the actual power state matches + * with the target state + * + * @hdac: the HDAC device + * @nid: NID to send the command + * @target_state: target state to check for + * + * Return true if state matches, false if not + */ +bool snd_hdac_check_power_state(struct hdac_device *hdac, + hda_nid_t nid, unsigned int target_state) +{ + unsigned int state = codec_read(hdac, nid, 0, + AC_VERB_GET_POWER_STATE, 0); + + if (state & AC_PWRST_ERROR) + return true; + state = (state >> 4) & 0x0f; + return (state == target_state); +} +EXPORT_SYMBOL_GPL(snd_hdac_check_power_state); -- 2.4.3

2 2

[alsa-devel] [PATCH] ASoC: Document DAI signal polarity
by Anatol Pomozov 08 Oct '15

08 Oct '15

Currently there is no clear definition of what FSYNC polarity is. Different drivers use its own definition of what is "normal" and what is "inverted" fsync. This leads to compatibility problems between drivers. For example TegraX1 driver assumes that DSP-A format with frames starting at rising FSYNC edge has "inverted" polarity, while RT5677 assumes it is "normal" polarity. Explicitly specify meaning of BCLK/FSYNC polarity to avoid future compatibility problems. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- include/sound/soc-dai.h | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..91e2e61 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,25 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * * Specifies whether the DAI can also support inverted clocks for the specified * format. + * + * BCLK: + * - "normal" polarity means signal is available at rising edge of BCLK + * - "inverted" polarity means signal is available at falling edge of BCLK + * + * FSYNC "normal" polarity depends on the frame format: + * - I2S: frame consists of left then right channel data. Left channel starts + * with falling FSYNC edge, right channel starts with rising FSYNC edge. + * - Left/Right Justified: frame consists of left then right channel data. + * Left channel starts with rising FSYNC edge, right channel starts with + * falling FSYNC edge. + * - DSP A/B: Frame starts with rising FSYNC edge. + * - AC97: Frame starts with rising FSYNC edge. + * + * "Negative" FSYNC polarity is the one opposite of "normal" polarity. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- 2.6.0.rc2.230.g3dd15c0

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: da7213: Add DT support for codec
by Adam Thomson 08 Oct '15

08 Oct '15

This patch set adds DT support to the codec driver, and supporting bindings documentation. Patch set is based against v4.3-rc4 Changes in v2: - Split mclk data support changes into separate patch - Use of_match_ptr() for match table assignment - Only call clk_prepare_enable/disable() if mclk data provided (not NULL). - Add error checking for clk_prepare_enable() usage. Adam Thomson (3): ASoC: da7213: Add DT support to codec driver ASoC: da7213: Add support to handle mclk data provided to driver ASoC: da7213: Add bindings documentation for codec driver Documentation/devicetree/bindings/sound/da7213.txt | 41 +++++ include/sound/da7213.h | 3 - sound/soc/codecs/da7213.c | 190 +++++++++++++++++++-- sound/soc/codecs/da7213.h | 8 +- 4 files changed, 221 insertions(+), 21 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/da7213.txt -- 1.9.3

5 9

[alsa-devel] USB audio device startup question
by Ricard Wanderlof 08 Oct '15

08 Oct '15

(This is in reference to getting the Zoom R16 to work, but the main query in this post is of a more general nature). In trying to get a Zoom R16 in audio interface mode to work in Linux I'm trying to understand how the USB-audio (snd-usb-audio module) subsystem works in Linux. I've looked in the USB audio version 1 and 2 documents, as well as the USB 2.0 standard, but there are some things that don't seem to be well defined there, like exactly how the audio data is packaged in the isochronous transfer packets. I've been using Wireshark to visualize the USB communication, and also comparing with the corresponding transfers on a Windows box by dumping the traffic with USBPcap. The only other USB audio interface I have access to here is an Edirol UA-5. I've noted that when plugging in the UA-5, there's period of about 15 seconds during which capture and playback are set up and audio data transferred, after which the interface is shut down again. When enabling the R16, during the corresponding initial period I see playback data but no capture data during this phase. I don't know if that is significant or not, for instance, if the lack of capture data during this phase is because the playback is not working as it should. My question is: what governs this startup phase, can someone point to a specific part of the code or some documentation describing what is supposed to go on here? Is it some sort of device test period to determine that the device actually works? The problem with the R16 is that although capture works fine, playback doesn't result in any audio. With the correct quirk set up (most importantly, using a 24 bit format), aplay appears to work fine with no errors, neither are there any complaints from dmesg, and looking at the communication I can see outgoing isochronous data from the host being transferred fine. More work needs to be done, but I've looked at the startup of the Windows driver which is deceptively simple: it basically just sets the appropriate capture and playback interfaces, together with sending off some sample rate USB control commands, the isochronous link comes up and that seems to be it. I've tried to emulate the specific USB control commands in Linux, as well as add some delays that there appear to be in the Windows driver, but so far it hasn't made a difference. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] ASoC: rt5645: Remove __initdata annotation from dmi_platform_intel_broadwell
by Jarkko Nikula 08 Oct '15

08 Oct '15

Fix the "Section mismatch" warning when this driver is built as module by removing the __initdata annotation from dmi_platform_intel_broadwell[]. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.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 4c4fe6b456d5..89ba4ffdbee7 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3244,7 +3244,7 @@ static int buddy_quirk_cb(const struct dmi_system_id *id) return 1; } -static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { +static struct dmi_system_id dmi_platform_intel_broadwell[] = { { .ident = "Chrome Buddy", .callback = buddy_quirk_cb, -- 2.6.1

1 1

[alsa-devel] [tinycompress] [PATCH] build: Fix make distcheck
by arun＠accosted.net 08 Oct '15

08 Oct '15

From: Arun Raghavan <git(a)arunraghavan.net> srcdir may not be the same as builddir, so use the correct variable for the include path. --- src/lib/Makefile.am | 2 +- src/utils/Makefile.am | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/src/lib/Makefile.am b/src/lib/Makefile.am index 6f900ac..4baeab5 100644 --- a/src/lib/Makefile.am +++ b/src/lib/Makefile.am @@ -2,4 +2,4 @@ tinycompressdir = $(libdir) tinycompress_LTLIBRARIES = libtinycompress.la libtinycompress_la_SOURCES = compress.c -libtinycompress_la_CFLAGS = -I $(top_builddir)/include +libtinycompress_la_CFLAGS = -I$(top_srcdir)/include diff --git a/src/utils/Makefile.am b/src/utils/Makefile.am index 6a89c45..5f685e3 100644 --- a/src/utils/Makefile.am +++ b/src/utils/Makefile.am @@ -3,8 +3,8 @@ bin_PROGRAMS = cplay crecord cplay_SOURCES = cplay.c crecord_SOURCES = crec.c -cplay_CFLAGS = -I$(top_builddir)/include -crecord_CFLAGS = -I$(top_builddir)/include +cplay_CFLAGS = -I$(top_srcdir)/include +crecord_CFLAGS = -I$(top_srcdir)/include cplay_LDADD = $(top_builddir)/src/lib/libtinycompress.la -- 2.4.3

2 1

[alsa-devel] [PATCH 0/2] ASoC: rockchip: i2s: add 8 channels capture and lrck-mode
by Sugar Zhang 08 Oct '15

08 Oct '15

Support max 8 channels capture. support lrck clk mode configuration, there are 3 modes: - txrx: lrck_tx and lrck_rx are different. - tx_share: lrck_tx is shared with lrck_rx. - rx_share: lrck_rx is shared with lrck_tx. Sugar Zhang (2): ASoC: rockchip: i2s: add 8 channels capture and lrck-mode support Documentation: DT bindings: rockchip-i2s: add capture and lrck-mode .../devicetree/bindings/sound/rockchip-i2s.txt | 5 +++ sound/soc/rockchip/rockchip_i2s.c | 48 +++++++++++++++++++++- sound/soc/rockchip/rockchip_i2s.h | 23 +++++++++++ 3 files changed, 74 insertions(+), 2 deletions(-) -- 2.3.6

4 16

[alsa-devel] [PATCH 0/2] *** wm8962 regmap related fix ***
by Jiada Wang 08 Oct '15

08 Oct '15

This patch set aims to fix issues in wm8962 codec driver related to regmap, currently any attempt to read from ALC Coefficient register will fail when wm8962 is in suspend mode. As ALC2 register is volatile register, it can't be read when cache_only flag is set. Another issue is, if wm8962's regulator is set to 'regulator-always-on' mode, then after wm8962 is resumed from suspend, wm8962 codec is reset, but cache_dirty flag isn't set, this cause difference between actual wm8962 HW and regmap cache. Jiada Wang (2): ASoC: WM8962: mark cache_dirty flag after software reset in pm_resume ASoC: Codec: wm8962: declare ALC Coefficients as 4 separate registers sound/soc/codecs/wm8962.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) -- 2.4.5

2 6

[alsa-devel] Audio routing in the tlv320aic3104
by Rick Mann 07 Oct '15

07 Oct '15

I've managed to get a hacky, fragile config working, in the sense that it sends my tlv320aic3104 samples from speaker-test, and the clocks and data all look good on the oscilloscope, and it ping-pongs between the left and right channel. But it's outputting the signal on the HPLOUT/HPROUT pins, not the L&R Line Out pins. My DTS has this: fragment@3 { target = <&ocp>; __overlay__ { sound { compatible = "ti,da830-evm-audio"; ti,model = "DA830 EVM"; ti,audio-codec = <&tlv320aic3x>; ti,mcasp-controller = <&mcasp0>; ti,codec-clock-rate = <12000000>; ti,audio-routing = "Line Out", "LLOUT", "Line Out", "RLOUT"; }; }; }; The original had: "Headphone Jack", "HPLOUT", "Headphone Jack", "HPROUT", "LINE1L", "Line In", "LINE1R", "Line In"; I also have an asound.state file I got from the Audio Cape: https://raw.githubusercontent.com/CircuitCo/BeagleBone-Audio/files/asound.s… I don't really know how to interpret that file. One resource online said it held the current state of the system. I guess that suggests I should play with amixer. But how does the DTS routing come into play? Thanks, -- Rick Mann rmann(a)latencyzero.com

2 3

[alsa-devel] [GIT PULL] ASoC fixes for v4.3
by Mark Brown 07 Oct '15

07 Oct '15

The following changes since commit 049e6dde7e57f0054fdc49102e7ef4830c698b46: Linux 4.3-rc4 (2015-10-04 16:57:17 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.3-rc4 for you to fetch changes up to e4fc141d2a022a63c87a4851b3c688eca6a1647b: Merge remote-tracking branches 'asoc/fix/tlv320aic3x' and 'asoc/fix/wm8962' into asoc-linus (2015-10-07 16:07:50 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.3 Quite a few fixes here but they're all very small and driver specific, none of them really stand out if you aren't using the relevant hardware but they're all useful if you do happen to have an affected device. ---------------------------------------------------------------- Andreas Dannenberg (1): ASoC: tas2552: fix dBscale-min declaration Benoît Thébaudeau (1): ASoC: imx-ssi: Fix DAI hardware signal inversions Gianluca Renzi (2): ASoC: sgtl5000: fix error message output for MicBias voltage ASoC: sgtl5000: fix wrong register MIC_BIAS_VOLTAGE setup on probe Jiada Wang (1): ASoC: wm8962: balance pm_runtime_enable Lars-Peter Clausen (1): ASoC: db1200: Fix DAI link format for db1300 and db1550 Mark Brown (3): MAINTAINERS: Remove wm97xx entry Merge remote-tracking branches 'asoc/fix/db1200', 'asoc/fix/dwc', 'asoc/fix/imx-ssi', 'asoc/fix/maintainers', 'asoc/fix/rt5645', 'asoc/fix/sgtl5000' and 'asoc/fix/tas2552' into asoc-linus Merge remote-tracking branches 'asoc/fix/tlv320aic3x' and 'asoc/fix/wm8962' into asoc-linus Oder Chiou (1): ASoC: rt5645: Correct the naming and setting of ADC Boost Volume Control Rick Mann (1): ASoC: tlv320aic3x: Prevent writing reserved registers on tlv320aic3104 CODECs Yitian Bu (1): ASoC: dwc: correct irq clear method yitian (1): ASoC: dwc: fix dma stop transferring issue MAINTAINERS | 9 --------- sound/soc/au1x/db1200.c | 4 ++++ sound/soc/codecs/rt5645.c | 6 +++--- sound/soc/codecs/rt5645.h | 16 +++++++++------- sound/soc/codecs/sgtl5000.c | 6 +++--- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 19 +++++++++++-------- sound/soc/codecs/wm8962.c | 5 ++++- sound/soc/dwc/designware_i2s.c | 19 ++++++++++++++----- sound/soc/fsl/imx-ssi.c | 19 +++++++++---------- 10 files changed, 58 insertions(+), 47 deletions(-)

1 0

[alsa-devel] [PATCH v4 0/8] Add DSP topology management for SKL
by Vinod Koul 07 Oct '15

07 Oct '15

The SKL driver does not code DSP topology in driver. With this series the ASoC topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, compute resources for modules, initialize and bind the pipe modules. It uses the SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules. Last patch in this series removes the unused dais. changes in v4: - rename resource allocators and add resource deallocator - remove bind_unbind and add in callng fn - fix some 80char warns - some minor cleanup at odd places changes in v3: - removeno/less logic wrappers - remove most of debug prints, few kept - use asoc widget list macros instead of using list_for_xxx - fix one instance of no empty line between two functions - fix tplg event ID > 255 - add more comments for locking, usage of widget list and few other places, fixed few error prints, add file name etc changes in v2: - add more explanation in changelog and code - add explanation of MCPS - fix whitespace issues - remove skl widget check on event handler as it is redundant - add switch for depth - add else for direction based code - remove cast for NHLT blob query - make debug prints to error - remove dump info - clean some debug prints Jeeja KP (5): ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table Vinod Koul (3): ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers sound/soc/intel/Kconfig | 1 + sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-pcm.c | 141 ++- sound/soc/intel/skylake/skl-topology.c | 1238 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 28 + sound/soc/intel/skylake/skl-tplg-interface.h | 80 ++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 8 files changed, 1495 insertions(+), 36 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c -- 2.4.3

3 15

[alsa-devel] [PATCH v3 0/8] Add DSP topology management for SKL
by Vinod Koul 07 Oct '15

07 Oct '15

The SKL driver does not code DSP topology in driver. With this series the ASoC topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, compute resources for modules, initialize and bind the pipe modules. It uses the SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules. Last patch in this series removes the unused dais. changes in v3: - removeno/less logic wrappers - remove most of debug prints, few kept - use asoc widget list macros instead of using list_for_xxx - fix one instance of no empty line between two functions - fix tplg event ID > 255 - add more comments for locking, usage of widget list and few other places, fixed few error prints, add file name etc changes in v2: - add more explanation in changelog and code - add explanation of MCPS - fix whitespace issues - remove skl widget check on event handler as it is redundant - add switch for depth - add else for direction based code - remove cast for NHLT blob query - make debug prints to error - remove dump info - clean some debug prints Jeeja KP (6): ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table Vinod Koul (2): ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers sound/soc/intel/Kconfig | 1 + sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-pcm.c | 143 +++- sound/soc/intel/skylake/skl-topology.c | 1185 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 28 + sound/soc/intel/skylake/skl-tplg-interface.h | 80 ++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 8 files changed, 1443 insertions(+), 37 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c -- 1.9.1

3 11

[alsa-devel] [PATCH 00/13] Add DSP topology management for SKL
by Subhransu S. Prusty 07 Oct '15

07 Oct '15

The SKL driver does not code DSP topology in driver. With this series the soc topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, add building blocks to instanatiate modules and pipes. It uses SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules, create pipeline and manage them. Last patch in this series removes the unused dais. Hardik T Shah (1): ASoC: Intel: Skylake: Add helpers for SRC and converter modules Jeeja KP (12): ASoC: Intel: Skylake: Add helpers for DSP module configuration ASoC: Intel: Skylake: Add DSP module init and binding routines ASoC: Intel: Skylake: Add pipe management helpers ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Remove CPU dai that is not used sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-messages.c | 751 +++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 173 ++-- sound/soc/intel/skylake/skl-topology.c | 1264 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 307 +++++++ sound/soc/intel/skylake/skl-tplg-interface.h | 166 ++++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 8 files changed, 2636 insertions(+), 68 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c create mode 100644 sound/soc/intel/skylake/skl-topology.h create mode 100644 sound/soc/intel/skylake/skl-tplg-interface.h -- 2.4.3

2 17

[alsa-devel] [PATCH] ALSA: bebob: support Firewire I/O card of Mackie Onyx 1220/1620/1640
by Takashi Sakamoto 07 Oct '15

07 Oct '15

Current ALSA BeBoB drivers has an entry for this model, while the value of vendor ID seems to be wrong according to an user's report. The vendor had released no updated firmware, thus we can judge that this model had not changed the content of its config ROM. It's reasonable to fix the ID according to the report. $ ./linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 0425720f bus_info_length 4, crc_length 37, crc 29199 404 31333934 bus_name "1394" 408 f0646122 irmc 1, cmc 1, isc 1, bmc 1, pmc 0, cyc_clk_acc 100, max_rec 6 (128), max_rom 1, gen 2, spd 2 (S400) 40c 00000ff2 company_id 00000f | 410 00004697 device_id f200004697 | EUI-64 00000ff200004697 root directory ----------------------------------------------------------------- 414 000859be directory_length 8, crc 22974 418 04000082 hardware version 41c 0c0083c0 node capabilities per IEEE 1394 420 03000ff2 vendor 424 8100000a --> descriptor leaf at 44c 428 17010065 model 42c 8100000d --> descriptor leaf at 460 430 13000910 version 434 d1000001 --> unit directory at 438 unit directory at 438 ----------------------------------------------------------------- 438 0004ccec directory_length 4, crc 52460 43c 1200a02d specifier id: 1394 TA 440 13010001 version: AV/C 444 17010065 model 448 8100000d --> descriptor leaf at 47c descriptor leaf at 44c ----------------------------------------------------------------- 44c 0004152a leaf_length 4, crc 5418 450 00000000 textual descriptor 454 00000000 minimal ASCII 458 4d61636b "Mack" 45c 69650000 "ie" descriptor leaf at 460 ----------------------------------------------------------------- 460 000612b5 leaf_length 6, crc 4789 464 00000000 textual descriptor 468 00000000 minimal ASCII 46c 4f6e7978 "Onyx" 470 20466972 " Fir" 474 65776972 "ewir" 478 65000000 "e" descriptor leaf at 47c ----------------------------------------------------------------- 47c 000612b5 leaf_length 6, crc 4789 480 00000000 textual descriptor 484 00000000 minimal ASCII 488 4f6e7978 "Onyx" 48c 20466972 " Fir" 490 65776972 "ewir" 494 65000000 "e" $ cat /proc/asound/card3/firewire/firmware Manufacturer: bridgeCo Protocol Ver: 1 Build Ver: 0 GUID: 0x00000FF200004697 Model ID: 0x82 Model Rev: 1 Firmware Date: 20040430 Firmware Time: 131527 Firmware ID: 0x10065 Firmware Ver: 2320 Base Addr: 0x20080000 Max Size: 1572864 Loader Date: 20040430 Loader Time: 112036 Reported-by: Andrzej Gansiniec <andrzej(a)gansiniec.pl> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 27a04ac..0ed5e5f 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -41,7 +41,8 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_EDIROL 0x000040ab #define VEN_PRESONUS 0x00000a92 #define VEN_BRIDGECO 0x000007f5 -#define VEN_MACKIE 0x0000000f +#define VEN_MACKIE1 0x0000000f +#define VEN_MACKIE2 0x00000ff2 #define VEN_STANTON 0x00001260 #define VEN_TASCAM 0x0000022e #define VEN_BEHRINGER 0x00001564 @@ -360,9 +361,9 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* BridgeCo, Audio5 */ SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal), /* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */ - SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010065, &spec_normal), + SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal), /* Mackie, d.2 (Firewire Option) */ - SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010067, &spec_normal), + SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal), /* Stanton, ScratchAmp */ SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal), /* Tascam, IF-FW DM */ -- 2.1.4

3 3

[alsa-devel] Applied "ASoC: rsnd: Remove obsolete platform data support" to the asoc tree
by Mark Brown 07 Oct '15

07 Oct '15

The patch ASoC: rsnd: Remove obsolete platform data support has been applied to the asoc tree at git://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 dcc448e619194098b24477a6d56af50c57f26f1d Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven <geert+renesas(a)glider.be> Date: Wed, 7 Oct 2015 12:42:22 +0200 Subject: [PATCH] ASoC: rsnd: Remove obsolete platform data support Since commit 3d7608e4c169af03 ("ARM: shmobile: bockw: remove legacy board file and config"), Renesas R-Car SoCs are only supported in generic DT-only ARM multi-platform builds. The driver doesn't need to use platform data anymore, hence remove platform data configuration. Move <sound/rcar_snd.h> to sound/soc/sh/rcar/, as it's no longer needed by platform code. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/core.c | 19 +++++-------------- {include/sound => sound/soc/sh/rcar}/rcar_snd.h | 0 sound/soc/sh/rcar/rsnd.h | 3 ++- sound/soc/sh/rcar/ssi.c | 3 --- 4 files changed, 7 insertions(+), 18 deletions(-) rename {include/sound => sound/soc/sh/rcar}/rcar_snd.h (100%) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index eec294d..6ef9a88 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1236,20 +1236,11 @@ static int rsnd_probe(struct platform_device *pdev) }; int ret, i; - info = NULL; - of_data = NULL; - if (of_id) { - info = devm_kzalloc(&pdev->dev, - sizeof(struct rcar_snd_info), GFP_KERNEL); - of_data = of_id->data; - } else { - info = pdev->dev.platform_data; - } - - if (!info) { - dev_err(dev, "driver needs R-Car sound information\n"); - return -ENODEV; - } + info = devm_kzalloc(&pdev->dev, sizeof(struct rcar_snd_info), + GFP_KERNEL); + if (!info) + return -ENOMEM; + of_data = of_id->data; /* * init priv data diff --git a/include/sound/rcar_snd.h b/sound/soc/sh/rcar/rcar_snd.h similarity index 100% rename from include/sound/rcar_snd.h rename to sound/soc/sh/rcar/rcar_snd.h diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e4068d7..e9fef53 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -21,10 +21,11 @@ #include <linux/of_irq.h> #include <linux/sh_dma.h> #include <linux/workqueue.h> -#include <sound/rcar_snd.h> #include <sound/soc.h> #include <sound/pcm_params.h> +#include "rcar_snd.h" + /* * pseudo register * diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 5e05f942..842a35b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -700,9 +700,6 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, struct device *dev = &pdev->dev; int nr, i; - if (!of_data) - return; - node = rsnd_ssi_of_node(priv); if (!node) return; -- 2.5.0

1 0

[alsa-devel] Applied "ASoC: da7213: Add DT support to codec driver" to the asoc tree
by Mark Brown 07 Oct '15

07 Oct '15

The patch ASoC: da7213: Add DT support to codec driver has been applied to the asoc tree at git://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 e90996a3ea224fbeb459b8052ecd366d7990e1f3 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Wed, 7 Oct 2015 14:27:08 +0100 Subject: [PATCH] ASoC: da7213: Add DT support to codec driver This patch adds support for DT bindings in the codec driver. As part of this support, the mclk data can now be provided and used to control the mclk during codec operation. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7213.c | 123 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 118 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index a9c86ef..ab1486b 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1387,10 +1387,118 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, return 0; } +/* DT */ +static const struct of_device_id da7213_of_match[] = { + { .compatible = "dlg,da7213", }, + { } +}; +MODULE_DEVICE_TABLE(of, da7213_of_match); + +static enum da7213_micbias_voltage + da7213_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1600: + return DA7213_MICBIAS_1_6V; + case 2200: + return DA7213_MICBIAS_2_2V; + case 2500: + return DA7213_MICBIAS_2_5V; + case 3000: + return DA7213_MICBIAS_3_0V; + default: + dev_warn(codec->dev, "Invalid micbias level\n"); + return DA7213_MICBIAS_2_2V; + } +} + +static enum da7213_dmic_data_sel + da7213_of_dmic_data_sel(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "lrise_rfall")) { + return DA7213_DMIC_DATA_LRISE_RFALL; + } else if (!strcmp(str, "lfall_rrise")) { + return DA7213_DMIC_DATA_LFALL_RRISE; + } else { + dev_warn(codec->dev, "Invalid DMIC data select type\n"); + return DA7213_DMIC_DATA_LRISE_RFALL; + } +} + +static enum da7213_dmic_samplephase + da7213_of_dmic_samplephase(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "on_clkedge")) { + return DA7213_DMIC_SAMPLE_ON_CLKEDGE; + } else if (!strcmp(str, "between_clkedge")) { + return DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE; + } else { + dev_warn(codec->dev, "Invalid DMIC sample phase\n"); + return DA7213_DMIC_SAMPLE_ON_CLKEDGE; + } +} + +static enum da7213_dmic_clk_rate + da7213_of_dmic_clkrate(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1500000: + return DA7213_DMIC_CLK_1_5MHZ; + case 3000000: + return DA7213_DMIC_CLK_3_0MHZ; + default: + dev_warn(codec->dev, "Invalid DMIC clock rate\n"); + return DA7213_DMIC_CLK_1_5MHZ; + } +} + +static struct da7213_platform_data + *da7213_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct da7213_platform_data *pdata; + const char *of_str; + u32 of_val32; + + pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) { + dev_warn(codec->dev, "Failed to allocate memory for pdata\n"); + return NULL; + } + + if (of_property_read_u32(np, "dlg,micbias1-lvl", &of_val32) >= 0) + pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, of_val32); + else + pdata->micbias1_lvl = DA7213_MICBIAS_2_2V; + + if (of_property_read_u32(np, "dlg,micbias2-lvl", &of_val32) >= 0) + pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, of_val32); + else + pdata->micbias2_lvl = DA7213_MICBIAS_2_2V; + + if (!of_property_read_string(np, "dlg,dmic-data-sel", &of_str)) + pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, of_str); + else + pdata->dmic_data_sel = DA7213_DMIC_DATA_LRISE_RFALL; + + if (!of_property_read_string(np, "dlg,dmic-samplephase", &of_str)) + pdata->dmic_samplephase = + da7213_of_dmic_samplephase(codec, of_str); + else + pdata->dmic_samplephase = DA7213_DMIC_SAMPLE_ON_CLKEDGE; + + if (of_property_read_u32(np, "dlg,dmic-clkrate", &of_val32) >= 0) + pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, of_val32); + else + pdata->dmic_clk_rate = DA7213_DMIC_CLK_3_0MHZ; + + return pdata; +} + + static int da7213_probe(struct snd_soc_codec *codec) { struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); - struct da7213_platform_data *pdata = da7213->pdata; /* Default to using ALC auto offset calibration mode. */ snd_soc_update_bits(codec, DA7213_ALC_CTRL1, @@ -1450,8 +1558,15 @@ static int da7213_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, DA7213_LINE_CTRL, DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE); + /* Handle DT/Platform data */ + if (codec->dev->of_node) + da7213->pdata = da7213_of_to_pdata(codec); + else + da7213->pdata = dev_get_platdata(codec->dev); + /* Set platform data values */ if (da7213->pdata) { + struct da7213_platform_data *pdata = da7213->pdata; u8 micbias_lvl = 0, dmic_cfg = 0; /* Set Mic Bias voltages */ @@ -1507,6 +1622,7 @@ static int da7213_probe(struct snd_soc_codec *codec) /* Set MCLK squaring */ da7213->mclk_squarer_en = pdata->mclk_squaring; } + return 0; } @@ -1537,7 +1653,6 @@ static int da7213_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct da7213_priv *da7213; - struct da7213_platform_data *pdata = dev_get_platdata(&i2c->dev); int ret; da7213 = devm_kzalloc(&i2c->dev, sizeof(struct da7213_priv), @@ -1545,9 +1660,6 @@ static int da7213_i2c_probe(struct i2c_client *i2c, if (!da7213) return -ENOMEM; - if (pdata) - da7213->pdata = pdata; - i2c_set_clientdata(i2c, da7213); da7213->regmap = devm_regmap_init_i2c(i2c, &da7213_regmap_config); @@ -1582,6 +1694,7 @@ MODULE_DEVICE_TABLE(i2c, da7213_i2c_id); static struct i2c_driver da7213_i2c_driver = { .driver = { .name = "da7213", + .of_match_table = of_match_ptr(da7213_of_match), }, .probe = da7213_i2c_probe, .remove = da7213_remove, -- 2.5.0

1 0

[alsa-devel] Applied "ASoC: da7213: Add support to handle mclk data provided to driver" to the asoc tree
by Mark Brown 07 Oct '15

07 Oct '15

The patch ASoC: da7213: Add support to handle mclk data provided to driver has been applied to the asoc tree at git://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 6e7c444318699496e6e6f30c875cf67534aeccc6 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Wed, 7 Oct 2015 14:27:11 +0100 Subject: [PATCH] ASoC: da7213: Add support to handle mclk data provided to driver Driver now can make use of mclk data, if provided, to set, enable and disable the clock source. As part of this, the choice to enable clock squaring is dealt with as part of dai_sysclk() call rather than as platform data. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/da7213.h | 3 --- sound/soc/codecs/da7213.c | 67 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/da7213.h | 8 ++++-- 3 files changed, 62 insertions(+), 16 deletions(-) diff --git a/include/sound/da7213.h b/include/sound/da7213.h index 673f5c3..e7eac89 100644 --- a/include/sound/da7213.h +++ b/include/sound/da7213.h @@ -44,9 +44,6 @@ struct da7213_platform_data { enum da7213_dmic_data_sel dmic_data_sel; enum da7213_dmic_samplephase dmic_samplephase; enum da7213_dmic_clk_rate dmic_clk_rate; - - /* MCLK squaring config */ - bool mclk_squaring; }; #endif /* _DA7213_PDATA_H */ diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index ab1486b..7278f93 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -12,6 +12,7 @@ * option) any later version. */ +#include <linux/clk.h> #include <linux/delay.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -1222,23 +1223,44 @@ static int da7213_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if ((da7213->clk_src == clk_id) && (da7213->mclk_rate == freq)) + return 0; + + if (((freq < 5000000) && (freq != 32768)) || (freq > 54000000)) { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } switch (clk_id) { case DA7213_CLKSRC_MCLK: - if ((freq == 32768) || - ((freq >= 5000000) && (freq <= 54000000))) { - da7213->mclk_rate = freq; - return 0; - } else { - dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", - freq); - return -EINVAL; - } + da7213->mclk_squarer_en = false; + break; + case DA7213_CLKSRC_MCLK_SQR: + da7213->mclk_squarer_en = true; break; default: dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); return -EINVAL; } + + da7213->clk_src = clk_id; + + if (da7213->mclk) { + freq = clk_round_rate(da7213->mclk, freq); + ret = clk_set_rate(da7213->mclk, freq); + if (ret) { + dev_err(codec_dai->dev, "Failed to set clock rate %d\n", + freq); + return ret; + } + } + + da7213->mclk_rate = freq; + + return 0; } /* Supported PLL input frequencies are 5MHz - 54MHz. */ @@ -1366,12 +1388,25 @@ static struct snd_soc_dai_driver da7213_dai = { static int da7213_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* MCLK */ + if (da7213->mclk) { + ret = clk_prepare_enable(da7213->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable mclk\n"); + return ret; + } + } + /* Enable VMID reference & master bias */ snd_soc_update_bits(codec, DA7213_REFERENCES, DA7213_VMID_EN | DA7213_BIAS_EN, @@ -1382,6 +1417,10 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, /* Disable VMID reference & master bias */ snd_soc_update_bits(codec, DA7213_REFERENCES, DA7213_VMID_EN | DA7213_BIAS_EN, 0); + + /* MCLK */ + if (da7213->mclk) + clk_disable_unprepare(da7213->mclk); break; } return 0; @@ -1618,9 +1657,15 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_DMIC_DATA_SEL_MASK | DA7213_DMIC_SAMPLEPHASE_MASK | DA7213_DMIC_CLK_RATE_MASK, dmic_cfg); + } - /* Set MCLK squaring */ - da7213->mclk_squarer_en = pdata->mclk_squaring; + /* Check if MCLK provided */ + da7213->mclk = devm_clk_get(codec->dev, "mclk"); + if (IS_ERR(da7213->mclk)) { + if (PTR_ERR(da7213->mclk) != -ENOENT) + return PTR_ERR(da7213->mclk); + else + da7213->mclk = NULL; } return 0; diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 9cb9ddd..030fd69 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -13,6 +13,7 @@ #ifndef _DA7213_H #define _DA7213_H +#include <linux/clk.h> #include <linux/regmap.h> #include <sound/da7213.h> @@ -504,14 +505,17 @@ #define DA7213_PLL_INDIV_20_40_MHZ_VAL 8 #define DA7213_PLL_INDIV_40_54_MHZ_VAL 16 -enum clk_src { - DA7213_CLKSRC_MCLK +enum da7213_clk_src { + DA7213_CLKSRC_MCLK = 0, + DA7213_CLKSRC_MCLK_SQR, }; /* Codec private data */ struct da7213_priv { struct regmap *regmap; + struct clk *mclk; unsigned int mclk_rate; + int clk_src; bool master; bool mclk_squarer_en; bool srm_en; -- 2.5.0

1 0

[alsa-devel] [PATCH 0/2] ASoC: da7213: Add DT support for codec
by Adam Thomson 07 Oct '15

07 Oct '15

This patch set adds DT support to the codec driver, and supporting bindings documentation. Patch set is based against v4.3-rc4 Adam Thomson (2): ASoC: da7213: Add DT support to codec driver ASoC: da7213: Add bindings documentation for codec driver Documentation/devicetree/bindings/sound/da7213.txt | 41 +++++ sound/soc/codecs/da7213.c | 181 +++++++++++++++++++-- sound/soc/codecs/da7213.h | 8 +- 3 files changed, 212 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/da7213.txt -- 1.9.3

3 5

[alsa-devel] Applied "ASoC: rsnd: Remove obsolete platform data support" to the asoc tree
by Mark Brown 07 Oct '15

07 Oct '15

The patch ASoC: rsnd: Remove obsolete platform data support has been applied to the asoc tree at git://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 971a41f034ce46c82f3ab7abebe14c2d1bc54e17 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven <geert+renesas(a)glider.be> Date: Wed, 7 Oct 2015 12:42:22 +0200 Subject: [PATCH] ASoC: rsnd: Remove obsolete platform data support Since commit 3d7608e4c169af03 ("ARM: shmobile: bockw: remove legacy board file and config"), Renesas R-Car SoCs are only supported in generic DT-only ARM multi-platform builds. The driver doesn't need to use platform data anymore, hence remove platform data configuration. Move <sound/rcar_snd.h> to sound/soc/sh/rcar/, as it's no longer needed by platform code. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/core.c | 19 +++++-------------- {include/sound => sound/soc/sh/rcar}/rcar_snd.h | 0 sound/soc/sh/rcar/rsnd.h | 3 ++- sound/soc/sh/rcar/ssi.c | 3 --- 4 files changed, 7 insertions(+), 18 deletions(-) rename {include/sound => sound/soc/sh/rcar}/rcar_snd.h (100%) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f3feed5..00bf1ff 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1224,20 +1224,11 @@ static int rsnd_probe(struct platform_device *pdev) }; int ret, i; - info = NULL; - of_data = NULL; - if (of_id) { - info = devm_kzalloc(&pdev->dev, - sizeof(struct rcar_snd_info), GFP_KERNEL); - of_data = of_id->data; - } else { - info = pdev->dev.platform_data; - } - - if (!info) { - dev_err(dev, "driver needs R-Car sound information\n"); - return -ENODEV; - } + info = devm_kzalloc(&pdev->dev, sizeof(struct rcar_snd_info), + GFP_KERNEL); + if (!info) + return -ENOMEM; + of_data = of_id->data; /* * init priv data diff --git a/include/sound/rcar_snd.h b/sound/soc/sh/rcar/rcar_snd.h similarity index 100% rename from include/sound/rcar_snd.h rename to sound/soc/sh/rcar/rcar_snd.h diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 7a0e52b..f604f69 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -21,10 +21,11 @@ #include <linux/of_irq.h> #include <linux/sh_dma.h> #include <linux/workqueue.h> -#include <sound/rcar_snd.h> #include <sound/soc.h> #include <sound/pcm_params.h> +#include "rcar_snd.h" + /* * pseudo register * diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index d45b9a7..3f5f195 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -697,9 +697,6 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, struct device *dev = &pdev->dev; int nr, i; - if (!of_data) - return; - node = rsnd_ssi_of_node(priv); if (!node) return; -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: sun4i-codec: Remove the routing property
by Maxime Ripard 07 Oct '15

07 Oct '15

Most of the boards have their headphone jack directly connected to the matching pins of the SoCs. Since most of the time we will have the same routing path, it makes no sense to put that in the DTS, since it will only be some useless duplication there. It also fixes the following warning messages that were seen so far, on boards where we were using the bindings in the documentation example. sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Left -> direct -> Headphone Jack sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Right -> direct -> Headphone Jack Reported-by: Priit Laes <plaes(a)plaes.org> Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- Documentation/devicetree/bindings/sound/sun4i-codec.txt | 6 ------ sound/soc/sunxi/sun4i-codec.c | 7 ------- 2 files changed, 13 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt index 680144b74ae9..c92966bd5488 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt @@ -13,10 +13,6 @@ Required properties: - clock-names: should contain followings: - "apb": the parent APB clock for this controller - "codec": the parent module clock -- 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. - Example: codec: codec@01c22c00 { @@ -28,6 +24,4 @@ codec: codec@01c22c00 { clock-names = "apb", "codec"; dmas = <&dma 0 19>, <&dma 0 19>; dma-names = "rx", "tx"; - routing = "Headphone Jack", "HP Right", - "Headphone Jack", "HP Left"; }; diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 47780552dcd0..bcbf4da168b6 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -571,7 +571,6 @@ static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev, static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) { struct snd_soc_card *card; - int ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) @@ -584,12 +583,6 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) card->dev = dev; card->name = "sun4i-codec"; - ret = snd_soc_of_parse_audio_routing(card, "routing"); - if (ret) { - dev_err(dev, "Failed to create our audio routing\n"); - return NULL; - } - return card; }; -- 2.5.3

2 1

[alsa-devel] [PATCH 0/2] ASoC: da7219: Improve mclk error checking, use of_match_ptr
by Adam Thomson 07 Oct '15

07 Oct '15

This patch set contains a couple of small updates/fixes: - Use of of_match_ptr() for assigning match table to driver structure - Only enable/disable mclk if it has been provided to the driver. - Add error checking of clk_prepare_enable. Patch set is based on changes introduced under patch: 'ASoC: da7219: Improve error handling for regulator supplies' (Commit 4e929134eb8271abc9c52c371e056debfea6898b , kernel/git/broonie/sound.git) Adam Thomson (2): ASoC: da7219: Use of_match_ptr() when assigning match table ASoC: da7219: Improve error checking of mclk enable/disable sound/soc/codecs/da7219.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) -- 1.9.3

2 4

[alsa-devel] [PATCH 00/18] ALSA: hdac: move hdac helpers to core
by Vinod Koul 07 Oct '15

07 Oct '15

Hi Takashi, Here is the second version which updates the users. This update was done using coccinelle scripts. I also noticed that the coccinelle is reformaing the change lines, but changes looked good to me so I have kept them, please do let me know if you would like these to be removed. Subhransu S. Prusty (1): ALSA: hdac: Copy codec helpers to core Vinod Koul (17): ALSA: hda: move hda beep to use hdac helpers ALSA: hda: move hda_eld to use hdac helpers ALSA: hda: move hda_generic to use hdac helpers ALSA: hda: move hda_hwdep to use hdac helpers ALSA: hda: move hda_jack to use hdac helpers ALSA: hda: move hda_proc to use hdac helpers ALSA: hda: move patch_analog to use hdac helpers ALSA: hda: move patch_ca0132.c to use hdac helpers ALSA: hda: move patch_cirrus to use hdac helpers ALSA: hda: move to use hdac helpers ALSA: hda: move patch_hdmi to use hdac helpers ALSA: hda: move patch_realtek to use hdac helpers ALSA: hda: move patch_si3054 to use hdac helpers ALSA: hda: move patch_sigmatel to use hdac helpers ALSA: hda: move patch_via to use hdac helpers ALSA: hda: move hda_codec to use hdac helpers ALSA: hda: remove the old helpers include/sound/hdaudio.h | 6 + sound/hda/hdac_device.c | 81 ++++++++++++ sound/pci/hda/hda_beep.c | 4 +- sound/pci/hda/hda_codec.c | 128 +++++++----------- sound/pci/hda/hda_codec.h | 5 - sound/pci/hda/hda_eld.c | 65 +++++++--- sound/pci/hda/hda_generic.c | 9 +- sound/pci/hda/hda_hwdep.c | 4 +- sound/pci/hda/hda_jack.c | 8 +- sound/pci/hda/hda_local.h | 13 -- sound/pci/hda/hda_proc.c | 80 ++++++------ sound/pci/hda/patch_analog.c | 10 +- sound/pci/hda/patch_ca0132.c | 223 ++++++++++++++++---------------- sound/pci/hda/patch_cirrus.c | 52 ++++---- sound/pci/hda/patch_conexant.c | 20 +-- sound/pci/hda/patch_hdmi.c | 285 +++++++++++++++++++++-------------------- sound/pci/hda/patch_realtek.c | 136 +++++++++++--------- sound/pci/hda/patch_si3054.c | 10 +- sound/pci/hda/patch_sigmatel.c | 78 +++++------ sound/pci/hda/patch_via.c | 14 +- 20 files changed, 660 insertions(+), 571 deletions(-) -- 2.4.3

3 20

[alsa-devel] [PATCH] ASoC: rsnd: Remove obsolete platform data support
by Geert Uytterhoeven 07 Oct '15

07 Oct '15

Since commit 3d7608e4c169af03 ("ARM: shmobile: bockw: remove legacy board file and config"), Renesas R-Car SoCs are only supported in generic DT-only ARM multi-platform builds. The driver doesn't need to use platform data anymore, hence remove platform data configuration. Move <sound/rcar_snd.h> to sound/soc/sh/rcar/, as it's no longer needed by platform code. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Commit 3d7608e4c169af03 is now in arm-soc/for-next. --- sound/soc/sh/rcar/core.c | 19 +++++-------------- {include/sound => sound/soc/sh/rcar}/rcar_snd.h | 0 sound/soc/sh/rcar/rsnd.h | 3 ++- sound/soc/sh/rcar/ssi.c | 3 --- 4 files changed, 7 insertions(+), 18 deletions(-) rename {include/sound => sound/soc/sh/rcar}/rcar_snd.h (100%) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index eec294da81e3..6ef9a884ca7c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1236,20 +1236,11 @@ static int rsnd_probe(struct platform_device *pdev) }; int ret, i; - info = NULL; - of_data = NULL; - if (of_id) { - info = devm_kzalloc(&pdev->dev, - sizeof(struct rcar_snd_info), GFP_KERNEL); - of_data = of_id->data; - } else { - info = pdev->dev.platform_data; - } - - if (!info) { - dev_err(dev, "driver needs R-Car sound information\n"); - return -ENODEV; - } + info = devm_kzalloc(&pdev->dev, sizeof(struct rcar_snd_info), + GFP_KERNEL); + if (!info) + return -ENOMEM; + of_data = of_id->data; /* * init priv data diff --git a/include/sound/rcar_snd.h b/sound/soc/sh/rcar/rcar_snd.h similarity index 100% rename from include/sound/rcar_snd.h rename to sound/soc/sh/rcar/rcar_snd.h diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e4068d78616c..e9fef53968b4 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -21,10 +21,11 @@ #include <linux/of_irq.h> #include <linux/sh_dma.h> #include <linux/workqueue.h> -#include <sound/rcar_snd.h> #include <sound/soc.h> #include <sound/pcm_params.h> +#include "rcar_snd.h" + /* * pseudo register * diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 5e05f9422073..842a35b1363a 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -700,9 +700,6 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, struct device *dev = &pdev->dev; int nr, i; - if (!of_data) - return; - node = rsnd_ssi_of_node(priv); if (!node) return; -- 1.9.1

1 0

[alsa-devel] Applied "ASoC: Intel: use dw_dmac autoconfiguration" to the asoc tree
by Mark Brown 07 Oct '15

07 Oct '15

The patch ASoC: Intel: use dw_dmac autoconfiguration has been applied to the asoc tree at git://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 b5e5a4549c8b88a880fa3866fa3803ea9396ba03 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Date: Mon, 5 Oct 2015 14:53:50 +0300 Subject: [PATCH] ASoC: Intel: use dw_dmac autoconfiguration Instead of hardconding a platform data for dw_dmac let's use it's own autoconfiguration feature. Thus, remove hardcoded values. Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/common/sst-firmware.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index ebcca6d..1636a1e 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -26,7 +26,6 @@ #include <linux/acpi.h> /* supported DMA engine drivers */ -#include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> #include <asm/page.h> @@ -169,12 +168,6 @@ err: return ret; } -static struct dw_dma_platform_data dw_pdata = { - .is_private = 1, - .chan_allocation_order = CHAN_ALLOCATION_ASCENDING, - .chan_priority = CHAN_PRIORITY_ASCENDING, -}; - static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem, int irq) { @@ -195,7 +188,8 @@ static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem, return ERR_PTR(err); chip->dev = dev; - err = dw_dma_probe(chip, &dw_pdata); + + err = dw_dma_probe(chip, NULL); if (err) return ERR_PTR(err); -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: nau8825: Add driver for headset chip Nuvoton 8825
by Anatol Pomozov 07 Oct '15

07 Oct '15

Sponsored-by: Google Chromium project Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/nau8825.txt | 102 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 1107 ++++++++++++++++++++ sound/soc/codecs/nau8825.h | 323 ++++++ 5 files changed, 1538 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 0000000..d337423 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,102 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-threshold-num: Number of buttons supported + - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) + where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold calculation. + + - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. + + - nuvoton,sar-voltage: Reference voltage for button impedance measurement. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-compare-time: SAR compare time + 0 - 500 ns + 1 - 1 us + 2 - 2 us + 3 - 4 us + + - nuvoton,sar-sampling-time: SAR sampling time + 0 - 2 us + 1 - 4 us + 2 - 8 us + 3 - 16 us + + - nuvoton,short-key-debounce: Button short key press debounce time. + 0 - 30 ms + 1 - 50 ms + 2 - 100 ms + 3 - 30 ms + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the + clocks described in clock-names + - clock-names: should include "mclk" for the MCLK master clock + +Example: + + headset: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,sar-compare-time = <0>; + nuvoton,sar-sampling-time = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + + clock-names = "mclk"; + clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 70e5a75..e1103e2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 @@ -897,6 +898,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be1491a..02284b9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -73,6 +73,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o @@ -265,6 +266,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..f31a500 --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,1107 @@ +/* + * Nuvoton NAU8825 audio codec driver + * + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * Copyright 2015 Nuvoton Technology Corp. + * Co-author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/clk.h> + +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + + +#include "nau8825.h" + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static int nau8825_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Prevent startup click by letting charge pump to ramp up */ + msleep(10); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char * const nau8825_adc_decimation[] = { + "32", "64", "128", "256" +}; + +static const struct soc_enum nau8825_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8825_REG_ADC_RATE, NAU8825_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8825_adc_decimation), nau8825_adc_decimation); + +static const char * const nau8825_dac_oversampl[] = { + "64", "256", "128", "", "32" +}; + +static const struct soc_enum nau8825_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8825_dac_oversampl), nau8825_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); +static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -9600, 2400); + +static const struct snd_kcontrol_new nau8825_controls[] = { + SOC_SINGLE_TLV("Mic Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Headphone Bypass Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("Headphone Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Volume", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), + SOC_DOUBLE_TLV("Headphone Crosstalk Volume", NAU8825_REG_DAC_DGAIN_CTRL, + 0, 8, 0xff, 0, crosstalk_vol_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum), +}; + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, + 15, 1), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, + 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_pump_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_PGA("Output Driver R Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver R Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"ADC", NULL, "ADC Power"}, + {"AIFTX", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver L Stage 1", NULL, "ADACL"}, + {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"Output DACL", NULL, "Output Driver L Stage 3"}, + {"Output DACR", NULL, "Output Driver R Stage 3"}, + {"HPOL", NULL, "Output DACL"}, + {"HPOR", NULL, "Output DACR"}, + {"HPOL", NULL, "Charge Pump"}, + {"HPOR", NULL, "Charge Pump"}, +}; + +static int nau8825_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 nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + nau8825->jack = jack; + + /* Ground HP Outputs[1:0], needed for headset auto detection + * Enable Automatic Mic/Gnd switching reading on insert interrupt[6] + */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons = 0; + + /* The chip supports up to 8 buttons, but ALSA defines only 6 buttons */ + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + if (value & BIT(4)) + buttons |= SND_JACK_BTN_4; + if (value & BIT(5)) + buttons |= SND_JACK_BTN_5; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type = 0; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + if (type & SND_JACK_HEADPHONE) { + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + * lower 8 bits - for long pressed buttons + */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_TRACKING_GAIN_MASK, + nau8825->sar_voltage << NAU8825_SAR_TRACKING_GAIN_SFT); + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_COMPARE_TIME_MASK, + nau8825->sar_compare_time << NAU8825_SAR_COMPARE_TIME_SFT); + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_SAMPLING_TIME_MASK, + nau8825->sar_sampling_time << NAU8825_SAR_SAMPLING_TIME_SFT); + + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_LEVELS_NR_MASK, + (nau8825->sar_threshold_num - 1) << NAU8825_KEYDET_LEVELS_NR_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_HYSTERESIS_MASK, + nau8825->sar_hysteresis << NAU8825_KEYDET_HYSTERESIS_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK, + nau8825->key_debounce << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT); + + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, + (nau8825->sar_threshold[0] << 8) | nau8825->sar_threshold[1]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, + (nau8825->sar_threshold[2] << 8) | nau8825->sar_threshold[3]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_3, + (nau8825->sar_threshold[4] << 8) | nau8825->sar_threshold[5]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_4, + (nau8825->sar_threshold[6] << 8) | nau8825->sar_threshold[7]); + + /* Enable short press and release interruptions */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_KEY_SHORT_PRESS_EN | NAU8825_IRQ_KEY_RELEASE_EN, + 0); +} + +static void nau8825_init_regs(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Enable Bias/Vmid */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_GLOBAL_BIAS_EN, NAU8825_GLOBAL_BIAS_EN); + + /* VMID Tieoff */ + regmap_update_bits(regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID_SEL_MASK, + nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT); + /* Disable Boost Driver, Automatic Short circuit protection enable */ + regmap_update_bits(regmap, NAU8825_REG_BOOST, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN); + + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_OUTPUT_EN, + nau8825->jkdet_enable ? 0 : NAU8825_JKDET_OUTPUT_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_EN, + nau8825->jkdet_pull_enable ? 0 : NAU8825_JKDET_PULL_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_UP, + nau8825->jkdet_pull_up ? NAU8825_JKDET_PULL_UP : 0); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_POLARITY, + /* jkdet_polarity - 1 is for active-low */ + nau8825->jkdet_polarity ? 0 : NAU8825_JACK_POLARITY); + + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_INSERT_DEBOUNCE_MASK, + nau8825->jack_insert_debounce << NAU8825_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_EJECT_DEBOUNCE_MASK, + nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT); + + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff); + + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_VOLTAGE_MASK, nau8825->micbias_voltage); + + if (nau8825->sar_threshold_num) + nau8825_setup_buttons(nau8825); + + /* Default oversampling/decimations settings are unusable + * (audible hiss). Set it to something better. + */ + regmap_update_bits(regmap, NAU8825_REG_ADC_RATE, + NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128); + regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, + NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + + nau8825->dapm = dapm; + + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. + */ + snd_soc_dapm_force_enable_pin(dapm, "DDACR"); + snd_soc_dapm_sync(dapm); + + /* Unmask interruptions. Handler uses dapm object so we can enable + * interruptions only after dapm is fully initialized. + */ + regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, + unsigned int freq) +{ + struct regmap *regmap = nau8825->regmap; + int ret; + + switch (clk_id) { + case NAU8825_CLK_MCLK: + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + break; + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + return ret; + } + } + + break; + case NAU8825_CLK_INTERNAL: + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, + NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + default: + dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8825->dev, "Sysclk is %dHz and clock id is %d\n", freq, + clk_id); + return 0; +} + +static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + return nau8825_configure_sysclk(nau8825, clk_id, freq); +} + +static int nau8825_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + if (nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + ret = regcache_sync(nau8825->regmap); + if (ret) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + break; + + case SND_SOC_BIAS_OFF: + if (nau8825->mclk_freq) + clk_disable_unprepare(nau8825->mclk); + + regcache_mark_dirty(nau8825->regmap); + break; + } + return 0; +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + .set_sysclk = nau8825_set_sysclk, + .set_bias_level = nau8825_set_bias_level, + .suspend_bias_off = true, + + .controls = nau8825_controls, + .num_controls = ARRAY_SIZE(nau8825_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_read_device_properties(struct device *dev, + struct nau8825 *nau8825) { + + nau8825->jkdet_enable = device_property_read_bool(dev, + "nuvoton,jkdet-enable"); + nau8825->jkdet_pull_enable = device_property_read_bool(dev, + "nuvoton,jkdet-pull-enable"); + nau8825->jkdet_pull_up = device_property_read_bool(dev, + "nuvoton,jkdet-pull-up"); + device_property_read_u32(dev, "nuvoton,jkdet-polarity", + &nau8825->jkdet_polarity); + device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8825->micbias_voltage); + device_property_read_u32(dev, "nuvoton,vref-impedance", + &nau8825->vref_impedance); + device_property_read_u32(dev, "nuvoton,sar-threshold-num", + &nau8825->sar_threshold_num); + device_property_read_u32_array(dev, "nuvoton,sar-threshold", + nau8825->sar_threshold, nau8825->sar_threshold_num); + device_property_read_u32(dev, "nuvoton,sar-hysteresis", + &nau8825->sar_hysteresis); + device_property_read_u32(dev, "nuvoton,sar-voltage", + &nau8825->sar_voltage); + device_property_read_u32(dev, "nuvoton,sar-compare-time", + &nau8825->sar_compare_time); + device_property_read_u32(dev, "nuvoton,sar-sampling-time", + &nau8825->sar_sampling_time); + device_property_read_u32(dev, "nuvoton,short-key-debounce", + &nau8825->key_debounce); + device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8825->jack_insert_debounce); + device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8825->jack_eject_debounce); + + nau8825->mclk = devm_clk_get(dev, "mclk"); + if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { + return -EPROBE_DEFER; + } else if (PTR_ERR(nau8825->mclk) == -ENOENT) { + /* The MCLK is managed externally or not used at all */ + nau8825->mclk = NULL; + dev_info(dev, "No 'mclk' clock found, assume MCLK is managed externally"); + } else if (IS_ERR(nau8825->mclk)) { + return -EINVAL; + } + + return 0; +} + +static int nau8825_setup_irq(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + int ret; + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable DDACR needed for interrupts + * It is the same as force_enable_pin("DDACR") we do later + */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(nau8825->dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + + return 0; +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825 = dev_get_platdata(&i2c->dev); + int ret, value; + + if (!nau8825) { + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + ret = nau8825_read_device_properties(dev, nau8825); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->dev = dev; + nau8825->irq = i2c->irq; + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + nau8825_init_regs(nau8825); + + if (i2c->irq) + nau8825_setup_irq(nau8825); + + return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; + +#ifdef CONFIG_OF +static const struct of_device_id nau8825_of_ids[] = { + { .compatible = "nuvoton,nau8825", }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8825_of_ids); +#endif + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(nau8825_of_ids), + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..8774923 --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,323 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov(a)chrominium.org> + * + * 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 __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* FLL6 (0x9) */ +#define NAU8825_DCO_EN (1 << 15) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +/* 0 - short to GND, 1 - open */ +#define NAU8825_SPKR_DWN1R (1 << 1) +#define NAU8825_SPKR_DWN1L (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8825_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_EJECT_DEBOUNCE_SFT) +#define NAU8825_JACK_POLARITY (1 << 1) /* 0 - active low, 1 - active high */ + +/* INTERRUPT_MASK (0xf) */ +#define NAU8825_IRQ_OUTPUT_EN (1 << 11) +#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) +#define NAU8825_IRQ_EJECT_EN (1 << 2) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* INTERRUPT_DIS_CTRL (0x12) */ +#define NAU8825_IRQ_HEADSET_COMPLETE_DIS (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) +#define NAU8825_IRQ_EJECT_DIS (1 << 2) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) +#define NAU8825_SAR_TRACKING_GAIN_SFT 8 +#define NAU8825_SAR_TRACKING_GAIN_MASK (0x7 << NAU8825_SAR_TRACKING_GAIN_SFT) +#define NAU8825_SAR_COMPARE_TIME_SFT 2 +#define NAU8825_SAR_COMPARE_TIME_MASK (3 << 2) +#define NAU8825_SAR_SAMPLING_TIME_SFT 0 +#define NAU8825_SAR_SAMPLING_TIME_MASK (3 << 0) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT 12 +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK (0x3 << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT) +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) +#define NAU8825_KEYDET_HYSTERESIS_SFT 0 +#define NAU8825_KEYDET_HYSTERESIS_MASK 0xf + +/* GPIO12_CTRL (0x1a) */ +#define NAU8825_JKDET_PULL_UP (1 << 11) /* 0 - pull down, 1 - pull up */ +#define NAU8825_JKDET_PULL_EN (1 << 9) /* 0 - enable pull, 1 - disable */ +#define NAU8825_JKDET_OUTPUT_EN (1 << 8) /* 0 - enable input, 1 - enable output */ + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - normal mode, 1 - Hi-Z output */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SYNC_DOWN_SFT 0 +#define NAU8825_ADC_SYNC_DOWN_MASK 0x3 +#define NAU8825_ADC_SYNC_DOWN_32 0 +#define NAU8825_ADC_SYNC_DOWN_64 1 +#define NAU8825_ADC_SYNC_DOWN_128 2 +#define NAU8825_ADC_SYNC_DOWN_256 3 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_SFT 0 +#define NAU8825_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8825_DAC_OVERSAMPLE_64 0 +#define NAU8825_DAC_OVERSAMPLE_256 1 +#define NAU8825_DAC_OVERSAMPLE_128 2 +#define NAU8825_DAC_OVERSAMPLE_32 4 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_SFT 4 +#define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 +#define NAU8825_MICBIAS_VOLTAGE_SFT 0 +#define NAU8825_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_G_DIS (1 << 8) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_INTEGR_R (1 << 5) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) + +/* CHARGE_PUMP (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + struct snd_soc_jack *jack; + struct clk *mclk; + int irq; + int mclk_freq; /* 0 - mclk is disabled */ + int button_pressed; + int micbias_voltage; + int vref_impedance; + bool jkdet_enable; + bool jkdet_pull_enable; + bool jkdet_pull_up; + int jkdet_polarity; + int sar_threshold_num; + int sar_threshold[8]; + int sar_hysteresis; + int sar_voltage; + int sar_compare_time; + int sar_sampling_time; + int key_debounce; + int jack_insert_debounce; + int jack_eject_debounce; +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- 2.6.0.rc2.230.g3dd15c0

3 5

[alsa-devel] [PATCH v2 0/8] ASoC: topology: ABI - Add support for PCM, BE & CC link
by mengdong.lin＠intel.com 07 Oct '15

07 Oct '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This series includes topology ABI update to support PCM (FE DAI/DAI links), BE and CC DAI links. Bump the ABI version to 4. Mengdong Lin (2): ASoC: topology: ABI - Change stream formats to a bitwise flag ASoC: topology: ABI - Remove tdm_slot & dai_fmt from snd_soc_tplg_stream Vedang Patel (6): ASoC: topology: ABI - Add PCM Support and bump ABI version to 4 ASoC: topology: ABI - Remove unused struct snd_soc_tplg_stream_config ASoC: topology: ABI - Use __le32 instead of __u32 in snd_soc_tplg_dapm_widget ASoC: topology: ABI - Add name element to snd_soc_tplg_stream ASoC: topology: ABI - Add the type for BE DAI link ASoC: topology: ABI - Add configuration for BE & Codec-Codec DAI Links include/uapi/sound/asoc.h | 76 +++++++++++++++++++++++++---------------------- sound/soc/soc-topology.c | 4 +-- 2 files changed, 43 insertions(+), 37 deletions(-) -- 1.9.1

7 21

[alsa-devel] [PATCH RFC V2 0/5] another generic audio hdmi codec proposal
by Arnaud Pouliquen 07 Oct '15

07 Oct '15

Version 2: This version integrates missing features upgraded to be aligned when possible with patch set: [PATCH RFC v4 0/8] Implement generic ASoC HDMI codec and use it in tda998x It does not support the abort callback, the mute and syclk ops. This could be added in V3 or separate patch if justified. [RFC 1/5]video: hdmi: add help function for N and CTS - Replace real frequency value by enum in switch case Notice that this patch is independant from the rest of the implentation [RFC 2/5]drm: add helper functions to add audio capabilities for bridge - extend audio bridge ops [RFC 3/5]ASoC: codec: hdmi drm codec driver aligned when possible with hdmi-codec.c implementation . ELD management for alsa contraints . Support of SPDIF and I2S mode . hwparam ond set_fmt functions implemented [RFC 4/5]drm: sti: connect audio driver => Implementation for STI platform [RFC 5/5]DT: sti: add audio HDMI dai link in audio card => Devicetree example Version 1: This patch set is a tentative to implement a generic code for the HDMI audio. Main concept are aligned with solution proposeded for TI platform. - ASoC codec driver registered by DRM driver - ASOC driver is a generic driver. - compatible with simple card Difference is that i propose a DRM generic interface based on bridge structure. Advantage is that all data exchanges are done through the DRM API. I think also that some helper functions could been used for N and CTS parameters calculation, as suggested by Russell King in a previous mail. I full aware that some features (like ELD and info frame) are partially or not implemented in my patches. This patch set is more a skeleton than a full implementation... I just post it to suggest a possible DRM API. Arnaud Pouliquen (5): video: hdmi: add helper function for N and CTS drm: add helper function to add audio capabilities for bridge ASoC: codec: hdmi drm codec driver drm: sti: connect audio driver DT: sti: add audio HDMI dai link in audio card arch/arm/boot/dts/stih410.dtsi | 6 +- arch/arm/boot/dts/stihxxx-b2120.dtsi | 21 ++ drivers/gpu/drm/drm_bridge.c | 137 +++++++++++++- drivers/gpu/drm/sti/sti_hdmi.c | 184 ++++++++++++++++-- drivers/gpu/drm/sti/sti_hdmi.h | 3 + drivers/video/hdmi.c | 147 ++++++++++++++ include/drm/drm_crtc.h | 62 ++++++ include/drm/drm_modes.h | 12 ++ include/linux/hdmi.h | 12 ++ include/sound/hdmi_drm.h | 15 ++ sound/soc/codecs/Kconfig | 8 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi_drm.c | 358 +++++++++++++++++++++++++++++++++++ 13 files changed, 951 insertions(+), 16 deletions(-) create mode 100644 include/sound/hdmi_drm.h create mode 100644 sound/soc/codecs/hdmi_drm.c -- 1.9.1

6 16

[alsa-devel] Zoom R16/24 playback slient
by Ricard Wanderlof 07 Oct '15

07 Oct '15

Bringing up an old thread from 2014 since I've gotten access to a Zoom R16 and am keen on getting it to work in full duplex under Linux. Apparently (not yet verified here yet, rebuilding a kernel with the patch applied at the moment), 8 channel capture works with the quirk, but playback doesn't (well, it appears to work, but no sound comes out) and neither does the mixer. > At Sun, 30 Nov 2014 12:08:38 +0200, > Panu Matilainen wrote: > > > > On 11/29/2014 09:39 PM, Takashi Iwai wrote: > > > At Sat, 29 Nov 2014 21:35:04 +0200, > > > Panu Matilainen wrote: > > >> > > >> This makes the midi interface and capture work out of the box with > > >> R16 (and presumably R24 too but untested). Playback stream would also > > >> seem to function fine except for one caveat: no sound is produced, > > >> so it is disabled for now. > > > > [ ... ] > > > > In the meanwhile I started looking at the mixer - this is just guessing > > but maybe its starting with outputs muted which could explain the > > playback stream running but without sound. > > > > The descriptors for interface 0 are garbage but what is there would seem > > to map to logical elements by subtype and terminaltype, after those the> > data doesn't seem to add up (eg number of input/output channels > > missing/wrongly placed etc): > > > > ** UNRECOGNIZED: 0b 24 01 00 01 35 00 03 01 02 03 > > -> Header? > > ** UNRECOGNIZED: 0c 24 02 05 01 01 00 02 03 00 00 00 > > -> USB streaming input terminal? > > ** UNRECOGNIZED: 09 24 03 08 01 03 00 05 00 > > -> Speaker output terminal? > > ** UNRECOGNIZED: 0c 24 02 09 01 02 00 08 00 00 00 00 > > -> Microphone input terminal? > > ** UNRECOGNIZED: 09 24 03 0c 01 01 00 09 00 > > -> USB streaming output terminal? > > > > Anything there that might ring a bell, or other ideas where to start > > poking at this thing? > > Yeah, most of these look almost sane, so any off-by-one firmware > issue? Searching the archives, I could find no further posts on the subject of the R16. Panu, did you do any more work on this? Where does the above output come from, and where could I find information on what type of data is expected? I assume what's expected is something one would get from an USB mixer device? In an earlier incarnation of a patch for the R16 (on the 10th of March 2014), there's a discussion between Jason Mancine and Takashi regarding the .format field in the quirks table (http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074117.html) which didn't seem to reach a conclusion; the problem then seemed to be that even when setting the .format to SNDRV_PCM_FMTBIT_S24_3LE, ALSA still setup up a 32 bit transfer. However, Panu's patch also added a small delay in snd_usb_ctl_msg_quirk(), so that perhaps was the real solution to the alleged format problem? (The final patch is much smaller than in this particular thread, possibly due to the use of QUIRK_ANY_INTERFACE since kernel version >3.11 ?) /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 17

[alsa-devel] [PATCH] ASoC: ad193x-spi: Fixup ad193x_spi_id table
by Axel Lin 07 Oct '15

07 Oct '15

AD1939 is missed from the table, so add it. AD1936 and AD1937 are controlled by I2C interface, so remove them. Fixes: e5224f58e3ef ("ASoC: ad193x: add support to ad1934") Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ad193x-spi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 364e8211..b55055a 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -39,9 +39,8 @@ static const struct spi_device_id ad193x_spi_id[] = { { "ad193x", AD193X }, { "ad1933", AD1933 }, { "ad1934", AD1934 }, - { "ad1936", AD193X }, - { "ad1937", AD193X }, { "ad1938", AD193X }, + { "ad1939", AD193X }, { } }; MODULE_DEVICE_TABLE(spi, ad193x_spi_id); -- 2.1.4

3 2

Re: [alsa-devel] [PATCH 4/5] ALSA: firewire-lib: add throttle for MIDI data rate
by Takashi Sakamoto 07 Oct '15

07 Oct '15

On Oct 7 2015 09:32, kbuild test robot wrote: > Hi Takashi, > > [auto build test ERROR on v4.3-rc4 -- if it's inappropriate base, please ignore] > > config: i386-randconfig-x009-201540 (attached as .config) > reproduce: > # save the attached .config to linux build tree > make ARCH=i386 > > Note: the linux-review/Takashi-Sakamoto/ALSA-add-helper-functions-for-MIDI-message-transmission-on-IEEE-1394-asynchronous-transaction HEAD f058371b5831baa9749177001cbbdcdf47753f3d builds fine. > It only hurts bisectibility. > > All errors (new ones prefixed by >>): > > sound/firewire/lib.c: In function 'snd_fw_async_midi_port_init': >>> sound/firewire/lib.c:181:6: error: 'struct snd_fw_async_midi_port' has no member named 'error' > port->error = false; > ^ > > vim +181 sound/firewire/lib.c > > 175 > 176 port->parent = fw_parent_device(unit); > 177 port->addr = addr; > 178 port->fill = fill; > 179 port->idling = true; > 180 port->next_ktime = ktime_set(0, 0); > > 181 port->error = false; > 182 > 183 INIT_WORK(&port->work, midi_port_work); > 184 Oops! I had wrong rebase operations to generate this patch. This line should be in the next patch, sorry... Regards Takashi Sakamoto

1 0

[alsa-devel] [PATCH 0/5] ALSA: add helper functions for MIDI message transmission on IEEE 1394 asynchronous transaction
by Takashi Sakamoto 07 Oct '15

07 Oct '15

Hi, This patchset updates a part of my former post (patch 03-07): [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html This patchset adds some helper functions for MIDI transmission via IEEE 1394 asynchronous transaction. Changes: * Use workqueue instead of tasklet for better usage of system resources. * Use ktime_t instead of jiffies for better resolution of time counting. * Some minor fixes. Takashi Sakamoto (5): ALSA: firewire-lib: add helper functions for asynchronous transactions to transfer MIDI messages ALSA: firewire-lib: add a restriction for a transaction at once ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error sound/firewire/lib.c | 138 +++++++++++++++++++++++++++++++++++++++++++++++++++ sound/firewire/lib.h | 56 +++++++++++++++++++++ 2 files changed, 194 insertions(+) -- 2.1.4

1 5

[alsa-devel] Supported (microphone!) external usb sound card?
by Frank Hartung 06 Oct '15

06 Oct '15

Hey guys, I bought the "ASUS Xonar U7" USB sound card about 2 years ago. When i switched from "Windows" to "Linux Mint Cinnamon", i noticed that ALSA wouldn't use the microphone in of my card. So I tried another external sound card this time, the "Creative Sound Blaster X-fi HD". Connected to my laptop (also computer) and having updated my Linux Mint Cinnamon to the version 17.1, ALSA wouldn't use the microphone in either! Audio out (headphone) is working fine for both. ------ Is there ANY (decent) external (usb) sound card supported by ALSA (headset AND microphone)? Please notice that I am really not a specialist in Linux or ALSA. Regards Frank

1 0

[alsa-devel] [PATCH 0/5] Add ASoC support for AMD APUs [v4]
by Alex Deucher 06 Oct '15

06 Oct '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create mfd cells which we hang the i2s codec and DMA engine on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review V3 changes: - Update the designware driver to handle slave mode, amd specific features - Use the designware driver directly for i2s - Move the DMA handling from the GPU driver into the AMD ASoC driver - Change the license on the ASoC driver to GPL V4 changes: - get the master vs slave mode for the designware codec from registers - copy designware quirks from platform data in local struct Patch 4 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream4 Thanks, Alex Maruthi Bayyavarapu (1): drm/amd: add ACP driver support Maruthi Srinivas Bayyavarapu (4): ASoC : dwc : support dw i2s in slave mode ASoC : dwc : support dw i2s in AMD platform ASoC : AMD : add ACP 2.2 register headers ASoC: AMD: add AMD ASoC ACP-I2S driver drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 10 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 127 ++ drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 269 +++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 41 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + include/linux/mfd/amd_acp.h | 43 + include/sound/designware_i2s.h | 5 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 4 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 518 ++++++ sound/soc/amd/acp.c | 736 +++++++++ sound/soc/amd/acp.h | 147 ++ sound/soc/amd/include/acp_2_2_d.h | 609 +++++++ sound/soc/amd/include/acp_2_2_enum.h | 1068 ++++++++++++ sound/soc/amd/include/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++++++++ sound/soc/dwc/designware_i2s.c | 276 ++-- 24 files changed, 6146 insertions(+), 102 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 include/linux/mfd/amd_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp.c create mode 100644 sound/soc/amd/acp.h create mode 100644 sound/soc/amd/include/acp_2_2_d.h create mode 100644 sound/soc/amd/include/acp_2_2_enum.h create mode 100644 sound/soc/amd/include/acp_2_2_sh_mask.h -- 1.8.3.1

3 8

[alsa-devel] [PATCH] ASoC: codecs: tas2552: fix dBscale-min declaration
by Andreas Dannenberg 06 Oct '15

06 Oct '15

The minimum volume level for the TAS2552 (control register value 0x00) is -7dB however the driver declares it as -0.07dB. Running amixer before the patch reports: dBscale-min=-0.07dB,step=1.00dB,mute=0 Running amixer with the patch applied reports: dBscale-min=-7.00dB,step=1.00dB,mute=0 Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> --- sound/soc/codecs/tas2552.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index e3a0bca..cc1d398 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = { /* * DAC digital volumes. From -7 to 24 dB in 1 dB steps */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0); static const char * const tas2552_din_source_select[] = { "Muted", -- 1.9.1

2 1

[alsa-devel] Applied "MAINTAINERS: Remove wm97xx entry" to the asoc tree
by Mark Brown 06 Oct '15

06 Oct '15

The patch MAINTAINERS: Remove wm97xx entry has been applied to the asoc tree at git://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 ff6afc8d0df0347901e862e622b1efaeb6f1fe5d Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Fri, 2 Oct 2015 18:07:09 +0100 Subject: [PATCH] MAINTAINERS: Remove wm97xx entry Neither myself or Liam is especially interested in this driver any more and the devices are already covered by the general ex-Wolfson entry so just remove this. Signed-off-by: Mark Brown <broonie(a)kernel.org> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- MAINTAINERS | 9 --------- 1 file changed, 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 0581f47..07790df 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -11363,15 +11363,6 @@ W: http://oops.ghostprotocols.net:81/blog S: Maintained F: drivers/net/wireless/wl3501* -WM97XX TOUCHSCREEN DRIVERS -M: Mark Brown <broonie(a)kernel.org> -M: Liam Girdwood <lrg(a)slimlogic.co.uk> -L: linux-input(a)vger.kernel.org -W: https://github.com/CirrusLogic/linux-drivers/wiki -S: Supported -F: drivers/input/touchscreen/*wm97* -F: include/linux/wm97xx.h - WOLFSON MICROELECTRONICS DRIVERS L: patches(a)opensource.wolfsonmicro.com T: git https://github.com/CirrusLogic/linux-drivers.git -- 2.5.0

1 0

[alsa-devel] [PATCH RFC] ASoC: simple-card: Update clocks binding for simple-card DAI subnodes
by Jyri Sarha 06 Oct '15

06 Oct '15

The updated binding provides a way to set clock-ID and direction parameters for DAI drivers set_sysclk() call back. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- I proposed something similar about a year ago, but Mark rejected that at the time. This RFC is to start that discussion again. This time before I do any code changes. Best regards, Jyri Documentation/devicetree/bindings/sound/simple-card.txt | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index cf3979e..d10bf2d 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -76,6 +76,7 @@ Optional CPU/CODEC subnodes properties: - dai-tdm-slot-num : Please refer to tdm-slot.txt. - dai-tdm-slot-width : Please refer to tdm-slot.txt. - clocks / system-clock-frequency : specify subnode's clock if needed. + it can be specified via "clocks" if system has clock node (= common clock), or "system-clock-frequency" (if system doens't support common clock) @@ -83,7 +84,21 @@ Optional CPU/CODEC subnodes properties: enabled with clk_prepare_enable() in dai startup() and disabled with clk_disable_unprepare() in dai - shutdown(). + shutdown(). "system-clock-frequency" + can also be an array if more than one + clock is described. +- clock-ids : An array of clock ID integers, + preferrably defined in DT header. + Each entry corresponds to the same + index postion first in "clocks" and + after the end of clocks array to + "system-clock-frequency" array. +- clock-dirs : An array of integers describing + clock directions: CLK_IN (= 0) or + OUT (= 1). Entries in the array + refer to clocks in the same way as + in clock-ids property. + Example 1 - single DAI link: -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: rockchip: namespace rockchip i2s module name
by Sjoerd Simons 06 Oct '15

06 Oct '15

Change the rockchip i2s object name (and thus module name) from the rather generic snd-soc-i2s to the more specific snd-soc-rockchip-i2s Signed-off-by: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> --- sound/soc/rockchip/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 1bc1dc3..e9ba558 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,7 +1,7 @@ # ROCKCHIP Platform Support -snd-soc-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-i2s-objs := rockchip_i2s.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o -- 2.6.0

2 1

[alsa-devel] Applied "ASoC: tas2552: fix dBscale-min declaration" to the asoc tree
by Mark Brown 06 Oct '15

06 Oct '15

The patch ASoC: tas2552: fix dBscale-min declaration has been applied to the asoc tree at git://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 e2600460bc3aa14ca1df86318a327cbbabedf9a8 Mon Sep 17 00:00:00 2001 From: Andreas Dannenberg <dannenberg(a)ti.com> Date: Mon, 5 Oct 2015 15:00:14 -0500 Subject: [PATCH] ASoC: tas2552: fix dBscale-min declaration The minimum volume level for the TAS2552 (control register value 0x00) is -7dB however the driver declares it as -0.07dB. Running amixer before the patch reports: dBscale-min=-0.07dB,step=1.00dB,mute=0 Running amixer with the patch applied reports: dBscale-min=-7.00dB,step=1.00dB,mute=0 Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/tas2552.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index e3a0bca..cc1d398 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = { /* * DAC digital volumes. From -7 to 24 dB in 1 dB steps */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0); static const char * const tas2552_din_source_select[] = { "Muted", -- 2.5.0

1 0

[alsa-devel] [PATCH v2] ASoC: rt5645: Add the HW EQ for the customized speaker output of Google Celes
by Oder Chiou 06 Oct '15

06 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5645.h | 1 + sound/soc/codecs/rt5645.c | 70 ++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5645.h | 5 ++++ 3 files changed, 75 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf615..cf944e4 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,7 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Invert JD when jack insert */ bool jd_invert; + unsigned int eq_mode; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c4fe6b..def9caa 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -224,6 +224,36 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5647_eq_parameter { + unsigned int reg; + unsigned int val; +}; + +#define EQ_REG_NUM 56 +struct hweq_s { + struct rt5647_eq_parameter par[EQ_REG_NUM]; + unsigned int ctrl; +}; + +static const struct hweq_s hweq_param[] = { + { /* EQ_NONE */ + { + {0, 0}, + }, + 0x0000, + }, + { /* EQ_GOOGLE_CELES */ + { + {0x1c0, 0x04b8}, + {0x1c1, 0xfd02}, + {0x1c2, 0x04b8}, + {0x1c3, 0xfd02}, + {0, 0}, + }, + 0x0020, + }, +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -631,6 +661,27 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_update_eqmode(struct snd_soc_codec *codec, + unsigned int mode) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int i; + + for (i = 0; i < EQ_REG_NUM; i++) { + if (hweq_param[mode].par[i].reg) + regmap_write(rt5645->regmap, + hweq_param[mode].par[i].reg, + hweq_param[mode].par[i].val); + else + break; + } + + snd_soc_update_bits(codec, RT5645_EQ_CTRL2, 0x33fe, + hweq_param[mode].ctrl); + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1532,9 +1583,11 @@ static int rt5645_spk_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 rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: + rt5645_update_eqmode(codec, rt5645->pdata.eq_mode); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1596,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + rt5645_update_eqmode(codec, EQ_NONE); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3205,6 +3259,20 @@ static int strago_quirk_cb(const struct dmi_system_id *id) return 1; } +static struct rt5645_platform_data celes_platform_data = { + .dmic1_data_pin = RT5645_DMIC1_DISABLE, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, + .eq_mode = EQ_GOOGLE_CELES, +}; + +static int celes_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &celes_platform_data; + + return 1; +} + static const struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", @@ -3215,7 +3283,7 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { }, { .ident = "Google Celes", - .callback = strago_quirk_cb, + .callback = celes_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 13ac732..89f25b8 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2187,6 +2187,11 @@ enum { RT5645_AD_MONO_R_FILTER = (0x1 << 5), }; +enum { + EQ_NONE, + EQ_GOOGLE_CELES, +}; + int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec, unsigned int filter_mask, unsigned int clk_src); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: Document DAI signal polarity
by Anatol Pomozov 06 Oct '15

06 Oct '15

Currently there is no clear definition of what is FSYNC polarity. Different drivers use its own definition of what is "normal" and what is "inverted" fsync. This leads to compatibility problems between drivers. For example TegraX1 driver assumes that DSP-A format with frames starting at rising FSYNC edge has "negative" polarity, while RT5677 assumes it is "positive" polarity. Explicitly specify meaning of BCLK/FSYNC polarity to avoid future compatibility problems. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- include/sound/soc-dai.h | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..45198db 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,22 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * - * Specifies whether the DAI can also support inverted clocks for the specified - * format. + * BCLK: + * - "normal" polarity means signal is available at rising edge of BCLK + * - "inverted" polarity means signal is available at falling edge of BCLK + * + * FSYNC "normal" polarity depends on the frame format: + * - I2S: frame consists of left then right channel data. Left channel starts + * with falling FSYNC edge, right channel starts with rising FSYNC edge. + * - I2S right/left justified: frame consists of left then right channel data. + * Left channel starts with rising FSYNC edge, right channel starts with + * falling FSYNC edge. + * - DSP A/B: Frame starts with rising FSYNC edge. + * - AC97: Frame starts with rising FSYNC edge. + * + * "Negative" FSYNC polarity is the one opposite of "normal" polarity. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- 2.6.0.rc2.230.g3dd15c0

3 4

[alsa-devel] [PATCH alsa-lib] hwdep: add FireWire digi00x/tascam driver support
by Takashi Sakamoto 05 Oct '15

05 Oct '15

Linux 4.4 newly support these drivers. Their hardware dependent interfaces give functionalities to get hardware information, to lock/unlock kernel streaming and to receive lock status event. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/hwdep.h | 4 +++- include/sound/asound.h | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/include/hwdep.h b/include/hwdep.h index 3d3c31b..2fe78cd 100644 --- a/include/hwdep.h +++ b/include/hwdep.h @@ -74,8 +74,10 @@ typedef enum _snd_hwdep_iface { SND_HWDEP_IFACE_FW_FIREWORKS, /**< Echo Audio Fireworks based device */ SND_HWDEP_IFACE_FW_BEBOB, /**< BridgeCo BeBoB based device */ SND_HWDEP_IFACE_FW_OXFW, /**< Oxford OXFW970/971 based device */ + SND_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ + SND_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */ - SND_HWDEP_IFACE_LAST = SND_HWDEP_IFACE_FW_OXFW /**< last known hwdep interface */ + SND_HWDEP_IFACE_LAST = SND_HWDEP_IFACE_FW_TASCAM /**< last known hwdep interface */ } snd_hwdep_iface_t; /** open for reading */ diff --git a/include/sound/asound.h b/include/sound/asound.h index a45be6b..a82108e 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -100,9 +100,11 @@ enum { SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */ SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */ + SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ + SNDRV_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_OXFW + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_TASCAM }; struct snd_hwdep_info { -- 2.1.4

2 1

[alsa-devel] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered
by Rick Mann 05 Oct '15

05 Oct '15

UPDATE: I was able to read the McASP registers (turns out, its clocks are only enabled while speaker-test is running, so that's the only time the registers are mapped). AHCLKXCTL is 0x00008000, which corresponds to having it set as an output, not inverted, and clock divide-by-1. This seems wrong. I'm having trouble determining what the AUXCLK that feeds the McASP is, but it's not 12 MHz, and without division, I don't see how to get there. For reference, here are the actual values of the McASP transmit registers: GBLCTL 0x00000000 Lots of stuff held in reset (red flag?) XGBLCTL 0x00000000 Same XMASK 0x0000FFFF I think this masks out 16-bits? XFMT 0x00008074 MSB bit first, slot size 16-bit, rot right 16 bits AFSXCTL 0x00000100 2-slot TDM (I2S mode) ACLKXCTL 0x00000000 Sync xmit, ACLKX is from pin AHCLKXCTL 0x00008000 Internal high-freq clock from clock divider XTDM 0x00000003 2 TDM slots active XINTCTL 0x00000000 XSTAT 0x0000010C Xmit error occurred, cur TDM slot is even, xmit clock failure XSLOT 0x0000017F Current time slot (shouldn't this be 0 or 1?) XCLKCHK 0x00000000 XEVTCTL 0x00000000 Thanks, Rick ---- I've been at this for weeks and I'm still unable to get linux to work correctly with my TI codec. This is on a Beaglebone Black running a 4.1.4 kernel (later versions are worse). I've tried multiple versions of the kernel trying to narrow it down. I keep coming back to 4.1.4 as the most recent that at least generates some audio clocks. The AM335x mcasp0 should be speaking i2s with the tlv320aic3104. I have my own build from source of the BBB 4.1.4-ti-r9 release (mainline plus TI plus Beaglebone stuff). I've sprinkled some logging around the audio drivers, but so far I don't see much, and I really don't understand how all the pieces fit together. One thing I found with my logging is the mcasp MCLK (AHCLKXDIV) is never set. I have logging calls in __davinci_mcasp_set_clkdiv() (sound/soc/davinci/davinci-mcasp.c), which is the only code I could find that sets the McASP clock, and it's never called. I tried to use devmem2 to read the registers in the SoC to see if they were set correctly, but I just get a bus error (other reads do work). Errors in dmesg when I load the device tree overlay for my cape suggest I still don't have something configured correctly (see below). I'd sure appreciate any insight into these latest issues. Thanks! Symptoms: MCLK should be 12 MHz, but is 24.00 MHz. Buffer underflows when running speaker-test. Errors to console upon loading cape: [ 38.502299] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.517917] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.530425] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.536996] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.564279] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT dmesg upon loading cape overlay (https://github.com/JetForMe/podtique/blob/v1/bbb/cape/Podtique1/BB-ENABLE-P…): [ 568.323587] bone_capemgr bone_capemgr: part_number 'BB-ENABLE-PRU', version 'N/A' [ 568.323629] bone_capemgr bone_capemgr: slot #4: override [ 568.323646] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 568.323663] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-ENABLE-PRU' [ 568.323804] bone_capemgr:capemgr_load_slot:1148: bone_capemgr bone_capemgr: slot #4: Requesting part number/version based 'BB-ENABLE-PRU-00A0.dtbo [ 568.323825] bone_capemgr:capemgr_load_slot:1162: bone_capemgr bone_capemgr: slot #4: Requesting firmware 'BB-ENABLE-PRU-00A0.dtbo' for board-name 'Override Board Name', version '00A0' [ 568.330369] bone_capemgr:capemgr_load_slot:1171: bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; converting to live tree [ 568.341685] gpio-of-helper ocp:gpio_helper: ready [ 568.344695] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; overlay id #0 [ 568.387640] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.387695] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.387712] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.394284] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.410560] snd_soc_core:snd_soc_register_codec:3064: tlv320aic3x-codec 2-0018: codec register 2-0018 [ 568.410614] snd_soc_core:snd_soc_register_dais:2590: tlv320aic3x-codec 2-0018: ASoC: dai register 2-0018 #1 [ 568.410633] snd_soc_core:snd_soc_register_dais:2634: tlv320aic3x-codec 2-0018: ASoC: Registered DAI 'tlv320aic3x-hifi' [ 568.410653] snd_soc_core:snd_soc_register_codec:3138: tlv320aic3x-codec 2-0018: ASoC: Registered codec 'tlv320aic3x-codec.2-0018' [ 568.421529] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.421588] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.421605] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.431455] **** RLM davinci-mcasp.c:1734: davinci_mcasp_probe [ 568.431507] base:of_property_match_string:1398: comparing tx with tx [ 568.431544] base:of_property_match_string:1398: comparing rx with tx [ 568.431555] base:of_property_match_string:1398: comparing rx with rx [ 568.431608] base:of_property_match_string:1398: comparing common with tx [ 568.431621] base:of_property_match_string:1398: comparing common with rx [ 568.431633] base:of_property_match_string:1398: comparing rx with tx [ 568.431643] base:of_property_match_string:1398: comparing rx with rx [ 568.434643] base:of_property_match_string:1398: comparing tx with tx [ 568.437971] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.446651] snd_soc_core:snd_soc_register_dais:2590: davinci-mcasp 48038000.mcasp: ASoC: dai register 48038000.mcasp #1 [ 568.446682] snd_soc_core:snd_soc_register_dais:2634: davinci-mcasp 48038000.mcasp: ASoC: Registered DAI '48038000.mcasp' [ 568.446840] snd_soc_core:snd_soc_add_platform:2892: davinci-mcasp 48038000.mcasp: ASoC: Registered platform '48038000.mcasp' [ 568.450046] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.450100] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.450425] snd_soc_core:snd_soc_dapm_new_dai_widgets:3494: tlv320aic3x-codec 2-0018: ASoC: adding Playback widget [ 568.450448] snd_soc_core:snd_soc_dapm_new_dai_widgets:3513: tlv320aic3x-codec 2-0018: ASoC: adding Capture widget [ 568.450995] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -2 [ 568.451015] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -1 [ 568.451030] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 0 [ 568.451045] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 1 [ 568.451059] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 2 [ 568.451099] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT [ 568.465530] snd_soc_core:soc_new_pcm:2510: davinci_evm ocp:sound: ASoC: registered pcm #0 AIC3X tlv320aic3x-hifi-0 [ 568.466155] davinci_evm ocp:sound: tlv320aic3x-hifi <-> 48038000.mcasp mapping ok [ 568.466185] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Right ADC -> Capture [ 568.466206] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Left ADC -> Capture [ 568.466226] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Right DAC [ 568.466241] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Left DAC upon starting speaker-test: [ 572.786071] snd_soc_core:soc_pcm_open:535: ASoC: tlv320aic3x-hifi <-> 48038000.mcasp info: [ 572.786103] snd_soc_core:soc_pcm_open:536: ASoC: rate mask 0x7fe [ 572.786114] snd_soc_core:soc_pcm_open:538: ASoC: min ch 2 max ch 2 [ 572.786126] snd_soc_core:soc_pcm_open:540: ASoC: min rate 8000 max rate 96000 [ 572.841514] davinci-mcasp 48038000.mcasp: Transmit buffer underflow ...repeated [ 574.087162] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 579.135283] snd_soc_core:close_delayed_work:602: TLV320AIC3X: ASoC: pop wq checking: Playback status: inactive waiting: yes -- Rick Mann rmann(a)latencyzero.com

2 2

[alsa-devel] [asoc:topic/fsl-mega-fast 4/4] sound/soc/fsl/fsl_esai.c:903:5: warning: "CONFIG_PM_SLEEP" is not defined
by kbuild test robot 05 Oct '15

05 Oct '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/fsl-mega-fast head: c64c60763b4e3c72a3520c8d51be858cd67bacb5 commit: c64c60763b4e3c72a3520c8d51be858cd67bacb5 [4/4] ASoC: fsl_esai: Add driver suspend and resume to support MEGA Fast config: x86_64-randconfig-x015-201540 (attached as .config) reproduce: git checkout c64c60763b4e3c72a3520c8d51be858cd67bacb5 # save the attached .config to linux build tree make ARCH=x86_64 All warnings (new ones prefixed by >>): >> sound/soc/fsl/fsl_esai.c:903:5: warning: "CONFIG_PM_SLEEP" is not defined [-Wundef] #if CONFIG_PM_SLEEP ^ vim +/CONFIG_PM_SLEEP +903 sound/soc/fsl/fsl_esai.c 887 } 888 889 ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); 890 if (ret) 891 dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); 892 893 return ret; 894 } 895 896 static const struct of_device_id fsl_esai_dt_ids[] = { 897 { .compatible = "fsl,imx35-esai", }, 898 { .compatible = "fsl,vf610-esai", }, 899 {} 900 }; 901 MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); 902 > 903 #if CONFIG_PM_SLEEP 904 static int fsl_esai_suspend(struct device *dev) 905 { 906 struct fsl_esai *esai = dev_get_drvdata(dev); 907 908 regcache_cache_only(esai->regmap, true); 909 regcache_mark_dirty(esai->regmap); 910 911 return 0; --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 1/4] ASoC: imx-spdif: add snd_soc_pm_ops for spdif machine driver
by Zidan Wang 05 Oct '15

05 Oct '15

Add snd_soc_pm_ops in machine driver to make the trigger suspend/resume be called in suspend/resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/imx-spdif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 33da26a..a407e83 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -89,6 +89,7 @@ MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); static struct platform_driver imx_spdif_driver = { .driver = { .name = "imx-spdif", + .pm = &snd_soc_pm_ops, .of_match_table = imx_spdif_dt_ids, }, .probe = imx_spdif_audio_probe, -- 1.9.1

3 9

[alsa-devel] [PATCH v3 1/4] ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 05 Oct '15

05 Oct '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..5c737f1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -802,10 +805,40 @@ static const struct of_device_id fsl_sai_ids[] = { { /* sentinel */ } }; +#if CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; -- 1.9.1

4 15

[alsa-devel] [PATCH linux-next v3 1/1] ASoC: ad193x: add support to ad1934
by Cyrille Pitchen 05 Oct '15

05 Oct '15

The AD1934 codec has no DAC feature. Hence it register mapping is slightly different from the register mapping of other members of the AD193x family. Some ASoC controls and widgets are related to the DAC feature so are not relevant in the case of an AD1934 codec. Signed-off-by: Cyrille Pitchen <cyrille.pitchen(a)atmel.com> --- Hi all, This patch adds support to the AD1934 codec, which is DAC only. Best Regards, Cyrille ChangeLog v3: - rebase patch for linux-next: don't use the CODEC's dapm field, which no longer exists, but call snd_soc_codec_get_dapm() instead. v2: - replace AD193x by AD193X in ad193x-i2c.c - add ad193x_ prefix to adc_audio_paths --- sound/soc/codecs/ad193x-i2c.c | 8 +-- sound/soc/codecs/ad193x-spi.c | 16 +++++- sound/soc/codecs/ad193x.c | 122 ++++++++++++++++++++++++++++++++---------- sound/soc/codecs/ad193x.h | 9 +++- 4 files changed, 121 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c index df3a1a415825..171313664bc8 100644 --- a/sound/soc/codecs/ad193x-i2c.c +++ b/sound/soc/codecs/ad193x-i2c.c @@ -15,8 +15,8 @@ #include "ad193x.h" static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, { } }; MODULE_DEVICE_TABLE(i2c, ad193x_id); @@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client, config.val_bits = 8; config.reg_bits = 8; - return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); + return ad193x_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 390cef9b9dc2..364e82112e8c 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -16,6 +16,7 @@ static int ad193x_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct regmap_config config; config = ad193x_regmap_config; @@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi) config.read_flag_mask = 0x09; config.write_flag_mask = 0x08; - return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_spi_remove(struct spi_device *spi) @@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id ad193x_spi_id[] = { + { "ad193x", AD193X }, + { "ad1933", AD1933 }, + { "ad1934", AD1934 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, + { "ad1938", AD193X }, + { } +}; +MODULE_DEVICE_TABLE(spi, ad193x_spi_id); + static struct spi_driver ad193x_spi_driver = { .driver = { .name = "ad193x", @@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = { }, .probe = ad193x_spi_probe, .remove = ad193x_spi_remove, + .id_table = ad193x_spi_id, }; module_spi_driver(ad193x_spi_driver); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 17c953595660..76d7966bd72c 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,6 +23,7 @@ /* codec private data */ struct ad193x_priv { struct regmap *regmap; + enum ad193x_type type; int sysclk; }; @@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, - AD193X_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, - AD193X_ADCR2_MUTE, 1, 1), - /* DAC switch control */ SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, AD193X_DACR1_MUTE, 1, 1), @@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, AD193X_DACR4_MUTE, 1, 1), + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = { + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, AD193X_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_INPUT("ADC1IN"), SND_SOC_DAPM_INPUT("ADC2IN"), }; @@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, { "DAC Output", NULL, "DAC" }, { "DAC Output", NULL, "VMID" }, - { "ADC", NULL, "SYSCLK" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC Output" }, { "DAC2OUT", NULL, "DAC Output" }, { "DAC3OUT", NULL, "DAC Output" }, { "DAC4OUT", NULL, "DAC Output" }, + { "SYSCLK", NULL, "PLL_PWR" }, +}; + +static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = { + { "ADC", NULL, "SYSCLK" }, + { "ADC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, - { "SYSCLK", NULL, "PLL_PWR" }, }; +static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) +{ + switch (ad193x->type) { + case AD1933: + case AD1934: + return false; + default: + break; + } + + return true; +} + /* * DAI ops entries */ @@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: - return -EINVAL; + if (ad193x_has_adc(ad193x)) + return -EINVAL; + break; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_SERFMT_MASK, adc_serfmt); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_FMT_MASK, adc_fmt); + if (ad193x_has_adc(ad193x)) { + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + } regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, dac_fmt); @@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_WORD_LEN_MASK, word_len); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -326,6 +354,8 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int num, ret; /* default setting for ad193x */ @@ -335,14 +365,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* dac in tdm mode */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* high-pass filter enable */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + } + /* pll input: mclki/xi */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* add adc controls */ + num = ARRAY_SIZE(ad193x_adc_snd_controls); + ret = snd_soc_add_codec_controls(codec, + ad193x_adc_snd_controls, + num); + if (ret) + return ret; + + /* add adc widgets */ + num = ARRAY_SIZE(ad193x_adc_widgets); + ret = snd_soc_dapm_new_controls(dapm, + ad193x_adc_widgets, + num); + if (ret) + return ret; + + /* add adc routes */ + num = ARRAY_SIZE(ad193x_adc_audio_paths); + ret = snd_soc_dapm_add_routes(dapm, + ad193x_adc_audio_paths, + num); + if (ret) + return ret; + } + return 0; } @@ -367,7 +429,8 @@ const struct regmap_config ad193x_regmap_config = { }; EXPORT_SYMBOL_GPL(ad193x_regmap_config); -int ad193x_probe(struct device *dev, struct regmap *regmap) +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type) { struct ad193x_priv *ad193x; @@ -379,6 +442,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; ad193x->regmap = regmap; + ad193x->type = type; dev_set_drvdata(dev, ad193x); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index ab9a998f15be..8b1e65f928d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -13,8 +13,15 @@ struct device; +enum ad193x_type { + AD193X, + AD1933, + AD1934, +}; + extern const struct regmap_config ad193x_regmap_config; -int ad193x_probe(struct device *dev, struct regmap *regmap); +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type); #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -- 1.8.2.2

3 2

[alsa-devel] [PATCH 0/4] ALSA: hdac: fixes and support for hda codecs
by Vinod Koul 05 Oct '15

05 Oct '15

We had two fixes found on testing. First one checks if stream is in use or not before releasing and second fixs wrong stread id map Then we add structure defination for hdac_ext_dma_params and lastly add helpers for read/write for codecs Jeeja KP (1): ALSA: hdac: Fix to check if stream not in use in release Subhransu S. Prusty (3): ALSA: hdac: Fix incorrect update of stream id mapping ALSA: hdac: structure definition for ext_dma_params ALSA: hdac: Add codec read/write and check power state for widgets include/sound/hdaudio.h | 6 ++++++ include/sound/hdaudio_ext.h | 6 ++++++ sound/hda/ext/hdac_ext_stream.c | 9 ++++---- sound/hda/hdac_device.c | 47 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 63 insertions(+), 5 deletions(-) -- 2.4.3

3 9

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Prevent the weird sound of the headphone while rebooting
by Oder Chiou 05 Oct '15

05 Oct '15

The patch adds the codec reset setting in the shutdown function to prevent the weird sound of the headphone happened by rebooting. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index f436f50..4c4fe6b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3517,6 +3517,8 @@ static void rt5645_i2c_shutdown(struct i2c_client *i2c) RT5645_CBJ_MN_JD); regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, 0); + msleep(20); + regmap_write(rt5645->regmap, RT5645_RESET, 0); } static struct i2c_driver rt5645_i2c_driver = { -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] Applied "ASoC: da7219: Improve error handling for regulator supplies" to the asoc tree
by Mark Brown 05 Oct '15

05 Oct '15

The patch ASoC: da7219: Improve error handling for regulator supplies has been applied to the asoc tree at git://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 4e929134eb8271abc9c52c371e056debfea6898b Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Mon, 5 Oct 2015 15:10:04 +0100 Subject: [PATCH] ASoC: da7219: Improve error handling for regulator supplies Currently if bulk_enable() of supplies fails, the code still goes on to try and put the device into active state, and set expected IO voltage of the device. This doesn't really make sense so code now returns on bulk_enable() failure, with an error message. Also, to help with debug, failure to get supplies also provides an error message. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7219.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 76f8fc2..c86a833 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1554,8 +1554,10 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES, da7219->supplies); - if (ret) + if (ret) { + dev_err(codec->dev, "Failed to get supplies"); return ret; + } /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; @@ -1567,6 +1569,10 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) /* Enable main supplies */ ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies"); + return ret; + } /* Ensure device in active mode */ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); @@ -1574,7 +1580,7 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) /* Update IO voltage level range */ snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl); - return ret; + return 0; } static void da7219_handle_pdata(struct snd_soc_codec *codec) -- 2.5.0

1 0

[alsa-devel] sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0'
by Michael Shell 05 Oct '15

05 Oct '15

Greetings, I recently upgraded (an admittedly ancient K6-2 PCI/ISA machine) to kernel version 4.2.2 (compiled from source). It went surprising well, but I did run into a minor issue which did not occur under the older kernel 3.16.3. Maybe the info here will help someone else out in the future or even reveal a bug that might have greater implications that have not yet been seen. The ancient machine has one of those ISA SoundBlaster cards whose drivers are compiled as modules (to allow isapnp to operate) before module loading. Upon issuing a /sbin/modprobe snd-sbawe port=0x220 mpu_port=0x330 awe_port=0x620 irq=5 dma8=1 dma16=5 mic_agc=1 csp=0 dmesg (under 4.2.2) spits out: EMU8000 [0x620]: 4096 KiB on-board DRAM detected ------------[ cut here ]------------ WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80() sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0' Modules linked in: snd_emu8000_synth(+) snd_emux_synth snd_seq_virmidi snd_util_mem snd_opl3_synth snd_seq_midi_emul snd_sbawe snd_opl3_lib snd_sb16_dsp snd_sb16_csp snd_sb_common snd_mpu401_uart lp parport_pc parport CPU: 0 PID: 2791 Comm: modprobe Not tainted 4.2.2 #1 . . ---- (full output with trace at the end of this message) The /sys/devices/isa/sbawe.0 directory did not exist prior to the attempt to load snd-sbawe and there are no other sound drivers in the system other than those for the Soundblaster. I do still have OSS emulation selected in the kernel config - I was surpised that still is the recommended default. I probably could avoid this bug by deselecting that. I doubt I have any software that still requires OSS emulation, which is probably going to be removed from future kernels anyway (and Debian systems don't even carry it anymore). Nevertheless, a bug is a bug and this problem should not happen. I think snd_seq_oss and snd_emux_synth are both trying to register the same sysfs name or a single driver is trying to register the same thing twice. For the record, what is the correct procedure to find out exactly which drivers are in conflict in duplicate sysfs filename cases like this? Cheers and thanks in advance, Mike Shell ---- EMU8000 [0x620]: 4096 KiB on-board DRAM detected ------------[ cut here ]------------ WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80() sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0' Modules linked in: snd_emu8000_synth(+) snd_emux_synth snd_seq_virmidi snd_util_mem snd_opl3_synth snd_seq_midi_emul snd_sbawe snd_opl3_lib snd_sb16_dsp snd_sb16_csp snd_sb_common snd_mpu401_uart lp parport_pc parport CPU: 0 PID: 2791 Comm: modprobe Not tainted 4.2.2 #1 c1604e78 c1031c40 c175c544 dd6dfc48 00000ae7 c1758c38 0000001f c1103391 c1103391 df493000 dd6ef2c0 dd6ec6b0 dd6bfe9c c1031d13 00000009 dd6dfc30 c175c544 dd6dfc48 c1103391 c1758c38 0000001f c175c544 df493fdf dd6ef2c0 Call Trace: [<c1604e78>] ? dump_stack+0xa/0x13 [<c1031c40>] ? warn_slowpath_common+0x80/0xc0 [<c1103391>] ? sysfs_warn_dup+0x51/0x80 [<c1103391>] ? sysfs_warn_dup+0x51/0x80 [<c1031d13>] ? warn_slowpath_fmt+0x33/0x40 [<c1103391>] ? sysfs_warn_dup+0x51/0x80 [<c110346b>] ? sysfs_create_dir_ns+0x6b/0x80 [<c11c156c>] ? kobject_add_internal+0x6c/0x2e0 [<c11c86cb>] ? vsnprintf+0xcb/0x400 [<c11c1956>] ? kobject_add+0x36/0x80 [<c134de21>] ? device_add+0xc1/0x500 [<c1441669>] ? snd_device_new+0x29/0xa0 [<c1460a56>] ? snd_seq_device_new+0xb6/0x100 [<c146096c>] ? snd_seq_device_dev_register+0xc/0x40 [<c1441604>] ? snd_device_register+0x44/0x80 [<e0bd1fc4>] ? snd_emux_init_seq_oss+0x84/0xa0 [snd_emux_synth] [<e0bcd1e3>] ? snd_emux_register+0x103/0x140 [snd_emux_synth] [<e0bcd000>] ? 0xe0bcd000 [<e0bcd020>] ? sf_sample_new+0x20/0x20 [snd_emux_synth] [<e0bcd040>] ? sf_sample_free+0x20/0x20 [snd_emux_synth] [<e0bdf113>] ? snd_emu8000_probe+0xd3/0x180 [snd_emu8000_synth] [<c11036dd>] ? sysfs_create_link+0x1d/0x40 [<c13504fb>] ? driver_probe_device+0x15b/0x220 [<c1350631>] ? __driver_attach+0x71/0x80 [<c13505c0>] ? driver_probe_device+0x220/0x220 [<c134f000>] ? bus_for_each_dev+0x40/0x80 [<c134ff7a>] ? driver_attach+0x1a/0x20 [<c13505c0>] ? driver_probe_device+0x220/0x220 [<c134fcf4>] ? bus_add_driver+0xd4/0x200 [<c1350b91>] ? driver_register+0x51/0xe0 [<e0be3000>] ? 0xe0be3000 [<e0be3000>] ? 0xe0be3000 [<c10003f2>] ? do_one_initcall+0x72/0x1a0 [<e0be3000>] ? 0xe0be3000 [<c14ac646>] ? netlink_broadcast_filtered+0x166/0x3c0 [<c106c4a5>] ? do_init_module+0x45/0x200 [<c10a8425>] ? __vunmap+0x85/0xe0 [<c106dfaa>] ? load_module+0x186a/0x1ea0 [<c106e6aa>] ? SyS_init_module+0xca/0x100 [<c160a2dd>] ? syscall_call+0x7/0x7 ---[ end trace 826ff8353957cad3 ]--- ------------[ cut here ]------------ WARNING: CPU: 0 PID: 2791 at lib/kobject.c:240 kobject_add_internal+0x252/0x2e0() kobject_add_internal failed for seq-oss-0-0 with -EEXIST, don't try to register things with the same name in the same directory. Modules linked in: snd_emu8000_synth(+) snd_emux_synth snd_seq_virmidi snd_util_mem snd_opl3_synth snd_seq_midi_emul snd_sbawe snd_opl3_lib snd_sb16_dsp snd_sb16_csp snd_sb_common snd_mpu401_uart lp parport_pc parport CPU: 0 PID: 2791 Comm: modprobe Tainted: G W 4.2.2 #1 c1604e78 c1031c40 c1768d78 dd6dfc84 00000ae7 c1768e8e 000000f0 c11c1752 c11c1752 dd66c274 00000000 dd66c274 dd6bfe9c c1031d13 00000009 dd6dfc6c c1768d78 dd6dfc84 c11c1752 c1768e8e 000000f0 c1768d78 c162bbd4 dd6ef2c0 Call Trace: [<c1604e78>] ? dump_stack+0xa/0x13 [<c1031c40>] ? warn_slowpath_common+0x80/0xc0 [<c11c1752>] ? kobject_add_internal+0x252/0x2e0 [<c11c1752>] ? kobject_add_internal+0x252/0x2e0 [<c1031d13>] ? warn_slowpath_fmt+0x33/0x40 [<c11c1752>] ? kobject_add_internal+0x252/0x2e0 [<c11c1956>] ? kobject_add+0x36/0x80 [<c134de21>] ? device_add+0xc1/0x500 [<c1441669>] ? snd_device_new+0x29/0xa0 [<c1460a56>] ? snd_seq_device_new+0xb6/0x100 [<c146096c>] ? snd_seq_device_dev_register+0xc/0x40 [<c1441604>] ? snd_device_register+0x44/0x80 [<e0bd1fc4>] ? snd_emux_init_seq_oss+0x84/0xa0 [snd_emux_synth] [<e0bcd1e3>] ? snd_emux_register+0x103/0x140 [snd_emux_synth] [<e0bcd000>] ? 0xe0bcd000 [<e0bcd020>] ? sf_sample_new+0x20/0x20 [snd_emux_synth] [<e0bcd040>] ? sf_sample_free+0x20/0x20 [snd_emux_synth] [<e0bdf113>] ? snd_emu8000_probe+0xd3/0x180 [snd_emu8000_synth] [<c11036dd>] ? sysfs_create_link+0x1d/0x40 [<c13504fb>] ? driver_probe_device+0x15b/0x220 [<c1350631>] ? __driver_attach+0x71/0x80 [<c13505c0>] ? driver_probe_device+0x220/0x220 [<c134f000>] ? bus_for_each_dev+0x40/0x80 [<c134ff7a>] ? driver_attach+0x1a/0x20 [<c13505c0>] ? driver_probe_device+0x220/0x220 [<c134fcf4>] ? bus_add_driver+0xd4/0x200 [<c1350b91>] ? driver_register+0x51/0xe0 [<e0be3000>] ? 0xe0be3000 [<e0be3000>] ? 0xe0be3000 [<c10003f2>] ? do_one_initcall+0x72/0x1a0 [<e0be3000>] ? 0xe0be3000 [<c14ac646>] ? netlink_broadcast_filtered+0x166/0x3c0 [<c106c4a5>] ? do_init_module+0x45/0x200 [<c10a8425>] ? __vunmap+0x85/0xe0 [<c106dfaa>] ? load_module+0x186a/0x1ea0 [<c106e6aa>] ? SyS_init_module+0xca/0x100 [<c160a2dd>] ? syscall_call+0x7/0x7 ---[ end trace 826ff8353957cad4 ]---

2 3

[alsa-devel] [PATCH] ASoC: sun4i-codec: Rework and fix headphone routing
by Maxime Ripard 05 Oct '15

05 Oct '15

Most of the boards have their headphone jack directly connected to the matching pins of the SoCs. Since most of the time we will have the same routing path, it makes sense to put that in the driver, and only have a property describing whether that route is enabled or not. It also fixes the following warning messages that were seen so far: sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Left -> direct -> Headphone Jack sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Right -> direct -> Headphone Jack Reported-by: Priit Laes <plaes(a)plaes.org> Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- .../devicetree/bindings/sound/sun4i-codec.txt | 9 ++++---- sound/soc/sunxi/sun4i-codec.c | 25 +++++++++++++++++----- 2 files changed, 24 insertions(+), 10 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt index 680144b74ae9..470823845698 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt @@ -13,10 +13,10 @@ Required properties: - clock-names: should contain followings: - "apb": the parent APB clock for this controller - "codec": the parent module clock -- 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. +Optional Properties: +- allwinner,hp-jack-attached: Indicates that the headphone jack on the + board is directly routed to the SoC pins Example: codec: codec@01c22c00 { @@ -28,6 +28,5 @@ codec: codec@01c22c00 { clock-names = "apb", "codec"; dmas = <&dma 0 19>, <&dma 0 19>; dma-names = "rx", "tx"; - routing = "Headphone Jack", "HP Right", - "Headphone Jack", "HP Left"; + allwinner,hp-jack-attached; }; diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 47780552dcd0..e86b8fcede5a 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -568,10 +568,18 @@ static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev, return link; }; +static const struct snd_soc_dapm_widget sun4i_codec_sink_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static const struct snd_soc_dapm_route sun4i_codec_sink_routes[] = { + { "Headphone Jack", NULL, "HP Right" }, + { "Headphone Jack", NULL, "HP Left" }, +}; + static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) { struct snd_soc_card *card; - int ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) @@ -584,10 +592,17 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) card->dev = dev; card->name = "sun4i-codec"; - ret = snd_soc_of_parse_audio_routing(card, "routing"); - if (ret) { - dev_err(dev, "Failed to create our audio routing\n"); - return NULL; + /* + * FIXME: This will probably require more logic once we will + * support other sinks (for capture for example), or + * encounter more complicated setups where the + * headphone jack is not directly connected to the SoC. + */ + if (of_property_read_bool(dev->of_node, "allwinner,hp-jack-attached")) { + card->dapm_widgets = sun4i_codec_sink_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_sink_widgets); + card->dapm_routes = sun4i_codec_sink_routes; + card->num_dapm_routes = ARRAY_SIZE(sun4i_codec_sink_routes); } return card; -- 2.5.3

2 3

[alsa-devel] [PATCH 1/1] ASoC: da7219: Improve error handling for regulator supplies
by Adam Thomson 05 Oct '15

05 Oct '15

Currently if bulk_enable() of supplies fails, the code still goes on to try and put the device into active state, and set expected IO voltage of the device. This doesn't really make sense so code now returns on bulk_enable() failure, with an error message. Also, to help with debug, failure to get supplies also provides an error message. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- This patch is based on changes introduced under patch: 'ASoC: codecs: Add da7219 codec driver' (Commit 6d817c0e9fd7536be76690bfdee88e8a81c16f7d, kernel/git/broonie/sound.git) sound/soc/codecs/da7219.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 76f8fc2..c86a833 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1554,8 +1554,10 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES, da7219->supplies); - if (ret) + if (ret) { + dev_err(codec->dev, "Failed to get supplies"); return ret; + } /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; @@ -1567,6 +1569,10 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) /* Enable main supplies */ ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies"); + return ret; + } /* Ensure device in active mode */ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); @@ -1574,7 +1580,7 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) /* Update IO voltage level range */ snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl); - return ret; + return 0; } static void da7219_handle_pdata(struct snd_soc_codec *codec) -- 1.9.3

1 0

[alsa-devel] [PATCH v2 0/2] ASoC: add driver for Atmel I2S controller
by Cyrille Pitchen 05 Oct '15

05 Oct '15

This series of patches adds support to the new Atmel I2S controller embedded on sama5d2 SoCs. ChangeLog v2: - initialize dev->dev before calling dev->caps->mck_init(). Cyrille Pitchen (2): ASoC: atmel-i2s: add DT bindings for I2S controller ASoC: atmel-i2s: add driver for the new Atmel I2S controller .../devicetree/bindings/sound/atmel-i2s.txt | 43 ++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-i2s.c | 760 +++++++++++++++++++++ 4 files changed, 815 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-i2s.txt create mode 100644 sound/soc/atmel/atmel-i2s.c -- 1.8.2.2

2 3

[alsa-devel] Missing return check of of_property_read_*()
by Takashi Iwai 05 Oct '15

05 Oct '15

Hi, I hoped that I wouldn't need to write this kind of post, but since nothing happened so far, so here we go: The following compile warnings are present for quite some time. They are due to the lack of return check from of_property_read*(). sound/soc/codecs/cs35l32.c: In function ‘cs35l32_i2c_probe’: sound/soc/codecs/cs35l32.c:278:2: warning: ‘val’ may be used uninitialized in this function [-Wmaybe-uninitialized] switch (val) { ^ sound/soc/codecs/cs35l32.c:272:15: note: ‘val’ was declared here unsigned int val; ^ sound/soc/sti/uniperif_player.c: In function ‘uni_player_init’: sound/soc/sti/uniperif_player.c:1006:6: warning: ‘mode’ may be used uninitialized in this function [-Wmaybe-uninitialized] if (strcasecmp(mode, "hdmi") == 0) ^ sound/soc/sti/uniperif_player.c:985:14: note: ‘mode’ was declared here const char *mode; ^ Both should be trivial to fix, but I won't bother to fix these things by myself, as it results always in a matter of taste how to write. So, please fix them appropriately. Although not shown as compile warning like the above, lots of codec drivers have no return value checks from of_property_read*(), too. A quick grep reveals: % git grep -B1 '^[[:space:]]*of_property_read_' sound/soc sound/soc/codecs/adau1701.c- sound/soc/codecs/adau1701.c: of_property_read_u32(dev->of_node, "adi,pll-clkdiv", -- sound/soc/codecs/adau1701.c- sound/soc/codecs/adau1701.c: of_property_read_u8_array(dev->of_node, "adi,pin-config", -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,boost-manager", &val); -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,sdout-datacfg", &val); -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,battery-threshold", &val); -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,battery-recovery", &val); -- sound/soc/codecs/sta32x.c- sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,output-conf", sound/soc/codecs/sta32x.c- &pdata->output_conf); sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,ch1-output-mapping", sound/soc/codecs/sta32x.c- &pdata->ch1_output_mapping); sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,ch2-output-mapping", sound/soc/codecs/sta32x.c- &pdata->ch2_output_mapping); sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,ch3-output-mapping", -- sound/soc/codecs/sta32x.c- tmp = 140; sound/soc/codecs/sta32x.c: of_property_read_u16(np, "st,drop-compensation-ns", &tmp); -- sound/soc/codecs/sta350.c- sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,output-conf", sound/soc/codecs/sta350.c- &pdata->output_conf); sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,ch1-output-mapping", sound/soc/codecs/sta350.c- &pdata->ch1_output_mapping); sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,ch2-output-mapping", sound/soc/codecs/sta350.c- &pdata->ch2_output_mapping); sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,ch3-output-mapping", -- sound/soc/codecs/sta350.c- tmp = 140; sound/soc/codecs/sta350.c: of_property_read_u16(np, "st,drop-compensation-ns", &tmp); -- sound/soc/codecs/tas5086.c- sound/soc/codecs/tas5086.c: of_property_read_u32(of_node, "ti,charge-period", -- sound/soc/codecs/tlv320aic31xx.c- sound/soc/codecs/tlv320aic31xx.c: of_property_read_u32(np, "ai31xx-micbias-vg", &value); -- sound/soc/codecs/twl4030.c- sound/soc/codecs/twl4030.c: of_property_read_u32(node, "ti,digimic_delay", sound/soc/codecs/twl4030.c- &pdata->digimic_delay); sound/soc/codecs/twl4030.c: of_property_read_u32(node, "ti,ramp_delay_value", sound/soc/codecs/twl4030.c- &pdata->ramp_delay_value); sound/soc/codecs/twl4030.c: of_property_read_u32(node, "ti,offset_cncl_path", -- sound/soc/fsl/fsl_esai.c- esai_priv->synchronous = sound/soc/fsl/fsl_esai.c: of_property_read_bool(np, "fsl,esai-synchronous"); -- sound/soc/omap/omap-abe-twl6040.c- priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); sound/soc/omap/omap-abe-twl6040.c: of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); -- sound/soc/samsung/i2s.c- } sound/soc/samsung/i2s.c: of_property_read_string_index(dev->of_node, -- sound/soc/sh/rcar/rsrc-card.c- /* sampling rate convert */ sound/soc/sh/rcar/rsrc-card.c: of_property_read_u32(node, "convert-rate", &priv->convert_rate); -- sound/soc/sti/uniperif_player.c- sound/soc/sti/uniperif_player.c: of_property_read_u32(pnode, "version", &player->ver); -- sound/soc/sti/uniperif_player.c- sound/soc/sti/uniperif_player.c: of_property_read_u32(pnode, "uniperiph-id", &info->id); -- sound/soc/sti/uniperif_player.c- /* Read the device mode property */ sound/soc/sti/uniperif_player.c: of_property_read_string(pnode, "mode", &mode); -- sound/soc/sti/uniperif_reader.c- sound/soc/sti/uniperif_reader.c: of_property_read_u32(node, "version", &reader->ver); Many of them look really buggy (while some have the default value to fall back). And many codes have even no value checks. These should be fixed as well. thanks, Takashi

6 26

[alsa-devel] Fwd: Fwd: [PATCH 1/1] SPI : spi-pxa2xx : fix spi init of WM510205 codec via ACPI (resend)
by Christian Hartmann 05 Oct '15

05 Oct '15

Hi, this is a resend with all recipients in cc now ---------- Forwarded message ---------- From: Christian Hartmann <cornogle(a)googlemail.com> Date: 2015-08-28 9:41 GMT+02:00 Subject: Re: [alsa-devel] Fwd: [PATCH 1/1] SPI : spi-pxa2xx : fix spi init of WM510205 codec via ACPI (resend) To: Mark Brown <broonie(a)kernel.org> hi Mark et al., 2015-08-27 16:56 GMT+02:00 Mark Brown <broonie(a)kernel.org>: > I can't see how that would work, the interrupt output isn't a GPIO. yes I think also that this is the GPIO pin number. I am not familiar with the interrupt handling, have again to look (in other drivers or similar), to understand how things should work, never done that before. I have added three dev_info() into spi-pxa2xx pxa2xx_spi_probe() to found out more whats wrong with the slave/master: commit 4b6ff73610deef478e53b958a02617f024bbef22 Author: Christian Hartmann <cornogle(a)googlemail.com> Date: Thu Aug 27 14:51:48 2015 +0200 spi/spi-pxa2xx.c : added three new dev_infos, which prints irq, bus_num and num_chipselect Signed-off-by: Christian Hartmann <cornogle(a)googlemail.com> diff --git a/drivers/spi/spi-pxa2xx.c b/drivers/spi/spi-pxa2xx.c index 6d795a5..cf02094 100644 --- a/drivers/spi/spi-pxa2xx.c +++ b/drivers/spi/spi-pxa2xx.c @@ -1389,6 +1389,9 @@ static int pxa2xx_spi_probe(struct platform_device *pdev) master->unprepare_transfer_hardware = pxa2xx_spi_unprepare_transfer; master->auto_runtime_pm = true; + dev_info(&pdev->dev, "bus_num = %d\n",master->bus_num); + dev_info(&pdev->dev, "num_chipselect = %d\n",master->num_chipselect); + drv_data->ssp_type = ssp->type; drv_data->null_dma_buf = (u32 *)PTR_ALIGN(&drv_data[1], DMA_ALIGNMENT); @@ -1423,6 +1426,8 @@ static int pxa2xx_spi_probe(struct platform_device *pdev) } + dev_info(&pdev->dev, "IRQ %d\n", ssp->irq); + /* Setup DMA if requested */ drv_data->tx_channel = -1; drv_data->rx_channel = -1; and thats what I got: [ 6.182903] pxa2xx-spi 80860F0E:00: bus_num = -1 [ 6.182909] pxa2xx-spi 80860F0E:00: num_chipselect = 2 [ 6.182928] pxa2xx-spi 80860F0E:00: IRQ 194 [ 6.183049] pxa2xx-spi 80860F0E:00: no DMA channels available, using PIO [ 6.183111] pxa2xx-spi 80860F0E:00: registered master spi32766 (dynamic) [ 6.192761] spi spi-WM510205:00: 8333333 Hz actual, PIO .... the rest of the dmesg is currently the same.... Q: this thread is about the patch for the pxa2xx-spi where the num_chipselect was raised by one to so that the slave connects here at all. otherwise the cs1 >= max will be hit. Do you think the bus_number is ok ? And shouldn't it be set the cs1=0 and num_chipselect=1 ?? The DSDT has wrongly SPI=1 as chipselect, either the firmware (DSDT) must be updated or a correct patch enables the spi master/slave init here. I hope or expect that when the spi slave will be init correctly, that the device would also work. The ACPI stuff is already done here, and I am running out of time. In the last resort I have to give up or have to do that at night :) cheers chris

2 9

[alsa-devel] Regarding ASoC: rt286: Fix the runtime error in the booting
by Axel Lin 05 Oct '15

05 Oct '15

Hi Order, I just notice your patch on the ML: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/098390.ht… I didn't subscribe the ML and I'm not in CC so I cannot directly reply to you. I think you should make a copy of memory for index_cache rather than directly use the rt286_index_def if the index_cache is necessary. diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1fbdb4f..210ab415 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -37,7 +37,7 @@ #define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { - const struct reg_default *index_cache; + struct reg_default *index_cache; int index_cache_size; struct regmap *regmap; struct snd_soc_codec *codec; @@ -1160,8 +1160,12 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return -ENODEV; } - rt286->index_cache = rt286_index_def; rt286->index_cache_size = INDEX_CACHE_SIZE; + rt286->index_cache = devm_kmalloc(&i2c->dev, INDEX_CACHE_SIZE, + GFP_KERNEL); + if (!rt286->index_cache) + return -ENOMEM; + rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286);

1 1

[alsa-devel] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered
by Rick Mann 05 Oct '15

05 Oct '15

I've been at this for weeks and I'm still unable to get linux to work correctly with my TI codec. This is on a Beaglebone Black running a 4.1.4 kernel (later versions are worse). I've tried multiple versions of the kernel trying to narrow it down. I keep coming back to 4.1.4 as the most recent that at least generates some audio clocks. The AM335x mcasp0 should be speaking i2s with the tlv320aic3104. I have my own build from source of the BBB 4.1.4-ti-r9 release (mainline plus TI plus Beaglebone stuff). I've sprinkled some logging around the audio drivers, but so far I don't see much, and I really don't understand how all the pieces fit together. One thing I found with my logging is the mcasp MCLK (AHCLKXDIV) is never set. I have logging calls in __davinci_mcasp_set_clkdiv() (sound/soc/davinci/davinci-mcasp.c), which is the only code I could find that sets the McASP clock, and it's never called. I tried to use devmem2 to read the registers in the SoC to see if they were set correctly, but I just get a bus error (other reads do work). Errors in dmesg when I load the device tree overlay for my cape suggest I still don't have something configured correctly (see below). I'd sure appreciate any insight into these latest issues. Thanks! Symptoms: MCLK should be 12 MHz, but is 24.00 MHz. Buffer underflows when running speaker-test. Errors to console upon loading cape: [ 38.502299] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.517917] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.530425] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.536996] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.564279] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT dmesg upon loading cape overlay (https://github.com/JetForMe/podtique/blob/v1/bbb/cape/Podtique1/BB-ENABLE-P…): [ 568.323587] bone_capemgr bone_capemgr: part_number 'BB-ENABLE-PRU', version 'N/A' [ 568.323629] bone_capemgr bone_capemgr: slot #4: override [ 568.323646] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 568.323663] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-ENABLE-PRU' [ 568.323804] bone_capemgr:capemgr_load_slot:1148: bone_capemgr bone_capemgr: slot #4: Requesting part number/version based 'BB-ENABLE-PRU-00A0.dtbo [ 568.323825] bone_capemgr:capemgr_load_slot:1162: bone_capemgr bone_capemgr: slot #4: Requesting firmware 'BB-ENABLE-PRU-00A0.dtbo' for board-name 'Override Board Name', version '00A0' [ 568.330369] bone_capemgr:capemgr_load_slot:1171: bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; converting to live tree [ 568.341685] gpio-of-helper ocp:gpio_helper: ready [ 568.344695] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; overlay id #0 [ 568.387640] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.387695] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.387712] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.394284] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.410560] snd_soc_core:snd_soc_register_codec:3064: tlv320aic3x-codec 2-0018: codec register 2-0018 [ 568.410614] snd_soc_core:snd_soc_register_dais:2590: tlv320aic3x-codec 2-0018: ASoC: dai register 2-0018 #1 [ 568.410633] snd_soc_core:snd_soc_register_dais:2634: tlv320aic3x-codec 2-0018: ASoC: Registered DAI 'tlv320aic3x-hifi' [ 568.410653] snd_soc_core:snd_soc_register_codec:3138: tlv320aic3x-codec 2-0018: ASoC: Registered codec 'tlv320aic3x-codec.2-0018' [ 568.421529] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.421588] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.421605] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.431455] **** RLM davinci-mcasp.c:1734: davinci_mcasp_probe [ 568.431507] base:of_property_match_string:1398: comparing tx with tx [ 568.431544] base:of_property_match_string:1398: comparing rx with tx [ 568.431555] base:of_property_match_string:1398: comparing rx with rx [ 568.431608] base:of_property_match_string:1398: comparing common with tx [ 568.431621] base:of_property_match_string:1398: comparing common with rx [ 568.431633] base:of_property_match_string:1398: comparing rx with tx [ 568.431643] base:of_property_match_string:1398: comparing rx with rx [ 568.434643] base:of_property_match_string:1398: comparing tx with tx [ 568.437971] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.446651] snd_soc_core:snd_soc_register_dais:2590: davinci-mcasp 48038000.mcasp: ASoC: dai register 48038000.mcasp #1 [ 568.446682] snd_soc_core:snd_soc_register_dais:2634: davinci-mcasp 48038000.mcasp: ASoC: Registered DAI '48038000.mcasp' [ 568.446840] snd_soc_core:snd_soc_add_platform:2892: davinci-mcasp 48038000.mcasp: ASoC: Registered platform '48038000.mcasp' [ 568.450046] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.450100] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.450425] snd_soc_core:snd_soc_dapm_new_dai_widgets:3494: tlv320aic3x-codec 2-0018: ASoC: adding Playback widget [ 568.450448] snd_soc_core:snd_soc_dapm_new_dai_widgets:3513: tlv320aic3x-codec 2-0018: ASoC: adding Capture widget [ 568.450995] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -2 [ 568.451015] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -1 [ 568.451030] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 0 [ 568.451045] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 1 [ 568.451059] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 2 [ 568.451099] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT [ 568.465530] snd_soc_core:soc_new_pcm:2510: davinci_evm ocp:sound: ASoC: registered pcm #0 AIC3X tlv320aic3x-hifi-0 [ 568.466155] davinci_evm ocp:sound: tlv320aic3x-hifi <-> 48038000.mcasp mapping ok [ 568.466185] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Right ADC -> Capture [ 568.466206] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Left ADC -> Capture [ 568.466226] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Right DAC [ 568.466241] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Left DAC upon starting speaker-test: [ 572.786071] snd_soc_core:soc_pcm_open:535: ASoC: tlv320aic3x-hifi <-> 48038000.mcasp info: [ 572.786103] snd_soc_core:soc_pcm_open:536: ASoC: rate mask 0x7fe [ 572.786114] snd_soc_core:soc_pcm_open:538: ASoC: min ch 2 max ch 2 [ 572.786126] snd_soc_core:soc_pcm_open:540: ASoC: min rate 8000 max rate 96000 [ 572.841514] davinci-mcasp 48038000.mcasp: Transmit buffer underflow ...repeated [ 574.087162] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 579.135283] snd_soc_core:close_delayed_work:602: TLV320AIC3X: ASoC: pop wq checking: Playback status: inactive waiting: yes -- Rick Mann rmann(a)latencyzero.com

1 0

[alsa-devel] [PATCH v2] ASoC: rt286: Fix the runtime error in the booting
by Oder Chiou 05 Oct '15

05 Oct '15

The struct rt286_index_def was used by the cache function, so it cannot be declared as const. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt286.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1fbdb4f..a7178ac 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -37,7 +37,7 @@ #define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { - const struct reg_default *index_cache; + struct reg_default *index_cache; int index_cache_size; struct regmap *regmap; struct snd_soc_codec *codec; @@ -49,7 +49,7 @@ struct rt286_priv { int clk_id; }; -static const struct reg_default rt286_index_def[] = { +static struct reg_default rt286_index_def[] = { { 0x01, 0xaaaa }, { 0x02, 0x8aaa }, { 0x03, 0x0002 }, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt286: Fix the runtime error in the booting
by Oder Chiou 05 Oct '15

05 Oct '15

The struct rt286_index_def was used by the cache function, so it cannot be declared as const. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1fbdb4f..79107dc 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -49,7 +49,7 @@ struct rt286_priv { int clk_id; }; -static const struct reg_default rt286_index_def[] = { +static struct reg_default rt286_index_def[] = { { 0x01, 0xaaaa }, { 0x02, 0x8aaa }, { 0x03, 0x0002 }, -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] baytrail crv2 audio mic no data
by Lemon 05 Oct '15

05 Oct '15

Hello everyone, I have a Z3735F board which use ssp0 connect to rt5640, I using linux mainline 4.2 and fw_sst_0f28.bin from https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/?h=byt the audio sound is ok, I tried many alsa config but the analog mic record data is all 0, kernel no error message. I don't know why mic record data is 0, does anyone knows ? regards LemonZou ======================================== arecord -f dat -t wav -d 5 foobar.wav hexdump foobar.wav 0000000 4952 4646 a624 000e 4157 4556 6d66 2074 0000010 0010 0000 0001 0002 bb80 0000 ee00 0002 0000020 0004 0010 6164 6174 a600 000e 0000 0000 0000030 0000 0000 0000 0000 0000 0000 0000 0000 * 00ea62c my board mic connect to alc5640 in2p, my main config include: "Stereo ADC MIXL ADC1" on "Stereo ADC1 Mux" "ADCL" "RECMIXL INL" on "RECMIXL BST2" on "pcm1_out mix 0 codec_in0" on "pcm1_out mix 0 codec_in1" on kernel message: [ 5329.874448] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp0-port [ 5329.979365] intel_sst_acpi 80860F28:00: Alloc for str 3 pipe 0xe [ 5335.250574] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp0-port [ 5335.253528] intel_sst_acpi 80860F28:00: Free for str 3 pipe 0xe ========================================

1 0

[alsa-devel] [REANNOUNCE] Audio Mini Summit 2015, 8th October, Dublin
by Mark Brown 05 Oct '15

05 Oct '15

Like previous years, we're going to hold a meeting to discuss lowlevel audio on Linux. This will be held the day after ELC Europe on 8th October at CCD (The Convention Centre) in Dublin. If you're interested please sign up in the attendee list: https://docs.google.com/spreadsheets/d/1bi-8GHsqlzt41FR20WiMuWILe_cZowFireV… and please follow up to this mail with any topics you'd like to see raised so we can start collecting them (probably another Google doc will be forthcoming for them). Previously suggested topics include: - Reducing kernel size for IoT applications - ACPI simple-card - Testing tools Thanks, Mark

9 25

[alsa-devel] [PATCH] ALSA: hda - Disable power_save_node for IDT 92HD73xx chips
by Takashi Iwai 04 Oct '15

04 Oct '15

The recent widget power saving introduced some unavoidable click noises on old IDT 92HD73xx chips while it still seems working on the compatible new chips. In the bugzilla, we tried lots of tests and workarounds, but they didn't help much. So, let's disable the feature for these specific chips as the least (but safest) fix. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=104981 Cc: <stable(a)vger.kernel.org> # v4.1+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9d947aef2c8b..def5cc8dff02 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4520,7 +4520,11 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_save_node = 1; + /* enable power_save_node only for new 92HD89xx chips, as it causes + * click noises on old 92HD73xx chips. + */ + if ((codec->core.vendor_id & 0xfffffff0) != 0x111d7670) + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; -- 2.5.3

1 0

[alsa-devel] [PATCH 0/7 v3] firewire-tascam: new driver for TASCAM FireWire series
by Takashi Sakamoto 04 Oct '15

04 Oct '15

Hi, This patchset updates a part of my former post: [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html This patchset adds support for a part of TASCAM FireWire series. This patchset supports their streaming and PCM functionalities, while their MIDI functionalities are not supported yet. In future work, the functionalities will be supported. Takashi Sakamoto (7): ALSA: firewire-tascam: add skeleton for TASCAM FireWire series ALSA: firewire-tascam: add a structure for model-dependent parameters. ALSA: firewire-tascam: add proc node to show firmware information ALSA: firewire-tascam: add data block processing layer ALSA: firewire-tascam: add streaming functionality ALSA: firewire-tascam: add PCM functionality ALSA: firewire-tascam: add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 12 + sound/firewire/Makefile | 1 + sound/firewire/tascam/Makefile | 3 + sound/firewire/tascam/amdtp-tascam.c | 243 +++++++++++++++++ sound/firewire/tascam/tascam-hwdep.c | 201 ++++++++++++++ sound/firewire/tascam/tascam-pcm.c | 312 +++++++++++++++++++++ sound/firewire/tascam/tascam-proc.c | 88 ++++++ sound/firewire/tascam/tascam-stream.c | 496 ++++++++++++++++++++++++++++++++++ sound/firewire/tascam/tascam.c | 213 +++++++++++++++ sound/firewire/tascam/tascam.h | 117 ++++++++ 12 files changed, 1689 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/tascam/Makefile create mode 100644 sound/firewire/tascam/amdtp-tascam.c create mode 100644 sound/firewire/tascam/tascam-hwdep.c create mode 100644 sound/firewire/tascam/tascam-pcm.c create mode 100644 sound/firewire/tascam/tascam-proc.c create mode 100644 sound/firewire/tascam/tascam-stream.c create mode 100644 sound/firewire/tascam/tascam.c create mode 100644 sound/firewire/tascam/tascam.h -- 2.1.4

5 19

[alsa-devel] [PATCH V5 1/2] ACPI / EC: Fix broken 64bit big-endian users of 'global_lock'
by Viresh Kumar 04 Oct '15

04 Oct '15

global_lock is defined as an unsigned long and accessing only its lower 32 bits from sysfs is incorrect, as we need to consider other 32 bits for big endian 64-bit systems. There are no such platforms yet, but the code needs to be robust for such a case. Fix that by changing type of 'global_lock' to u32. Signed-off-by: Viresh Kumar <viresh.kumar(a)linaro.org> --- BCC'd a lot of people (rather than cc'ing them) to make sure - the series reaches them - mailing lists do not block the patchset due to long cc list - and we don't spam the BCC'd people for every reply V4->V5: - Switch back to the original solution of making global_lock u32. --- drivers/acpi/ec_sys.c | 2 +- drivers/acpi/internal.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/acpi/ec_sys.c b/drivers/acpi/ec_sys.c index b4c216bab22b..bea8e425a8de 100644 --- a/drivers/acpi/ec_sys.c +++ b/drivers/acpi/ec_sys.c @@ -128,7 +128,7 @@ static int acpi_ec_add_debugfs(struct acpi_ec *ec, unsigned int ec_device_count) if (!debugfs_create_x32("gpe", 0444, dev_dir, (u32 *)&first_ec->gpe)) goto error; if (!debugfs_create_bool("use_global_lock", 0444, dev_dir, - (u32 *)&first_ec->global_lock)) + &first_ec->global_lock)) goto error; if (write_support) diff --git a/drivers/acpi/internal.h b/drivers/acpi/internal.h index 9e426210c2a8..9db196de003c 100644 --- a/drivers/acpi/internal.h +++ b/drivers/acpi/internal.h @@ -138,7 +138,7 @@ struct acpi_ec { unsigned long gpe; unsigned long command_addr; unsigned long data_addr; - unsigned long global_lock; + u32 global_lock; unsigned long flags; unsigned long reference_count; struct mutex mutex; -- 2.4.0

3 3

[alsa-devel] [RFC] MIDI over Bluetooth Low Energy
by Felipe Tonello 03 Oct '15

03 Oct '15

Hi guys, I am planning to start the support of MIDI BLE profile[1]. This profile is not officially supported yet, but it will most likely be very similar, so development efforts are still valid. I suggest two main goals: * To be transparent to applications, i.e., use rawmidi and sequencer ALSA interfaces to interact. * To support peripheral and central BLE roles. My question is: what is the best way possible of doing it? My initial though is to write a GATT BlueZ profile plugin that will load snd-virtmidi module with id and midi_devs parameters, then read and write seq events from/to it. I am not sure if this is really possible. Another way of implementing is as a rawmidi and a seq plugin using the BlueZ GATT D-Bus interface. IMO this is not ideal because it requires a lot more work (rawmidi and seq plugins, maybe even a library to avoid code duplication) and has an overhead of using dbus. It is also possible to write a kernel module to handle ALSA card/device setup and reads and writes from the bluez plugin (perhaps this simplifies things because it has less dependencies). They all have the problem of context switching between bluez plugin and alsa midi driver. I would prefer to use a shared ring buffer between ALSA e BlueZ. Any ideas and comments? [1] https://developer.apple.com/bluetooth/Apple-Bluetooth-Low-Energy-MIDI-Speci… PS: I am at #bluez and #alsa as ftonello. Felipe Tonello

5 15

[alsa-devel] [PATCH] ALSA: hda - Apply SPDIF pin ctl to MacBookPro 12, 1
by Takashi Iwai 03 Oct '15

03 Oct '15

From: John Flatness <john(a)zerocrates.org> The MacBookPro 12,1 has the same setup as the 11 for controlling the status of the optical audio light. Simply apply the existing workaround to the subsystem ID for the 12,1. [sorted the fixup entry by tiwai] Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=105401 Signed-off-by: John Flatness <john(a)zerocrates.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 584a0343ab0c..85813de26da8 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -633,6 +633,7 @@ static const struct snd_pci_quirk cs4208_mac_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x5e00, "MacBookPro 11,2", CS4208_MBP11), SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6), SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6), + SND_PCI_QUIRK(0x106b, 0x7b00, "MacBookPro 12,1", CS4208_MBP11), {} /* terminator */ }; -- 2.5.3

1 0

[alsa-devel] [PATCH] ALSA: hda: Add dock support for ThinkPad T550
by Laura Abbott 03 Oct '15

03 Oct '15

Much like all the other Lenovo laptops, add a quirk to make sound work with docking. Reported-and-tested-by: lacknerflo(a)gmail.com Signed-off-by: Laura Abbott <labbott(a)fedoraproject.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index afec6dc..16b8dcb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5306,6 +5306,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.4.3

2 1

[alsa-devel] [RESEND PATCH v2 1/1] ASoC: dwc: fix dma stop transferring issue
by yitian 03 Oct '15

03 Oct '15

Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music continues to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally causes music playing will be stopped once XRUN occurs for the first time. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- changes in V2: - add definition for i and irq --- sound/soc/dwc/designware_i2s.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..76b2e19 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -141,13 +141,22 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { - + u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 1.7.12.4

2 6

[alsa-devel] Kernel Device Tree entries for simple-audio-card
by Rick Mann 03 Oct '15

03 Oct '15

Hi. I've built a Beaglebone Black Cape that has, among other things, a TI Audio CODEC on it (TLV320AIC3104). This is the same CODEC as the BBB Audio Cape (http://elinux.org/CircuitCo:Audio_Cape_RevB), and it's connected in almost the exact same way. The last few days I've been figuring out the Device Tree overlay needed to get Linux (4.1.4) to recognize it. It's finding the device and instantiating the low-level driver (tlv320aic3x-codec), although I'm not sure at this point if it's actually talking to it (I have verified that low-level i2c commands do work). But when it comes to the sound card setup, I'm having trouble getting the Device Tree right. I'm hoping someone on this list can offer some insight. The DT is here: https://github.com/JetForMe/podtique/blob/master/bbb/cape/Podtique1/BB-ENAB… And this is dmesg when I try to load that (note the name of my overlay is BB-ENABLE-PRU, which is an artifact of the experimentation I've been doing; it'll change to something more appropriate): [ 65.859275] bone_capemgr bone_capemgr: part_number 'BB-ENABLE-PRU', version 'N/A' [ 65.859318] bone_capemgr bone_capemgr: slot #4: override [ 65.859336] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 65.859355] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-ENABLE-PRU' [ 65.859488] bone_capemgr: bone_capemgr bone_capemgr: slot #4: Requesting part number/version based 'BB-ENABLE-PRU-00A0.dtbo [ 65.859506] bone_capemgr: bone_capemgr bone_capemgr: slot #4: Requesting firmware 'BB-ENABLE-PRU-00A0.dtbo' for board-name 'Override Board Name', version '00A0' [ 65.861105] bone_capemgr: bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; converting to live tree [ 65.875618] gpio-of-helper ocp:gpio_helper: ready [ 65.876327] gpiolib_of: of_get_named_gpiod_flags: parsed 'gpio' property of node '//fixedregulator@1[0]' - status (0) [ 65.876619] core: lz-codec-reg: no parameters [ 65.876713] fixed: reg-fixed-voltage fixedregulator@1: lz-codec-reg supplying 0uV [ 65.883284] pruss_uio 4a300000.pruss: pins are not configured from the driver [ 65.900080] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; overlay id #0 [ 65.946254] gpiolib_of: of_get_named_gpiod_flags: parsed 'gpio-reset' property of node '/ocp/i2c@4819c000/tlv320aic3104@0[0]' - status (0) [ 65.946317] core: tlv320aic3x-codec 2-0018: Looking up IOVDD-supply from device tree [ 65.946405] core: tlv320aic3x-codec 2-0018: Looking up DVDD-supply from device tree [ 65.946468] core: tlv320aic3x-codec 2-0018: Looking up AVDD-supply from device tree [ 65.946544] core: tlv320aic3x-codec 2-0018: Looking up DRVDD-supply from device tree [ 65.946606] snd_soc_core: tlv320aic3x-codec 2-0018: codec register 2-0018 [ 65.946640] snd_soc_core: tlv320aic3x-codec 2-0018: ASoC: dai register 2-0018 #1 [ 65.946655] snd_soc_core: tlv320aic3x-codec 2-0018: ASoC: Registered DAI 'tlv320aic3x-hifi' [ 65.946671] snd_soc_core: tlv320aic3x-codec 2-0018: ASoC: Registered codec 'tlv320aic3x-codec.2-0018' [ 65.956491] //sound/simple-audio-card,cpu: could not get #sound-dai-cells for /ocp/mcasp@48038000 [ 65.978759] snd_soc_core: davinci-mcasp 48038000.mcasp: ASoC: dai register 48038000.mcasp #1 [ 65.978795] snd_soc_core: davinci-mcasp 48038000.mcasp: ASoC: Registered DAI '48038000.mcasp' [ 65.978954] snd_soc_core: davinci-mcasp 48038000.mcasp: ASoC: Registered platform '48038000.mcasp' [ 66.012427] asoc-simple-card sound: parse error -22 [ 66.042609] asoc-simple-card: probe of sound failed with error -22 Note the bit about "#sound-dai-cells". I'm not sure what's wrong here, and it seems like it might be something in the distro-provided DTB, not my own. aplay lists nothing (I think in part because I have yet to put in a proper asound.conf file): # aplay -l aplay: device_list:268: no soundcards found... Can someone suggest some diagnostics I can do to see what the state of things is? I'm just not sure what to look for. Thanks! -- Rick Mann rmann(a)latencyzero.com

2 18

[alsa-devel] [PATCH MC Next Gen 00/20] Update ALSA, and au0828 drivers to use Managed Media Controller API
by Shuah Khan 03 Oct '15

03 Oct '15

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API to share tuner. Please note that Managed Media Controller API and DVB and V4L2 drivers updates to use Media Controller API have been added in a prior patch series. Media Controller API is enhanced with two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. Please find the graphs generated using mc_nextgen_test tool at various points during testing at: https://drive.google.com/folderview?id=0B0NIL0BQg-Alb3JFb2diMXRoQlU&usp=sha… This patch series is the port of the same work that was done in the following patch series to use Media Controller Next Gen API. Patch v3 - Media Controller API: https://www.mail-archive.com/linux-media@vger.kernel.org/msg92572.html References and History: Changes since v2: http://www.spinics.net/lists/linux-media/msg91926.html 1. An important change in this patch series is made to ALSA to check for resources and hold in snd_usb_hw_params(), and release from snd_usb_hw_free(). This change fixed the lockdep warnings seen when resources were held in TRIGGER_START and released from TRIGGER_STOP which could run in IRQ context. I acknowledge Clemens Ladisch for suggesting the correct places to hold/free resources to avoid IRQ path complications. 2. With the above change, the patch series is simpler without the need to change the graph_mutex into a spinlock. 3. I split the patches up differently for easy reviews - no code bloat from v2. 4. A second important change is now the Bridge driver (au0828) owns and drives the graph creation as well as enabling and disabling tuner. It also keeps state information to avoid graph walks in enable_source and disable_source handler. I acknowledge Hans Verkuil for his suggestions and ideas for this change. History: This patch series has been updated to address comments from 3 previous versions of this series. Links to v3 version for reference are: https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89491.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg89492.html https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89493.html Shuah Khan (20): media: Media Controller register/unregister entity_notify API media: Add ALSA Media Controller function entities media: Media Controller enable/disable source handler API media: Media Controller fix to not let stream_count go negative media: Media Controller export non locking __media_entity_setup_link() media: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add v4l_enable/disable_media_tuner() helper functions media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: au0828 Use au8522_media_pads enum for pad defines media: au0828 fix au0828_create_media_graph() entity checks media: Change v4l-core to check for tuner availability media: dvb-frontend invoke enable/disable_source handlers media: au0828 video remove au0828_enable_analog_tuner() media: au0828 video change to use v4l_enable_media_tuner() media: au0828 change to use Managed Media Controller API media: au0828 change to register/unregister entity_notify hook media: au0828 implement enable_source and disable_source handlers media: dvb-core create tuner to demod pad link in disabled state sound/usb: Update ALSA driver to use Managed Media Controller API drivers/media/dvb-core/dvb_frontend.c | 139 ++--------- drivers/media/dvb-core/dvb_frontend.h | 3 + drivers/media/dvb-core/dvbdev.c | 3 +- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 43 ++++ drivers/media/media-entity.c | 64 +++-- drivers/media/usb/au0828/au0828-core.c | 337 ++++++++++++++++++++------- drivers/media/usb/au0828/au0828-video.c | 75 +----- drivers/media/usb/au0828/au0828.h | 8 + drivers/media/v4l2-core/v4l2-dev.c | 27 +++ drivers/media/v4l2-core/v4l2-fh.c | 1 + drivers/media/v4l2-core/v4l2-ioctl.c | 29 +++ drivers/media/v4l2-core/videobuf2-core.c | 3 + include/media/media-device.h | 44 ++++ include/media/media-entity.h | 3 + include/media/v4l2-dev.h | 4 + include/uapi/linux/media.h | 9 +- sound/usb/Makefile | 15 +- sound/usb/card.c | 5 + sound/usb/card.h | 1 + sound/usb/media.c | 201 ++++++++++++++++ sound/usb/media.h | 53 +++++ sound/usb/mixer.c | 1 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 1 + 30 files changed, 823 insertions(+), 304 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

1 20

[alsa-devel] [PATCH 0/8] Add support for Wolfson WM8998 and WM1814 codecs
by Richard Fitzgerald 02 Oct '15

02 Oct '15

This set of patches adds support for the Wolfson Microelectronics WM8998 and WM1814 codecs. Richard Fitzgerald (8): mfd: arizona: Split INx_MODE into two fields ASoC: arizona: add defines for single-input gain control mfd: arizona: Add support for WM8998 and WM1814 gpio: arizona: add support for WM8998 and WM1814 regulator: arizona: add support for WM8998 and WM1814 extcon: arizona: Add support for WM8998 and WM1814 ASoC: wm8998: Initial WM8998 codec driver Documentation: Add WM8998/WM1814 device tree bindings Documentation/devicetree/bindings/mfd/arizona.txt | 3 + drivers/extcon/extcon-arizona.c | 33 +- drivers/gpio/gpio-arizona.c | 2 + drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 106 ++- drivers/mfd/arizona-i2c.c | 8 + drivers/mfd/arizona-irq.c | 9 + drivers/mfd/arizona.h | 5 + drivers/mfd/wm8998-tables.c | 1592 +++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 2 + include/linux/mfd/arizona/core.h | 3 + include/linux/mfd/arizona/pdata.h | 5 +- include/linux/mfd/arizona/registers.h | 247 +++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 7 + sound/soc/codecs/arizona.h | 7 +- sound/soc/codecs/wm8998.c | 1392 ++++++++++++++++++ sound/soc/codecs/wm8998.h | 23 + 20 files changed, 3432 insertions(+), 29 deletions(-) create mode 100644 drivers/mfd/wm8998-tables.c create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h -- 1.7.2.5

6 22

[alsa-devel] [PATCH v2 0/3] Add support for Wolfson Micro WM8998 codec
by Richard Fitzgerald 02 Oct '15

02 Oct '15

This set of patches is based on broonie/regulator.git/for-next and broonie/sound.git/for-next branches. Richard Fitzgerald (3): regulator: arizona: add support for WM8998 and WM1814 ASoC: arizona: add defines for single-input gain control ASoC: wm8998: Initial WM8998 codec driver drivers/regulator/arizona-ldo1.c | 2 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 33 + sound/soc/codecs/arizona.h | 12 +- sound/soc/codecs/wm8998.c | 1430 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8998.h | 23 + 7 files changed, 1507 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h -- 1.9.1

3 6

[alsa-devel] [PATCH 0/8] ASoC: topology: ABI - Add support for PCM, BE & CC link
by mengdong.lin＠intel.com 02 Oct '15

02 Oct '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This series includes topology ABI update to support PCM (FE DAI/DAI links), BE and CC DAI links. Bump the ABI version to 4. Mengdong Lin (2): ASoC: topology: ABI - Change stream formats to a bitwise flag ASoC: topology: ABI - Remove tdm_slot & dai_fmt from snd_soc_tplg_stream Vedang Patel (6): ASoC: topology: ABI - Add PCM Support and bump ABI version to 4 ASoC: topology: ABI - Remove unused struct snd_soc_tplg_stream_config ASoC: topology: ABI - Use __le32 instead of __u32 in snd_soc_tplg_dapm_widget ASoC: topology: ABI - Add name element to snd_soc_tplg_stream ASoC: topology: ABI - Add the type for BE DAI link ASoC: topology: ABI - Add configuration for BE & Codec-Codec DAI Links include/uapi/sound/asoc.h | 76 +++++++++++++++++++++++++---------------------- sound/soc/soc-topology.c | 4 +-- 2 files changed, 43 insertions(+), 37 deletions(-) -- 1.9.1

3 17

[alsa-devel] Applied "ASoC: dwc: fix dma stop transferring issue" to the asoc tree
by Mark Brown 02 Oct '15

02 Oct '15

The patch ASoC: dwc: fix dma stop transferring issue has been applied to the asoc tree at git://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 924eb475126fd6bc23c475ac742a69fc466c0b75 Mon Sep 17 00:00:00 2001 From: yitian <yitian.bu(a)tangramtek.com> Date: Tue, 29 Sep 2015 22:43:17 +0800 Subject: [PATCH] ASoC: dwc: fix dma stop transferring issue Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music continues to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally causes music playing will be stopped once XRUN occurs for the first time. [On list discussion suggests this may be partly a race condition on slow systems -- broonie] Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/dwc/designware_i2s.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 0d28e3b..ba34252 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -141,13 +141,22 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { - + u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 2.5.0

1 0

[alsa-devel] Applied "ASoC: codecs: Add da7219 codec driver" to the asoc tree
by Mark Brown 02 Oct '15

02 Oct '15

The patch ASoC: codecs: Add da7219 codec driver has been applied to the asoc tree at git://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 6d817c0e9fd7536be76690bfdee88e8a81c16f7d Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Tue, 29 Sep 2015 16:44:01 +0100 Subject: [PATCH] ASoC: codecs: Add da7219 codec driver This adds support for the DA7219 audio codec with built-in advanced accessory detect features. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/da7219-aad.h | 99 +++ include/sound/da7219.h | 55 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 823 +++++++++++++++++ sound/soc/codecs/da7219-aad.h | 212 +++++ sound/soc/codecs/da7219.c | 1940 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 +++++++++++++++++ 8 files changed, 3955 insertions(+) create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h diff --git a/include/sound/da7219-aad.h b/include/sound/da7219-aad.h new file mode 100644 index 0000000..17802fb --- /dev/null +++ b/include/sound/da7219-aad.h @@ -0,0 +1,99 @@ +/* + * da7219-aad.h - DA7322 ASoC Codec AAD Driver Platform Data + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_AAD_PDATA_H +#define __DA7219_AAD_PDATA_H + +enum da7219_aad_micbias_pulse_lvl { + DA7219_AAD_MICBIAS_PULSE_LVL_OFF = 0, + DA7219_AAD_MICBIAS_PULSE_LVL_2_8V = 6, + DA7219_AAD_MICBIAS_PULSE_LVL_2_9V, +}; + +enum da7219_aad_btn_cfg { + DA7219_AAD_BTN_CFG_2MS = 1, + DA7219_AAD_BTN_CFG_5MS, + DA7219_AAD_BTN_CFG_10MS, + DA7219_AAD_BTN_CFG_50MS, + DA7219_AAD_BTN_CFG_100MS, + DA7219_AAD_BTN_CFG_200MS, + DA7219_AAD_BTN_CFG_500MS, +}; + +enum da7219_aad_mic_det_thr { + DA7219_AAD_MIC_DET_THR_200_OHMS = 0, + DA7219_AAD_MIC_DET_THR_500_OHMS, + DA7219_AAD_MIC_DET_THR_750_OHMS, + DA7219_AAD_MIC_DET_THR_1000_OHMS, +}; + +enum da7219_aad_jack_ins_deb { + DA7219_AAD_JACK_INS_DEB_5MS = 0, + DA7219_AAD_JACK_INS_DEB_10MS, + DA7219_AAD_JACK_INS_DEB_20MS, + DA7219_AAD_JACK_INS_DEB_50MS, + DA7219_AAD_JACK_INS_DEB_100MS, + DA7219_AAD_JACK_INS_DEB_200MS, + DA7219_AAD_JACK_INS_DEB_500MS, + DA7219_AAD_JACK_INS_DEB_1S, +}; + +enum da7219_aad_jack_det_rate { + DA7219_AAD_JACK_DET_RATE_32_64MS = 0, + DA7219_AAD_JACK_DET_RATE_64_128MS, + DA7219_AAD_JACK_DET_RATE_128_256MS, + DA7219_AAD_JACK_DET_RATE_256_512MS, +}; + +enum da7219_aad_jack_rem_deb { + DA7219_AAD_JACK_REM_DEB_1MS = 0, + DA7219_AAD_JACK_REM_DEB_5MS, + DA7219_AAD_JACK_REM_DEB_10MS, + DA7219_AAD_JACK_REM_DEB_20MS, +}; + +enum da7219_aad_btn_avg { + DA7219_AAD_BTN_AVG_1 = 0, + DA7219_AAD_BTN_AVG_2, + DA7219_AAD_BTN_AVG_4, + DA7219_AAD_BTN_AVG_8, +}; + +enum da7219_aad_adc_1bit_rpt { + DA7219_AAD_ADC_1BIT_RPT_1 = 0, + DA7219_AAD_ADC_1BIT_RPT_2, + DA7219_AAD_ADC_1BIT_RPT_4, + DA7219_AAD_ADC_1BIT_RPT_8, +}; + +struct da7219_aad_pdata { + int irq; + + enum da7219_aad_micbias_pulse_lvl micbias_pulse_lvl; + u32 micbias_pulse_time; + enum da7219_aad_btn_cfg btn_cfg; + enum da7219_aad_mic_det_thr mic_det_thr; + enum da7219_aad_jack_ins_deb jack_ins_deb; + enum da7219_aad_jack_det_rate jack_det_rate; + enum da7219_aad_jack_rem_deb jack_rem_deb; + + u8 a_d_btn_thr; + u8 d_b_btn_thr; + u8 b_c_btn_thr; + u8 c_mic_btn_thr; + + enum da7219_aad_btn_avg btn_avg; + enum da7219_aad_adc_1bit_rpt adc_1bit_rpt; +}; + +#endif /* __DA7219_AAD_PDATA_H */ diff --git a/include/sound/da7219.h b/include/sound/da7219.h new file mode 100644 index 0000000..3f39e13 --- /dev/null +++ b/include/sound/da7219.h @@ -0,0 +1,55 @@ +/* + * da7219.h - DA7219 ASoC Codec Driver Platform Data + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_PDATA_H +#define __DA7219_PDATA_H + +/* LDO */ +enum da7219_ldo_lvl_sel { + DA7219_LDO_LVL_SEL_1_05V = 0, + DA7219_LDO_LVL_SEL_1_10V, + DA7219_LDO_LVL_SEL_1_20V, + DA7219_LDO_LVL_SEL_1_40V, +}; + +/* Mic Bias */ +enum da7219_micbias_voltage { + DA7219_MICBIAS_1_8V = 1, + DA7219_MICBIAS_2_0V, + DA7219_MICBIAS_2_2V, + DA7219_MICBIAS_2_4V, + DA7219_MICBIAS_2_6V, +}; + +/* Mic input type */ +enum da7219_mic_amp_in_sel { + DA7219_MIC_AMP_IN_SEL_DIFF = 0, + DA7219_MIC_AMP_IN_SEL_SE_P, + DA7219_MIC_AMP_IN_SEL_SE_N, +}; + +struct da7219_aad_pdata; + +struct da7219_pdata { + /* Internal LDO */ + enum da7219_ldo_lvl_sel ldo_lvl_sel; + + /* Mic */ + enum da7219_micbias_voltage micbias_lvl; + enum da7219_mic_amp_in_sel mic_amp_in_sel; + + /* AAD */ + struct da7219_aad_pdata *aad_pdata; +}; + +#endif /* __DA7219_PDATA_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..c9a895c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C + select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC @@ -430,6 +431,9 @@ config SND_SOC_DA7210 config SND_SOC_DA7213 tristate +config SND_SOC_DA7219 + tristate + config SND_SOC_DA732X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077..eb10a34 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -49,6 +49,7 @@ snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o +snd-soc-da7219-objs := da7219.o da7219-aad.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-bt-sco-objs := bt-sco.o @@ -241,6 +242,7 @@ obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o +obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c new file mode 100644 index 0000000..9459593 --- /dev/null +++ b/sound/soc/codecs/da7219-aad.c @@ -0,0 +1,823 @@ +/* + * da7219-aad.c - Dialog DA7219 ALSA SoC AAD Driver + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> +#include <linux/pm_wakeirq.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/workqueue.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/da7219.h> + +#include "da7219.h" +#include "da7219-aad.h" + + +/* + * Detection control + */ + +void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + da7219->aad->jack = jack; + da7219->aad->jack_inserted = false; + + /* Send an initial empty report */ + snd_soc_jack_report(jack, 0, DA7219_AAD_REPORT_ALL_MASK); + + /* Enable/Disable jack detection */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_ACCDET_EN_MASK, + (jack ? DA7219_ACCDET_EN_MASK : 0)); +} +EXPORT_SYMBOL_GPL(da7219_aad_jack_det); + +/* + * Button/HPTest work + */ + +static void da7219_aad_btn_det_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, btn_det_work); + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + u8 statusa, micbias_ctrl; + bool micbias_up = false; + int retries = 0; + + /* Drive headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK, + DA7219_HP_L_AMP_OE_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK, + DA7219_HP_R_AMP_OE_MASK); + + /* Make sure mic bias is up */ + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + + do { + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); + if (statusa & DA7219_MICBIAS_UP_STS_MASK) + micbias_up = true; + else if (retries++ < DA7219_AAD_MICBIAS_CHK_RETRIES) + msleep(DA7219_AAD_MICBIAS_CHK_DELAY); + } while ((!micbias_up) && (retries < DA7219_AAD_MICBIAS_CHK_RETRIES)); + + if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES) + dev_warn(codec->dev, "Mic bias status check timed out"); + + /* + * Mic bias pulse required to enable mic, must be done before enabling + * button detection to prevent erroneous button readings. + */ + if (da7219_aad->micbias_pulse_lvl && da7219_aad->micbias_pulse_time) { + /* Pulse higher level voltage */ + micbias_ctrl = snd_soc_read(codec, DA7219_MICBIAS_CTRL); + snd_soc_update_bits(codec, DA7219_MICBIAS_CTRL, + DA7219_MICBIAS1_LEVEL_MASK, + da7219_aad->micbias_pulse_lvl); + msleep(da7219_aad->micbias_pulse_time); + snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_ctrl); + + } + + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, + da7219_aad->btn_cfg); +} + +static void da7219_aad_hptest_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, hptest_work); + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + u16 tonegen_freq_hptest; + u8 accdet_cfg8; + int report = 0; + + /* Lock DAPM and any Kcontrols that are affected by this test */ + snd_soc_dapm_mutex_lock(dapm); + mutex_lock(&da7219->lock); + + /* Bypass cache so it saves current settings */ + regcache_cache_bypass(da7219->regmap, true); + + /* Make sure Tone Generator is disabled */ + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0); + + /* Enable HPTest block, 1KOhms check */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8, + DA7219_HPTEST_EN_MASK | DA7219_HPTEST_RES_SEL_MASK, + DA7219_HPTEST_EN_MASK | + DA7219_HPTEST_RES_SEL_1KOHMS); + + /* Set gains to 0db */ + snd_soc_write(codec, DA7219_DAC_L_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB); + snd_soc_write(codec, DA7219_DAC_R_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB); + snd_soc_write(codec, DA7219_HP_L_GAIN, DA7219_HP_AMP_GAIN_0DB); + snd_soc_write(codec, DA7219_HP_R_GAIN, DA7219_HP_AMP_GAIN_0DB); + + /* Disable DAC filters, EQs and soft mute */ + snd_soc_update_bits(codec, DA7219_DAC_FILTERS1, DA7219_HPF_MODE_MASK, + 0); + snd_soc_update_bits(codec, DA7219_DAC_FILTERS4, DA7219_DAC_EQ_EN_MASK, + 0); + snd_soc_update_bits(codec, DA7219_DAC_FILTERS5, + DA7219_DAC_SOFTMUTE_EN_MASK, 0); + + /* Enable HP left & right paths */ + snd_soc_update_bits(codec, DA7219_CP_CTRL, DA7219_CP_EN_MASK, + DA7219_CP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DIG_ROUTING_DAC, + DA7219_DAC_L_SRC_MASK | DA7219_DAC_R_SRC_MASK, + DA7219_DAC_L_SRC_TONEGEN | + DA7219_DAC_R_SRC_TONEGEN); + snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, + DA7219_DAC_L_EN_MASK | DA7219_DAC_L_MUTE_EN_MASK, + DA7219_DAC_L_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, + DA7219_DAC_R_EN_MASK | DA7219_DAC_R_MUTE_EN_MASK, + DA7219_DAC_R_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_L_MIX_SELECT_MASK, + DA7219_MIXOUT_L_MIX_SELECT_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_R_SELECT, + DA7219_MIXOUT_R_MIX_SELECT_MASK, + DA7219_MIXOUT_R_MIX_SELECT_MASK); + snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1L, + DA7219_OUTFILT_ST_1L_SRC_MASK, + DA7219_DMIX_ST_SRC_OUTFILT1L); + snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1R, + DA7219_OUTFILT_ST_1R_SRC_MASK, + DA7219_DMIX_ST_SRC_OUTFILT1R); + snd_soc_update_bits(codec, DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_MASK, + DA7219_MIXOUT_L_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_MASK, + DA7219_MIXOUT_R_AMP_EN_MASK); + snd_soc_write(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK); + snd_soc_write(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK); + + /* Configure & start Tone Generator */ + snd_soc_write(codec, DA7219_TONE_GEN_ON_PER, DA7219_BEEP_ON_PER_MASK); + tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ); + regmap_raw_write(da7219->regmap, DA7219_TONE_GEN_FREQ1_L, + &tonegen_freq_hptest, sizeof(tonegen_freq_hptest)); + snd_soc_update_bits(codec, DA7219_TONE_GEN_CFG2, + DA7219_SWG_SEL_MASK | DA7219_TONE_GEN_GAIN_MASK, + DA7219_SWG_SEL_SRAMP | + DA7219_TONE_GEN_GAIN_MINUS_15DB); + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, DA7219_START_STOPN_MASK); + + msleep(DA7219_AAD_HPTEST_PERIOD); + + /* Grab comparator reading */ + accdet_cfg8 = snd_soc_read(codec, DA7219_ACCDET_CONFIG_8); + if (accdet_cfg8 & DA7219_HPTEST_COMP_MASK) + report |= SND_JACK_HEADPHONE; + else + report |= SND_JACK_LINEOUT; + + /* Stop tone generator */ + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0); + + msleep(DA7219_AAD_HPTEST_PERIOD); + + /* Restore original settings from cache */ + regcache_mark_dirty(da7219->regmap); + regcache_sync_region(da7219->regmap, DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DROUTING_ST_OUTFILT_1L, + DA7219_DROUTING_ST_OUTFILT_1R); + regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_R_SELECT); + regcache_sync_region(da7219->regmap, DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DIG_ROUTING_DAC, + DA7219_DIG_ROUTING_DAC); + regcache_sync_region(da7219->regmap, DA7219_CP_CTRL, DA7219_CP_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS5, + DA7219_DAC_FILTERS5); + regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS4, + DA7219_DAC_FILTERS1); + regcache_sync_region(da7219->regmap, DA7219_HP_L_GAIN, + DA7219_HP_R_GAIN); + regcache_sync_region(da7219->regmap, DA7219_DAC_L_GAIN, + DA7219_DAC_R_GAIN); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_ON_PER, + DA7219_TONE_GEN_ON_PER); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_FREQ1_L, + DA7219_TONE_GEN_FREQ1_U); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_CFG1, + DA7219_TONE_GEN_CFG2); + + regcache_cache_bypass(da7219->regmap, false); + + /* Disable HPTest block */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8, + DA7219_HPTEST_EN_MASK, 0); + + /* Drive Headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK, + DA7219_HP_L_AMP_OE_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK, + DA7219_HP_R_AMP_OE_MASK); + + mutex_unlock(&da7219->lock); + snd_soc_dapm_mutex_unlock(dapm); + + /* + * Only send report if jack hasn't been removed during process, + * otherwise it's invalid and we drop it. + */ + if (da7219_aad->jack_inserted) + snd_soc_jack_report(da7219_aad->jack, report, + SND_JACK_HEADSET | SND_JACK_LINEOUT); +} + + +/* + * IRQ + */ + +static irqreturn_t da7219_aad_irq_thread(int irq, void *data) +{ + struct da7219_aad_priv *da7219_aad = data; + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 events[DA7219_AAD_IRQ_REG_MAX]; + u8 statusa; + int i, report = 0, mask = 0; + + /* Read current IRQ events */ + regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + + if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) + return IRQ_NONE; + + /* Read status register for jack insertion & type status */ + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); + + /* Clear events */ + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + + dev_dbg(codec->dev, "IRQ events = 0x%x|0x%x, status = 0x%x\n", + events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B], + statusa); + + if (statusa & DA7219_JACK_INSERTION_STS_MASK) { + /* Jack Insertion */ + if (events[DA7219_AAD_IRQ_REG_A] & + DA7219_E_JACK_INSERTED_MASK) { + report |= SND_JACK_MECHANICAL; + mask |= SND_JACK_MECHANICAL; + da7219_aad->jack_inserted = true; + } + + /* Jack type detection */ + if (events[DA7219_AAD_IRQ_REG_A] & + DA7219_E_JACK_DETECT_COMPLETE_MASK) { + /* + * If 4-pole, then enable button detection, else perform + * HP impedance test to determine output type to report. + * + * We schedule work here as the tasks themselves can + * take time to complete, and in particular for hptest + * we want to be able to check if the jack was removed + * during the procedure as this will invalidate the + * result. By doing this as work, the IRQ thread can + * handle a removal, and we can check at the end of + * hptest if we have a valid result or not. + */ + if (statusa & DA7219_JACK_TYPE_STS_MASK) { + report |= SND_JACK_HEADSET; + mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT; + schedule_work(&da7219_aad->btn_det_work); + } else { + schedule_work(&da7219_aad->hptest_work); + } + } + + /* Button support for 4-pole jack */ + if (statusa & DA7219_JACK_TYPE_STS_MASK) { + for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) { + /* Button Press */ + if (events[DA7219_AAD_IRQ_REG_B] & + (DA7219_E_BUTTON_A_PRESSED_MASK << i)) { + report |= SND_JACK_BTN_0 >> i; + mask |= SND_JACK_BTN_0 >> i; + } + } + snd_soc_jack_report(da7219_aad->jack, report, mask); + + for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) { + /* Button Release */ + if (events[DA7219_AAD_IRQ_REG_B] & + (DA7219_E_BUTTON_A_RELEASED_MASK >> i)) { + report &= ~(SND_JACK_BTN_0 >> i); + mask |= SND_JACK_BTN_0 >> i; + } + } + } + } else { + /* Jack removal */ + if (events[DA7219_AAD_IRQ_REG_A] & DA7219_E_JACK_REMOVED_MASK) { + report = 0; + mask |= DA7219_AAD_REPORT_ALL_MASK; + da7219_aad->jack_inserted = false; + + /* Un-drive headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK, 0); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK, 0); + + /* Ensure button detection disabled */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, 0); + + /* Disable mic bias */ + snd_soc_dapm_disable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + + /* Cancel any pending work */ + cancel_work_sync(&da7219_aad->btn_det_work); + cancel_work_sync(&da7219_aad->hptest_work); + } + } + + snd_soc_jack_report(da7219_aad->jack, report, mask); + + return IRQ_HANDLED; +} + +/* + * DT to pdata conversion + */ + +static enum da7219_aad_micbias_pulse_lvl + da7219_aad_of_micbias_pulse_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 2800: + return DA7219_AAD_MICBIAS_PULSE_LVL_2_8V; + case 2900: + return DA7219_AAD_MICBIAS_PULSE_LVL_2_9V; + default: + dev_warn(codec->dev, "Invalid micbias pulse level"); + return DA7219_AAD_MICBIAS_PULSE_LVL_OFF; + } +} + +static enum da7219_aad_btn_cfg + da7219_aad_of_btn_cfg(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 2: + return DA7219_AAD_BTN_CFG_2MS; + case 5: + return DA7219_AAD_BTN_CFG_5MS; + case 10: + return DA7219_AAD_BTN_CFG_10MS; + case 50: + return DA7219_AAD_BTN_CFG_50MS; + case 100: + return DA7219_AAD_BTN_CFG_100MS; + case 200: + return DA7219_AAD_BTN_CFG_200MS; + case 500: + return DA7219_AAD_BTN_CFG_500MS; + default: + dev_warn(codec->dev, "Invalid button config"); + return DA7219_AAD_BTN_CFG_10MS; + } +} + +static enum da7219_aad_mic_det_thr + da7219_aad_of_mic_det_thr(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 200: + return DA7219_AAD_MIC_DET_THR_200_OHMS; + case 500: + return DA7219_AAD_MIC_DET_THR_500_OHMS; + case 750: + return DA7219_AAD_MIC_DET_THR_750_OHMS; + case 1000: + return DA7219_AAD_MIC_DET_THR_1000_OHMS; + default: + dev_warn(codec->dev, "Invalid mic detect threshold"); + return DA7219_AAD_MIC_DET_THR_500_OHMS; + } +} + +static enum da7219_aad_jack_ins_deb + da7219_aad_of_jack_ins_deb(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 5: + return DA7219_AAD_JACK_INS_DEB_5MS; + case 10: + return DA7219_AAD_JACK_INS_DEB_10MS; + case 20: + return DA7219_AAD_JACK_INS_DEB_20MS; + case 50: + return DA7219_AAD_JACK_INS_DEB_50MS; + case 100: + return DA7219_AAD_JACK_INS_DEB_100MS; + case 200: + return DA7219_AAD_JACK_INS_DEB_200MS; + case 500: + return DA7219_AAD_JACK_INS_DEB_500MS; + case 1000: + return DA7219_AAD_JACK_INS_DEB_1S; + default: + dev_warn(codec->dev, "Invalid jack insert debounce"); + return DA7219_AAD_JACK_INS_DEB_20MS; + } +} + +static enum da7219_aad_jack_det_rate + da7219_aad_of_jack_det_rate(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "32ms_64ms")) { + return DA7219_AAD_JACK_DET_RATE_32_64MS; + } else if (!strcmp(str, "64ms_128ms")) { + return DA7219_AAD_JACK_DET_RATE_64_128MS; + } else if (!strcmp(str, "128ms_256ms")) { + return DA7219_AAD_JACK_DET_RATE_128_256MS; + } else if (!strcmp(str, "256ms_512ms")) { + return DA7219_AAD_JACK_DET_RATE_256_512MS; + } else { + dev_warn(codec->dev, "Invalid jack detect rate"); + return DA7219_AAD_JACK_DET_RATE_256_512MS; + } +} + +static enum da7219_aad_jack_rem_deb + da7219_aad_of_jack_rem_deb(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_JACK_REM_DEB_1MS; + case 5: + return DA7219_AAD_JACK_REM_DEB_5MS; + case 10: + return DA7219_AAD_JACK_REM_DEB_10MS; + case 20: + return DA7219_AAD_JACK_REM_DEB_20MS; + default: + dev_warn(codec->dev, "Invalid jack removal debounce"); + return DA7219_AAD_JACK_REM_DEB_1MS; + } +} + +static enum da7219_aad_btn_avg + da7219_aad_of_btn_avg(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_BTN_AVG_1; + case 2: + return DA7219_AAD_BTN_AVG_2; + case 4: + return DA7219_AAD_BTN_AVG_4; + case 8: + return DA7219_AAD_BTN_AVG_8; + default: + dev_warn(codec->dev, "Invalid button average value"); + return DA7219_AAD_BTN_AVG_2; + } +} + +static enum da7219_aad_adc_1bit_rpt + da7219_aad_of_adc_1bit_rpt(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_ADC_1BIT_RPT_1; + case 2: + return DA7219_AAD_ADC_1BIT_RPT_2; + case 4: + return DA7219_AAD_ADC_1BIT_RPT_4; + case 8: + return DA7219_AAD_ADC_1BIT_RPT_8; + default: + dev_warn(codec->dev, "Invalid ADC 1-bit repeat value"); + return DA7219_AAD_ADC_1BIT_RPT_1; + } +} + +static struct da7219_aad_pdata *da7219_aad_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct device_node *aad_np = of_find_node_by_name(np, "da7219_aad"); + struct da7219_aad_pdata *aad_pdata; + const char *of_str; + u32 of_val32; + + if (!aad_np) + return NULL; + + aad_pdata = devm_kzalloc(codec->dev, sizeof(*aad_pdata), GFP_KERNEL); + if (!aad_pdata) + goto out; + + aad_pdata->irq = irq_of_parse_and_map(np, 0); + + if (of_property_read_u32(aad_np, "dlg,micbias-pulse-lvl", + &of_val32) >= 0) + aad_pdata->micbias_pulse_lvl = + da7219_aad_of_micbias_pulse_lvl(codec, of_val32); + else + aad_pdata->micbias_pulse_lvl = DA7219_AAD_MICBIAS_PULSE_LVL_OFF; + + if (of_property_read_u32(aad_np, "dlg,micbias-pulse-time", + &of_val32) >= 0) + aad_pdata->micbias_pulse_time = of_val32; + + if (of_property_read_u32(aad_np, "dlg,btn-cfg", &of_val32) >= 0) + aad_pdata->btn_cfg = da7219_aad_of_btn_cfg(codec, of_val32); + else + aad_pdata->btn_cfg = DA7219_AAD_BTN_CFG_10MS; + + if (of_property_read_u32(aad_np, "dlg,mic-det-thr", &of_val32) >= 0) + aad_pdata->mic_det_thr = + da7219_aad_of_mic_det_thr(codec, of_val32); + else + aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS; + + if (of_property_read_u32(aad_np, "dlg,jack-ins-deb", &of_val32) >= 0) + aad_pdata->jack_ins_deb = + da7219_aad_of_jack_ins_deb(codec, of_val32); + else + aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS; + + if (!of_property_read_string(aad_np, "dlg,jack-det-rate", &of_str)) + aad_pdata->jack_det_rate = + da7219_aad_of_jack_det_rate(codec, of_str); + else + aad_pdata->jack_det_rate = DA7219_AAD_JACK_DET_RATE_256_512MS; + + if (of_property_read_u32(aad_np, "dlg,jack-rem-deb", &of_val32) >= 0) + aad_pdata->jack_rem_deb = + da7219_aad_of_jack_rem_deb(codec, of_val32); + else + aad_pdata->jack_rem_deb = DA7219_AAD_JACK_REM_DEB_1MS; + + if (of_property_read_u32(aad_np, "dlg,a-d-btn-thr", &of_val32) >= 0) + aad_pdata->a_d_btn_thr = (u8) of_val32; + else + aad_pdata->a_d_btn_thr = 0xA; + + if (of_property_read_u32(aad_np, "dlg,d-b-btn-thr", &of_val32) >= 0) + aad_pdata->d_b_btn_thr = (u8) of_val32; + else + aad_pdata->d_b_btn_thr = 0x16; + + if (of_property_read_u32(aad_np, "dlg,b-c-btn-thr", &of_val32) >= 0) + aad_pdata->b_c_btn_thr = (u8) of_val32; + else + aad_pdata->b_c_btn_thr = 0x21; + + if (of_property_read_u32(aad_np, "dlg,c-mic-btn-thr", &of_val32) >= 0) + aad_pdata->c_mic_btn_thr = (u8) of_val32; + else + aad_pdata->c_mic_btn_thr = 0x3E; + + if (of_property_read_u32(aad_np, "dlg,btn-avg", &of_val32) >= 0) + aad_pdata->btn_avg = da7219_aad_of_btn_avg(codec, of_val32); + else + aad_pdata->btn_avg = DA7219_AAD_BTN_AVG_2; + + if (of_property_read_u32(aad_np, "dlg,adc-1bit-rpt", &of_val32) >= 0) + aad_pdata->adc_1bit_rpt = + da7219_aad_of_adc_1bit_rpt(codec, of_val32); + else + aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1; + +out: + of_node_put(aad_np); + + return aad_pdata; +} + +static void da7219_aad_handle_pdata(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + struct da7219_pdata *pdata = da7219->pdata; + + if ((pdata) && (pdata->aad_pdata)) { + struct da7219_aad_pdata *aad_pdata = pdata->aad_pdata; + u8 cfg, mask; + + da7219_aad->irq = aad_pdata->irq; + + switch (aad_pdata->micbias_pulse_lvl) { + case DA7219_AAD_MICBIAS_PULSE_LVL_2_8V: + case DA7219_AAD_MICBIAS_PULSE_LVL_2_9V: + da7219_aad->micbias_pulse_lvl = + (aad_pdata->micbias_pulse_lvl << + DA7219_MICBIAS1_LEVEL_SHIFT); + break; + default: + break; + } + + da7219_aad->micbias_pulse_time = aad_pdata->micbias_pulse_time; + + switch (aad_pdata->btn_cfg) { + case DA7219_AAD_BTN_CFG_2MS: + case DA7219_AAD_BTN_CFG_5MS: + case DA7219_AAD_BTN_CFG_10MS: + case DA7219_AAD_BTN_CFG_50MS: + case DA7219_AAD_BTN_CFG_100MS: + case DA7219_AAD_BTN_CFG_200MS: + case DA7219_AAD_BTN_CFG_500MS: + da7219_aad->btn_cfg = (aad_pdata->btn_cfg << + DA7219_BUTTON_CONFIG_SHIFT); + } + + cfg = 0; + mask = 0; + switch (aad_pdata->mic_det_thr) { + case DA7219_AAD_MIC_DET_THR_200_OHMS: + case DA7219_AAD_MIC_DET_THR_500_OHMS: + case DA7219_AAD_MIC_DET_THR_750_OHMS: + case DA7219_AAD_MIC_DET_THR_1000_OHMS: + cfg |= (aad_pdata->mic_det_thr << + DA7219_MIC_DET_THRESH_SHIFT); + mask |= DA7219_MIC_DET_THRESH_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, mask, cfg); + + cfg = 0; + mask = 0; + switch (aad_pdata->jack_ins_deb) { + case DA7219_AAD_JACK_INS_DEB_5MS: + case DA7219_AAD_JACK_INS_DEB_10MS: + case DA7219_AAD_JACK_INS_DEB_20MS: + case DA7219_AAD_JACK_INS_DEB_50MS: + case DA7219_AAD_JACK_INS_DEB_100MS: + case DA7219_AAD_JACK_INS_DEB_200MS: + case DA7219_AAD_JACK_INS_DEB_500MS: + case DA7219_AAD_JACK_INS_DEB_1S: + cfg |= (aad_pdata->jack_ins_deb << + DA7219_JACKDET_DEBOUNCE_SHIFT); + mask |= DA7219_JACKDET_DEBOUNCE_MASK; + } + switch (aad_pdata->jack_det_rate) { + case DA7219_AAD_JACK_DET_RATE_32_64MS: + case DA7219_AAD_JACK_DET_RATE_64_128MS: + case DA7219_AAD_JACK_DET_RATE_128_256MS: + case DA7219_AAD_JACK_DET_RATE_256_512MS: + cfg |= (aad_pdata->jack_det_rate << + DA7219_JACK_DETECT_RATE_SHIFT); + mask |= DA7219_JACK_DETECT_RATE_MASK; + } + switch (aad_pdata->jack_rem_deb) { + case DA7219_AAD_JACK_REM_DEB_1MS: + case DA7219_AAD_JACK_REM_DEB_5MS: + case DA7219_AAD_JACK_REM_DEB_10MS: + case DA7219_AAD_JACK_REM_DEB_20MS: + cfg |= (aad_pdata->jack_rem_deb << + DA7219_JACKDET_REM_DEB_SHIFT); + mask |= DA7219_JACKDET_REM_DEB_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_2, mask, cfg); + + snd_soc_write(codec, DA7219_ACCDET_CONFIG_3, + aad_pdata->a_d_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_4, + aad_pdata->d_b_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_5, + aad_pdata->b_c_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_6, + aad_pdata->c_mic_btn_thr); + + cfg = 0; + mask = 0; + switch (aad_pdata->btn_avg) { + case DA7219_AAD_BTN_AVG_1: + case DA7219_AAD_BTN_AVG_2: + case DA7219_AAD_BTN_AVG_4: + case DA7219_AAD_BTN_AVG_8: + cfg |= (aad_pdata->btn_avg << + DA7219_BUTTON_AVERAGE_SHIFT); + mask |= DA7219_BUTTON_AVERAGE_MASK; + } + switch (aad_pdata->adc_1bit_rpt) { + case DA7219_AAD_ADC_1BIT_RPT_1: + case DA7219_AAD_ADC_1BIT_RPT_2: + case DA7219_AAD_ADC_1BIT_RPT_4: + case DA7219_AAD_ADC_1BIT_RPT_8: + cfg |= (aad_pdata->adc_1bit_rpt << + DA7219_ADC_1_BIT_REPEAT_SHIFT); + mask |= DA7219_ADC_1_BIT_REPEAT_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_7, mask, cfg); + } +} + + +/* + * Init/Exit + */ + +int da7219_aad_init(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad; + u8 mask[DA7219_AAD_IRQ_REG_MAX]; + int ret; + + da7219_aad = devm_kzalloc(codec->dev, sizeof(*da7219_aad), GFP_KERNEL); + if (!da7219_aad) + return -ENOMEM; + + da7219->aad = da7219_aad; + da7219_aad->codec = codec; + + /* Handle any DT/platform data */ + if ((codec->dev->of_node) && (da7219->pdata)) + da7219->pdata->aad_pdata = da7219_aad_of_to_pdata(codec); + + da7219_aad_handle_pdata(codec); + + /* Disable button detection */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, 0); + + INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work); + INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); + + ret = request_threaded_irq(da7219_aad->irq, NULL, + da7219_aad_irq_thread, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "da7219-aad", da7219_aad); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + return ret; + } + + /* Unmask AAD IRQs */ + memset(mask, 0, DA7219_AAD_IRQ_REG_MAX); + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, + &mask, DA7219_AAD_IRQ_REG_MAX); + + return 0; +} +EXPORT_SYMBOL_GPL(da7219_aad_init); + +void da7219_aad_exit(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + u8 mask[DA7219_AAD_IRQ_REG_MAX]; + + /* Mask off AAD IRQs */ + memset(mask, DA7219_BYTE_MASK, DA7219_AAD_IRQ_REG_MAX); + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, + mask, DA7219_AAD_IRQ_REG_MAX); + + free_irq(da7219_aad->irq, da7219_aad); + + cancel_work_sync(&da7219_aad->btn_det_work); + cancel_work_sync(&da7219_aad->hptest_work); +} +EXPORT_SYMBOL_GPL(da7219_aad_exit); + +MODULE_DESCRIPTION("ASoC DA7219 AAD Driver"); +MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h new file mode 100644 index 0000000..4fccf67 --- /dev/null +++ b/sound/soc/codecs/da7219-aad.h @@ -0,0 +1,212 @@ +/* + * da7219-aad.h - DA7322 ASoC AAD Driver + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_AAD_H +#define __DA7219_AAD_H + +#include <linux/timer.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/da7219-aad.h> + +/* + * Registers + */ + +#define DA7219_ACCDET_STATUS_A 0xC0 +#define DA7219_ACCDET_STATUS_B 0xC1 +#define DA7219_ACCDET_IRQ_EVENT_A 0xC2 +#define DA7219_ACCDET_IRQ_EVENT_B 0xC3 +#define DA7219_ACCDET_IRQ_MASK_A 0xC4 +#define DA7219_ACCDET_IRQ_MASK_B 0xC5 +#define DA7219_ACCDET_CONFIG_1 0xC6 +#define DA7219_ACCDET_CONFIG_2 0xC7 +#define DA7219_ACCDET_CONFIG_3 0xC8 +#define DA7219_ACCDET_CONFIG_4 0xC9 +#define DA7219_ACCDET_CONFIG_5 0xCA +#define DA7219_ACCDET_CONFIG_6 0xCB +#define DA7219_ACCDET_CONFIG_7 0xCC +#define DA7219_ACCDET_CONFIG_8 0xCD + + +/* + * Bit Fields + */ + +/* DA7219_ACCDET_STATUS_A = 0xC0 */ +#define DA7219_JACK_INSERTION_STS_SHIFT 0 +#define DA7219_JACK_INSERTION_STS_MASK (0x1 << 0) +#define DA7219_JACK_TYPE_STS_SHIFT 1 +#define DA7219_JACK_TYPE_STS_MASK (0x1 << 1) +#define DA7219_JACK_PIN_ORDER_STS_SHIFT 2 +#define DA7219_JACK_PIN_ORDER_STS_MASK (0x1 << 2) +#define DA7219_MICBIAS_UP_STS_SHIFT 3 +#define DA7219_MICBIAS_UP_STS_MASK (0x1 << 3) + +/* DA7219_ACCDET_STATUS_B = 0xC1 */ +#define DA7219_BUTTON_TYPE_STS_SHIFT 0 +#define DA7219_BUTTON_TYPE_STS_MASK (0xFF << 0) + +/* DA7219_ACCDET_IRQ_EVENT_A = 0xC2 */ +#define DA7219_E_JACK_INSERTED_SHIFT 0 +#define DA7219_E_JACK_INSERTED_MASK (0x1 << 0) +#define DA7219_E_JACK_REMOVED_SHIFT 1 +#define DA7219_E_JACK_REMOVED_MASK (0x1 << 1) +#define DA7219_E_JACK_DETECT_COMPLETE_SHIFT 2 +#define DA7219_E_JACK_DETECT_COMPLETE_MASK (0x1 << 2) + +/* DA7219_ACCDET_IRQ_EVENT_B = 0xC3 */ +#define DA7219_E_BUTTON_A_PRESSED_SHIFT 0 +#define DA7219_E_BUTTON_A_PRESSED_MASK (0x1 << 0) +#define DA7219_E_BUTTON_B_PRESSED_SHIFT 1 +#define DA7219_E_BUTTON_B_PRESSED_MASK (0x1 << 1) +#define DA7219_E_BUTTON_C_PRESSED_SHIFT 2 +#define DA7219_E_BUTTON_C_PRESSED_MASK (0x1 << 2) +#define DA7219_E_BUTTON_D_PRESSED_SHIFT 3 +#define DA7219_E_BUTTON_D_PRESSED_MASK (0x1 << 3) +#define DA7219_E_BUTTON_D_RELEASED_SHIFT 4 +#define DA7219_E_BUTTON_D_RELEASED_MASK (0x1 << 4) +#define DA7219_E_BUTTON_C_RELEASED_SHIFT 5 +#define DA7219_E_BUTTON_C_RELEASED_MASK (0x1 << 5) +#define DA7219_E_BUTTON_B_RELEASED_SHIFT 6 +#define DA7219_E_BUTTON_B_RELEASED_MASK (0x1 << 6) +#define DA7219_E_BUTTON_A_RELEASED_SHIFT 7 +#define DA7219_E_BUTTON_A_RELEASED_MASK (0x1 << 7) + +/* DA7219_ACCDET_IRQ_MASK_A = 0xC4 */ +#define DA7219_M_JACK_INSERTED_SHIFT 0 +#define DA7219_M_JACK_INSERTED_MASK (0x1 << 0) +#define DA7219_M_JACK_REMOVED_SHIFT 1 +#define DA7219_M_JACK_REMOVED_MASK (0x1 << 1) +#define DA7219_M_JACK_DETECT_COMPLETE_SHIFT 2 +#define DA7219_M_JACK_DETECT_COMPLETE_MASK (0x1 << 2) + +/* DA7219_ACCDET_IRQ_MASK_B = 0xC5 */ +#define DA7219_M_BUTTON_A_PRESSED_SHIFT 0 +#define DA7219_M_BUTTON_A_PRESSED_MASK (0x1 << 0) +#define DA7219_M_BUTTON_B_PRESSED_SHIFT 1 +#define DA7219_M_BUTTON_B_PRESSED_MASK (0x1 << 1) +#define DA7219_M_BUTTON_C_PRESSED_SHIFT 2 +#define DA7219_M_BUTTON_C_PRESSED_MASK (0x1 << 2) +#define DA7219_M_BUTTON_D_PRESSED_SHIFT 3 +#define DA7219_M_BUTTON_D_PRESSED_MASK (0x1 << 3) +#define DA7219_M_BUTTON_D_RELEASED_SHIFT 4 +#define DA7219_M_BUTTON_D_RELEASED_MASK (0x1 << 4) +#define DA7219_M_BUTTON_C_RELEASED_SHIFT 5 +#define DA7219_M_BUTTON_C_RELEASED_MASK (0x1 << 5) +#define DA7219_M_BUTTON_B_RELEASED_SHIFT 6 +#define DA7219_M_BUTTON_B_RELEASED_MASK (0x1 << 6) +#define DA7219_M_BUTTON_A_RELEASED_SHIFT 7 +#define DA7219_M_BUTTON_A_RELEASED_MASK (0x1 << 7) + +/* DA7219_ACCDET_CONFIG_1 = 0xC6 */ +#define DA7219_ACCDET_EN_SHIFT 0 +#define DA7219_ACCDET_EN_MASK (0x1 << 0) +#define DA7219_BUTTON_CONFIG_SHIFT 1 +#define DA7219_BUTTON_CONFIG_MASK (0x7 << 1) +#define DA7219_MIC_DET_THRESH_SHIFT 4 +#define DA7219_MIC_DET_THRESH_MASK (0x3 << 4) +#define DA7219_JACK_TYPE_DET_EN_SHIFT 6 +#define DA7219_JACK_TYPE_DET_EN_MASK (0x1 << 6) +#define DA7219_PIN_ORDER_DET_EN_SHIFT 7 +#define DA7219_PIN_ORDER_DET_EN_MASK (0x1 << 7) + +/* DA7219_ACCDET_CONFIG_2 = 0xC7 */ +#define DA7219_ACCDET_PAUSE_SHIFT 0 +#define DA7219_ACCDET_PAUSE_MASK (0x1 << 0) +#define DA7219_JACKDET_DEBOUNCE_SHIFT 1 +#define DA7219_JACKDET_DEBOUNCE_MASK (0x7 << 1) +#define DA7219_JACK_DETECT_RATE_SHIFT 4 +#define DA7219_JACK_DETECT_RATE_MASK (0x3 << 4) +#define DA7219_JACKDET_REM_DEB_SHIFT 6 +#define DA7219_JACKDET_REM_DEB_MASK (0x3 << 6) + +/* DA7219_ACCDET_CONFIG_3 = 0xC8 */ +#define DA7219_A_D_BUTTON_THRESH_SHIFT 0 +#define DA7219_A_D_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_4 = 0xC9 */ +#define DA7219_D_B_BUTTON_THRESH_SHIFT 0 +#define DA7219_D_B_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_5 = 0xCA */ +#define DA7219_B_C_BUTTON_THRESH_SHIFT 0 +#define DA7219_B_C_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_6 = 0xCB */ +#define DA7219_C_MIC_BUTTON_THRESH_SHIFT 0 +#define DA7219_C_MIC_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_7 = 0xCC */ +#define DA7219_BUTTON_AVERAGE_SHIFT 0 +#define DA7219_BUTTON_AVERAGE_MASK (0x3 << 0) +#define DA7219_ADC_1_BIT_REPEAT_SHIFT 2 +#define DA7219_ADC_1_BIT_REPEAT_MASK (0x3 << 2) +#define DA7219_PIN_ORDER_FORCE_SHIFT 4 +#define DA7219_PIN_ORDER_FORCE_MASK (0x1 << 4) +#define DA7219_JACK_TYPE_FORCE_SHIFT 5 +#define DA7219_JACK_TYPE_FORCE_MASK (0x1 << 5) + +/* DA7219_ACCDET_CONFIG_8 = 0xCD */ +#define DA7219_HPTEST_EN_SHIFT 0 +#define DA7219_HPTEST_EN_MASK (0x1 << 0) +#define DA7219_HPTEST_RES_SEL_SHIFT 1 +#define DA7219_HPTEST_RES_SEL_MASK (0x3 << 1) +#define DA7219_HPTEST_RES_SEL_1KOHMS (0x0 << 1) +#define DA7219_HPTEST_COMP_SHIFT 4 +#define DA7219_HPTEST_COMP_MASK (0x1 << 4) + + +#define DA7219_AAD_MAX_BUTTONS 4 +#define DA7219_AAD_REPORT_ALL_MASK (SND_JACK_MECHANICAL | \ + SND_JACK_HEADSET | SND_JACK_LINEOUT | \ + SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +#define DA7219_AAD_MICBIAS_CHK_DELAY 10 +#define DA7219_AAD_MICBIAS_CHK_RETRIES 5 + +#define DA7219_AAD_HPTEST_RAMP_FREQ 0x28 +#define DA7219_AAD_HPTEST_PERIOD 65 + +enum da7219_aad_event_regs { + DA7219_AAD_IRQ_REG_A = 0, + DA7219_AAD_IRQ_REG_B, + DA7219_AAD_IRQ_REG_MAX, +}; + +/* Private data */ +struct da7219_aad_priv { + struct snd_soc_codec *codec; + int irq; + + u8 micbias_pulse_lvl; + u32 micbias_pulse_time; + + u8 btn_cfg; + + struct work_struct btn_det_work; + struct work_struct hptest_work; + + struct snd_soc_jack *jack; + bool jack_inserted; +}; + +/* AAD control */ +void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +/* Init/Exit */ +int da7219_aad_init(struct snd_soc_codec *codec); +void da7219_aad_exit(struct snd_soc_codec *codec); + +#endif /* __DA7219_AAD_H */ diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c new file mode 100644 index 0000000..76f8fc2 --- /dev/null +++ b/sound/soc/codecs/da7219.c @@ -0,0 +1,1940 @@ +/* + * da7219.c - DA7219 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/pm.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/regulator/consumer.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 <asm/div64.h> + +#include <sound/da7219.h> +#include "da7219.h" +#include "da7219-aad.h" + + +/* + * TLVs and Enums + */ + +/* Input TLVs */ +static const DECLARE_TLV_DB_SCALE(da7219_mic_gain_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7219_adc_dig_gain_tlv, -8325, 75, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_ana_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_sidetone_gain_tlv, -4200, 300, 0); +static const DECLARE_TLV_DB_SCALE(da7219_tonegen_gain_tlv, -4500, 300, 0); + +/* Output TLVs */ +static const DECLARE_TLV_DB_SCALE(da7219_dac_eq_band_tlv, -1050, 150, 0); + +static const DECLARE_TLV_DB_RANGE(da7219_dac_dig_gain_tlv, + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -77.25dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7725, 75, 0) +); + +static const DECLARE_TLV_DB_SCALE(da7219_dac_ng_threshold_tlv, -10200, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_hp_gain_tlv, -5700, 100, 0); + +/* Input Enums */ +static const char * const da7219_alc_attack_rate_txt[] = { + "7.33/fs", "14.66/fs", "29.32/fs", "58.64/fs", "117.3/fs", "234.6/fs", + "469.1/fs", "938.2/fs", "1876/fs", "3753/fs", "7506/fs", "15012/fs", + "30024/fs" +}; + +static const struct soc_enum da7219_alc_attack_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_ATTACK_SHIFT, + DA7219_ALC_ATTACK_MAX, da7219_alc_attack_rate_txt); + +static const char * const da7219_alc_release_rate_txt[] = { + "28.66/fs", "57.33/fs", "114.6/fs", "229.3/fs", "458.6/fs", "917.1/fs", + "1834/fs", "3668/fs", "7337/fs", "14674/fs", "29348/fs" +}; + +static const struct soc_enum da7219_alc_release_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_RELEASE_SHIFT, + DA7219_ALC_RELEASE_MAX, da7219_alc_release_rate_txt); + +static const char * const da7219_alc_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da7219_alc_hold_time = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_HOLD_SHIFT, + DA7219_ALC_HOLD_MAX, da7219_alc_hold_time_txt); + +static const char * const da7219_alc_env_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da7219_alc_env_attack_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_ATTACK_SHIFT, + DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt); + +static const struct soc_enum da7219_alc_env_release_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_RELEASE_SHIFT, + DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt); + +static const char * const da7219_alc_anticlip_step_txt[] = { + "0.034dB/fs", "0.068dB/fs", "0.136dB/fs", "0.272dB/fs" +}; + +static const struct soc_enum da7219_alc_anticlip_step = + SOC_ENUM_SINGLE(DA7219_ALC_ANTICLIP_CTRL, + DA7219_ALC_ANTICLIP_STEP_SHIFT, + DA7219_ALC_ANTICLIP_STEP_MAX, + da7219_alc_anticlip_step_txt); + +/* Input/Output Enums */ +static const char * const da7219_gain_ramp_rate_txt[] = { + "Nominal Rate * 8", "Nominal Rate", "Nominal Rate / 8", + "Nominal Rate / 16" +}; + +static const struct soc_enum da7219_gain_ramp_rate = + SOC_ENUM_SINGLE(DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_SHIFT, + DA7219_GAIN_RAMP_RATE_MAX, da7219_gain_ramp_rate_txt); + +static const char * const da7219_hpf_mode_txt[] = { + "Disabled", "Audio", "Voice" +}; + +static const unsigned int da7219_hpf_mode_val[] = { + DA7219_HPF_DISABLED, DA7219_HPF_AUDIO_EN, DA7219_HPF_VOICE_EN, +}; + +static const struct soc_enum da7219_adc_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_ADC_FILTERS1, DA7219_HPF_MODE_SHIFT, + DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX, + da7219_hpf_mode_txt, da7219_hpf_mode_val); + +static const struct soc_enum da7219_dac_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_DAC_FILTERS1, DA7219_HPF_MODE_SHIFT, + DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX, + da7219_hpf_mode_txt, da7219_hpf_mode_val); + +static const char * const da7219_audio_hpf_corner_txt[] = { + "2Hz", "4Hz", "8Hz", "16Hz" +}; + +static const struct soc_enum da7219_adc_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1, + DA7219_ADC_AUDIO_HPF_CORNER_SHIFT, + DA7219_AUDIO_HPF_CORNER_MAX, + da7219_audio_hpf_corner_txt); + +static const struct soc_enum da7219_dac_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1, + DA7219_DAC_AUDIO_HPF_CORNER_SHIFT, + DA7219_AUDIO_HPF_CORNER_MAX, + da7219_audio_hpf_corner_txt); + +static const char * const da7219_voice_hpf_corner_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7219_adc_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1, + DA7219_ADC_VOICE_HPF_CORNER_SHIFT, + DA7219_VOICE_HPF_CORNER_MAX, + da7219_voice_hpf_corner_txt); + +static const struct soc_enum da7219_dac_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1, + DA7219_DAC_VOICE_HPF_CORNER_SHIFT, + DA7219_VOICE_HPF_CORNER_MAX, + da7219_voice_hpf_corner_txt); + +static const char * const da7219_tonegen_dtmf_key_txt[] = { + "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D", + "*", "#" +}; + +static const struct soc_enum da7219_tonegen_dtmf_key = + SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG1, DA7219_DTMF_REG_SHIFT, + DA7219_DTMF_REG_MAX, da7219_tonegen_dtmf_key_txt); + +static const char * const da7219_tonegen_swg_sel_txt[] = { + "Sum", "SWG1", "SWG2", "SWG1_1-Cos" +}; + +static const struct soc_enum da7219_tonegen_swg_sel = + SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG2, DA7219_SWG_SEL_SHIFT, + DA7219_SWG_SEL_MAX, da7219_tonegen_swg_sel_txt); + +/* Output Enums */ +static const char * const da7219_dac_softmute_rate_txt[] = { + "1 Sample", "2 Samples", "4 Samples", "8 Samples", "16 Samples", + "32 Samples", "64 Samples" +}; + +static const struct soc_enum da7219_dac_softmute_rate = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS5, DA7219_DAC_SOFTMUTE_RATE_SHIFT, + DA7219_DAC_SOFTMUTE_RATE_MAX, + da7219_dac_softmute_rate_txt); + +static const char * const da7219_dac_ng_setup_time_txt[] = { + "256 Samples", "512 Samples", "1024 Samples", "2048 Samples" +}; + +static const struct soc_enum da7219_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_SETUP_TIME_SHIFT, + DA7219_DAC_NG_SETUP_TIME_MAX, + da7219_dac_ng_setup_time_txt); + +static const char * const da7219_dac_ng_rampup_txt[] = { + "0.22ms/dB", "0.0138ms/dB" +}; + +static const struct soc_enum da7219_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_RAMPUP_RATE_SHIFT, + DA7219_DAC_NG_RAMP_RATE_MAX, + da7219_dac_ng_rampup_txt); + +static const char * const da7219_dac_ng_rampdown_txt[] = { + "0.88ms/dB", "14.08ms/dB" +}; + +static const struct soc_enum da7219_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_RAMPDN_RATE_SHIFT, + DA7219_DAC_NG_RAMP_RATE_MAX, + da7219_dac_ng_rampdown_txt); + + +static const char * const da7219_cp_track_mode_txt[] = { + "Largest Volume", "DAC Volume", "Signal Magnitude" +}; + +static const unsigned int da7219_cp_track_mode_val[] = { + DA7219_CP_MCHANGE_LARGEST_VOL, DA7219_CP_MCHANGE_DAC_VOL, + DA7219_CP_MCHANGE_SIG_MAG +}; + +static const struct soc_enum da7219_cp_track_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_CP_CTRL, DA7219_CP_MCHANGE_SHIFT, + DA7219_CP_MCHANGE_REL_MASK, DA7219_CP_MCHANGE_MAX, + da7219_cp_track_mode_txt, + da7219_cp_track_mode_val); + + +/* + * Control Functions + */ + +/* Locked Kcontrol calls */ +static int da7219_volsw_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_get_volsw(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_volsw_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_put_volsw(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_enum_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_get_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_put_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +/* ALC */ +static void da7219_alc_calib(struct snd_soc_codec *codec) +{ + u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl; + + /* Save current state of mic control register */ + mic_ctrl = snd_soc_read(codec, DA7219_MIC_1_CTRL); + + /* Save current state of input mixer control register */ + mixin_ctrl = snd_soc_read(codec, DA7219_MIXIN_L_CTRL); + + /* Save current state of input ADC control register */ + adc_ctrl = snd_soc_read(codec, DA7219_ADC_L_CTRL); + + /* Enable then Mute MIC PGAs */ + snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK, + DA7219_MIC_1_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_MUTE_EN_MASK, + DA7219_MIC_1_AMP_MUTE_EN_MASK); + + /* Enable input mixers unmuted */ + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_MASK | + DA7219_MIXIN_L_AMP_MUTE_EN_MASK, + DA7219_MIXIN_L_AMP_EN_MASK); + + /* Enable input filters unmuted */ + snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, + DA7219_ADC_L_MUTE_EN_MASK | DA7219_ADC_L_EN_MASK, + DA7219_ADC_L_EN_MASK); + + /* Perform auto calibration */ + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_AUTO_CALIB_EN_MASK, + DA7219_ALC_AUTO_CALIB_EN_MASK); + do { + calib_ctrl = snd_soc_read(codec, DA7219_ALC_CTRL1); + } while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK); + + /* If auto calibration fails, disable DC offset, hybrid ALC */ + if (calib_ctrl & DA7219_ALC_CALIB_OVERFLOW_MASK) { + dev_warn(codec->dev, + "ALC auto calibration failed with overflow\n"); + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK, 0); + } else { + /* Enable DC offset cancellation, hybrid mode */ + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK); + } + + /* Restore input filter control register to original state */ + snd_soc_write(codec, DA7219_ADC_L_CTRL, adc_ctrl); + + /* Restore input mixer control registers to original state */ + snd_soc_write(codec, DA7219_MIXIN_L_CTRL, mixin_ctrl); + + /* Restore MIC control registers to original states */ + snd_soc_write(codec, DA7219_MIC_1_CTRL, mic_ctrl); +} + +static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + + /* + * If ALC in operation and value of control has been updated, + * make sure calibrated offsets are updated. + */ + if ((ret == 1) && (da7219->alc_en)) + da7219_alc_calib(codec); + + return ret; +} + +static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + + /* Force ALC offset calibration if enabling ALC */ + if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) { + da7219_alc_calib(codec); + da7219->alc_en = false; + } else { + da7219->alc_en = false; + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* ToneGen */ +static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + int ret; + + mutex_lock(&da7219->lock); + ret = regmap_raw_read(da7219->regmap, reg, &val, sizeof(val)); + mutex_unlock(&da7219->lock); + + if (ret) + return ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to host endianness here. + */ + ucontrol->value.integer.value[0] = le16_to_cpu(val); + + return 0; +} + +static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + int ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to little endian here to align with + * HW registers. + */ + val = cpu_to_le16(ucontrol->value.integer.value[0]); + + mutex_lock(&da7219->lock); + ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val)); + mutex_unlock(&da7219->lock); + + return ret; +} + + +/* + * KControls + */ + +static const struct snd_kcontrol_new da7219_snd_controls[] = { + /* Mics */ + SOC_SINGLE_TLV("Mic Volume", DA7219_MIC_1_GAIN, + DA7219_MIC_1_AMP_GAIN_SHIFT, DA7219_MIC_1_AMP_GAIN_MAX, + DA7219_NO_INVERT, da7219_mic_gain_tlv), + SOC_SINGLE("Mic Switch", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + + /* Mixer Input */ + SOC_SINGLE_EXT_TLV("Mixin Volume", DA7219_MIXIN_L_GAIN, + DA7219_MIXIN_L_AMP_GAIN_SHIFT, + DA7219_MIXIN_L_AMP_GAIN_MAX, DA7219_NO_INVERT, + snd_soc_get_volsw, da7219_mixin_gain_put, + da7219_mixin_gain_tlv), + SOC_SINGLE("Mixin Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + SOC_SINGLE("Mixin Gain Ramp Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + SOC_SINGLE("Mixin ZC Gain Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_ZC_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + + /* ADC */ + SOC_SINGLE_TLV("Capture Digital Volume", DA7219_ADC_L_GAIN, + DA7219_ADC_L_DIGITAL_GAIN_SHIFT, + DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, + da7219_adc_dig_gain_tlv), + SOC_SINGLE("Capture Digital Switch", DA7219_ADC_L_CTRL, + DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + SOC_SINGLE("Capture Digital Gain Ramp Switch", DA7219_ADC_L_CTRL, + DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + + /* ALC */ + SOC_ENUM("ALC Attack Rate", da7219_alc_attack_rate), + SOC_ENUM("ALC Release Rate", da7219_alc_release_rate), + SOC_ENUM("ALC Hold Time", da7219_alc_hold_time), + SOC_ENUM("ALC Envelope Attack Rate", da7219_alc_env_attack_rate), + SOC_ENUM("ALC Envelope Release Rate", da7219_alc_env_release_rate), + SOC_SINGLE_TLV("ALC Noise Threshold", DA7219_ALC_NOISE, + DA7219_ALC_NOISE_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Min Threshold", DA7219_ALC_TARGET_MIN, + DA7219_ALC_THRESHOLD_MIN_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold", DA7219_ALC_TARGET_MAX, + DA7219_ALC_THRESHOLD_MAX_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS, + DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Volume", DA7219_ALC_GAIN_LIMITS, + DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Min Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS, + DA7219_ALC_ANA_GAIN_MIN_SHIFT, + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Max Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS, + DA7219_ALC_ANA_GAIN_MAX_SHIFT, + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), + SOC_ENUM("ALC Anticlip Step", da7219_alc_anticlip_step), + SOC_SINGLE("ALC Anticlip Switch", DA7219_ALC_ANTICLIP_CTRL, + DA7219_ALC_ANTIPCLIP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + SOC_SINGLE_EXT("ALC Switch", DA7219_ALC_CTRL1, DA7219_ALC_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT, + snd_soc_get_volsw, da7219_alc_sw_put), + + /* Input High-Pass Filters */ + SOC_ENUM("ADC HPF Mode", da7219_adc_hpf_mode), + SOC_ENUM("ADC HPF Corner Audio", da7219_adc_audio_hpf_corner), + SOC_ENUM("ADC HPF Corner Voice", da7219_adc_voice_hpf_corner), + + /* Sidetone Filter */ + SOC_SINGLE_TLV("Sidetone Volume", DA7219_SIDETONE_GAIN, + DA7219_SIDETONE_GAIN_SHIFT, DA7219_SIDETONE_GAIN_MAX, + DA7219_NO_INVERT, da7219_sidetone_gain_tlv), + SOC_SINGLE("Sidetone Switch", DA7219_SIDETONE_CTRL, + DA7219_SIDETONE_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + + /* Tone Generator */ + SOC_SINGLE_EXT_TLV("ToneGen Volume", DA7219_TONE_GEN_CFG2, + DA7219_TONE_GEN_GAIN_SHIFT, DA7219_TONE_GEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put, da7219_tonegen_gain_tlv), + SOC_ENUM_EXT("ToneGen DTMF Key", da7219_tonegen_dtmf_key, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_SINGLE_EXT("ToneGen DTMF Switch", DA7219_TONE_GEN_CFG1, + DA7219_DTMF_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_ENUM_EXT("ToneGen Sinewave Gen Type", da7219_tonegen_swg_sel, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_SINGLE_EXT("ToneGen Sinewave1 Freq", DA7219_TONE_GEN_FREQ1_L, + DA7219_FREQ1_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT, + da7219_tonegen_freq_get, da7219_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen Sinewave2 Freq", DA7219_TONE_GEN_FREQ2_L, + DA7219_FREQ2_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT, + da7219_tonegen_freq_get, da7219_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen On Time", DA7219_TONE_GEN_ON_PER, + DA7219_BEEP_ON_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_SINGLE("ToneGen Off Time", DA7219_TONE_GEN_OFF_PER, + DA7219_BEEP_OFF_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX, + DA7219_NO_INVERT), + + /* Gain ramping */ + SOC_ENUM("Gain Ramp Rate", da7219_gain_ramp_rate), + + /* DAC High-Pass Filter */ + SOC_ENUM_EXT("DAC HPF Mode", da7219_dac_hpf_mode, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_ENUM("DAC HPF Corner Audio", da7219_dac_audio_hpf_corner), + SOC_ENUM("DAC HPF Corner Voice", da7219_dac_voice_hpf_corner), + + /* DAC 5-Band Equaliser */ + SOC_SINGLE_TLV("DAC EQ Band1 Volume", DA7219_DAC_FILTERS2, + DA7219_DAC_EQ_BAND1_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band2 Volume", DA7219_DAC_FILTERS2, + DA7219_DAC_EQ_BAND2_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band3 Volume", DA7219_DAC_FILTERS3, + DA7219_DAC_EQ_BAND3_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band4 Volume", DA7219_DAC_FILTERS3, + DA7219_DAC_EQ_BAND4_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band5 Volume", DA7219_DAC_FILTERS4, + DA7219_DAC_EQ_BAND5_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_EXT("DAC EQ Switch", DA7219_DAC_FILTERS4, + DA7219_DAC_EQ_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + + /* DAC Softmute */ + SOC_ENUM("DAC Soft Mute Rate", da7219_dac_softmute_rate), + SOC_SINGLE_EXT("DAC Soft Mute Switch", DA7219_DAC_FILTERS5, + DA7219_DAC_SOFTMUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + + /* DAC Noise Gate */ + SOC_ENUM("DAC NG Setup Time", da7219_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da7219_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da7219_dac_ng_rampdown_rate), + SOC_SINGLE_TLV("DAC NG Off Threshold", DA7219_DAC_NG_OFF_THRESH, + DA7219_DAC_NG_OFF_THRESHOLD_SHIFT, + DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT, + da7219_dac_ng_threshold_tlv), + SOC_SINGLE_TLV("DAC NG On Threshold", DA7219_DAC_NG_ON_THRESH, + DA7219_DAC_NG_ON_THRESHOLD_SHIFT, + DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT, + da7219_dac_ng_threshold_tlv), + SOC_SINGLE("DAC NG Switch", DA7219_DAC_NG_CTRL, DA7219_DAC_NG_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + + /* DACs */ + SOC_DOUBLE_R_EXT_TLV("Playback Digital Volume", DA7219_DAC_L_GAIN, + DA7219_DAC_R_GAIN, DA7219_DAC_L_DIGITAL_GAIN_SHIFT, + DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put, + da7219_dac_dig_gain_tlv), + SOC_DOUBLE_R_EXT("Playback Digital Switch", DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL, DA7219_DAC_L_MUTE_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put), + SOC_DOUBLE_R("Playback Digital Gain Ramp Switch", DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + + /* CP */ + SOC_ENUM("Charge Pump Track Mode", da7219_cp_track_mode), + SOC_SINGLE("Charge Pump Threshold", DA7219_CP_VOL_THRESHOLD1, + DA7219_CP_THRESH_VDD2_SHIFT, DA7219_CP_THRESH_VDD2_MAX, + DA7219_NO_INVERT), + + /* Headphones */ + SOC_DOUBLE_R_EXT_TLV("Headphone Volume", DA7219_HP_L_GAIN, + DA7219_HP_R_GAIN, DA7219_HP_L_AMP_GAIN_SHIFT, + DA7219_HP_AMP_GAIN_MAX, DA7219_NO_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put, + da7219_hp_gain_tlv), + SOC_DOUBLE_R_EXT("Headphone Switch", DA7219_HP_L_CTRL, DA7219_HP_R_CTRL, + DA7219_HP_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_DOUBLE_R("Headphone Gain Ramp Switch", DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL, DA7219_HP_L_AMP_RAMP_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + SOC_DOUBLE_R("Headphone ZC Gain Switch", DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL, DA7219_HP_L_AMP_ZC_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + + +/* + * DAPM Mux Controls + */ + +static const char * const da7219_out_sel_txt[] = { + "ADC", "Tone Generator", "DAIL", "DAIR" +}; + +static const struct soc_enum da7219_out_dail_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI, + DA7219_DAI_L_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dail_sel_mux = + SOC_DAPM_ENUM("Out DAIL Mux", da7219_out_dail_sel); + +static const struct soc_enum da7219_out_dair_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI, + DA7219_DAI_R_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dair_sel_mux = + SOC_DAPM_ENUM("Out DAIR Mux", da7219_out_dair_sel); + +static const struct soc_enum da7219_out_dacl_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC, + DA7219_DAC_L_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dacl_sel_mux = + SOC_DAPM_ENUM("Out DACL Mux", da7219_out_dacl_sel); + +static const struct soc_enum da7219_out_dacr_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC, + DA7219_DAC_R_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dacr_sel_mux = + SOC_DAPM_ENUM("Out DACR Mux", da7219_out_dacr_sel); + + +/* + * DAPM Mixer Controls + */ + +static const struct snd_kcontrol_new da7219_mixin_controls[] = { + SOC_DAPM_SINGLE("Mic Switch", DA7219_MIXIN_L_SELECT, + DA7219_MIXIN_L_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +static const struct snd_kcontrol_new da7219_mixout_l_controls[] = { + SOC_DAPM_SINGLE("DACL Switch", DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_L_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +static const struct snd_kcontrol_new da7219_mixout_r_controls[] = { + SOC_DAPM_SINGLE("DACR Switch", DA7219_MIXOUT_R_SELECT, + DA7219_MIXOUT_R_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +#define DA7219_DMIX_ST_CTRLS(reg) \ + SOC_DAPM_SINGLE("Out FilterL Switch", reg, \ + DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \ + SOC_DAPM_SINGLE("Out FilterR Switch", reg, \ + DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \ + SOC_DAPM_SINGLE("Sidetone Switch", reg, \ + DA7219_DMIX_ST_SRC_SIDETONE_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT) \ + +static const struct snd_kcontrol_new da7219_st_out_filtl_mix_controls[] = { + DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1L), +}; + +static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = { + DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1R), +}; + + +/* + * DAPM Events + */ + +static int da7219_dai_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 da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 pll_ctrl, pll_status; + int i = 0; + bool srm_lock = false; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (da7219->master) + /* Enable DAI clks for master mode */ + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, + DA7219_DAI_CLK_EN_MASK); + + /* PC synchronised to DAI */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, 0); + + /* Slave mode, if SRM not enabled no need for status checks */ + pll_ctrl = snd_soc_read(codec, DA7219_PLL_CTRL); + if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) + return 0; + + /* Check SRM has locked */ + do { + pll_status = snd_soc_read(codec, DA7219_PLL_SRM_STS); + if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { + srm_lock = true; + } else { + ++i; + msleep(50); + } + } while ((i < DA7219_SRM_CHECK_RETRIES) & (!srm_lock)); + + if (!srm_lock) + dev_warn(codec->dev, "SRM failed to lock\n"); + + return 0; + case SND_SOC_DAPM_POST_PMD: + /* PC free-running */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, + DA7219_PC_FREERUN_MASK); + + /* Disable DAI clks if in master mode */ + if (da7219->master) + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, 0); + return 0; + default: + return -EINVAL; + } +} + + +/* + * DAPM Widgets + */ + +static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = { + /* Input Supplies */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7219_MICBIAS_CTRL, + DA7219_MICBIAS1_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Input Filters */ + SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT, + DA7219_NO_INVERT), + + /* Tone Generator */ + SND_SOC_DAPM_SIGGEN("TONE"), + SND_SOC_DAPM_PGA("Tone Generator", DA7219_TONE_GEN_CFG1, + DA7219_START_STOPN_SHIFT, DA7219_NO_INVERT, NULL, 0), + + /* Sidetone Input */ + SND_SOC_DAPM_ADC("Sidetone Filter", NULL, DA7219_SIDETONE_CTRL, + DA7219_SIDETONE_EN_SHIFT, DA7219_NO_INVERT), + + /* Input Mixer Supply */ + SND_SOC_DAPM_SUPPLY("Mixer In Supply", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_MIX_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Input Mixer */ + SND_SOC_DAPM_MIXER("Mixer In", SND_SOC_NOPM, 0, 0, + da7219_mixin_controls, + ARRAY_SIZE(da7219_mixin_controls)), + + /* Input Muxes */ + SND_SOC_DAPM_MUX("Out DAIL Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dail_sel_mux), + SND_SOC_DAPM_MUX("Out DAIR Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dair_sel_mux), + + /* DAI Supply */ + SND_SOC_DAPM_SUPPLY("DAI", DA7219_DAI_CTRL, DA7219_DAI_EN_SHIFT, + DA7219_NO_INVERT, da7219_dai_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* DAI */ + SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DAIIN", "Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Muxes */ + SND_SOC_DAPM_MUX("Out DACL Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dacl_sel_mux), + SND_SOC_DAPM_MUX("Out DACR Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dacr_sel_mux), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7219_mixout_l_controls, + ARRAY_SIZE(da7219_mixout_l_controls)), + SND_SOC_DAPM_MIXER("Mixer Out FilterR", SND_SOC_NOPM, 0, 0, + da7219_mixout_r_controls, + ARRAY_SIZE(da7219_mixout_r_controls)), + + /* Sidetone Mixers */ + SND_SOC_DAPM_MIXER("ST Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7219_st_out_filtl_mix_controls, + ARRAY_SIZE(da7219_st_out_filtl_mix_controls)), + SND_SOC_DAPM_MIXER("ST Mixer Out FilterR", SND_SOC_NOPM, 0, + 0, da7219_st_out_filtr_mix_controls, + ARRAY_SIZE(da7219_st_out_filtr_mix_controls)), + + /* DACs */ + SND_SOC_DAPM_DAC("DACL", NULL, DA7219_DAC_L_CTRL, DA7219_DAC_L_EN_SHIFT, + DA7219_NO_INVERT), + SND_SOC_DAPM_DAC("DACR", NULL, DA7219_DAC_R_CTRL, DA7219_DAC_R_EN_SHIFT, + DA7219_NO_INVERT), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("Mixout Left PGA", DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixout Right PGA", DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left PGA", DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right PGA", DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + + /* Output Supplies */ + SND_SOC_DAPM_SUPPLY("Charge Pump", DA7219_CP_CTRL, DA7219_CP_EN_SHIFT, + DA7219_NO_INVERT, NULL, 0), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + + +/* + * DAPM Mux Routes + */ + +#define DA7219_OUT_DAI_MUX_ROUTES(name) \ + {name, "ADC", "Mixer In"}, \ + {name, "Tone Generator", "Tone Generator"}, \ + {name, "DAIL", "DAIOUT"}, \ + {name, "DAIR", "DAIOUT"} + +#define DA7219_OUT_DAC_MUX_ROUTES(name) \ + {name, "ADC", "Mixer In"}, \ + {name, "Tone Generator", "Tone Generator"}, \ + {name, "DAIL", "DAIIN"}, \ + {name, "DAIR", "DAIIN"} + +/* + * DAPM Mixer Routes + */ + +#define DA7219_DMIX_ST_ROUTES(name) \ + {name, "Out FilterL Switch", "Mixer Out FilterL"}, \ + {name, "Out FilterR Switch", "Mixer Out FilterR"}, \ + {name, "Sidetone Switch", "Sidetone Filter"} + + +/* + * DAPM audio route definition + */ + +static const struct snd_soc_dapm_route da7219_audio_map[] = { + /* Input paths */ + {"MIC", NULL, "Mic Bias"}, + {"Mic PGA", NULL, "MIC"}, + {"Mixin PGA", NULL, "Mic PGA"}, + {"ADC", NULL, "Mixin PGA"}, + + {"Sidetone Filter", NULL, "ADC"}, + {"Mixer In", NULL, "Mixer In Supply"}, + {"Mixer In", "Mic Switch", "ADC"}, + + {"Tone Generator", NULL, "TONE"}, + + DA7219_OUT_DAI_MUX_ROUTES("Out DAIL Mux"), + DA7219_OUT_DAI_MUX_ROUTES("Out DAIR Mux"), + + {"DAIOUT", NULL, "Out DAIL Mux"}, + {"DAIOUT", NULL, "Out DAIR Mux"}, + {"DAIOUT", NULL, "DAI"}, + + /* Output paths */ + {"DAIIN", NULL, "DAI"}, + + DA7219_OUT_DAC_MUX_ROUTES("Out DACL Mux"), + DA7219_OUT_DAC_MUX_ROUTES("Out DACR Mux"), + + {"Mixer Out FilterL", "DACL Switch", "Out DACL Mux"}, + {"Mixer Out FilterR", "DACR Switch", "Out DACR Mux"}, + + DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterL"), + DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterR"), + + {"DACL", NULL, "ST Mixer Out FilterL"}, + {"DACR", NULL, "ST Mixer Out FilterR"}, + + {"Mixout Left PGA", NULL, "DACL"}, + {"Mixout Right PGA", NULL, "DACR"}, + + {"Headphone Left PGA", NULL, "Mixout Left PGA"}, + {"Headphone Right PGA", NULL, "Mixout Right PGA"}, + + {"HPL", NULL, "Headphone Left PGA"}, + {"HPR", NULL, "Headphone Right PGA"}, + + {"HPL", NULL, "Charge Pump"}, + {"HPR", NULL, "Charge Pump"}, +}; + + +/* + * DAI operations + */ + +static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) + return 0; + + if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + + switch (clk_id) { + case DA7219_CLKSRC_MCLK_SQR: + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_MCLK_SQR_EN_MASK, + DA7219_PLL_MCLK_SQR_EN_MASK); + break; + case DA7219_CLKSRC_MCLK: + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_MCLK_SQR_EN_MASK, 0); + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } + + da7219->clk_src = clk_id; + + if (da7219->mclk) { + freq = clk_round_rate(da7219->mclk, freq); + ret = clk_set_rate(da7219->mclk, freq); + if (ret) { + dev_err(codec_dai->dev, "Failed to set clock rate %d\n", + freq); + return ret; + } + } + + da7219->mclk_rate = freq; + + return 0; +} + +static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + u8 pll_ctrl, indiv_bits, indiv; + u8 pll_frac_top, pll_frac_bot, pll_integer; + u32 freq_ref; + u64 frac_div; + + /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ + if (da7219->mclk_rate == 32768) { + indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; + indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; + } else if (da7219->mclk_rate < 2000000) { + dev_err(codec->dev, "PLL input clock %d below valid range\n", + da7219->mclk_rate); + return -EINVAL; + } else if (da7219->mclk_rate <= 5000000) { + indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; + indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; + } else if (da7219->mclk_rate <= 10000000) { + indiv_bits = DA7219_PLL_INDIV_5_10_MHZ; + indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL; + } else if (da7219->mclk_rate <= 20000000) { + indiv_bits = DA7219_PLL_INDIV_10_20_MHZ; + indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL; + } else if (da7219->mclk_rate <= 40000000) { + indiv_bits = DA7219_PLL_INDIV_20_40_MHZ; + indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7219->mclk_rate <= 54000000) { + indiv_bits = DA7219_PLL_INDIV_40_54_MHZ; + indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL; + } else { + dev_err(codec->dev, "PLL input clock %d above valid range\n", + da7219->mclk_rate); + return -EINVAL; + } + freq_ref = (da7219->mclk_rate / indiv); + pll_ctrl = indiv_bits; + + /* Configure PLL */ + switch (source) { + case DA7219_SYSCLK_MCLK: + pll_ctrl |= DA7219_PLL_MODE_BYPASS; + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_INDIV_MASK | + DA7219_PLL_MODE_MASK, pll_ctrl); + return 0; + case DA7219_SYSCLK_PLL: + pll_ctrl |= DA7219_PLL_MODE_NORMAL; + break; + case DA7219_SYSCLK_PLL_SRM: + pll_ctrl |= DA7219_PLL_MODE_SRM; + break; + case DA7219_SYSCLK_PLL_32KHZ: + pll_ctrl |= DA7219_PLL_MODE_32KHZ; + break; + default: + dev_err(codec->dev, "Invalid PLL config\n"); + return -EINVAL; + } + + /* Calculate dividers for PLL */ + pll_integer = fout / freq_ref; + frac_div = (u64)(fout % freq_ref) * 8192ULL; + do_div(frac_div, freq_ref); + pll_frac_top = (frac_div >> DA7219_BYTE_SHIFT) & DA7219_BYTE_MASK; + pll_frac_bot = (frac_div) & DA7219_BYTE_MASK; + + /* Write PLL config & dividers */ + snd_soc_write(codec, DA7219_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA7219_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA7219_PLL_INTEGER, pll_integer); + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK, + pll_ctrl); + + return 0; +} + +static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 dai_clk_mode = 0, dai_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + da7219->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + da7219->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7219_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV | + DA7219_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_ctrl |= DA7219_DAI_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + dai_ctrl |= DA7219_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_ctrl |= DA7219_DAI_FORMAT_RIGHT_J; + break; + case SND_SOC_DAIFMT_DSP_B: + dai_ctrl |= DA7219_DAI_FORMAT_DSP; + break; + default: + return -EINVAL; + } + + /* By default 64 BCLKs per WCLK is supported */ + dai_clk_mode |= DA7219_DAI_BCLKS_PER_WCLK_64; + + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK | + DA7219_DAI_CLK_POL_MASK | DA7219_DAI_WCLK_POL_MASK, + dai_clk_mode); + snd_soc_update_bits(codec, DA7219_DAI_CTRL, DA7219_DAI_FORMAT_MASK, + dai_ctrl); + + return 0; +} + +static int da7219_set_dai_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 da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 dai_bclks_per_wclk; + u16 offset; + u32 frame_size; + + /* No channels enabled so disable TDM, revert to 64-bit frames */ + if (!tx_mask) { + snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL, + DA7219_DAI_TDM_CH_EN_MASK | + DA7219_DAI_TDM_MODE_EN_MASK, 0); + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + DA7219_DAI_BCLKS_PER_WCLK_64); + return 0; + } + + /* Check we have valid slots */ + if (fls(tx_mask) > DA7219_DAI_TDM_MAX_SLOTS) { + dev_err(codec->dev, "Invalid number of slots, max = %d\n", + DA7219_DAI_TDM_MAX_SLOTS); + return -EINVAL; + } + + /* Check we have a valid offset given */ + if (rx_mask > DA7219_DAI_OFFSET_MAX) { + dev_err(codec->dev, "Invalid slot offset, max = %d\n", + DA7219_DAI_OFFSET_MAX); + return -EINVAL; + } + + /* Calculate & validate frame size based on slot info provided. */ + frame_size = slots * slot_width; + switch (frame_size) { + case 32: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; + break; + case 64: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; + break; + case 128: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; + break; + case 256: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; + break; + default: + dev_err(codec->dev, "Invalid frame size %d\n", frame_size); + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + dai_bclks_per_wclk); + + offset = cpu_to_le16(rx_mask); + regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, + &offset, sizeof(offset)); + + snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL, + DA7219_DAI_TDM_CH_EN_MASK | + DA7219_DAI_TDM_MODE_EN_MASK, + (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | + DA7219_DAI_TDM_MODE_EN_MASK); + + return 0; +} + +static int da7219_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; + u8 dai_ctrl = 0, fs; + unsigned int channels; + + switch (params_width(params)) { + case 16: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S16_LE; + break; + case 20: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S20_LE; + break; + case 24: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S24_LE; + break; + case 32: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S32_LE; + break; + default: + return -EINVAL; + } + + channels = params_channels(params); + if ((channels < 1) | (channels > DA7219_DAI_CH_NUM_MAX)) { + dev_err(codec->dev, + "Invalid number of channels, only 1 to %d supported\n", + DA7219_DAI_CH_NUM_MAX); + return -EINVAL; + } + dai_ctrl |= channels << DA7219_DAI_CH_NUM_SHIFT; + + switch (params_rate(params)) { + case 8000: + fs = DA7219_SR_8000; + break; + case 11025: + fs = DA7219_SR_11025; + break; + case 12000: + fs = DA7219_SR_12000; + break; + case 16000: + fs = DA7219_SR_16000; + break; + case 22050: + fs = DA7219_SR_22050; + break; + case 24000: + fs = DA7219_SR_24000; + break; + case 32000: + fs = DA7219_SR_32000; + break; + case 44100: + fs = DA7219_SR_44100; + break; + case 48000: + fs = DA7219_SR_48000; + break; + case 88200: + fs = DA7219_SR_88200; + break; + case 96000: + fs = DA7219_SR_96000; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7219_DAI_CTRL, + DA7219_DAI_WORD_LENGTH_MASK | + DA7219_DAI_CH_NUM_MASK, + dai_ctrl); + snd_soc_write(codec, DA7219_SR, fs); + + return 0; +} + +static const struct snd_soc_dai_ops da7219_dai_ops = { + .hw_params = da7219_hw_params, + .set_sysclk = da7219_set_dai_sysclk, + .set_pll = da7219_set_dai_pll, + .set_fmt = da7219_set_dai_fmt, + .set_tdm_slot = da7219_set_dai_tdm_slot, +}; + +#define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver da7219_dai = { + .name = "da7219-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = DA7219_DAI_CH_NUM_MAX, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7219_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = DA7219_DAI_CH_NUM_MAX, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7219_FORMATS, + }, + .ops = &da7219_dai_ops, + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, +}; + + +/* + * DT + */ + +static const struct of_device_id da7219_of_match[] = { + { .compatible = "dlg,da7219", }, + { } +}; +MODULE_DEVICE_TABLE(of, da7219_of_match); + +static enum da7219_ldo_lvl_sel da7219_of_ldo_lvl(struct snd_soc_codec *codec, + u32 val) +{ + switch (val) { + case 1050: + return DA7219_LDO_LVL_SEL_1_05V; + case 1100: + return DA7219_LDO_LVL_SEL_1_10V; + case 1200: + return DA7219_LDO_LVL_SEL_1_20V; + case 1400: + return DA7219_LDO_LVL_SEL_1_40V; + default: + dev_warn(codec->dev, "Invalid LDO level"); + return DA7219_LDO_LVL_SEL_1_05V; + } +} + +static enum da7219_micbias_voltage + da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1800: + return DA7219_MICBIAS_1_8V; + case 2000: + return DA7219_MICBIAS_2_0V; + case 2200: + return DA7219_MICBIAS_2_2V; + case 2400: + return DA7219_MICBIAS_2_4V; + case 2600: + return DA7219_MICBIAS_2_6V; + default: + dev_warn(codec->dev, "Invalid micbias level"); + return DA7219_MICBIAS_2_2V; + } +} + +static enum da7219_mic_amp_in_sel + da7219_of_mic_amp_in_sel(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "diff")) { + return DA7219_MIC_AMP_IN_SEL_DIFF; + } else if (!strcmp(str, "se_p")) { + return DA7219_MIC_AMP_IN_SEL_SE_P; + } else if (!strcmp(str, "se_n")) { + return DA7219_MIC_AMP_IN_SEL_SE_N; + } else { + dev_warn(codec->dev, "Invalid mic input type selection"); + return DA7219_MIC_AMP_IN_SEL_DIFF; + } +} + +static struct da7219_pdata *da7219_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct da7219_pdata *pdata; + const char *of_str; + u32 of_val32; + + pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return NULL; + + if (of_property_read_u32(np, "dlg,ldo-lvl", &of_val32) >= 0) + pdata->ldo_lvl_sel = da7219_of_ldo_lvl(codec, of_val32); + + if (of_property_read_u32(np, "dlg,micbias-lvl", &of_val32) >= 0) + pdata->micbias_lvl = da7219_of_micbias_lvl(codec, of_val32); + else + pdata->micbias_lvl = DA7219_MICBIAS_2_2V; + + if (!of_property_read_string(np, "dlg,mic-amp-in-sel", &of_str)) + pdata->mic_amp_in_sel = da7219_of_mic_amp_in_sel(codec, of_str); + else + pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + + return pdata; +} + + +/* + * Codec driver functions + */ + +static int da7219_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* MCLK */ + clk_prepare_enable(da7219->mclk); + + /* Master bias */ + snd_soc_update_bits(codec, DA7219_REFERENCES, + DA7219_BIAS_EN_MASK, + DA7219_BIAS_EN_MASK); + + /* Enable Internal Digital LDO */ + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_EN_MASK, + DA7219_LDO_EN_MASK); + } + break; + case SND_SOC_BIAS_OFF: + /* Only disable if jack detection not active */ + if (!da7219->aad->jack) { + /* Bypass Internal Digital LDO */ + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_EN_MASK, 0); + + /* Master bias */ + snd_soc_update_bits(codec, DA7219_REFERENCES, + DA7219_BIAS_EN_MASK, 0); + } + + /* MCLK */ + clk_disable_unprepare(da7219->mclk); + break; + } + + return 0; +} + +static const char *da7219_supply_names[DA7219_NUM_SUPPLIES] = { + [DA7219_SUPPLY_VDD] = "VDD", + [DA7219_SUPPLY_VDDMIC] = "VDDMIC", + [DA7219_SUPPLY_VDDIO] = "VDDIO", +}; + +static int da7219_handle_supplies(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct regulator *vddio; + u8 io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V; + int i, ret; + + /* Get required supplies */ + for (i = 0; i < DA7219_NUM_SUPPLIES; ++i) + da7219->supplies[i].supply = da7219_supply_names[i]; + + ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES, + da7219->supplies); + if (ret) + return ret; + + /* Determine VDDIO voltage provided */ + vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; + ret = regulator_get_voltage(vddio); + if (ret < 1200000) + dev_warn(codec->dev, "Invalid VDDIO voltage\n"); + else if (ret < 2800000) + io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; + + /* Enable main supplies */ + ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); + + /* Ensure device in active mode */ + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); + + /* Update IO voltage level range */ + snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl); + + return ret; +} + +static void da7219_handle_pdata(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_pdata *pdata = da7219->pdata; + + if (pdata) { + u8 micbias_lvl = 0; + + /* Internal LDO */ + switch (pdata->ldo_lvl_sel) { + case DA7219_LDO_LVL_SEL_1_05V: + case DA7219_LDO_LVL_SEL_1_10V: + case DA7219_LDO_LVL_SEL_1_20V: + case DA7219_LDO_LVL_SEL_1_40V: + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_LEVEL_SELECT_MASK, + (pdata->ldo_lvl_sel << + DA7219_LDO_LEVEL_SELECT_SHIFT)); + break; + } + + /* Mic Bias voltages */ + switch (pdata->micbias_lvl) { + case DA7219_MICBIAS_1_8V: + case DA7219_MICBIAS_2_0V: + case DA7219_MICBIAS_2_2V: + case DA7219_MICBIAS_2_4V: + case DA7219_MICBIAS_2_6V: + micbias_lvl |= (pdata->micbias_lvl << + DA7219_MICBIAS1_LEVEL_SHIFT); + break; + } + + snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_lvl); + + /* Mic */ + switch (pdata->mic_amp_in_sel) { + case DA7219_MIC_AMP_IN_SEL_DIFF: + case DA7219_MIC_AMP_IN_SEL_SE_P: + case DA7219_MIC_AMP_IN_SEL_SE_N: + snd_soc_write(codec, DA7219_MIC_1_SELECT, + pdata->mic_amp_in_sel); + break; + } + } +} + +static int da7219_probe(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_init(&da7219->lock); + + /* Regulator configuration */ + ret = da7219_handle_supplies(codec); + if (ret) + return ret; + + /* Handle DT/Platform data */ + if (codec->dev->of_node) + da7219->pdata = da7219_of_to_pdata(codec); + else + da7219->pdata = dev_get_platdata(codec->dev); + + da7219_handle_pdata(codec); + + /* Check if MCLK provided */ + da7219->mclk = devm_clk_get(codec->dev, "mclk"); + if (IS_ERR(da7219->mclk)) { + if (PTR_ERR(da7219->mclk) != -ENOENT) + return PTR_ERR(da7219->mclk); + else + da7219->mclk = NULL; + } + + /* Default PC counter to free-running */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, + DA7219_PC_FREERUN_MASK); + + /* Default gain ramping */ + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_RAMP_EN_MASK, + DA7219_MIXIN_L_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, DA7219_ADC_L_RAMP_EN_MASK, + DA7219_ADC_L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, DA7219_DAC_L_RAMP_EN_MASK, + DA7219_DAC_L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, DA7219_DAC_R_RAMP_EN_MASK, + DA7219_DAC_R_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_RAMP_EN_MASK, + DA7219_HP_L_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_RAMP_EN_MASK, + DA7219_HP_R_AMP_RAMP_EN_MASK); + + /* Default infinite tone gen, start/stop by Kcontrol */ + snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK); + + /* Initialise AAD block */ + return da7219_aad_init(codec); +} + +static int da7219_remove(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + da7219_aad_exit(codec); + + /* Supplies */ + return regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies); +} + +#ifdef CONFIG_PM +static int da7219_suspend(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + + /* Put device into standby mode if jack detection disabled */ + if (!da7219->aad->jack) + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, 0); + + return 0; +} + +static int da7219_resume(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + /* Put device into active mode if previously pushed to standby */ + if (!da7219->aad->jack) + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, + DA7219_SYSTEM_ACTIVE_MASK); + + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define da7219_suspend NULL +#define da7219_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_da7219 = { + .probe = da7219_probe, + .remove = da7219_remove, + .suspend = da7219_suspend, + .resume = da7219_resume, + .set_bias_level = da7219_set_bias_level, + + .controls = da7219_snd_controls, + .num_controls = ARRAY_SIZE(da7219_snd_controls), + + .dapm_widgets = da7219_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7219_dapm_widgets), + .dapm_routes = da7219_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7219_audio_map), +}; + + +/* + * Regmap configs + */ + +static struct reg_default da7219_reg_defaults[] = { + { DA7219_MIC_1_SELECT, 0x00 }, + { DA7219_CIF_TIMEOUT_CTRL, 0x01 }, + { DA7219_SR_24_48, 0x00 }, + { DA7219_SR, 0x0A }, + { DA7219_CIF_I2C_ADDR_CFG, 0x02 }, + { DA7219_PLL_CTRL, 0x10 }, + { DA7219_PLL_FRAC_TOP, 0x00 }, + { DA7219_PLL_FRAC_BOT, 0x00 }, + { DA7219_PLL_INTEGER, 0x20 }, + { DA7219_DIG_ROUTING_DAI, 0x10 }, + { DA7219_DAI_CLK_MODE, 0x01 }, + { DA7219_DAI_CTRL, 0x28 }, + { DA7219_DAI_TDM_CTRL, 0x40 }, + { DA7219_DIG_ROUTING_DAC, 0x32 }, + { DA7219_DAI_OFFSET_LOWER, 0x00 }, + { DA7219_DAI_OFFSET_UPPER, 0x00 }, + { DA7219_REFERENCES, 0x00 }, + { DA7219_MIXIN_L_SELECT, 0x00 }, + { DA7219_MIXIN_L_GAIN, 0x03 }, + { DA7219_ADC_L_GAIN, 0x6F }, + { DA7219_ADC_FILTERS1, 0x80 }, + { DA7219_MIC_1_GAIN, 0x01 }, + { DA7219_SIDETONE_CTRL, 0x40 }, + { DA7219_SIDETONE_GAIN, 0x0E }, + { DA7219_DROUTING_ST_OUTFILT_1L, 0x01 }, + { DA7219_DROUTING_ST_OUTFILT_1R, 0x02 }, + { DA7219_DAC_FILTERS5, 0x00 }, + { DA7219_DAC_FILTERS2, 0x88 }, + { DA7219_DAC_FILTERS3, 0x88 }, + { DA7219_DAC_FILTERS4, 0x08 }, + { DA7219_DAC_FILTERS1, 0x80 }, + { DA7219_DAC_L_GAIN, 0x6F }, + { DA7219_DAC_R_GAIN, 0x6F }, + { DA7219_CP_CTRL, 0x20 }, + { DA7219_HP_L_GAIN, 0x39 }, + { DA7219_HP_R_GAIN, 0x39 }, + { DA7219_MIXOUT_L_SELECT, 0x00 }, + { DA7219_MIXOUT_R_SELECT, 0x00 }, + { DA7219_MICBIAS_CTRL, 0x03 }, + { DA7219_MIC_1_CTRL, 0x40 }, + { DA7219_MIXIN_L_CTRL, 0x40 }, + { DA7219_ADC_L_CTRL, 0x40 }, + { DA7219_DAC_L_CTRL, 0x40 }, + { DA7219_DAC_R_CTRL, 0x40 }, + { DA7219_HP_L_CTRL, 0x40 }, + { DA7219_HP_R_CTRL, 0x40 }, + { DA7219_MIXOUT_L_CTRL, 0x10 }, + { DA7219_MIXOUT_R_CTRL, 0x10 }, + { DA7219_CHIP_ID1, 0x23 }, + { DA7219_CHIP_ID2, 0x93 }, + { DA7219_CHIP_REVISION, 0x00 }, + { DA7219_LDO_CTRL, 0x00 }, + { DA7219_IO_CTRL, 0x00 }, + { DA7219_GAIN_RAMP_CTRL, 0x00 }, + { DA7219_PC_COUNT, 0x02 }, + { DA7219_CP_VOL_THRESHOLD1, 0x0E }, + { DA7219_DIG_CTRL, 0x00 }, + { DA7219_ALC_CTRL2, 0x00 }, + { DA7219_ALC_CTRL3, 0x00 }, + { DA7219_ALC_NOISE, 0x3F }, + { DA7219_ALC_TARGET_MIN, 0x3F }, + { DA7219_ALC_TARGET_MAX, 0x00 }, + { DA7219_ALC_GAIN_LIMITS, 0xFF }, + { DA7219_ALC_ANA_GAIN_LIMITS, 0x71 }, + { DA7219_ALC_ANTICLIP_CTRL, 0x00 }, + { DA7219_ALC_ANTICLIP_LEVEL, 0x00 }, + { DA7219_DAC_NG_SETUP_TIME, 0x00 }, + { DA7219_DAC_NG_OFF_THRESH, 0x00 }, + { DA7219_DAC_NG_ON_THRESH, 0x00 }, + { DA7219_DAC_NG_CTRL, 0x00 }, + { DA7219_TONE_GEN_CFG1, 0x00 }, + { DA7219_TONE_GEN_CFG2, 0x00 }, + { DA7219_TONE_GEN_CYCLES, 0x00 }, + { DA7219_TONE_GEN_FREQ1_L, 0x55 }, + { DA7219_TONE_GEN_FREQ1_U, 0x15 }, + { DA7219_TONE_GEN_FREQ2_L, 0x00 }, + { DA7219_TONE_GEN_FREQ2_U, 0x40 }, + { DA7219_TONE_GEN_ON_PER, 0x02 }, + { DA7219_TONE_GEN_OFF_PER, 0x01 }, + { DA7219_ACCDET_IRQ_MASK_A, 0x00 }, + { DA7219_ACCDET_IRQ_MASK_B, 0x00 }, + { DA7219_ACCDET_CONFIG_1, 0xD6 }, + { DA7219_ACCDET_CONFIG_2, 0x34 }, + { DA7219_ACCDET_CONFIG_3, 0x0A }, + { DA7219_ACCDET_CONFIG_4, 0x16 }, + { DA7219_ACCDET_CONFIG_5, 0x21 }, + { DA7219_ACCDET_CONFIG_6, 0x3E }, + { DA7219_ACCDET_CONFIG_7, 0x01 }, + { DA7219_SYSTEM_ACTIVE, 0x00 }, +}; + +static bool da7219_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA7219_MIC_1_GAIN_STATUS: + case DA7219_MIXIN_L_GAIN_STATUS: + case DA7219_ADC_L_GAIN_STATUS: + case DA7219_DAC_L_GAIN_STATUS: + case DA7219_DAC_R_GAIN_STATUS: + case DA7219_HP_L_GAIN_STATUS: + case DA7219_HP_R_GAIN_STATUS: + case DA7219_CIF_CTRL: + case DA7219_PLL_SRM_STS: + case DA7219_ALC_CTRL1: + case DA7219_SYSTEM_MODES_INPUT: + case DA7219_SYSTEM_MODES_OUTPUT: + case DA7219_ALC_OFFSET_AUTO_M_L: + case DA7219_ALC_OFFSET_AUTO_U_L: + case DA7219_TONE_GEN_CFG1: + case DA7219_ACCDET_STATUS_A: + case DA7219_ACCDET_STATUS_B: + case DA7219_ACCDET_IRQ_EVENT_A: + case DA7219_ACCDET_IRQ_EVENT_B: + case DA7219_ACCDET_CONFIG_8: + case DA7219_SYSTEM_STATUS: + return 1; + default: + return 0; + } +} + +static const struct regmap_config da7219_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DA7219_SYSTEM_ACTIVE, + .reg_defaults = da7219_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7219_reg_defaults), + .volatile_reg = da7219_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + + +/* + * I2C layer + */ + +static int da7219_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7219_priv *da7219; + int ret; + + da7219 = devm_kzalloc(&i2c->dev, sizeof(struct da7219_priv), + GFP_KERNEL); + if (!da7219) + return -ENOMEM; + + i2c_set_clientdata(i2c, da7219); + + da7219->regmap = devm_regmap_init_i2c(i2c, &da7219_regmap_config); + if (IS_ERR(da7219->regmap)) { + ret = PTR_ERR(da7219->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7219, + &da7219_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da7219 codec: %d\n", + ret); + } + return ret; +} + +static int da7219_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da7219_i2c_id[] = { + { "da7219", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7219_i2c_id); + +static struct i2c_driver da7219_i2c_driver = { + .driver = { + .name = "da7219", + .of_match_table = da7219_of_match, + }, + .probe = da7219_i2c_probe, + .remove = da7219_i2c_remove, + .id_table = da7219_i2c_id, +}; + +module_i2c_driver(da7219_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA7219 Codec Driver"); +MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h new file mode 100644 index 0000000..b514268 --- /dev/null +++ b/sound/soc/codecs/da7219.h @@ -0,0 +1,820 @@ +/* + * da7219.h - DA7219 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_H +#define __DA7219_H + +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/da7219.h> + +/* + * Registers + */ + +#define DA7219_MIC_1_GAIN_STATUS 0x6 +#define DA7219_MIXIN_L_GAIN_STATUS 0x8 +#define DA7219_ADC_L_GAIN_STATUS 0xA +#define DA7219_DAC_L_GAIN_STATUS 0xC +#define DA7219_DAC_R_GAIN_STATUS 0xD +#define DA7219_HP_L_GAIN_STATUS 0xE +#define DA7219_HP_R_GAIN_STATUS 0xF +#define DA7219_MIC_1_SELECT 0x10 +#define DA7219_CIF_TIMEOUT_CTRL 0x12 +#define DA7219_CIF_CTRL 0x13 +#define DA7219_SR_24_48 0x16 +#define DA7219_SR 0x17 +#define DA7219_CIF_I2C_ADDR_CFG 0x1B +#define DA7219_PLL_CTRL 0x20 +#define DA7219_PLL_FRAC_TOP 0x22 +#define DA7219_PLL_FRAC_BOT 0x23 +#define DA7219_PLL_INTEGER 0x24 +#define DA7219_PLL_SRM_STS 0x25 +#define DA7219_DIG_ROUTING_DAI 0x2A +#define DA7219_DAI_CLK_MODE 0x2B +#define DA7219_DAI_CTRL 0x2C +#define DA7219_DAI_TDM_CTRL 0x2D +#define DA7219_DIG_ROUTING_DAC 0x2E +#define DA7219_ALC_CTRL1 0x2F +#define DA7219_DAI_OFFSET_LOWER 0x30 +#define DA7219_DAI_OFFSET_UPPER 0x31 +#define DA7219_REFERENCES 0x32 +#define DA7219_MIXIN_L_SELECT 0x33 +#define DA7219_MIXIN_L_GAIN 0x34 +#define DA7219_ADC_L_GAIN 0x36 +#define DA7219_ADC_FILTERS1 0x38 +#define DA7219_MIC_1_GAIN 0x39 +#define DA7219_SIDETONE_CTRL 0x3A +#define DA7219_SIDETONE_GAIN 0x3B +#define DA7219_DROUTING_ST_OUTFILT_1L 0x3C +#define DA7219_DROUTING_ST_OUTFILT_1R 0x3D +#define DA7219_DAC_FILTERS5 0x40 +#define DA7219_DAC_FILTERS2 0x41 +#define DA7219_DAC_FILTERS3 0x42 +#define DA7219_DAC_FILTERS4 0x43 +#define DA7219_DAC_FILTERS1 0x44 +#define DA7219_DAC_L_GAIN 0x45 +#define DA7219_DAC_R_GAIN 0x46 +#define DA7219_CP_CTRL 0x47 +#define DA7219_HP_L_GAIN 0x48 +#define DA7219_HP_R_GAIN 0x49 +#define DA7219_MIXOUT_L_SELECT 0x4B +#define DA7219_MIXOUT_R_SELECT 0x4C +#define DA7219_SYSTEM_MODES_INPUT 0x50 +#define DA7219_SYSTEM_MODES_OUTPUT 0x51 +#define DA7219_MICBIAS_CTRL 0x62 +#define DA7219_MIC_1_CTRL 0x63 +#define DA7219_MIXIN_L_CTRL 0x65 +#define DA7219_ADC_L_CTRL 0x67 +#define DA7219_DAC_L_CTRL 0x69 +#define DA7219_DAC_R_CTRL 0x6A +#define DA7219_HP_L_CTRL 0x6B +#define DA7219_HP_R_CTRL 0x6C +#define DA7219_MIXOUT_L_CTRL 0x6E +#define DA7219_MIXOUT_R_CTRL 0x6F +#define DA7219_CHIP_ID1 0x81 +#define DA7219_CHIP_ID2 0x82 +#define DA7219_CHIP_REVISION 0x83 +#define DA7219_LDO_CTRL 0x90 +#define DA7219_IO_CTRL 0x91 +#define DA7219_GAIN_RAMP_CTRL 0x92 +#define DA7219_PC_COUNT 0x94 +#define DA7219_CP_VOL_THRESHOLD1 0x95 +#define DA7219_CP_DELAY 0x96 +#define DA7219_DIG_CTRL 0x99 +#define DA7219_ALC_CTRL2 0x9A +#define DA7219_ALC_CTRL3 0x9B +#define DA7219_ALC_NOISE 0x9C +#define DA7219_ALC_TARGET_MIN 0x9D +#define DA7219_ALC_TARGET_MAX 0x9E +#define DA7219_ALC_GAIN_LIMITS 0x9F +#define DA7219_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA7219_ALC_ANTICLIP_CTRL 0xA1 +#define DA7219_ALC_ANTICLIP_LEVEL 0xA2 +#define DA7219_ALC_OFFSET_AUTO_M_L 0xA3 +#define DA7219_ALC_OFFSET_AUTO_U_L 0xA4 +#define DA7219_DAC_NG_SETUP_TIME 0xAF +#define DA7219_DAC_NG_OFF_THRESH 0xB0 +#define DA7219_DAC_NG_ON_THRESH 0xB1 +#define DA7219_DAC_NG_CTRL 0xB2 +#define DA7219_TONE_GEN_CFG1 0xB4 +#define DA7219_TONE_GEN_CFG2 0xB5 +#define DA7219_TONE_GEN_CYCLES 0xB6 +#define DA7219_TONE_GEN_FREQ1_L 0xB7 +#define DA7219_TONE_GEN_FREQ1_U 0xB8 +#define DA7219_TONE_GEN_FREQ2_L 0xB9 +#define DA7219_TONE_GEN_FREQ2_U 0xBA +#define DA7219_TONE_GEN_ON_PER 0xBB +#define DA7219_TONE_GEN_OFF_PER 0xBC +#define DA7219_SYSTEM_STATUS 0xE0 +#define DA7219_SYSTEM_ACTIVE 0xFD + + +/* + * Bit Fields + */ + +#define DA7219_SWITCH_EN_MAX 0x1 + +/* DA7219_MIC_1_GAIN_STATUS = 0x6 */ +#define DA7219_MIC_1_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_MIC_1_AMP_GAIN_STATUS_MASK (0x7 << 0) +#define DA7219_MIC_1_AMP_GAIN_MAX 0x7 + +/* DA7219_MIXIN_L_GAIN_STATUS = 0x8 */ +#define DA7219_MIXIN_L_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_MIXIN_L_AMP_GAIN_STATUS_MASK (0xF << 0) + +/* DA7219_ADC_L_GAIN_STATUS = 0xA */ +#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_DAC_L_GAIN_STATUS = 0xC */ +#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_DAC_R_GAIN_STATUS = 0xD */ +#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_HP_L_GAIN_STATUS = 0xE */ +#define DA7219_HP_L_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_HP_L_AMP_GAIN_STATUS_MASK (0x3F << 0) + +/* DA7219_HP_R_GAIN_STATUS = 0xF */ +#define DA7219_HP_R_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_HP_R_AMP_GAIN_STATUS_MASK (0x3F << 0) + +/* DA7219_MIC_1_SELECT = 0x10 */ +#define DA7219_MIC_1_AMP_IN_SEL_SHIFT 0 +#define DA7219_MIC_1_AMP_IN_SEL_MASK (0x3 << 0) + +/* DA7219_CIF_TIMEOUT_CTRL = 0x12 */ +#define DA7219_I2C_TIMEOUT_EN_SHIFT 0 +#define DA7219_I2C_TIMEOUT_EN_MASK (0x1 << 0) + +/* DA7219_CIF_CTRL = 0x13 */ +#define DA7219_CIF_I2C_WRITE_MODE_SHIFT 0 +#define DA7219_CIF_I2C_WRITE_MODE_MASK (0x1 << 0) +#define DA7219_CIF_REG_SOFT_RESET_SHIFT 7 +#define DA7219_CIF_REG_SOFT_RESET_MASK (0x1 << 7) + +/* DA7219_SR_24_48 = 0x16 */ +#define DA7219_SR_24_48_SHIFT 0 +#define DA7219_SR_24_48_MASK (0x1 << 0) + +/* DA7219_SR = 0x17 */ +#define DA7219_SR_SHIFT 0 +#define DA7219_SR_MASK (0xF << 0) +#define DA7219_SR_8000 (0x01 << 0) +#define DA7219_SR_11025 (0x02 << 0) +#define DA7219_SR_12000 (0x03 << 0) +#define DA7219_SR_16000 (0x05 << 0) +#define DA7219_SR_22050 (0x06 << 0) +#define DA7219_SR_24000 (0x07 << 0) +#define DA7219_SR_32000 (0x09 << 0) +#define DA7219_SR_44100 (0x0A << 0) +#define DA7219_SR_48000 (0x0B << 0) +#define DA7219_SR_88200 (0x0E << 0) +#define DA7219_SR_96000 (0x0F << 0) + +/* DA7219_CIF_I2C_ADDR_CFG = 0x1B */ +#define DA7219_CIF_I2C_ADDR_CFG_SHIFT 0 +#define DA7219_CIF_I2C_ADDR_CFG_MASK (0x3 << 0) + +/* DA7219_PLL_CTRL = 0x20 */ +#define DA7219_PLL_INDIV_SHIFT 2 +#define DA7219_PLL_INDIV_MASK (0x7 << 2) +#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2) +#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2) +#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2) +#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2) +#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2) +#define DA7219_PLL_MCLK_SQR_EN_SHIFT 5 +#define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5) +#define DA7219_PLL_MODE_SHIFT 6 +#define DA7219_PLL_MODE_MASK (0x3 << 6) +#define DA7219_PLL_MODE_BYPASS (0x0 << 6) +#define DA7219_PLL_MODE_NORMAL (0x1 << 6) +#define DA7219_PLL_MODE_SRM (0x2 << 6) +#define DA7219_PLL_MODE_32KHZ (0x3 << 6) + +/* DA7219_PLL_FRAC_TOP = 0x22 */ +#define DA7219_PLL_FBDIV_FRAC_TOP_SHIFT 0 +#define DA7219_PLL_FBDIV_FRAC_TOP_MASK (0x1F << 0) + +/* DA7219_PLL_FRAC_BOT = 0x23 */ +#define DA7219_PLL_FBDIV_FRAC_BOT_SHIFT 0 +#define DA7219_PLL_FBDIV_FRAC_BOT_MASK (0xFF << 0) + +/* DA7219_PLL_INTEGER = 0x24 */ +#define DA7219_PLL_FBDIV_INTEGER_SHIFT 0 +#define DA7219_PLL_FBDIV_INTEGER_MASK (0x7F << 0) + +/* DA7219_PLL_SRM_STS = 0x25 */ +#define DA7219_PLL_SRM_STATE_SHIFT 0 +#define DA7219_PLL_SRM_STATE_MASK (0xF << 0) +#define DA7219_PLL_SRM_STATUS_SHIFT 4 +#define DA7219_PLL_SRM_STATUS_MASK (0xF << 4) +#define DA7219_PLL_SRM_STS_SRM_LOCK (0x1 << 7) + +/* DA7219_DIG_ROUTING_DAI = 0x2A */ +#define DA7219_DAI_L_SRC_SHIFT 0 +#define DA7219_DAI_L_SRC_MASK (0x3 << 0) +#define DA7219_DAI_R_SRC_SHIFT 4 +#define DA7219_DAI_R_SRC_MASK (0x3 << 4) +#define DA7219_OUT_SRC_MAX 4 + +/* DA7219_DAI_CLK_MODE = 0x2B */ +#define DA7219_DAI_BCLKS_PER_WCLK_SHIFT 0 +#define DA7219_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_32 (0x0 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_64 (0x1 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_128 (0x2 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) +#define DA7219_DAI_CLK_POL_SHIFT 2 +#define DA7219_DAI_CLK_POL_MASK (0x1 << 2) +#define DA7219_DAI_CLK_POL_INV (0x1 << 2) +#define DA7219_DAI_WCLK_POL_SHIFT 3 +#define DA7219_DAI_WCLK_POL_MASK (0x1 << 3) +#define DA7219_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7219_DAI_WCLK_TRI_STATE_SHIFT 4 +#define DA7219_DAI_WCLK_TRI_STATE_MASK (0x1 << 4) +#define DA7219_DAI_CLK_EN_SHIFT 7 +#define DA7219_DAI_CLK_EN_MASK (0x1 << 7) + +/* DA7219_DAI_CTRL = 0x2C */ +#define DA7219_DAI_FORMAT_SHIFT 0 +#define DA7219_DAI_FORMAT_MASK (0x3 << 0) +#define DA7219_DAI_FORMAT_I2S (0x0 << 0) +#define DA7219_DAI_FORMAT_LEFT_J (0x1 << 0) +#define DA7219_DAI_FORMAT_RIGHT_J (0x2 << 0) +#define DA7219_DAI_FORMAT_DSP (0x3 << 0) +#define DA7219_DAI_WORD_LENGTH_SHIFT 2 +#define DA7219_DAI_WORD_LENGTH_MASK (0x3 << 2) +#define DA7219_DAI_WORD_LENGTH_S16_LE (0x0 << 2) +#define DA7219_DAI_WORD_LENGTH_S20_LE (0x1 << 2) +#define DA7219_DAI_WORD_LENGTH_S24_LE (0x2 << 2) +#define DA7219_DAI_WORD_LENGTH_S32_LE (0x3 << 2) +#define DA7219_DAI_CH_NUM_SHIFT 4 +#define DA7219_DAI_CH_NUM_MASK (0x3 << 4) +#define DA7219_DAI_CH_NUM_MAX 2 +#define DA7219_DAI_EN_SHIFT 7 +#define DA7219_DAI_EN_MASK (0x1 << 7) + +/* DA7219_DAI_TDM_CTRL = 0x2D */ +#define DA7219_DAI_TDM_CH_EN_SHIFT 0 +#define DA7219_DAI_TDM_CH_EN_MASK (0x3 << 0) +#define DA7219_DAI_OE_SHIFT 6 +#define DA7219_DAI_OE_MASK (0x1 << 6) +#define DA7219_DAI_TDM_MODE_EN_SHIFT 7 +#define DA7219_DAI_TDM_MODE_EN_MASK (0x1 << 7) +#define DA7219_DAI_TDM_MAX_SLOTS 2 + +/* DA7219_DIG_ROUTING_DAC = 0x2E */ +#define DA7219_DAC_L_SRC_SHIFT 0 +#define DA7219_DAC_L_SRC_MASK (0x3 << 0) +#define DA7219_DAC_L_SRC_TONEGEN (0x1 << 0) +#define DA7219_DAC_L_MONO_SHIFT 3 +#define DA7219_DAC_L_MONO_MASK (0x1 << 3) +#define DA7219_DAC_R_SRC_SHIFT 4 +#define DA7219_DAC_R_SRC_MASK (0x3 << 4) +#define DA7219_DAC_R_SRC_TONEGEN (0x1 << 4) +#define DA7219_DAC_R_MONO_SHIFT 7 +#define DA7219_DAC_R_MONO_MASK (0x1 << 7) + +/* DA7219_ALC_CTRL1 = 0x2F */ +#define DA7219_ALC_OFFSET_EN_SHIFT 0 +#define DA7219_ALC_OFFSET_EN_MASK (0x1 << 0) +#define DA7219_ALC_SYNC_MODE_SHIFT 1 +#define DA7219_ALC_SYNC_MODE_MASK (0x1 << 1) +#define DA7219_ALC_EN_SHIFT 3 +#define DA7219_ALC_EN_MASK (0x1 << 3) +#define DA7219_ALC_AUTO_CALIB_EN_SHIFT 4 +#define DA7219_ALC_AUTO_CALIB_EN_MASK (0x1 << 4) +#define DA7219_ALC_CALIB_OVERFLOW_SHIFT 5 +#define DA7219_ALC_CALIB_OVERFLOW_MASK (0x1 << 5) + +/* DA7219_DAI_OFFSET_LOWER = 0x30 */ +#define DA7219_DAI_OFFSET_LOWER_SHIFT 0 +#define DA7219_DAI_OFFSET_LOWER_MASK (0xFF << 0) + +/* DA7219_DAI_OFFSET_UPPER = 0x31 */ +#define DA7219_DAI_OFFSET_UPPER_SHIFT 0 +#define DA7219_DAI_OFFSET_UPPER_MASK (0x7 << 0) +#define DA7219_DAI_OFFSET_MAX 0x2FF + +/* DA7219_REFERENCES = 0x32 */ +#define DA7219_BIAS_EN_SHIFT 3 +#define DA7219_BIAS_EN_MASK (0x1 << 3) +#define DA7219_VMID_FAST_CHARGE_SHIFT 4 +#define DA7219_VMID_FAST_CHARGE_MASK (0x1 << 4) + +/* DA7219_MIXIN_L_SELECT = 0x33 */ +#define DA7219_MIXIN_L_MIX_SELECT_SHIFT 0 +#define DA7219_MIXIN_L_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_MIXIN_L_GAIN = 0x34 */ +#define DA7219_MIXIN_L_AMP_GAIN_SHIFT 0 +#define DA7219_MIXIN_L_AMP_GAIN_MASK (0xF << 0) +#define DA7219_MIXIN_L_AMP_GAIN_MAX 0xF + +/* DA7219_ADC_L_GAIN = 0x36 */ +#define DA7219_ADC_L_DIGITAL_GAIN_SHIFT 0 +#define DA7219_ADC_L_DIGITAL_GAIN_MASK (0x7F << 0) +#define DA7219_ADC_L_DIGITAL_GAIN_MAX 0x7F + +/* DA7219_ADC_FILTERS1 = 0x38 */ +#define DA7219_ADC_VOICE_HPF_CORNER_SHIFT 0 +#define DA7219_ADC_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7219_VOICE_HPF_CORNER_MAX 8 +#define DA7219_ADC_VOICE_EN_SHIFT 3 +#define DA7219_ADC_VOICE_EN_MASK (0x1 << 3) +#define DA7219_ADC_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7219_ADC_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7219_AUDIO_HPF_CORNER_MAX 4 +#define DA7219_ADC_HPF_EN_SHIFT 7 +#define DA7219_ADC_HPF_EN_MASK (0x1 << 7) +#define DA7219_HPF_MODE_SHIFT 0 +#define DA7219_HPF_DISABLED ((0x0 << 3) | (0x0 << 7)) +#define DA7219_HPF_AUDIO_EN ((0x0 << 3) | (0x1 << 7)) +#define DA7219_HPF_VOICE_EN ((0x1 << 3) | (0x1 << 7)) +#define DA7219_HPF_MODE_MASK ((0x1 << 3) | (0x1 << 7)) +#define DA7219_HPF_MODE_MAX 3 + +/* DA7219_MIC_1_GAIN = 0x39 */ +#define DA7219_MIC_1_AMP_GAIN_SHIFT 0 +#define DA7219_MIC_1_AMP_GAIN_MASK (0x7 << 0) + +/* DA7219_SIDETONE_CTRL = 0x3A */ +#define DA7219_SIDETONE_MUTE_EN_SHIFT 6 +#define DA7219_SIDETONE_MUTE_EN_MASK (0x1 << 6) +#define DA7219_SIDETONE_EN_SHIFT 7 +#define DA7219_SIDETONE_EN_MASK (0x1 << 7) + +/* DA7219_SIDETONE_GAIN = 0x3B */ +#define DA7219_SIDETONE_GAIN_SHIFT 0 +#define DA7219_SIDETONE_GAIN_MASK (0xF << 0) +#define DA7219_SIDETONE_GAIN_MAX 0xE + +/* DA7219_DROUTING_ST_OUTFILT_1L = 0x3C */ +#define DA7219_OUTFILT_ST_1L_SRC_SHIFT 0 +#define DA7219_OUTFILT_ST_1L_SRC_MASK (0x7 << 0) +#define DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT 0 +#define DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT 1 +#define DA7219_DMIX_ST_SRC_SIDETONE_SHIFT 2 +#define DA7219_DMIX_ST_SRC_OUTFILT1L (0x1 << 0) +#define DA7219_DMIX_ST_SRC_OUTFILT1R (0x1 << 1) + +/* DA7219_DROUTING_ST_OUTFILT_1R = 0x3D */ +#define DA7219_OUTFILT_ST_1R_SRC_SHIFT 0 +#define DA7219_OUTFILT_ST_1R_SRC_MASK (0x7 << 0) + +/* DA7219_DAC_FILTERS5 = 0x40 */ +#define DA7219_DAC_SOFTMUTE_RATE_SHIFT 4 +#define DA7219_DAC_SOFTMUTE_RATE_MASK (0x7 << 4) +#define DA7219_DAC_SOFTMUTE_RATE_MAX 7 +#define DA7219_DAC_SOFTMUTE_EN_SHIFT 7 +#define DA7219_DAC_SOFTMUTE_EN_MASK (0x1 << 7) + +/* DA7219_DAC_FILTERS2 = 0x41 */ +#define DA7219_DAC_EQ_BAND1_SHIFT 0 +#define DA7219_DAC_EQ_BAND1_MASK (0xF << 0) +#define DA7219_DAC_EQ_BAND2_SHIFT 4 +#define DA7219_DAC_EQ_BAND2_MASK (0xF << 4) +#define DA7219_DAC_EQ_BAND_MAX 0xF + +/* DA7219_DAC_FILTERS3 = 0x42 */ +#define DA7219_DAC_EQ_BAND3_SHIFT 0 +#define DA7219_DAC_EQ_BAND3_MASK (0xF << 0) +#define DA7219_DAC_EQ_BAND4_SHIFT 4 +#define DA7219_DAC_EQ_BAND4_MASK (0xF << 4) + +/* DA7219_DAC_FILTERS4 = 0x43 */ +#define DA7219_DAC_EQ_BAND5_SHIFT 0 +#define DA7219_DAC_EQ_BAND5_MASK (0xF << 0) +#define DA7219_DAC_EQ_EN_SHIFT 7 +#define DA7219_DAC_EQ_EN_MASK (0x1 << 7) + +/* DA7219_DAC_FILTERS1 = 0x44 */ +#define DA7219_DAC_VOICE_HPF_CORNER_SHIFT 0 +#define DA7219_DAC_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7219_DAC_VOICE_EN_SHIFT 3 +#define DA7219_DAC_VOICE_EN_MASK (0x1 << 3) +#define DA7219_DAC_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7219_DAC_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7219_DAC_HPF_EN_SHIFT 7 +#define DA7219_DAC_HPF_EN_MASK (0x1 << 7) + +/* DA7219_DAC_L_GAIN = 0x45 */ +#define DA7219_DAC_L_DIGITAL_GAIN_SHIFT 0 +#define DA7219_DAC_L_DIGITAL_GAIN_MASK (0x7F << 0) +#define DA7219_DAC_DIGITAL_GAIN_MAX 0x7F +#define DA7219_DAC_DIGITAL_GAIN_0DB (0x6F << 0) + +/* DA7219_DAC_R_GAIN = 0x46 */ +#define DA7219_DAC_R_DIGITAL_GAIN_SHIFT 0 +#define DA7219_DAC_R_DIGITAL_GAIN_MASK (0x7F << 0) + +/* DA7219_CP_CTRL = 0x47 */ +#define DA7219_CP_MCHANGE_SHIFT 4 +#define DA7219_CP_MCHANGE_MASK (0x3 << 4) +#define DA7219_CP_MCHANGE_REL_MASK 0x3 +#define DA7219_CP_MCHANGE_MAX 3 +#define DA7219_CP_MCHANGE_LARGEST_VOL 0x1 +#define DA7219_CP_MCHANGE_DAC_VOL 0x2 +#define DA7219_CP_MCHANGE_SIG_MAG 0x3 +#define DA7219_CP_EN_SHIFT 7 +#define DA7219_CP_EN_MASK (0x1 << 7) + +/* DA7219_HP_L_GAIN = 0x48 */ +#define DA7219_HP_L_AMP_GAIN_SHIFT 0 +#define DA7219_HP_L_AMP_GAIN_MASK (0x3F << 0) +#define DA7219_HP_AMP_GAIN_MAX 0x3F +#define DA7219_HP_AMP_GAIN_0DB (0x39 << 0) + +/* DA7219_HP_R_GAIN = 0x49 */ +#define DA7219_HP_R_AMP_GAIN_SHIFT 0 +#define DA7219_HP_R_AMP_GAIN_MASK (0x3F << 0) + +/* DA7219_MIXOUT_L_SELECT = 0x4B */ +#define DA7219_MIXOUT_L_MIX_SELECT_SHIFT 0 +#define DA7219_MIXOUT_L_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_MIXOUT_R_SELECT = 0x4C */ +#define DA7219_MIXOUT_R_MIX_SELECT_SHIFT 0 +#define DA7219_MIXOUT_R_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_SYSTEM_MODES_INPUT = 0x50 */ +#define DA7219_MODE_SUBMIT_SHIFT 0 +#define DA7219_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7219_ADC_MODE_SHIFT 1 +#define DA7219_ADC_MODE_MASK (0x7F << 1) + +/* DA7219_SYSTEM_MODES_OUTPUT = 0x51 */ +#define DA7219_MODE_SUBMIT_SHIFT 0 +#define DA7219_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7219_DAC_MODE_SHIFT 1 +#define DA7219_DAC_MODE_MASK (0x7F << 1) + +/* DA7219_MICBIAS_CTRL = 0x62 */ +#define DA7219_MICBIAS1_LEVEL_SHIFT 0 +#define DA7219_MICBIAS1_LEVEL_MASK (0x7 << 0) +#define DA7219_MICBIAS1_EN_SHIFT 3 +#define DA7219_MICBIAS1_EN_MASK (0x1 << 3) + +/* DA7219_MIC_1_CTRL = 0x63 */ +#define DA7219_MIC_1_AMP_RAMP_EN_SHIFT 5 +#define DA7219_MIC_1_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_MIC_1_AMP_MUTE_EN_SHIFT 6 +#define DA7219_MIC_1_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_MIC_1_AMP_EN_SHIFT 7 +#define DA7219_MIC_1_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXIN_L_CTRL = 0x65 */ +#define DA7219_MIXIN_L_MIX_EN_SHIFT 3 +#define DA7219_MIXIN_L_MIX_EN_MASK (0x1 << 3) +#define DA7219_MIXIN_L_AMP_ZC_EN_SHIFT 4 +#define DA7219_MIXIN_L_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT 5 +#define DA7219_MIXIN_L_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT 6 +#define DA7219_MIXIN_L_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_MIXIN_L_AMP_EN_SHIFT 7 +#define DA7219_MIXIN_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_ADC_L_CTRL = 0x67 */ +#define DA7219_ADC_L_BIAS_SHIFT 0 +#define DA7219_ADC_L_BIAS_MASK (0x3 << 0) +#define DA7219_ADC_L_RAMP_EN_SHIFT 5 +#define DA7219_ADC_L_RAMP_EN_MASK (0x1 << 5) +#define DA7219_ADC_L_MUTE_EN_SHIFT 6 +#define DA7219_ADC_L_MUTE_EN_MASK (0x1 << 6) +#define DA7219_ADC_L_EN_SHIFT 7 +#define DA7219_ADC_L_EN_MASK (0x1 << 7) + +/* DA7219_DAC_L_CTRL = 0x69 */ +#define DA7219_DAC_L_RAMP_EN_SHIFT 5 +#define DA7219_DAC_L_RAMP_EN_MASK (0x1 << 5) +#define DA7219_DAC_L_MUTE_EN_SHIFT 6 +#define DA7219_DAC_L_MUTE_EN_MASK (0x1 << 6) +#define DA7219_DAC_L_EN_SHIFT 7 +#define DA7219_DAC_L_EN_MASK (0x1 << 7) + +/* DA7219_DAC_R_CTRL = 0x6A */ +#define DA7219_DAC_R_RAMP_EN_SHIFT 5 +#define DA7219_DAC_R_RAMP_EN_MASK (0x1 << 5) +#define DA7219_DAC_R_MUTE_EN_SHIFT 6 +#define DA7219_DAC_R_MUTE_EN_MASK (0x1 << 6) +#define DA7219_DAC_R_EN_SHIFT 7 +#define DA7219_DAC_R_EN_MASK (0x1 << 7) + +/* DA7219_HP_L_CTRL = 0x6B */ +#define DA7219_HP_L_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7219_HP_L_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7219_HP_L_AMP_OE_SHIFT 3 +#define DA7219_HP_L_AMP_OE_MASK (0x1 << 3) +#define DA7219_HP_L_AMP_ZC_EN_SHIFT 4 +#define DA7219_HP_L_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_HP_L_AMP_RAMP_EN_SHIFT 5 +#define DA7219_HP_L_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_HP_L_AMP_MUTE_EN_SHIFT 6 +#define DA7219_HP_L_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_HP_L_AMP_EN_SHIFT 7 +#define DA7219_HP_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_HP_R_CTRL = 0x6C */ +#define DA7219_HP_R_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7219_HP_R_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7219_HP_R_AMP_OE_SHIFT 3 +#define DA7219_HP_R_AMP_OE_MASK (0x1 << 3) +#define DA7219_HP_R_AMP_ZC_EN_SHIFT 4 +#define DA7219_HP_R_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_HP_R_AMP_RAMP_EN_SHIFT 5 +#define DA7219_HP_R_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_HP_R_AMP_MUTE_EN_SHIFT 6 +#define DA7219_HP_R_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_HP_R_AMP_EN_SHIFT 7 +#define DA7219_HP_R_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXOUT_L_CTRL = 0x6E */ +#define DA7219_MIXOUT_L_AMP_EN_SHIFT 7 +#define DA7219_MIXOUT_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXOUT_R_CTRL = 0x6F */ +#define DA7219_MIXOUT_R_AMP_EN_SHIFT 7 +#define DA7219_MIXOUT_R_AMP_EN_MASK (0x1 << 7) + +/* DA7219_CHIP_ID1 = 0x81 */ +#define DA7219_CHIP_ID1_SHIFT 0 +#define DA7219_CHIP_ID1_MASK (0xFF << 0) + +/* DA7219_CHIP_ID2 = 0x82 */ +#define DA7219_CHIP_ID2_SHIFT 0 +#define DA7219_CHIP_ID2_MASK (0xFF << 0) + +/* DA7219_CHIP_REVISION = 0x83 */ +#define DA7219_CHIP_MINOR_SHIFT 0 +#define DA7219_CHIP_MINOR_MASK (0xF << 0) +#define DA7219_CHIP_MAJOR_SHIFT 4 +#define DA7219_CHIP_MAJOR_MASK (0xF << 4) + +/* DA7219_LDO_CTRL = 0x90 */ +#define DA7219_LDO_LEVEL_SELECT_SHIFT 4 +#define DA7219_LDO_LEVEL_SELECT_MASK (0x3 << 4) +#define DA7219_LDO_EN_SHIFT 7 +#define DA7219_LDO_EN_MASK (0x1 << 7) + +/* DA7219_IO_CTRL = 0x91 */ +#define DA7219_IO_VOLTAGE_LEVEL_SHIFT 0 +#define DA7219_IO_VOLTAGE_LEVEL_MASK (0x1 << 0) +#define DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V 0 +#define DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V 1 + +/* DA7219_GAIN_RAMP_CTRL = 0x92 */ +#define DA7219_GAIN_RAMP_RATE_SHIFT 0 +#define DA7219_GAIN_RAMP_RATE_MASK (0x3 << 0) +#define DA7219_GAIN_RAMP_RATE_MAX 4 + +/* DA7219_PC_COUNT = 0x94 */ +#define DA7219_PC_FREERUN_SHIFT 0 +#define DA7219_PC_FREERUN_MASK (0x1 << 0) +#define DA7219_PC_RESYNC_AUTO_SHIFT 1 +#define DA7219_PC_RESYNC_AUTO_MASK (0x1 << 1) + +/* DA7219_CP_VOL_THRESHOLD1 = 0x95 */ +#define DA7219_CP_THRESH_VDD2_SHIFT 0 +#define DA7219_CP_THRESH_VDD2_MASK (0x3F << 0) +#define DA7219_CP_THRESH_VDD2_MAX 0x3F + +/* DA7219_DIG_CTRL = 0x99 */ +#define DA7219_DAC_L_INV_SHIFT 3 +#define DA7219_DAC_L_INV_MASK (0x1 << 3) +#define DA7219_DAC_R_INV_SHIFT 7 +#define DA7219_DAC_R_INV_MASK (0x1 << 7) + +/* DA7219_ALC_CTRL2 = 0x9A */ +#define DA7219_ALC_ATTACK_SHIFT 0 +#define DA7219_ALC_ATTACK_MASK (0xF << 0) +#define DA7219_ALC_ATTACK_MAX 13 +#define DA7219_ALC_RELEASE_SHIFT 4 +#define DA7219_ALC_RELEASE_MASK (0xF << 4) +#define DA7219_ALC_RELEASE_MAX 11 + +/* DA7219_ALC_CTRL3 = 0x9B */ +#define DA7219_ALC_HOLD_SHIFT 0 +#define DA7219_ALC_HOLD_MASK (0xF << 0) +#define DA7219_ALC_HOLD_MAX 16 +#define DA7219_ALC_INTEG_ATTACK_SHIFT 4 +#define DA7219_ALC_INTEG_ATTACK_MASK (0x3 << 4) +#define DA7219_ALC_INTEG_RELEASE_SHIFT 6 +#define DA7219_ALC_INTEG_RELEASE_MASK (0x3 << 6) +#define DA7219_ALC_INTEG_MAX 4 + +/* DA7219_ALC_NOISE = 0x9C */ +#define DA7219_ALC_NOISE_SHIFT 0 +#define DA7219_ALC_NOISE_MASK (0x3F << 0) +#define DA7219_ALC_THRESHOLD_MAX 0x3F + +/* DA7219_ALC_TARGET_MIN = 0x9D */ +#define DA7219_ALC_THRESHOLD_MIN_SHIFT 0 +#define DA7219_ALC_THRESHOLD_MIN_MASK (0x3F << 0) + +/* DA7219_ALC_TARGET_MAX = 0x9E */ +#define DA7219_ALC_THRESHOLD_MAX_SHIFT 0 +#define DA7219_ALC_THRESHOLD_MAX_MASK (0x3F << 0) + +/* DA7219_ALC_GAIN_LIMITS = 0x9F */ +#define DA7219_ALC_ATTEN_MAX_SHIFT 0 +#define DA7219_ALC_ATTEN_MAX_MASK (0xF << 0) +#define DA7219_ALC_GAIN_MAX_SHIFT 4 +#define DA7219_ALC_GAIN_MAX_MASK (0xF << 4) +#define DA7219_ALC_ATTEN_GAIN_MAX 0xF + +/* DA7219_ALC_ANA_GAIN_LIMITS = 0xA0 */ +#define DA7219_ALC_ANA_GAIN_MIN_SHIFT 0 +#define DA7219_ALC_ANA_GAIN_MIN_MASK (0x7 << 0) +#define DA7219_ALC_ANA_GAIN_MIN 0x1 +#define DA7219_ALC_ANA_GAIN_MAX_SHIFT 4 +#define DA7219_ALC_ANA_GAIN_MAX_MASK (0x7 << 4) +#define DA7219_ALC_ANA_GAIN_MAX 0x7 + +/* DA7219_ALC_ANTICLIP_CTRL = 0xA1 */ +#define DA7219_ALC_ANTICLIP_STEP_SHIFT 0 +#define DA7219_ALC_ANTICLIP_STEP_MASK (0x3 << 0) +#define DA7219_ALC_ANTICLIP_STEP_MAX 4 +#define DA7219_ALC_ANTIPCLIP_EN_SHIFT 7 +#define DA7219_ALC_ANTIPCLIP_EN_MASK (0x1 << 7) + +/* DA7219_ALC_ANTICLIP_LEVEL = 0xA2 */ +#define DA7219_ALC_ANTICLIP_LEVEL_SHIFT 0 +#define DA7219_ALC_ANTICLIP_LEVEL_MASK (0x7F << 0) + +/* DA7219_ALC_OFFSET_AUTO_M_L = 0xA3 */ +#define DA7219_ALC_OFFSET_AUTO_M_L_SHIFT 0 +#define DA7219_ALC_OFFSET_AUTO_M_L_MASK (0xFF << 0) + +/* DA7219_ALC_OFFSET_AUTO_U_L = 0xA4 */ +#define DA7219_ALC_OFFSET_AUTO_U_L_SHIFT 0 +#define DA7219_ALC_OFFSET_AUTO_U_L_MASK (0xF << 0) + +/* DA7219_DAC_NG_SETUP_TIME = 0xAF */ +#define DA7219_DAC_NG_SETUP_TIME_SHIFT 0 +#define DA7219_DAC_NG_SETUP_TIME_MASK (0x3 << 0) +#define DA7219_DAC_NG_SETUP_TIME_MAX 4 +#define DA7219_DAC_NG_RAMPUP_RATE_SHIFT 2 +#define DA7219_DAC_NG_RAMPUP_RATE_MASK (0x1 << 2) +#define DA7219_DAC_NG_RAMPDN_RATE_SHIFT 3 +#define DA7219_DAC_NG_RAMPDN_RATE_MASK (0x1 << 3) +#define DA7219_DAC_NG_RAMP_RATE_MAX 2 + +/* DA7219_DAC_NG_OFF_THRESH = 0xB0 */ +#define DA7219_DAC_NG_OFF_THRESHOLD_SHIFT 0 +#define DA7219_DAC_NG_OFF_THRESHOLD_MASK (0x7 << 0) +#define DA7219_DAC_NG_THRESHOLD_MAX 0x7 + +/* DA7219_DAC_NG_ON_THRESH = 0xB1 */ +#define DA7219_DAC_NG_ON_THRESHOLD_SHIFT 0 +#define DA7219_DAC_NG_ON_THRESHOLD_MASK (0x7 << 0) + +/* DA7219_DAC_NG_CTRL = 0xB2 */ +#define DA7219_DAC_NG_EN_SHIFT 7 +#define DA7219_DAC_NG_EN_MASK (0x1 << 7) + +/* DA7219_TONE_GEN_CFG1 = 0xB4 */ +#define DA7219_DTMF_REG_SHIFT 0 +#define DA7219_DTMF_REG_MASK (0xF << 0) +#define DA7219_DTMF_REG_MAX 16 +#define DA7219_DTMF_EN_SHIFT 4 +#define DA7219_DTMF_EN_MASK (0x1 << 4) +#define DA7219_START_STOPN_SHIFT 7 +#define DA7219_START_STOPN_MASK (0x1 << 7) + +/* DA7219_TONE_GEN_CFG2 = 0xB5 */ +#define DA7219_SWG_SEL_SHIFT 0 +#define DA7219_SWG_SEL_MASK (0x3 << 0) +#define DA7219_SWG_SEL_MAX 4 +#define DA7219_SWG_SEL_SRAMP (0x3 << 0) +#define DA7219_TONE_GEN_GAIN_SHIFT 4 +#define DA7219_TONE_GEN_GAIN_MASK (0xF << 4) +#define DA7219_TONE_GEN_GAIN_MAX 0xF +#define DA7219_TONE_GEN_GAIN_MINUS_9DB (0x3 << 4) +#define DA7219_TONE_GEN_GAIN_MINUS_15DB (0x5 << 4) + +/* DA7219_TONE_GEN_CYCLES = 0xB6 */ +#define DA7219_BEEP_CYCLES_SHIFT 0 +#define DA7219_BEEP_CYCLES_MASK (0x7 << 0) + +/* DA7219_TONE_GEN_FREQ1_L = 0xB7 */ +#define DA7219_FREQ1_L_SHIFT 0 +#define DA7219_FREQ1_L_MASK (0xFF << 0) +#define DA7219_FREQ_MAX 0xFFFF + +/* DA7219_TONE_GEN_FREQ1_U = 0xB8 */ +#define DA7219_FREQ1_U_SHIFT 0 +#define DA7219_FREQ1_U_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_FREQ2_L = 0xB9 */ +#define DA7219_FREQ2_L_SHIFT 0 +#define DA7219_FREQ2_L_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_FREQ2_U = 0xBA */ +#define DA7219_FREQ2_U_SHIFT 0 +#define DA7219_FREQ2_U_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_ON_PER = 0xBB */ +#define DA7219_BEEP_ON_PER_SHIFT 0 +#define DA7219_BEEP_ON_PER_MASK (0x3F << 0) +#define DA7219_BEEP_ON_OFF_MAX 0x3F + +/* DA7219_TONE_GEN_OFF_PER = 0xBC */ +#define DA7219_BEEP_OFF_PER_SHIFT 0 +#define DA7219_BEEP_OFF_PER_MASK (0x3F << 0) + +/* DA7219_SYSTEM_STATUS = 0xE0 */ +#define DA7219_SC1_BUSY_SHIFT 0 +#define DA7219_SC1_BUSY_MASK (0x1 << 0) +#define DA7219_SC2_BUSY_SHIFT 1 +#define DA7219_SC2_BUSY_MASK (0x1 << 1) + +/* DA7219_SYSTEM_ACTIVE = 0xFD */ +#define DA7219_SYSTEM_ACTIVE_SHIFT 0 +#define DA7219_SYSTEM_ACTIVE_MASK (0x1 << 0) + + +/* + * General defines & data + */ + +/* Register inversion */ +#define DA7219_NO_INVERT 0 +#define DA7219_INVERT 1 + +/* Byte related defines */ +#define DA7219_BYTE_SHIFT 8 +#define DA7219_BYTE_MASK 0xFF + +/* PLL Output Frequencies */ +#define DA7219_PLL_FREQ_OUT_90316 90316800 +#define DA7219_PLL_FREQ_OUT_98304 98304000 + +/* PLL Frequency Dividers */ +#define DA7219_PLL_INDIV_2_5_MHZ_VAL 1 +#define DA7219_PLL_INDIV_5_10_MHZ_VAL 2 +#define DA7219_PLL_INDIV_10_20_MHZ_VAL 4 +#define DA7219_PLL_INDIV_20_40_MHZ_VAL 8 +#define DA7219_PLL_INDIV_40_54_MHZ_VAL 16 + +/* SRM */ +#define DA7219_SRM_CHECK_RETRIES 8 + +enum da7219_clk_src { + DA7219_CLKSRC_MCLK = 0, + DA7219_CLKSRC_MCLK_SQR, +}; + +enum da7219_sys_clk { + DA7219_SYSCLK_MCLK = 0, + DA7219_SYSCLK_PLL, + DA7219_SYSCLK_PLL_SRM, + DA7219_SYSCLK_PLL_32KHZ +}; + +/* Regulators */ +enum da7219_supplies { + DA7219_SUPPLY_VDD = 0, + DA7219_SUPPLY_VDDMIC, + DA7219_SUPPLY_VDDIO, + DA7219_NUM_SUPPLIES, +}; + +struct da7219_aad_priv; + +/* Private data */ +struct da7219_priv { + struct da7219_aad_priv *aad; + struct da7219_pdata *pdata; + + struct regulator_bulk_data supplies[DA7219_NUM_SUPPLIES]; + struct regmap *regmap; + struct mutex lock; + + struct clk *mclk; + unsigned int mclk_rate; + int clk_src; + + bool master; + bool alc_en; +}; + +#endif /* __DA7219_H */ -- 2.5.0

1 0

[alsa-devel] [PATCH v2 0/4] ASoC: Add support for DA7219 audio codec
by Adam Thomson 02 Oct '15

02 Oct '15

This patch set adds support for the DA7219 audio codec with built-in advanced accessory detection functionality. Patch set includes codec driver, associated DT bindings documentation and MAINTAINERS file updates to cover new bindings. This patch set is baselined against the v4.3-rc2 kernel version. Changes in v2: - Add SOC_DOUBLE_R_EXT support to ASoC core. - Use of upper-case naming for controls, where missing. - Use of Playback/Capture Digital naming for certain ADC/DAC controls. - CP mchange and vol_thresh DT bindings removed, now Kcontrols - Kcontrols and DAPM now locked as part of hptest work, and specific controls affected have had locking introduced. - Timeout for btn_det work when checking micbias status. - Added comments to describe reasoning for work use with hptest, btn_det. - Added comments to tonegen_freq_put/get to clarify endian conversion, and code now uses regmap_raw_read/write calls instead of bulk. - DT parsing code now prints dev_warn() for invalid settings. - SOC_VALUE_ENUM_SINGLE now used for hpf_mode handling, instead of customer get/put handlers. - DTMF key control now a SOC_ENUM to correctly describe supported keys, including non-numeric. - MIXIN_L_MIX_EN bit set as part of DAPM now (SUPPLY), rather than in probe. - Fix dai_sysclk() function to correctly check for no change, using clk_id as well as freq. - Error checking added for clk_set_rate() call. - Improved error checking for devm_clk_get() call. - Removed explicit use of device_may_wakeup() and enable/disable_irq_wake(). Now taken care of by I2C core so DT bindings updated accordingly to document this. - hptest work now uses cache to keep state, instead of reading registers individually. Cache is bypassed, then when test executed and complete, regcache_sync_region() used to restore settings to pre-test values. - Regulator consumer framework usage added, in particular to determine VDDIO level in use so device can be configured accordingly. DT documents updated to reflect this. Adam Thomson (4): ASoC: Add SOC_DOUBLE_R_EXT ASoC: codecs: Add da7219 codec driver ASoC: da7219: Add bindings documentation for DA7219 audio codec MAINTAINERS: da7219: Add entry to cover DA7219 bindings document Documentation/devicetree/bindings/sound/da7219.txt | 106 ++ MAINTAINERS | 1 + include/sound/da7219-aad.h | 99 + include/sound/da7219.h | 55 + include/sound/soc.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 823 +++++++++ sound/soc/codecs/da7219-aad.h | 212 +++ sound/soc/codecs/da7219.c | 1938 ++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 +++++++++ 11 files changed, 4067 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h -- 1.9.3

2 5

[alsa-devel] [PATCH 0/3] ASoC: Add support for DA7219 audio codec
by Adam Thomson 02 Oct '15

02 Oct '15

This patch set adds support for the DA7219 audio codec with built-in advanced accessory detection functionality. Patch set includes codec driver, associated DT bindings documentation and MAINTAINERS file updates to cover new bindings. Adam Thomson (3): ASoC: codecs: Add da7219 codec driver ASoC: da7219: Add bindings documentation for DA7219 audio codec MAINTAINERS: da7219: Add entry to cover DA7219 bindings document Documentation/devicetree/bindings/sound/da7219.txt | 106 ++ MAINTAINERS | 1 + include/sound/da7219-aad.h | 99 ++ include/sound/da7219.h | 76 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 800 +++++++++ sound/soc/codecs/da7219-aad.h | 209 +++ sound/soc/codecs/da7219.c | 1877 ++++++++++++++++++++ sound/soc/codecs/da7219.h | 807 +++++++++ 10 files changed, 3981 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h -- 1.9.3

3 6

[alsa-devel] [PATCH] ASoC: rt5645: Allow 4 channel recording on AIF1
by Ben Zhang 02 Oct '15

02 Oct '15

The codec supports 4 channel recording with TDM on AIF1. This patch modifies the DAI capability to allow it. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- 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 b0d96b6..0d092bf 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3101,7 +3101,7 @@ static struct snd_soc_dai_driver rt5645_dai[] = { .capture = { .stream_name = "AIF1 Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 4, .rates = RT5645_STEREO_RATES, .formats = RT5645_FORMATS, }, -- 2.6.0.rc2.230.g3dd15c0

2 3

[alsa-devel] ATMEL sam9g25 SSC in I2S slave mode on linux 4.1.1 play only right channel
by Paolo Tagliavini 02 Oct '15

02 Oct '15

I'm working on my based sam9g25 design. To have SPDIF output I used cs4265 codec instead of wm8731 as sem9g25 reference design. Now I'm working on porting of cs4265 codec driver. It work for play only right channel, left channel is mute and not supplied by SSC. SSC is configured as I2S slave : SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM Do you know this behaviour ? There is a patch for fix it ? Thank you in advance.

1 0

[alsa-devel] [PATCH 1/1] ASoc: dwc: correct irq clear method
by yitian 02 Oct '15

02 Oct '15

from Designware I2S datasheet, irq is cleared by reading from TOR/ROR registers, rather than by writing into them. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..0d28e3b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -131,10 +131,10 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, TOR(i), 0); + i2s_read_reg(dev->i2s_base, TOR(i)); } else { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, ROR(i), 0); + i2s_read_reg(dev->i2s_base, ROR(i)); } } -- 1.7.12.4

2 1

Re: [alsa-devel] Bug: LADSPA implementation - no up/downmix handling
by Takashi Iwai 02 Oct '15

02 Oct '15

On Fri, 25 Sep 2015 20:51:38 +0200, Andreas Hartmann wrote: > > Am Donnerstag, 24. September 2015, 15:07:14 schrieben Sie: > > On Thu, 24 Sep 2015 12:44:53 +0200, > > > > Andreas Hartmann wrote: > > > Am Mittwoch, 23. September 2015, 16:26:58 schrieb Takashi Iwai: > > > > On Sat, 12 Sep 2015 22:10:40 +0200, > > > > > > > > Andreas Hartmann wrote: > > > > > Hi, > > > > > > > > > > Using the surround_encoder of the swh-plugins I noticed, that the alsa > > > > > implementation only assumes an equal number of in- and output channels > > > > > for > > > > > ladspa plugins. This occurs from Line 752 onwards in pcm_ladspa.c and > > > > > leads to a Segfault shortly after. > > > > > There needs to be a check and correct handling for up/downmix > > > > > scenarios. > > > > > > > > Could you give the back trace? Or at best submit a fix patch? > > > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > _______________________________________________ > > > > Alsa-devel mailing list > > > > Alsa-devel(a)alsa-project.org > > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > > Sure here you go > > > > > > #0 0x00007ffff78623ef in snd_pcm_ladspa_allocate_memory (pcm=0x626fa0, > > > pcm=0x626fa0, pcm=0x626fa0, ladspa=0x621ad0) at pcm_ladspa.c:753 > > > #1 snd_pcm_ladspa_init (pcm=0x626fa0) at pcm_ladspa.c:834 > > > #2 0x00007ffff7842946 in snd_pcm_plugin_prepare (pcm=0x626fa0) at > > > pcm_plugin.c:171 > > > #3 0x00007ffff784290f in snd_pcm_plugin_prepare (pcm=0x62c760) at > > > pcm_plugin.c:162 > > > #4 0x000000000040256a in ?? () > > > #5 0x00007ffff7222ec5 in __libc_start_main (main=0x401d80, argc=4, > > > argv=0x7fffffffde28, init=<optimized out>, fini=<optimized out>, > > > rtld_fini=<optimized out>, > > > > > > stack_end=0x7fffffffde18) at libc-start.c:287 > > > > > > #6 0x0000000000402fdd in ?? () > > > > It looks rather like a simply typo. Does the patch below work? > > > > > > Takashi > > > > diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c > > index 631ee0f35b73..6552b4379ec0 100644 > > --- a/src/pcm/pcm_ladspa.c > > +++ b/src/pcm/pcm_ladspa.c > > @@ -749,7 +749,7 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t > > *pcm, snd_pcm_ladspa_t *lads return -ENOMEM; > > } > > for (idx = 0; idx < instance->input.channels.size; idx++) { > > - chn = instance->output.channels.array[idx]; > > + chn = instance->input.channels.array[idx]; > > if (pchannels[chn] == NULL && chn < ichannels) { > > instance->input.data[idx] = NULL; > > continue; > > It seems to work Sorry, I forgot to apply this to upstream. Now pushed out. thanks, Takashi

1 0

[alsa-devel] [RESEND PATCH 1/1] ASoC: dwc: correct irq clear method
by yitian 02 Oct '15

02 Oct '15

from Designware I2S datasheet, irq is cleared by reading from TOR/ROR registers, rather than by writing into them. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..0d28e3b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -131,10 +131,10 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, TOR(i), 0); + i2s_read_reg(dev->i2s_base, TOR(i)); } else { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, ROR(i), 0); + i2s_read_reg(dev->i2s_base, ROR(i)); } } -- 1.7.12.4

2 4

[alsa-devel] [PATCH 1/1] ASoC: dwc: correct irq clear method
by Yitian Bu 02 Oct '15

02 Oct '15

from Designware I2S datasheet, tx/rx XRUN irq is cleared by reading register TOR/ROR, rather than by writing into them. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..0d28e3b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -131,10 +131,10 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, TOR(i), 0); + i2s_read_reg(dev->i2s_base, TOR(i)); } else { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, ROR(i), 0); + i2s_read_reg(dev->i2s_base, ROR(i)); } } -- 1.7.12.4

1 0

[alsa-devel] [PATCH 0/2] CONFIG_OF vs DMA_OF
by Kuninori Morimoto 02 Oct '15

02 Oct '15

Hi Vinod, Mark Current OF base DMAEngine API will be compiled if .config has CONFIG_DMA_OF. but of-dma.h based on CONFIG_OF instead of CONFIG_DMA_OF. These patches tidyup this issue. Please check 1) patch carefully, since DMAEngine API has many dependency to other drivers. > Mark I think 2) patch is based on 1). Please check it Kuninori Morimoto (2): 1) dmaengine: OF DMAEngine API based on CONFIG_DMA_OF instead of CONFIG_OF 2) ASoC: rsnd: SND_SOC_RCAR doesn't depend on DMA_OF include/linux/of_dma.h | 2 +- sound/soc/sh/Kconfig | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) Best regards --- Kuninori Morimoto

2 9

[alsa-devel] Additional bugs in tlv320aic3x.c driver (register defaults)
by Rick Mann 01 Oct '15

01 Oct '15

Benoît, I've discovered some additional minor bugs in the tlv320aic3104 support in the tlv320aic3x driver. In this instance, the reg_default values aren't quite right. I've written a corrected list, but the set specified includes the reserved registers. I don't know if devm_regmap_init_i2c() attempts to write the default values, or uses them as the initial values in a cache (and attempts to avoid writes unless specific registers change). In my debugging, it does not appear to write the entire register map at any point to the IC, so I think it's the latter. I also don't know how the regmap is intended to be used. It is passed to devm_regmap_init_i2c() as .reg_defaults, implying the array provides starting values, but other parts of the code refer to the array as if the values are changed in that array as they're read and written. Documentation for regmap suggests regmaps can be "sparse," which I presume would be done by omitting the reserved registers from the defaults array. In any case, the fix is not trivial, and I don't want to go down that road not knowing if it's the right approach. Does this make sense? For reference, the registers with different default values (than the ones specified in that file) are 32, 33, 51, 58, 65, 72, 93. Thanks, -- Rick Mann rmann(a)latencyzero.com

2 1

[alsa-devel] Enabling kernel debugging in sound before devices are loaded
by Rick Mann 01 Oct '15

01 Oct '15

I'm trying to enable kernel debug logging like this: # echo "file sound* +pflm" > /sys/kernel/debug/dynamic_debug/control But it has no effect until after I load the Device Tree Overlay that creates my codec and sound card nodes. Unfortunately at this point, it's too late for the debugging to do any good. This is with kernel 4.1.9 on a Beaglebone Black Debian Jessie distro. I don't really know how the kernel dyndebug system works; I am able to enable other debugging locations, but none in sound/* Appreciate any guidance. Thank you. -- Rick Mann rmann(a)latencyzero.com

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Correct the naming and setting of ADC Boost Volume Control
by Oder Chiou 01 Oct '15

01 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 +++--- sound/soc/codecs/rt5645.h | 16 +++++++++------- 2 files changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3eb184c..080cc1c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -519,11 +519,11 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { RT5645_L_VOL_SFT + 1, RT5645_R_VOL_SFT + 1, 63, 0, adc_vol_tlv), /* ADC Boost Volume Control */ - SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1, + SOC_DOUBLE_TLV("ADC Boost Capture Volume", RT5645_ADC_BST_VOL1, RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1, - RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0, + SOC_DOUBLE_TLV("Mono ADC Boost Capture Volume", RT5645_ADC_BST_VOL2, + RT5645_MONO_ADC_L_BST_SFT, RT5645_MONO_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), /* I2S2 function select */ diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index f45861c..61bc8ab 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -39,8 +39,8 @@ #define RT5645_STO1_ADC_DIG_VOL 0x1c #define RT5645_MONO_ADC_DIG_VOL 0x1d #define RT5645_ADC_BST_VOL1 0x1e -/* Mixer - D-D */ #define RT5645_ADC_BST_VOL2 0x20 +/* Mixer - D-D */ #define RT5645_STO1_ADC_MIXER 0x27 #define RT5645_MONO_ADC_MIXER 0x28 #define RT5645_AD_DA_MIXER 0x29 @@ -315,12 +315,14 @@ #define RT5645_STO1_ADC_R_BST_SFT 12 #define RT5645_STO1_ADC_COMP_MASK (0x3 << 10) #define RT5645_STO1_ADC_COMP_SFT 10 -#define RT5645_STO2_ADC_L_BST_MASK (0x3 << 8) -#define RT5645_STO2_ADC_L_BST_SFT 8 -#define RT5645_STO2_ADC_R_BST_MASK (0x3 << 6) -#define RT5645_STO2_ADC_R_BST_SFT 6 -#define RT5645_STO2_ADC_COMP_MASK (0x3 << 4) -#define RT5645_STO2_ADC_COMP_SFT 4 + +/* ADC Boost Volume Control (0x20) */ +#define RT5645_MONO_ADC_L_BST_MASK (0x3 << 14) +#define RT5645_MONO_ADC_L_BST_SFT 14 +#define RT5645_MONO_ADC_R_BST_MASK (0x3 << 12) +#define RT5645_MONO_ADC_R_BST_SFT 12 +#define RT5645_MONO_ADC_COMP_MASK (0x3 << 10) +#define RT5645_MONO_ADC_COMP_SFT 10 /* Stereo2 ADC Mixer Control (0x26) */ #define RT5645_STO2_ADC_SRC_MASK (0x1 << 15) -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] Prevent writing reserved registers on tlv320aic3104 CODECs
by Rick Mann 01 Oct '15

01 Oct '15

The current code writes a set of registers that are reserved on the tlc320aic3104. The change skips those registers for that IC. Signed-off-by: Rick Mann <rmann(a)latencyzero.com> --- sound/soc/codecs/tlv320aic3x.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f1c9fffcd3a6..8d893e94e78d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1531,14 +1531,17 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); - /* Line2 Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + /* On tlv320aic3104, these registers are reserved and must not be written */ + if (aic3x->model != AIC3X_MODEL_3104) { + /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + /* Line2 Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + } switch (aic3x->model) { case AIC3X_MODEL_3X: -- 1.9.1

2 2

[alsa-devel] Driver for audio power amplifier ICs?
by Rick Mann 01 Oct '15

01 Oct '15

I've been working on this Beaglebone Black custom audio cape. It has a tlv320aic3104 codec (that I'm still trying to get working properly), and connected to its line outs is a tpa3140d2 analog 2x10W power amplifier. This chip has very little in the way of CPU interface, but it does have /shutdown and /fault lines. I'm wondering how ALSA becomes aware of that chip, in order to enable it when it enables certain routes in the tlv320aic3104, and to inform application software if the /fault line asserts. I don't mind writing a driver, if necessary, I just don't know where such a thing would fit. I also wonder what the device tree entries for the amp might look like. Thanks! -- Rick Mann rmann(a)latencyzero.com

2 3

[alsa-devel] aic31xx_set_bias_level not called
by Matthias Weißer 01 Oct '15

01 Oct '15

Hi I currently try to port an existing alsa driver to a new hardware. The only difference between these to board is the used codec. Where the original board (iMX6SX SDB from Freescale) uses a wm8962 our custom board uses a tlv320aic3110. So I took the file sound\soc\fsl\imx-wm8962.c and changed it according to the new codec. From what I understand the main things where just renaming. I then setup my device tree according to the new environment and the kernel seemed to setup stuff correctly. But when I try to play some sound using speaker-test I see some errors and don't hear anything. Tracking stuff down using dev_dbg outputs I see that aic31xx_set_bias_level is never called with a value > 1 which causes that the clocks of the codec are never enabled. Maybe someone can give me some hints on what can be wrong. Here are some things which might be useful: ~ # dmesg | grep -i -A 2 ALSA [ 5.058905] ALSA device list: [ 5.061885] #0: tlv320aic31xx-audio ~ # dmesg | grep tlv [ 1.598495] tlv320aic31xx-codec 2-0018: ## aic31xx_i2c_probe: tlv320aic3110 codec_type = 1 [ 1.613239] tlv320aic31xx-codec 2-0018: ## aic31xx_codec_probe [ 1.621895] tlv320aic31xx-codec 2-0018: ## aic31xx_set_dai_fmt: fmt = 0x1001 [ 1.630390] imx-tlv320aic31xx sound: tlv320aic31xx-hifi <-> 202c000.ssi mapping ok [ 1.638145] tlv320aic31xx-codec 2-0018: ## aic31xx_set_dai_sysclk: clk_id = 0, freq = 24000000, dir = 0 [ 1.663566] tlv320aic31xx-codec 2-0018: ## aic31xx_set_bias_level: 0 -> 1 [ 5.061885] #0: tlv320aic31xx-audio [ 36.860404] tlv320aic31xx-codec 2-0018: ## aic31xx_hw_params: width 16 rate 48000 [ 36.886777] tlv320aic31xx-codec 2-0018: pll 8.1920/2 dosr 128 n 8 m 2 aosr 128 n 8 m 2 bclk_n 16 # aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=tlv320aic31xxau tlv320aic31xx-audio, Default Audio Device sysdefault:CARD=tlv320aic31xxau tlv320aic31xx-audio, Default Audio Device # speaker-test -Ddefault speaker-test 1.0.29 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 32768 Period size range from 64 to 16384 Using max buffer size 32768 Periods = 4 was set period_size = 8192 was set buffer_size = 32768 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error # uname -a Linux imx6sx-g343-devel 4.1.4-dirty #49 Thu Oct 1 10:39:44 CEST 2015 armv7l GNU/Linux /* * Copyright 2013 Freescale Semiconductor, Inc. * * Based on imx-wm8962.c * Copyright 2012 Freescale Semiconductor, Inc. * Copyright 2012 Linaro Ltd. * * The code contained herein is licensed under the GNU General Public * License. You may obtain a copy of the GNU General Public License * Version 2 or later at the following locations: * * http://www.opensource.org/licenses/gpl-license.html * http://www.gnu.org/copyleft/gpl.html */ #define DEBUG #include <linux/module.h> #include <linux/of_platform.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/clk.h> #include <sound/soc.h> #include <sound/pcm_params.h> #include <sound/soc-dapm.h> #include <linux/pinctrl/consumer.h> #include "../codecs/tlv320aic31xx.h" #include "imx-audmux.h" #define DAI_NAME_SIZE 32 struct imx_tlv320aic31xx_data { struct snd_soc_dai_link dai; struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; struct clk *codec_clk; unsigned int clk_frequency; }; struct imx_priv { struct platform_device *pdev; }; static struct imx_priv card_priv; static const struct snd_soc_dapm_widget imx_tlv320aic31xx_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("AMIC", NULL), SND_SOC_DAPM_MIC("DMIC", NULL), }; static int sample_rate = 44100; static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { sample_rate = params_rate(params); sample_format = params_format(params); return 0; } static struct snd_soc_ops imx_hifi_ops = { .hw_params = imx_hifi_hw_params, }; static int imx_tlv320aic31xx_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; struct imx_tlv320aic31xx_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret = 0; ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->clk_frequency, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(dev, "failed to set sysclk in %s\n", __func__); return ret; } static int imx_tlv320aic31xx_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct imx_priv *priv = &card_priv; struct i2c_client *codec_dev; struct imx_tlv320aic31xx_data *data; int int_port, ext_port; int ret; priv->pdev = pdev; ret = of_property_read_u32(np, "mux-int-port", &int_port); if (ret) { dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); return ret; } ret = of_property_read_u32(np, "mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); return ret; } /* * The port numbering in the hardware manual starts at 1, while * the audmux API expects it starts at 0. */ int_port--; ext_port--; ret = imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | IMX_AUDMUX_V2_PTCR_TFSDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR, IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); if (ret) { dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { dev_err(&pdev->dev, "audmux external port setup failed\n"); return ret; } ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); if (!ssi_np || !codec_np) { dev_err(&pdev->dev, "phandle missing or invalid\n"); ret = -EINVAL; goto fail; } ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); ret = -EINVAL; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev || !codec_dev->dev.driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); ret = -EINVAL; goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { ret = -ENOMEM; goto fail; } data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); goto fail; } data->clk_frequency = clk_get_rate(data->codec_clk); ret = clk_prepare_enable(data->codec_clk); if (ret) { dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); goto fail; } data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; data->dai.codec_dai_name = "tlv320aic31xx-hifi"; data->dai.codec_of_node = codec_np; data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) goto clk_fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) goto clk_fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; data->card.dapm_widgets = imx_tlv320aic31xx_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_tlv320aic31xx_dapm_widgets); data->card.late_probe = imx_tlv320aic31xx_late_probe; platform_set_drvdata(pdev, &data->card); snd_soc_card_set_drvdata(&data->card, data); ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; } of_node_put(ssi_np); of_node_put(codec_np); return 0; clk_fail: clk_disable_unprepare(data->codec_clk); fail: of_node_put(ssi_np); of_node_put(codec_np); return ret; } static int imx_tlv320aic31xx_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct imx_tlv320aic31xx_data *data = snd_soc_card_get_drvdata(card); if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); return 0; } static const struct of_device_id imx_tlv320aic31xx_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, imx_tlv320aic31xx_dt_ids); static struct platform_driver imx_tlv320aic31xx_driver = { .driver = { .name = "imx-tlv320aic31xx", .pm = &snd_soc_pm_ops, .of_match_table = imx_tlv320aic31xx_dt_ids, }, .probe = imx_tlv320aic31xx_probe, .remove = imx_tlv320aic31xx_remove, }; module_platform_driver(imx_tlv320aic31xx_driver); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("Freescale i.MX tlv320aic31xx ASoC machine driver"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:imx-tlv320aic31xx");

1 0

[alsa-devel] [PATCH] ASoC: Document DAI signal polarity
by Anatol Pomozov 01 Oct '15

01 Oct '15

Per discussion at [1] currently there is no clear definition of what is FSYNC polarity. Different drivers use its own definition of what is "normal" and what is "inverted" fsync in different modes. This leads to compatibility problems between drivers. Explicitly specify meaning of BCLK/FSYNC polarity. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/097121.html Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- include/sound/soc-dai.h | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..c8170c4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,15 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * - * Specifies whether the DAI can also support inverted clocks for the specified - * format. + * For BCLK: + * - "normal" polarity means signal sensing happens at rising edge of BCLK + * - "inverted" polarity means signal sensing happens at falling edge of BCLK + * + * For FSYNC: + * - "normal" polarity means frame starts at rising edge of FSYNC + * - "inverted" polarity means frame starts at falling edge of FSYNC */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- 2.5.0.457.gab17608

6 10

[alsa-devel] [PATCH] ASoC: pxa: pxa-pcm-lib: switch over to snd-soc-dmaengine-pcm
by Robert Jarzmik 01 Oct '15

01 Oct '15

From: Daniel Mack <zonque(a)gmail.com> This patch removes the old PXA DMA API usage and switches over to generic functions provided by snd-soc-dmaengine-pcm. More cleanups may be done on top of this, and some function stubs can now be removed completetly. However, the intention here was to keep the transition as small as possible. This was tested on the mioa701 pxa27x board. Signed-off-by: Daniel Mack <zonque(a)gmail.com> [trivial change from mmp-dma to pxa-dma] Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- include/sound/pxa2xx-lib.h | 1 - sound/arm/pxa2xx-ac97.c | 13 ++- sound/arm/pxa2xx-pcm-lib.c | 201 ++++++++------------------------------------ sound/arm/pxa2xx-pcm.c | 12 +-- sound/arm/pxa2xx-pcm.h | 2 - sound/soc/pxa/pxa2xx-ac97.c | 49 +++++++---- sound/soc/pxa/pxa2xx-i2s.c | 3 + sound/soc/pxa/pxa2xx-pcm.c | 32 ------- 8 files changed, 85 insertions(+), 228 deletions(-) diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 56e818e4a1cb..6ef629bde164 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -12,7 +12,6 @@ extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd); extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream); extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); -extern void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id); extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream); extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 38590b322c54..fbd5dad0c484 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,7 +44,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_reset, }; -static unsigned long pxa2xx_ac97_pcm_out_req = 12; +static struct pxad_param pxa2xx_ac97_pcm_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 12, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -51,7 +56,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .filter_data = &pxa2xx_ac97_pcm_out_req, }; -static unsigned long pxa2xx_ac97_pcm_in_req = 11; +static struct pxad_param pxa2xx_ac97_pcm_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 11, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 01f8fdc42b1b..e9b98af6b52c 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/dma-mapping.h> #include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <sound/core.h> #include <sound/pcm.h> @@ -15,8 +16,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include <mach/dma.h> - #include "pxa2xx-pcm.h" static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { @@ -31,7 +30,7 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .period_bytes_min = 32, .period_bytes_max = 8192 - 32, .periods_min = 1, - .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc), + .periods_max = 256, .buffer_bytes_max = 128 * 1024, .fifo_size = 32, }; @@ -39,65 +38,29 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd = runtime->private_data; - size_t totsize = params_buffer_bytes(params); - size_t period = params_period_bytes(params); - pxa_dma_desc *dma_desc; - dma_addr_t dma_buff_phys, next_desc_phys; - u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_dmaengine_dai_dma_data *dma_params; + struct dma_slave_config config; + int ret; - /* temporary transition hack */ - switch (rtd->params->addr_width) { - case DMA_SLAVE_BUSWIDTH_1_BYTE: - dcmd |= DCMD_WIDTH1; - break; - case DMA_SLAVE_BUSWIDTH_2_BYTES: - dcmd |= DCMD_WIDTH2; - break; - case DMA_SLAVE_BUSWIDTH_4_BYTES: - dcmd |= DCMD_WIDTH4; - break; - default: - /* can't happen */ - break; - } + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; - switch (rtd->params->maxburst) { - case 8: - dcmd |= DCMD_BURST8; - break; - case 16: - dcmd |= DCMD_BURST16; - break; - case 32: - dcmd |= DCMD_BURST32; - break; - } + ret = snd_hwparams_to_dma_slave_config(substream, params, &config); + if (ret) + return ret; - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = totsize; + snd_dmaengine_pcm_set_config_from_dai_data(substream, + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream), + &config); - dma_desc = rtd->dma_desc_array; - next_desc_phys = rtd->dma_desc_array_phys; - dma_buff_phys = runtime->dma_addr; - do { - next_desc_phys += sizeof(pxa_dma_desc); - dma_desc->ddadr = next_desc_phys; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dma_desc->dsadr = dma_buff_phys; - dma_desc->dtadr = rtd->params->addr; - } else { - dma_desc->dsadr = rtd->params->addr; - dma_desc->dtadr = dma_buff_phys; - } - if (period > totsize) - period = totsize; - dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN; - dma_desc++; - dma_buff_phys += period; - } while (totsize -= period); - dma_desc[-1].ddadr = rtd->dma_desc_array_phys; + ret = dmaengine_slave_config(chan, &config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; } @@ -105,13 +68,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_params); int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - - if (rtd && rtd->params && rtd->params->filter_data) { - unsigned long req = *(unsigned long *) rtd->params->filter_data; - DRCMR(req) = 0; - } - snd_pcm_set_runtime_buffer(substream, NULL); return 0; } @@ -119,100 +75,36 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_free); int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; - DCSR(prtd->dma_ch) = DCSR_RUN; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - DCSR(prtd->dma_ch) &= ~DCSR_RUN; - break; - - case SNDRV_PCM_TRIGGER_RESUME: - DCSR(prtd->dma_ch) |= DCSR_RUN; - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; - DCSR(prtd->dma_ch) |= DCSR_RUN; - break; - - default: - ret = -EINVAL; - } - - return ret; + return snd_dmaengine_pcm_trigger(substream, cmd); } EXPORT_SYMBOL(pxa2xx_pcm_trigger); snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *prtd = runtime->private_data; - - dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch); - snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr); - - if (x == runtime->buffer_size) - x = 0; - return x; + return snd_dmaengine_pcm_pointer(substream); } EXPORT_SYMBOL(pxa2xx_pcm_pointer); int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - unsigned long req; - - if (!prtd || !prtd->params) - return 0; - - if (prtd->dma_ch == -1) - return -EINVAL; - - DCSR(prtd->dma_ch) &= ~DCSR_RUN; - DCSR(prtd->dma_ch) = 0; - DCMD(prtd->dma_ch) = 0; - req = *(unsigned long *) prtd->params->filter_data; - DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD; - return 0; } EXPORT_SYMBOL(__pxa2xx_pcm_prepare); -void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) -{ - struct snd_pcm_substream *substream = dev_id; - int dcsr; - - dcsr = DCSR(dma_ch); - DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN; - - if (dcsr & DCSR_ENDINTR) { - snd_pcm_period_elapsed(substream); - } else { - printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n", - dma_ch, dcsr); - snd_pcm_stop_xrun(substream); - } -} -EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); - int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd; + struct snd_dmaengine_dai_dma_data *dma_params; int ret; runtime->hw = pxa2xx_pcm_hardware; + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; + /* * For mysterious reasons (and despite what the manual says) * playback samples are lost if the DMA count is not a multiple @@ -221,48 +113,27 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (ret) - goto out; + return ret; ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); if (ret) - goto out; + return ret; ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) - goto out; - - ret = -ENOMEM; - rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); - if (!rtd) - goto out; - rtd->dma_desc_array = - dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE, - &rtd->dma_desc_array_phys, GFP_KERNEL); - if (!rtd->dma_desc_array) - goto err1; + return ret; - rtd->dma_ch = -1; - runtime->private_data = rtd; - return 0; - - err1: - kfree(rtd); - out: - return ret; + return snd_dmaengine_pcm_open_request_chan(substream, + pxad_filter_fn, + dma_params->filter_data); } EXPORT_SYMBOL(__pxa2xx_pcm_open); int __pxa2xx_pcm_close(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd = runtime->private_data; - - dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, - rtd->dma_desc_array, rtd->dma_desc_array_phys); - kfree(rtd); - return 0; + return snd_dmaengine_pcm_close_release_chan(substream); } EXPORT_SYMBOL(__pxa2xx_pcm_close); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 83be8e3f095e..83fcfac97739 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -46,17 +46,13 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? client->playback_params : client->capture_params; - ret = pxa_request_dma("dma", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - goto err2; - rtd->dma_ch = ret; ret = client->startup(substream); if (!ret) - goto out; + goto err2; + + return 0; - pxa_free_dma(rtd->dma_ch); err2: __pxa2xx_pcm_close(substream); out: @@ -66,9 +62,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) { struct pxa2xx_pcm_client *client = substream->private_data; - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - pxa_free_dma(rtd->dma_ch); client->shutdown(substream); return __pxa2xx_pcm_close(substream); diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h index 00330985beec..8fa2b7c9e6b8 100644 --- a/sound/arm/pxa2xx-pcm.h +++ b/sound/arm/pxa2xx-pcm.h @@ -13,8 +13,6 @@ struct pxa2xx_runtime_data { int dma_ch; struct snd_dmaengine_dai_dma_data *params; - struct pxa_dma_desc *dma_desc_array; - dma_addr_t dma_desc_array_phys; }; struct pxa2xx_pcm_client { diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9e4b04e0fbd1..f3de615aacd7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <sound/core.h> #include <sound/ac97_codec.h> @@ -49,7 +50,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11; +static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 11, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -57,7 +62,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, }; -static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12; +static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 12, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -65,7 +74,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10; +static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 10, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -73,7 +85,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9; +static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 9, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -81,7 +96,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8; +static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 8, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -89,9 +107,8 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; -static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -105,9 +122,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_aux_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -121,9 +137,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -139,15 +154,15 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_48000) static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = { - .hw_params = pxa2xx_ac97_hw_params, + .startup = pxa2xx_ac97_hifi_startup, }; static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = { - .hw_params = pxa2xx_ac97_hw_aux_params, + .startup = pxa2xx_ac97_aux_startup, }; static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { - .hw_params = pxa2xx_ac97_hw_mic_params, + .startup = pxa2xx_ac97_mic_startup, }; /* diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6b4e40036910..0389cf7b4b1e 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -319,6 +319,9 @@ static int pxa2xx_i2s_probe(struct snd_soc_dai *dai) /* Along with FIFO servicing */ SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + snd_soc_dai_init_dma_data(dai, &pxa2xx_i2s_pcm_stereo_out, + &pxa2xx_i2s_pcm_stereo_in); + return 0; } diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 831ee37d2e3e..76ba0e55870d 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -15,8 +15,6 @@ #include <linux/dmaengine.h> #include <linux/of.h> -#include <mach/dma.h> - #include <sound/core.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> @@ -27,11 +25,8 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_dmaengine_dai_dma_data *dma; - int ret; dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); @@ -40,40 +35,13 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, if (!dma) return 0; - /* this may get called several times by oss emulation - * with different params */ - if (prtd->params == NULL) { - prtd->params = dma; - ret = pxa_request_dma("name", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - return ret; - prtd->dma_ch = ret; - } else if (prtd->params != dma) { - pxa_free_dma(prtd->dma_ch); - prtd->params = dma; - ret = pxa_request_dma("name", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - return ret; - prtd->dma_ch = ret; - } - return __pxa2xx_pcm_hw_params(substream, params); } static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - __pxa2xx_pcm_hw_free(substream); - if (prtd->dma_ch >= 0) { - pxa_free_dma(prtd->dma_ch); - prtd->dma_ch = -1; - prtd->params = NULL; - } - return 0; } -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: imx-ssi: Fix DAI hardware signal inversions
by Benoît Thébaudeau 30 Sep '15

30 Sep '15

SND_SOC_DAIFMT_{IB|NB}_{IF|NF} are defined as inverting or not BCLK or FRM relatively to what is standard for the specified DAI hardware audio format. Consequently, the absolute polarities of these signals cannot be derived only from these settings as this driver did. The format has to be taken into account too. This fixes inverted left/right channels in I²S mode. Signed-off-by: Benoît Thébaudeau <benoit(a)wsystem.com> --- sound/soc/fsl/imx-ssi.c | 19 +++++++++---------- 1 file changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 48b2d24..b95132e 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -95,7 +95,8 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: /* data on rising edge of bclk, frame low 1clk before data */ - strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSI | + SSI_STCR_TEFS; scr |= SSI_SCR_NET; if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) { scr &= ~SSI_I2S_MODE_MASK; @@ -104,33 +105,31 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) break; case SND_SOC_DAIFMT_LEFT_J: /* data on rising edge of bclk, frame high with data */ - strcr |= SSI_STCR_TXBIT0; + strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP; break; case SND_SOC_DAIFMT_DSP_B: /* data on rising edge of bclk, frame high with data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0; + strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL; break; case SND_SOC_DAIFMT_DSP_A: /* data on rising edge of bclk, frame high 1clk before data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; + strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL | + SSI_STCR_TEFS; break; } /* DAI clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_IB_IF: - strcr |= SSI_STCR_TFSI; - strcr &= ~SSI_STCR_TSCKP; + strcr ^= SSI_STCR_TSCKP | SSI_STCR_TFSI; break; case SND_SOC_DAIFMT_IB_NF: - strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + strcr ^= SSI_STCR_TSCKP; break; case SND_SOC_DAIFMT_NB_IF: - strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + strcr ^= SSI_STCR_TFSI; break; case SND_SOC_DAIFMT_NB_NF: - strcr &= ~SSI_STCR_TFSI; - strcr |= SSI_STCR_TSCKP; break; } -- 2.1.4

5 6

[alsa-devel] [PATCH] ASoC: Codec: wm8962: balance pm_runtime_enable
by Jiada Wang 30 Sep '15

30 Sep '15

pm_runtime_enable is called in probe to enable runtime PM for wm8962 codec, but pm_runtime_disable isn't called in remove callback, nor is called in error path if probe fails after runtime PM is enabled, this causes unbalanced pm_runtime_enable. This patch Adds pm_runtime_disable in remove callback and error path, to balance pm_runtime_enable. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/codecs/wm8962.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 293e47a..2fbc6ef 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3760,7 +3760,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8962, &wm8962_dai, 1); if (ret < 0) - goto err_enable; + goto err_pm_runtime; regcache_cache_only(wm8962->regmap, true); @@ -3769,6 +3769,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, return 0; +err_pm_runtime: + pm_runtime_disable(&i2c->dev); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); err: @@ -3778,6 +3780,7 @@ err: static int wm8962_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); + pm_runtime_disable(&client->dev); return 0; } -- 2.4.5

3 3

[alsa-devel] [PATCH 1/2] ASoC: sun4i-codec: pass through clk_set_rate error
by Maxime Ripard 30 Sep '15

30 Sep '15

The current code, disregarding the clk_set_rate error code, was always returning -EINVAL. Fix that and return the code in order to have more clue about what's going on. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/soc/sunxi/sun4i-codec.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 6e83e62ef039..28834129bea3 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -283,7 +283,7 @@ static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); unsigned long clk_freq; - int hwrate; + int ret, hwrate; u32 val; if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) @@ -293,8 +293,9 @@ static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, if (!clk_freq) return -EINVAL; - if (clk_set_rate(scodec->clk_module, clk_freq)) - return -EINVAL; + ret = clk_set_rate(scodec->clk_module, clk_freq); + if (ret) + return ret; hwrate = sun4i_codec_get_hw_rate(params); if (hwrate < 0) -- 2.5.3

2 4

[alsa-devel] SoC/Codecs/SGTL5000 - Non working MIC_BIAS Voltage settings
by gianluca 30 Sep '15

30 Sep '15

Hello, in attachment there is a small patch to ensure the correct register value for MIC BIAS Voltage. Best regards -- Eurek s.r.l. | Electronic Engineering | http://www.eurek.it via Celletta 8/B, 40026 Imola, Italy | Phone: +39-(0)542-609120 p.iva 00690621206 - c.f. 04020030377 | Fax: +39-(0)542-609212

4 5

[alsa-devel] Found bugs in tlv320aic3x.c driver, how do I report them?
by Rick Mann 30 Sep '15

30 Sep '15

Hi. I came across some bugs in sound/soc/codecs/tlv320aic3x.c, wherein it writes non-zero values to reserved registers on the tlv320aic3104 (I found it in 4.1.4, but it's still in 4.3-rc3). I have an untested proposed patch. But I don't know anything about how to report it or offer it. What should I do? Thanks. For reference: $ git diff -- sound/soc/codecs/tlv320aic3x.c diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 51c4713..79c0ca0 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1509,14 +1509,24 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); - /* Line2 Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + if (aic3x->model == AIC3X_MODEL_3104) { + /* On tlv320aic3014, these registers are reserved and must be written 0 */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, 0); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, 0); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, 0); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, 0); + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, 0); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, 0); + } else { + /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + /* Line2 Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + } switch (aic3x->model) { case AIC3X_MODEL_3X: -- Rick Mann rmann(a)latencyzero.com

2 5

[alsa-devel] [PATCH 0/2] ASoC: Add support for the Allwinner A10 codec
by Maxime Ripard 30 Sep '15

30 Sep '15

Hi everyone, This patch set adds the support for what Allwinner calls the codec on their SoCs. This codec is actually a combination of a codec and DAI, tied together in a single memory-mapped IP. It is completely standalone, and outputs directly the analog signal. While it supports both playback and capture, the capture is not implemented in this patch, and will be posted eventually as a separate one. This set, in order to be functional, has a dependency on the audio clocks patch set posted separately. However, it doesn't needs this to compile properly, so I guess it can be merged without really caring for the merging status of the clock patches. Let me know what you think, Maxime Emilio López (1): ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs Maxime Ripard (1): Documentation: bindings: Add the Allwinner A10 codec bindings .../devicetree/bindings/sound/sun4i-codec.txt | 33 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-codec.c | 720 +++++++++++++++++++++ 6 files changed, 767 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt create mode 100644 sound/soc/sunxi/Kconfig create mode 100644 sound/soc/sunxi/Makefile create mode 100644 sound/soc/sunxi/sun4i-codec.c -- 2.5.1

5 11

[alsa-devel] [PATCH 0/7 v3] firewire-digi00x: new driver for Digidesign 002/003 family
by Takashi Sakamoto 30 Sep '15

30 Sep '15

This patchset updates a part of my previous post: [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html This patchset adds support for Digidesign Digi 002/003 family without MIDI functionality. Although most part of AMDTP stream functionality is clear, a mechanism of clock synchronization with internal clock source is not clear (see the last patch). Takashi Sakamoto (7): ALSA: firewire-digi00x: add skeleton for Digi 002/003 family ALSA: firewire-digi00x: add data block processing layer ALSA: firewire-digi00x: add stream functionality ALSA: firewire-digi00x: add proc node to show clock status ALSA: firewire-digi00x: add PCM functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 8 + sound/firewire/Kconfig | 15 + sound/firewire/Makefile | 1 + sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 330 +++++++++++++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 373 ++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 +++++++ sound/firewire/digi00x/digi00x-stream.c | 420 +++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 81 ++++++ sound/firewire/digi00x/digi00x.c | 169 +++++++++++ sound/firewire/digi00x/digi00x.h | 146 ++++++++++ 13 files changed, 1848 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/amdtp-dot.c create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x-transaction.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h -- 2.1.4

2 9

[alsa-devel] [PATCH] drm/i915: Add missing const to audio_rate_need_prog()
by Takashi Iwai 30 Sep '15

30 Sep '15

The lack of const leads to a compile warning after merging i915 upstream tree: drivers/gpu/drm/i915/intel_audio.c:147:13: note: expected 'struct drm_display_mode *' but argument is of type 'const struct drm_display_mode *' Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/drm/i915/intel_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index 30f6859dcb36..dffd0b4c5f17 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -144,7 +144,7 @@ static uint32_t audio_config_setup_n_reg(int n, uint32_t val) /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, - struct drm_display_mode *mode) + const struct drm_display_mode *mode) { if (((mode->clock == TMDS_297M) || (mode->clock == TMDS_296M)) && -- 2.5.3

2 2

[alsa-devel] [PATCH RFC 0/7] ASoC: tlv320aic26: Add device tree support and bug fixes
by Cormier, Jonathan 30 Sep '15

30 Sep '15

Hi, This is my first patch so I'm including the RFC to help identify if I've missed something. The following patches were created to get the tlv320aic26 working on our AM335x SoM. This patch series relies on commit 93d0ad8f374c "ASoC: tlv320aic26: Convert to params_width()" for proper operation. Thanks -Jonathan Cormier, Jonathan (7): ASoC: tlv320aic26: Add device tree binding ASoC: tlv320aic26: Fix module autoload ASoC: tlv320aic26: Make SND_SOC_TLV320AIC26 user-visible ASoC: tlv320aic26: Fix regmap by setting reg_defaults and reg_stride ASoC: tlv320aic26: hw_params was unintentionally clearing AIC26 master mode ASoC: tlv320aic26: Add support for DSP_B data format ASoC: tlv320aic26: Change Capture Mute to Capture Switch to match alsa documentation. .../devicetree/bindings/sound/tlv320aic26.txt | 65 ++++++++++++++++++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/tlv320aic26.c | 60 ++++++++++++++++++-- 3 files changed, 120 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic26.txt -- 1.9.1

3 19

[alsa-devel] speaker-test error using sgtl5000 and imx27
by Alexandre Schnegg 30 Sep '15

30 Sep '15

Hello, I would like to use the Freescale SGTl5000 audio codec with the Freescale imx27 processor. Unfortunately, I get this error when I try to run < speaker-test >: # speaker-test speaker-test 1.0.28 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 32768 Period size range from 64 to 16384 Using max buffer size 32768 Periods = 4 was set period_size = 8192 was set buffer_size = 32768 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error The codec is connected to SSI2 pins and I would like to use the SSI0 internal port in asynchronous mode (4 wires). The command interface is I2C1. The codec seems to work because it generates both bit and frame clocks (the codec is the master) but no data are coming from the processor. I'm using kernel 4.1.7 but I have got the same error with versions 3.19.8, 4.0.5 and 4.0.9. I have also tried the < imx-sgtl5000 > driver and I get the same error. I can use < alsamixer > with no issue. Here are my device tree bindings: sound { compatible = "simple-audio-card"; simple-audio-card,name = "sgtl5000-audio"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&dailink0_master>; simple-audio-card,frame-master = <&dailink0_master>; simple-audio-card,widgets = "Microphone", "Mic Jack", "Headphone", "Headphone Jack"; simple-audio-card,routing = "MIC_IN", "Mic Jack", "Mic Jack", "Mic Bias", "Headphone Jack", "HP_OUT"; simple-audio-card,cpu { sound-dai = <&ssi1 0>; }; dailink0_master: simple-audio-card,codec { sound-dai = <&sgtl5000>; clocks = <&mclk>; }; }; mclk: asvmpc { #clock-cells = <0>; compatible = "fixed-clock"; clock-frequency = <12288000>; }; &ssi1 { #sound-dai-cells = <1>; status = "okay"; fsl,audmux-port = <MX27_AUDMUX_PPCR2_SSI_PINS_2>; fsl,port-config = < (IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)) >; }; &audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_ssi2>; status = "okay"; fsl,audmux-port = <MX27_AUDMUX_HPCR1_SSI0>; fsl,port-config = < (IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR2_SSI_PINS_2) | IMX_AUDMUX_V1_PCR_TFSDIR | IMX_AUDMUX_V1_PCR_TCLKDIR | IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR2_SSI_PINS_2)) >; }; &i2c1 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c1>; status = "okay"; sgtl5000: sgtl5000@0a { #sound-dai-cells = <0>; compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&mclk>; micbias-resistor-k-ohms = <2>; micbias-voltage-m-volts = <2250>; VDDA-supply = <&reg_2p8v>; VDDIO-supply = <&reg_2p8v>; }; }; pinctrl_ssi2: ssi2grp { fsl,pins = < MX27_PAD_SSI2_FS__SSI2_FS 0x0 MX27_PAD_SSI2_RXDAT__SSI2_RXDAT 0x0 MX27_PAD_SSI2_TXDAT__SSI2_TXDAT 0x0 MX27_PAD_SSI2_CLK__SSI2_CLK 0x0 >; }; And the modules I have loaded before running < speaker test > : # modprobe snd-soc-sgtl5000 [ 23.274351] sgtl5000 0-000a: sgtl5000 revision 0x11 # modprobe snd-soc-simple-card # modprobe snd-soc-fsl-ssi [ 79.752288] sgtl5000 0-000a: Using internal LDO instead of VDDD # [ 79.853711] asoc-simple-card sound: sgtl5000 <-> 10010000.ssi mapping ok # modprobe snd-soc-imx-audmux Does anybody know what is the mistake? Thank you for your help. Best regards. Alexandre Schnegg

3 5

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: fix wrong register MIC_BIAS_VOLTAGE setup on probe
by Gianluca Renzi 29 Sep '15

29 Sep '15

Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Gianluca Renzi <gianlucarenzi(a)eurekelettronica.it> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- a/sound/soc/codecs/sgtl5000.c 2015-09-25 15:24:10.597051191 +0200 +++ b/sound/soc/codecs/sgtl5000.c 2015-09-25 21:26:07.102020194 +0200 @@ -1376,8 +1376,8 @@ static int sgtl5000_probe(struct snd_soc sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_MASK, - sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT); + SGTL5000_BIAS_VOLT_MASK, + sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT); /* * disable DAP * TODO:

2 2

[alsa-devel] [PATCH alsa-utils] alsactl: Manage both save and restore in a single unit
by Takashi Iwai 29 Sep '15

29 Sep '15

With RemainAfterExit=true, we can manage both save and restore of the card state in a single unit file. This will fix also the case where systemd reloads the service; with two individual units, it will restore the previous state before saving, and may lead to inconsistent state suddenly. Also fix alsa-state.service as well to make both start and stop working in a simpler way. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=929619 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/Makefile.am | 14 ++------------ alsactl/alsa-restore.service.in | 7 +++---- alsactl/alsa-state.service.in | 3 --- alsactl/alsa-store.service.in | 16 ---------------- 4 files changed, 5 insertions(+), 35 deletions(-) delete mode 100644 alsactl/alsa-store.service.in diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index 47f06e906ced..cac8094687de 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -25,20 +25,15 @@ if HAVE_SYSTEMD systemdsystemunit_DATA = \ alsa-state.service \ - alsa-restore.service \ - alsa-store.service + alsa-restore.service install-data-hook: $(MKDIR_P) -m 0755 \ - $(DESTDIR)$(systemdsystemunitdir)/basic.target.wants \ - $(DESTDIR)$(systemdsystemunitdir)/shutdown.target.wants + $(DESTDIR)$(systemdsystemunitdir)/basic.target.wants ( cd $(DESTDIR)$(systemdsystemunitdir)/basic.target.wants && \ rm -f alsa-state.service alsa-restore.service && \ $(LN_S) ../alsa-state.service alsa-state.service && \ $(LN_S) ../alsa-restore.service alsa-restore.service) - ( cd $(DESTDIR)$(systemdsystemunitdir)/shutdown.target.wants && \ - rm -f alsa-store.service && \ - $(LN_S) ../alsa-store.service alsa-store.service ) endif @@ -54,22 +49,17 @@ alsa-state.service: alsa-state.service.in alsa-restore.service: alsa-restore.service.in $(edit) -alsa-store.service: alsa-store.service.in - $(edit) - 90-alsa-restore.rules: 90-alsa-restore.rules.in $(edit) EXTRA_DIST += \ alsa-state.service.in \ alsa-restore.service.in \ - alsa-store.service.in \ 90-alsa-restore.rules.in CLEANFILES = \ alsa-state.service \ alsa-restore.service \ - alsa-store.service \ 90-alsa-restore.rules %.7: %.xml diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index d1a74d637457..2884098c26f0 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -4,15 +4,14 @@ # [Unit] -Description=Restore Sound Card State +Description=Save/Restore Sound Card State ConditionPathExists=!@daemonswitch@ ConditionPathExistsGlob=/dev/snd/control* -DefaultDependencies=no After=alsa-state.service -Before=shutdown.target -Conflicts=shutdown.target [Service] Type=oneshot +RemainAfterExit=true ExecStart=-@sbindir@/alsactl restore +ExecStop=-@sbindir@/alsactl store StandardOutput=syslog diff --git a/alsactl/alsa-state.service.in b/alsactl/alsa-state.service.in index 56c12c104565..f1321d63a0ca 100644 --- a/alsactl/alsa-state.service.in +++ b/alsactl/alsa-state.service.in @@ -6,10 +6,7 @@ [Unit] Description=Manage Sound Card State (restore and store) ConditionPathExists=@daemonswitch@ -DefaultDependencies=no After=sysinit.target -Before=shutdown.target -Conflicts=shutdown.target [Service] Type=simple diff --git a/alsactl/alsa-store.service.in b/alsactl/alsa-store.service.in deleted file mode 100644 index c89cfff04f35..000000000000 --- a/alsactl/alsa-store.service.in +++ /dev/null @@ -1,16 +0,0 @@ -# -# Note that two different ALSA card state management schemes exist and they -# can be switched using a file exist check - /etc/alsa/state-daemon.conf . -# - -[Unit] -Description=Store Sound Card State -ConditionPathExists=!@daemonswitch@ -ConditionPathExistsGlob=/dev/snd/control* -DefaultDependencies=no -Before=shutdown.target - -[Service] -Type=oneshot -ExecStart=@sbindir@/alsactl store -StandardOutput=syslog -- 2.5.3

1 0

[alsa-devel] [PATCH v3 0/4] ASoC: Add support for DA7219 audio codec
by Adam Thomson 29 Sep '15

29 Sep '15

This patch set adds support for the DA7219 audio codec with built-in advanced accessory detection functionality. Patch set includes codec driver, associated DT bindings documentation and MAINTAINERS file updates to cover new bindings. This patch set is baselined against the v4.3-rc3 kernel version. Changes in v3: - Actually set clk_src in dai_sysclk() function of codec, so future calls can correctly identify if work is to be done or not. Changes in v2: - Add SOC_DOUBLE_R_EXT support to ASoC core. - Use of upper-case naming for controls, where missing. - Use of Playback/Capture Digital naming for certain ADC/DAC controls. - CP mchange and vol_thresh DT bindings removed, now Kcontrols - Kcontrols and DAPM now locked as part of hptest work, and specific controls affected have had locking introduced. - Timeout for btn_det work when checking micbias status. - Added comments to describe reasoning for work use with hptest, btn_det. - Added comments to tonegen_freq_put/get to clarify endian conversion, and code now uses regmap_raw_read/write calls instead of bulk. - DT parsing code now prints dev_warn() for invalid settings. - SOC_VALUE_ENUM_SINGLE now used for hpf_mode handling, instead of customer get/put handlers. - DTMF key control now a SOC_ENUM to correctly describe supported keys, including non-numeric. - MIXIN_L_MIX_EN bit set as part of DAPM now (SUPPLY), rather than in probe. - Fix dai_sysclk() function to correctly check for no change, using clk_id as well as freq. - Error checking added for clk_set_rate() call. - Improved error checking for devm_clk_get() call. - Removed explicit use of device_may_wakeup() and enable/disable_irq_wake(). Now taken care of by I2C core so DT bindings updated accordingly to document this. - hptest work now uses cache to keep state, instead of reading registers individually. Cache is bypassed, then when test executed and complete, regcache_sync_region() used to restore settings to pre-test values. - Regulator consumer framework usage added, in particular to determine VDDIO level in use so device can be configured accordingly. DT documents updated to reflect this. Adam Thomson (4): ASoC: Add SOC_DOUBLE_R_EXT ASoC: codecs: Add da7219 codec driver ASoC: da7219: Add bindings documentation for DA7219 audio codec MAINTAINERS: da7219: Add entry to cover DA7219 bindings document Documentation/devicetree/bindings/sound/da7219.txt | 106 ++ MAINTAINERS | 1 + include/sound/da7219-aad.h | 99 + include/sound/da7219.h | 55 + include/sound/soc.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 823 +++++++++ sound/soc/codecs/da7219-aad.h | 212 +++ sound/soc/codecs/da7219.c | 1940 ++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 +++++++++ 11 files changed, 4069 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h -- 1.9.3

1 4

[alsa-devel] [linux-sunxi][PATCH 3/3] ASOC: sunxi: Add support for the spdif block
by codekipper＠gmail.com 29 Sep '15

29 Sep '15

From: Marcus Cooper <codekipper(a)gmail.com> The sun4i, sun6i and sun7i SoC families have an SPDIF block which is capable of playback and capture. This patch enables the playback of this block for the sun4i and sun7i families. Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> --- sound/soc/sunxi/Kconfig | 10 + sound/soc/sunxi/Makefile | 4 + sound/soc/sunxi/sunxi-machine-spdif.c | 110 +++++ sound/soc/sunxi/sunxi-spdif.c | 801 ++++++++++++++++++++++++++++++++++ 4 files changed, 925 insertions(+) create mode 100644 sound/soc/sunxi/sunxi-machine-spdif.c create mode 100644 sound/soc/sunxi/sunxi-spdif.c diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 84c72ec..053db02 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -8,4 +8,14 @@ config SND_SUN4I_CODEC Select Y or M to add support for the Codec embedded in the Allwinner A10 and affiliated SoCs. +config SND_SOC_SUNXI_DAI_SPDIF + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + +config SND_SOC_SUNXI_MACHINE_SPDIF + tristate "APB on-chip sun4i/sun5i/sun7i SPDIF" + select SND_SOC_SUNXI_DAI_SPDIF + help + Say Y if you want to add support for SoC S/PDIF audio as simple audio card. endmenu diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index ea8a08c..7849a75 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -1,2 +1,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o +snd-soc-sunxi-dai-spdif-objs := sunxi-spdif.o +obj-$(CONFIG_SND_SOC_SUNXI_DAI_SPDIF) += snd-soc-sunxi-dai-spdif.o +snd-soc-sunxi-machine-spdif-objs := sunxi-machine-spdif.o +obj-$(CONFIG_SND_SOC_SUNXI_MACHINE_SPDIF) += snd-soc-sunxi-machine-spdif.o diff --git a/sound/soc/sunxi/sunxi-machine-spdif.c b/sound/soc/sunxi/sunxi-machine-spdif.c new file mode 100644 index 0000000..f8f6bd8 --- /dev/null +++ b/sound/soc/sunxi/sunxi-machine-spdif.c @@ -0,0 +1,110 @@ +/* + * Copyright (C) 2015 Andrea Venturi <be17068(a)iperbole.bo.it> + * From code by (C) 2013 Freescale Semiconductor, Inc. + * (sound/soc/fsl/imx-spdif.c) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that 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. + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> + +struct sunxi_machine_spdif_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; +}; + +static int sunxi_machine_spdif_audio_probe(struct platform_device *pdev) +{ + struct device_node *spdif_np, *np = pdev->dev.of_node; + struct sunxi_machine_spdif_data *data; + int ret = 0; + + dev_dbg(&pdev->dev, "%s: Looking for spdif-controller\n", __func__); + spdif_np = of_parse_phandle(np, "spdif-controller", 0); + if (!spdif_np) { + dev_err(&pdev->dev, "failed to find spdif-controller\n"); + ret = -EINVAL; + goto end; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto end; + } + + data->dai.name = "S/PDIF PCM"; + data->dai.stream_name = "S/PDIF PCM"; + data->dai.codec_dai_name = "snd-soc-dummy-dai"; + data->dai.codec_name = "snd-soc-dummy"; + data->dai.cpu_of_node = spdif_np; + data->dai.platform_of_node = spdif_np; + data->dai.playback_only = true; + data->dai.capture_only = true; + + if (of_property_read_bool(np, "spdif-out")) + data->dai.capture_only = false; + + if (of_property_read_bool(np, "spdif-in")) + data->dai.playback_only = false; + + if (data->dai.playback_only && data->dai.capture_only) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto end; + } + + data->card.dev = &pdev->dev; + data->card.dai_link = &data->dai; + data->card.num_links = 1; + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto end; + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); + goto end; + } + + platform_set_drvdata(pdev, data); + +end: + of_node_put(spdif_np); + + return ret; +} + +#ifdef CONFIG_OF +static const struct of_device_id sunxi_machine_spdif_dt_ids[] = { + { .compatible = "allwinner,sunxi-audio-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sunxi_machine_spdif_dt_ids); +#endif + +static struct platform_driver sunxi_machine_spdif_driver = { + .driver = { + .name = "sunxi-machine-spdif", + .of_match_table = sunxi_machine_spdif_dt_ids, + }, + .probe = sunxi_machine_spdif_audio_probe, +}; + +module_platform_driver(sunxi_machine_spdif_driver); + +MODULE_AUTHOR("Marcus Cooper <codekipper(a)gmail.com>"); +MODULE_AUTHOR("Andrea Venturi, <be17068(a)iperbole.bo.it>"); +MODULE_DESCRIPTION("Allwinner sunxi S/PDIF machine driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/sunxi/sunxi-spdif.c b/sound/soc/sunxi/sunxi-spdif.c new file mode 100644 index 0000000..cae9261 --- /dev/null +++ b/sound/soc/sunxi/sunxi-spdif.c @@ -0,0 +1,801 @@ +/* + * ALSA SoC SPDIF Audio Layer + * + * Copyright 2015 Andrea Venturi <be17068(a)iperbole.bo.it> + * Copyright 2015 Marcus Cooper <codekipper(a)gmail.com> + * + * Based on the Allwinner SDK driver, released under the GPL. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that 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. + */ + +/* + * this is SPDIF sunxi simple audio card DAI driver that uses the codec + * "dummy driver" as per sound/soc/fsl/imx-spdif.c + */ +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/regmap.h> +#include <linux/of_address.h> +#include <linux/of_device.h> +#include <linux/ioport.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUNXI_SPDIF_CTL (0x00) + #define SUNXI_SPDIF_CTL_MCLKDIV(v) ((v) << 4) /* v even */ + #define SUNXI_SPDIF_CTL_MCLKOUTEN BIT(2) + #define SUNXI_SPDIF_CTL_GEN BIT(1) + #define SUNXI_SPDIF_CTL_RESET BIT(0) + +#define SUNXI_SPDIF_TXCFG (0x04) + #define SUNXI_SPDIF_TXCFG_SINGLEMOD BIT(31) + #define SUNXI_SPDIF_TXCFG_ASS BIT(17) + #define SUNXI_SPDIF_TXCFG_NONAUDIO BIT(16) + #define SUNXI_SPDIF_TXCFG_TXRATIO(v) ((v) << 4) + #define SUNXI_SPDIF_TXCFG_TXRATIO_MASK GENMASK(8, 4) + #define SUNXI_SPDIF_TXCFG_FMTRVD GENMASK(3, 2) + #define SUNXI_SPDIF_TXCFG_FMT16BIT (0 << 2) + #define SUNXI_SPDIF_TXCFG_FMT20BIT (1 << 2) + #define SUNXI_SPDIF_TXCFG_FMT24BIT (2 << 2) + #define SUNXI_SPDIF_TXCFG_CHSTMODE BIT(1) + #define SUNXI_SPDIF_TXCFG_TXEN BIT(0) + +#define SUNXI_SPDIF_RXCFG (0x08) + #define SUNXI_SPDIF_RXCFG_LOCKFLAG BIT(4) + #define SUNXI_SPDIF_RXCFG_CHSTSRC BIT(3) + #define SUNXI_SPDIF_RXCFG_CHSTCP BIT(1) + #define SUNXI_SPDIF_RXCFG_RXEN BIT(0) + +#define SUNXI_SPDIF_TXFIFO (0x0C) + +#define SUNXI_SPDIF_RXFIFO (0x10) + +#define SUNXI_SPDIF_FCTL (0x14) + #define SUNXI_SPDIF_FCTL_FIFOSRC BIT(31) + #define SUNXI_SPDIF_FCTL_FTX BIT(17) + #define SUNXI_SPDIF_FCTL_FRX BIT(16) + #define SUNXI_SPDIF_FCTL_TXTL(v) ((v) << 8) + #define SUNXI_SPDIF_FCTL_TXTL_MASK GENMASK(12, 8) + #define SUNXI_SPDIF_FCTL_RXTL(v) ((v) << 3) + #define SUNXI_SPDIF_FCTL_RXTL_MASK GENMASK(7, 3) + #define SUNXI_SPDIF_FCTL_TXIM BIT(2) + #define SUNXI_SPDIF_FCTL_RXOM(v) ((v) << 0) + #define SUNXI_SPDIF_FCTL_RXOM_MASK GENMASK(1, 0) + +#define SUNXI_SPDIF_FSTA (0x18) + #define SUNXI_SPDIF_FSTA_TXE BIT(14) + #define SUNXI_SPDIF_FSTA_TXECNTSHT (8) + #define SUNXI_SPDIF_FSTA_RXA BIT(6) + #define SUNXI_SPDIF_FSTA_RXACNTSHT (0) + +#define SUNXI_SPDIF_INT (0x1C) + #define SUNXI_SPDIF_INT_RXLOCKEN BIT(18) + #define SUNXI_SPDIF_INT_RXUNLOCKEN BIT(17) + #define SUNXI_SPDIF_INT_RXPARERREN BIT(16) + #define SUNXI_SPDIF_INT_TXDRQEN BIT(7) + #define SUNXI_SPDIF_INT_TXUIEN BIT(6) + #define SUNXI_SPDIF_INT_TXOIEN BIT(5) + #define SUNXI_SPDIF_INT_TXEIEN BIT(4) + #define SUNXI_SPDIF_INT_RXDRQEN BIT(2) + #define SUNXI_SPDIF_INT_RXOIEN BIT(1) + #define SUNXI_SPDIF_INT_RXAIEN BIT(0) + +#define SUNXI_SPDIF_ISTA (0x20) + #define SUNXI_SPDIF_ISTA_RXLOCKSTA BIT(18) + #define SUNXI_SPDIF_ISTA_RXUNLOCKSTA BIT(17) + #define SUNXI_SPDIF_ISTA_RXPARERRSTA BIT(16) + #define SUNXI_SPDIF_ISTA_TXUSTA BIT(6) + #define SUNXI_SPDIF_ISTA_TXOSTA BIT(5) + #define SUNXI_SPDIF_ISTA_TXESTA BIT(4) + #define SUNXI_SPDIF_ISTA_RXOSTA BIT(1) + #define SUNXI_SPDIF_ISTA_RXASTA BIT(0) + +#define SUNXI_SPDIF_TXCNT (0x24) + +#define SUNXI_SPDIF_RXCNT (0x28) + +#define SUNXI_SPDIF_TXCHSTA0 (0x2C) + #define SUNXI_SPDIF_TXCHSTA0_CLK(v) ((v) << 28) + #define SUNXI_SPDIF_TXCHSTA0_SAMFREQ(v) ((v) << 24) + #define SUNXI_SPDIF_TXCHSTA0_SAMFREQ_MASK GENMASK(27, 24) + #define SUNXI_SPDIF_TXCHSTA0_CHNUM(v) ((v) << 20) + #define SUNXI_SPDIF_TXCHSTA0_CHNUM_MASK GENMASK(23, 20) + #define SUNXI_SPDIF_TXCHSTA0_SRCNUM(v) ((v) << 16) + #define SUNXI_SPDIF_TXCHSTA0_CATACOD(v) ((v) << 8) + #define SUNXI_SPDIF_TXCHSTA0_MODE(v) ((v) << 6) + #define SUNXI_SPDIF_TXCHSTA0_EMPHASIS(v) ((v) << 3) + #define SUNXI_SPDIF_TXCHSTA0_CP BIT(2) + #define SUNXI_SPDIF_TXCHSTA0_AUDIO BIT(1) + #define SUNXI_SPDIF_TXCHSTA0_PRO BIT(0) + +#define SUNXI_SPDIF_TXCHSTA1 (0x30) + #define SUNXI_SPDIF_TXCHSTA1_CGMSA(v) ((v) << 8) + #define SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ(v) ((v) << 4) + #define SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ_MASK GENMASK(7, 4) + #define SUNXI_SPDIF_TXCHSTA1_SAMWORDLEN(v) ((v) << 1) + #define SUNXI_SPDIF_TXCHSTA1_MAXWORDLEN BIT(0) + +#define SUNXI_SPDIF_RXCHSTA0 (0x34) + #define SUNXI_SPDIF_RXCHSTA0_CLK(v) ((v) << 28) + #define SUNXI_SPDIF_RXCHSTA0_SAMFREQ(v) ((v) << 24) + #define SUNXI_SPDIF_RXCHSTA0_CHNUM(v) ((v) << 20) + #define SUNXI_SPDIF_RXCHSTA0_SRCNUM(v) ((v) << 16) + #define SUNXI_SPDIF_RXCHSTA0_CATACOD(v) ((v) << 8) + #define SUNXI_SPDIF_RXCHSTA0_MODE(v) ((v) << 6) + #define SUNXI_SPDIF_RXCHSTA0_EMPHASIS(v) ((v) << 3) + #define SUNXI_SPDIF_RXCHSTA0_CP BIT(2) + #define SUNXI_SPDIF_RXCHSTA0_AUDIO BIT(1) + #define SUNXI_SPDIF_RXCHSTA0_PRO BIT(0) + +#define SUNXI_SPDIF_RXCHSTA1 (0x38) + #define SUNXI_SPDIF_RXCHSTA1_CGMSA(v) ((v) << 8) + #define SUNXI_SPDIF_RXCHSTA1_ORISAMFREQ(v) ((v) << 4) + #define SUNXI_SPDIF_RXCHSTA1_SAMWORDLEN(v) ((v) << 1) + #define SUNXI_SPDIF_RXCHSTA1_MAXWORDLEN BIT(0) + +/* Defines for Sampling Frequency */ +enum sunxi_spdif_sampling_freq { + SUNXI_SPDIF_SAMFREQ_44_1KHZ = 0x0, + SUNXI_SPDIF_SAMFREQ_NOT_INDICATED = 0x1, + SUNXI_SPDIF_SAMFREQ_48KHZ = 0x2, + SUNXI_SPDIF_SAMFREQ_32KHZ = 0x3, + SUNXI_SPDIF_SAMFREQ_22_05KHZ = 0x4, + SUNXI_SPDIF_SAMFREQ_24KHZ = 0x6, + SUNXI_SPDIF_SAMFREQ_88_2KHZ = 0x8, + SUNXI_SPDIF_SAMFREQ_768KHZ = 0x9, + SUNXI_SPDIF_SAMFREQ_96KHZ = 0xa, + SUNXI_SPDIF_SAMFREQ_176_4KHZ = 0xc, + SUNXI_SPDIF_SAMFREQ_192KHZ = 0xe, +}; + +/* + * Original sampling frequency can be represented by inverting the value of the + * sampling frequency. + */ +#define ORIGINAL(v) ((~v) & 0xf) + +struct sunxi_spdif_dev { + struct platform_device *pdev; + struct clk *clk; + struct clk *apb_clk; + struct clk *audio_clk; + struct snd_soc_dai_driver cpu_dai_drv; + bool running; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct snd_dmaengine_dai_dma_data dma_params_rx; + bool playback_supported; + bool capture_supported; +}; + +static void sunxi_spdif_configure(struct sunxi_spdif_dev *host) +{ + u32 reg_val; + + /* soft reset SPDIF */ + regmap_write(host->regmap, SUNXI_SPDIF_CTL, SUNXI_SPDIF_CTL_RESET); + + /* MCLK OUTPUT enable */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_CTL, + SUNXI_SPDIF_CTL_MCLKOUTEN, SUNXI_SPDIF_CTL_MCLKOUTEN); + + /* flush TX FIFO */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_FCTL, SUNXI_SPDIF_FCTL_FTX, + SUNXI_SPDIF_FCTL_FTX); + + /* clear interrupt status */ + regmap_read(host->regmap, SUNXI_SPDIF_ISTA, &reg_val); + regmap_write(host->regmap, SUNXI_SPDIF_ISTA, reg_val); + + /* clear TX counter */ + regmap_write(host->regmap, SUNXI_SPDIF_TXCNT, 0); + +} + +void sunxi_snd_txctrl(struct snd_pcm_substream *substream, + struct sunxi_spdif_dev *host, int on) +{ + u32 tmp; + + regmap_read(host->regmap, SUNXI_SPDIF_TXCNT, &tmp); + if (substream->runtime->channels == 1) + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCFG, + SUNXI_SPDIF_TXCFG_SINGLEMOD, + SUNXI_SPDIF_TXCFG_SINGLEMOD); + + if (on) { + /* SPDIF TX ENABLE */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCFG, + SUNXI_SPDIF_TXCFG_TXEN, + SUNXI_SPDIF_TXCFG_TXEN); + + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_INT, + SUNXI_SPDIF_INT_TXDRQEN, + SUNXI_SPDIF_INT_TXDRQEN); + + /* Global enable */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_CTL, + SUNXI_SPDIF_CTL_GEN, + SUNXI_SPDIF_CTL_GEN); + } else { + /* SPDIF TX DISABLE */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCFG, + SUNXI_SPDIF_TXCFG_TXEN, 0); + + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_INT, + SUNXI_SPDIF_INT_TXDRQEN, 0); + + /* Global disable */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_CTL, + SUNXI_SPDIF_CTL_GEN, 0); + } +} + +static int sunxi_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai); + int ret; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + /* Reset module and interrupts only for first initialization */ + if (!cpu_dai->active) { + ret = clk_prepare_enable(host->clk); + if (ret) + return ret; + } + + host->running = true; + sunxi_spdif_configure(host); + + return 0; +} + +static void sunxi_spdif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return; + + clk_disable_unprepare(host->clk); + host->running = false; +} + +static int sunxi_spdif_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + u32 reg_val; + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + reg_val = 0; + reg_val &= ~SUNXI_SPDIF_TXCFG_SINGLEMOD; + reg_val |= SUNXI_SPDIF_TXCFG_ASS; + reg_val |= fmt; /* set non audio and bit depth */ + reg_val |= SUNXI_SPDIF_TXCFG_CHSTMODE; + regmap_write(host->regmap, SUNXI_SPDIF_TXCFG, reg_val); + reg_val = 0; + reg_val &= ~SUNXI_SPDIF_FCTL_FIFOSRC; + reg_val |= SUNXI_SPDIF_FCTL_TXTL_MASK; + reg_val |= SUNXI_SPDIF_FCTL_RXTL_MASK; + reg_val |= SUNXI_SPDIF_FCTL_TXIM; + reg_val |= SUNXI_SPDIF_FCTL_RXOM_MASK; + regmap_write(host->regmap, SUNXI_SPDIF_FCTL, reg_val); + reg_val = 0; + reg_val |= (SUNXI_SPDIF_TXCHSTA0_CHNUM(2)); + regmap_write(host->regmap, SUNXI_SPDIF_TXCHSTA0, reg_val); + + reg_val = 0; + reg_val |= (SUNXI_SPDIF_TXCHSTA1_SAMWORDLEN(1)); + if ((fmt & SUNXI_SPDIF_TXCFG_FMT24BIT) == SUNXI_SPDIF_TXCFG_FMT24BIT) + reg_val |= SUNXI_SPDIF_TXCHSTA1_MAXWORDLEN; + regmap_write(host->regmap, SUNXI_SPDIF_TXCHSTA1, reg_val); + return 0; +} + +static int sunxi_spdif_set_sysclk(struct snd_soc_dai *cpu_dai, unsigned int freq) +{ + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct platform_device *pdev = host->pdev; + + if (clk_set_rate(host->audio_clk, freq)) + dev_err(&pdev->dev, + "Setting pll2 clock rate for %d Hz failed!\n", freq); + if (clk_set_rate(host->clk, freq)) + dev_err(&pdev->dev, + "Setting SPDIF clock rate for %d Hz failed!\n", freq); + return 0; +} + +static int sunxi_spdif_set_clkdiv(struct snd_soc_dai *cpu_dai, + unsigned int rate, int div) +{ + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + int sample_freq, original_sample_freq; + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCHSTA0, + SUNXI_SPDIF_TXCHSTA0_SAMFREQ_MASK, 0); + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCHSTA1, + SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ_MASK, 0); + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCFG, + SUNXI_SPDIF_TXCFG_TXRATIO_MASK, + SUNXI_SPDIF_TXCFG_TXRATIO(div - 1)); + + if (rate == 24576000) { + switch (div) { + /* 24KHZ */ + case 8: + sample_freq = SUNXI_SPDIF_SAMFREQ_24KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_24KHZ); + break; + + /* 32KHZ */ + case 6: + sample_freq = SUNXI_SPDIF_SAMFREQ_32KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_32KHZ); + break; + + /* 48KHZ */ + case 4: + sample_freq = SUNXI_SPDIF_SAMFREQ_48KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_48KHZ); + break; + + /* 96KHZ */ + case 2: + sample_freq = SUNXI_SPDIF_SAMFREQ_96KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_96KHZ); + break; + + /* 192KHZ */ + case 1: + sample_freq = SUNXI_SPDIF_SAMFREQ_192KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_192KHZ); + break; + + default: + sample_freq = SUNXI_SPDIF_SAMFREQ_NOT_INDICATED; + original_sample_freq = 0; + break; + } + } else { + /* 22.5792MHz */ + switch (div) { + /* 22.05KHZ */ + case 8: + sample_freq = SUNXI_SPDIF_SAMFREQ_22_05KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_22_05KHZ); + break; + + /* 44.1KHZ */ + case 4: + sample_freq = SUNXI_SPDIF_SAMFREQ_44_1KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_44_1KHZ); + break; + + /* 88.2KHZ */ + case 2: + sample_freq = SUNXI_SPDIF_SAMFREQ_88_2KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_88_2KHZ); + break; + + /* 176.4KHZ */ + case 1: + sample_freq = SUNXI_SPDIF_SAMFREQ_176_4KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_176_4KHZ); + break; + + default: + sample_freq = SUNXI_SPDIF_SAMFREQ_NOT_INDICATED; + original_sample_freq = 0; + break; + } + } + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCHSTA0, + SUNXI_SPDIF_TXCHSTA0_SAMFREQ_MASK, + SUNXI_SPDIF_TXCHSTA0_SAMFREQ(sample_freq)); + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCHSTA1, + SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ_MASK, + SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ(original_sample_freq)); + + return 0; +} + +static int sunxi_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int ret = 0; + int fmt; + unsigned long rate = params_rate(params); + u32 mclk_div = 0; + unsigned int mclk = 0; + + /* Add the PCM and raw data select interface */ + switch (params_channels(params)) { + case 1: /* PCM mode */ + case 2: + fmt = 0; + break; + case 4: /* raw data mode */ + fmt = SUNXI_SPDIF_TXCFG_NONAUDIO; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + fmt |= SUNXI_SPDIF_TXCFG_FMT16BIT; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + fmt |= SUNXI_SPDIF_TXCFG_FMT20BIT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + fmt |= SUNXI_SPDIF_TXCFG_FMT24BIT; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + switch (rate) { + case 22050: + case 44100: + case 88200: + case 176400: + mclk = 22579200; + break; + case 24000: + case 32000: + case 48000: + case 96000: + case 192000: + mclk = 24576000; + break; + default: + return -EINVAL; + } + + ret = sunxi_spdif_set_sysclk(cpu_dai, mclk); + if (ret < 0) + return ret; + + switch (rate) { + case 22050: + case 24000: + mclk_div = 8; + break; + case 32000: + mclk_div = 6; + break; + case 44100: + case 48000: + mclk_div = 4; + break; + case 88200: + case 96000: + mclk_div = 2; + break; + case 176400: + case 192000: + mclk_div = 1; + break; + default: + return -EINVAL; + } + + ret = sunxi_spdif_set_clkdiv(cpu_dai, mclk, mclk_div); + if (ret < 0) + return ret; + + return 0; +} + +static int sunxi_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sunxi_snd_txctrl(substream, host, 1); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sunxi_snd_txctrl(substream, host, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sunxi_spdif_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &host->dma_params_tx, + &host->dma_params_rx); + return 0; +} + +static const struct snd_soc_dai_ops sunxi_spdif_dai_ops = { + .startup = sunxi_spdif_startup, + .shutdown = sunxi_spdif_shutdown, + .trigger = sunxi_spdif_trigger, + .hw_params = sunxi_spdif_hw_params, + .set_fmt = sunxi_spdif_set_fmt, +}; + +static const struct regmap_config sunxi_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUNXI_SPDIF_RXCHSTA1, +}; + +#define SUNXI_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) +#define SUNXI_FORMATS (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ + SNDRV_PCM_FORMAT_S24_LE) + +static struct snd_soc_dai_driver sunxi_spdif_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SUNXI_RATES, + .formats = SUNXI_FORMATS, + }, + .probe = sunxi_spdif_soc_dai_probe, + .ops = &sunxi_spdif_dai_ops, + .name = "spdif", +}; + +static const struct snd_soc_dapm_widget dit_widgets[] = { + SND_SOC_DAPM_OUTPUT("spdif-out"), +}; + +static const struct snd_soc_dapm_route dit_routes[] = { + { "spdif-out", NULL, "Playback" }, +}; + +#ifdef CONFIG_OF +static const struct of_device_id sunxi_spdif_of_match[] = { + { .compatible = "allwinner,sun4i-a10-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sunxi_spdif_of_match); +#endif + +static const struct snd_soc_component_driver sunxi_spdif_component = { + .name = "sunxi-spdif", +}; + +static int sunxi_spdif_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct sunxi_spdif_dev *host; + struct resource *res; + int ret; + const struct of_device_id *of_id; + void __iomem *base; + + dev_dbg(&pdev->dev, "Entered %s\n", __func__); + + if (!np) + return -ENODEV; + + of_id = of_match_device(sunxi_spdif_of_match, &pdev->dev); + if (!of_id) { + dev_err(&pdev->dev, "No matched devices found.\n"); + return -EINVAL; + } + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + host->pdev = pdev; + + /* Initialize this copy of the CPU DAI driver structure */ + memcpy(&host->cpu_dai_drv, &sunxi_spdif_dai, sizeof(sunxi_spdif_dai)); + host->cpu_dai_drv.name = dev_name(&pdev->dev); + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sunxi_spdif_regmap_config); + + /* Clocks */ + host->apb_clk = devm_clk_get(&pdev->dev, "apb"); + if (IS_ERR(host->apb_clk)) { + dev_err(&pdev->dev, "failed to get a apb clock.\n"); + return PTR_ERR(host->apb_clk); + } + + if (clk_prepare_enable(host->apb_clk)) { + dev_err(&pdev->dev, "try to enable apb_spdif_clk failed\n"); + return -EINVAL; + } + + host->audio_clk = devm_clk_get(&pdev->dev, "audio"); + if (IS_ERR(host->audio_clk)) { + dev_err(&pdev->dev, "failed to get an audio clock.\n"); + return PTR_ERR(host->audio_clk); + } + + host->clk = devm_clk_get(&pdev->dev, "spdif"); + if (IS_ERR(host->clk)) { + dev_err(&pdev->dev, "failed to get a spdif clock.\n"); + return PTR_ERR(host->clk); + } + + ret = clk_set_rate(host->audio_clk, 24576000); + if (ret) { + dev_err(&pdev->dev, "set spdif base clock failed!\n"); + return ret; + } + + if (clk_prepare_enable(host->audio_clk)) { + dev_err(&pdev->dev, "try to enable audio clk failed\n"); + ret = -EINVAL; + goto exit_clkdisable_apb_clk; + } + + host->playback_supported = false; + host->capture_supported = false; + + if (of_property_read_bool(np, "spdif-out")) + host->playback_supported = true; + + if (!host->playback_supported) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto exit_clkdisable_clk; + } + + host->dma_params_tx.addr = res->start + SUNXI_SPDIF_TXFIFO; + host->dma_params_tx.maxburst = 4; + host->dma_params_tx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + host->dma_params_rx.addr = res->start + SUNXI_SPDIF_RXFIFO; + host->dma_params_rx.maxburst = 4; + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + /* Register with ASoC */ + dev_set_drvdata(&pdev->dev, host); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sunxi_spdif_component, &sunxi_spdif_dai, 1); + if (ret) + goto exit_clkdisable_clk; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto exit_clkdisable_clk; + return 0; + +exit_clkdisable_clk: + clk_disable_unprepare(host->clk); +exit_clkdisable_apb_clk: + clk_disable_unprepare(host->apb_clk); + return ret; +} + +static int sunxi_spdif_remove(struct platform_device *pdev) +{ + struct sunxi_spdif_dev *host = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + if (!IS_ERR(host->clk)) { + clk_disable_unprepare(host->clk); + clk_disable_unprepare(host->apb_clk); + } + + return 0; +} + +#ifdef CONFIG_PM +static int sunxi_spdif_suspend(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct sunxi_spdif_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) + clk_disable(host->clk); + + return 0; +} + +static int sunxi_spdif_resume(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct sunxi_spdif_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) { + clk_enable(host->clk); + sunxi_spdif_configure(host); + } + return 0; +} + +static SIMPLE_DEV_PM_OPS(spdif_dev_pm_ops, sunxi_spdif_suspend, + sunxi_spdif_resume); + +#define SPDIF_DEV_PM_OPS (&spdif_dev_pm_ops) +#else +#define SPDIF_DEV_PM_OPS NULL + +#endif + +static struct platform_driver sunxi_spdif_driver = { + .driver = { + .name = "sunxi-spdif", + .owner = THIS_MODULE, + .pm = SPDIF_DEV_PM_OPS, + .of_match_table = of_match_ptr(sunxi_spdif_of_match), + }, + .probe = sunxi_spdif_probe, + .remove = sunxi_spdif_remove, +}; + +module_platform_driver(sunxi_spdif_driver); + +MODULE_AUTHOR("Marcus Cooper <codekipper(a)gmail.com>"); +MODULE_AUTHOR("Andrea Venturi <be17068(a)iperbole.bo.it>"); +MODULE_DESCRIPTION("Allwinner Sunxi SPDIF SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sunxi-spdif"); -- 1.9.1

4 5

[alsa-devel] [sound:topic/firewire-update 15/15] sound/firewire/amdtp-am824.c:402:14: sparse: symbol 'process_rx_data_blocks' was not declared. Should it be static?
by kbuild test robot 29 Sep '15

29 Sep '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/firewire-update head: df075feefbd347f13fba5198294cda619532c237 commit: df075feefbd347f13fba5198294cda619532c237 [15/15] ALSA: firewire-lib: complete AM824 data block processing layer reproduce: # apt-get install sparse git checkout df075feefbd347f13fba5198294cda619532c237 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/firewire/amdtp-am824.c:402:14: sparse: symbol 'process_rx_data_blocks' was not declared. Should it be static? >> sound/firewire/amdtp-am824.c:423:14: sparse: symbol 'process_tx_data_blocks' was not declared. Should it be static? 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

2 2

[alsa-devel] [RFC 0/5] another generic audio hdmi codec proposal
by Arnaud Pouliquen 29 Sep '15

29 Sep '15

I 'm working on HDMI implementation for sti platform. As some discussions are on going for audio on HDMI. Here is the begin of my work... This patch set is a tentative to implement a generic code for the HDMI audio. Main concept are aligned with solution proposeded for TI platform. - ASoC codec driver registered by DRM driver - ASOC driver is a generic driver. - compatible with simple card Difference is that i propose a DRM generic interface based on bridge structure. Advantage is that all data exchanges are done through the DRM API. I think also that some helper functions could been used for N and CTS parameters calculation, as suggested by Russell King in a previous mail. I full aware that some features (like ELD and info frame) are partially or not implemented in my patches. This patch set is more a skeleton than a full implementation... I just post it to suggest a possible DRM API. Arnaud Pouliquen (5): video: hdmi: add help function for N and cts drm: add helper functionto add audio capabilities for bridge ASoC: codec: hdmi drm codec driver drm: sti: connect audio driver DT: sti: add audio HDMI dai link in audio card arch/arm/boot/dts/stih410.dtsi | 4 +- arch/arm/boot/dts/stihxxx-b2120.dtsi | 21 +++++ arch/arm/configs/multi_v7_defconfig | 1 + drivers/gpu/drm/drm_bridge.c | 114 +++++++++++++++++++++++++++ drivers/gpu/drm/sti/sti_hdmi.c | 146 +++++++++++++++++++++++++++++++--- drivers/gpu/drm/sti/sti_hdmi.h | 3 + drivers/video/hdmi.c | 148 +++++++++++++++++++++++++++++++++++ include/drm/drm_crtc.h | 31 ++++++++ include/linux/hdmi.h | 24 ++++++ include/sound/hdmi_drm.h | 16 ++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi_drm.c | 125 +++++++++++++++++++++++++++++ 13 files changed, 625 insertions(+), 14 deletions(-) create mode 100644 include/sound/hdmi_drm.h create mode 100644 sound/soc/codecs/hdmi_drm.c -- 1.9.1

2 8

[alsa-devel] [PATCH 0/2] ASoc: add driver for Atmel I2S controller
by Cyrille Pitchen 29 Sep '15

29 Sep '15

This series of patches add support to the new Atmel I2S controller embedded on sama5d2 SoCs. Cyrille Pitchen (2): ASoC: atmel-i2s: add DT bindings for I2S controller ASoC: atmel-i2s: add driver for the new Atmel I2S controller .../devicetree/bindings/sound/atmel-i2s.txt | 43 ++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-i2s.c | 760 +++++++++++++++++++++ 4 files changed, 815 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-i2s.txt create mode 100644 sound/soc/atmel/atmel-i2s.c -- 1.8.2.2

1 3

[alsa-devel] [PATCH 00/15 v3] ALSA: firewire-lib: separate to packet streaming layer and data block processing layer
by Takashi Sakamoto 29 Sep '15

29 Sep '15

This patchset updates a part of my previous post (patch 01 and 02): [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html The aim of this patchset is to allow current amdtp implementation to handle several data block processing backends. For this aim, this patchset separates the implementation to packet streaming layer and AM824 data block processing layer. The communication between these two layers is done by callback function. This patchset adds an overhead to call the functions as the same frequently as 8,000 times per second. Takashi Sakamoto (15): ALSA: firewire-lib: arrange structure for AMDTP stream ALSA: firewire-lib: return error code when amdtp_stream_set_parameters() detects error ALSA: firewire-lib: add an argument for Dice's dual wire mode ALSA: firewire-lib: add a member of frame_multiplier instead of double_pcm_frames ALSA: firewire-lib: add helper functions as interfaces between packet streaming layer and data block processing layer ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' to prepare for functional separation ALSA: firewire-lib: add data block processing layer for AM824 format ALSA: firewire-lib: rename parameter setting function for AM824 with FDF field ALSA: firewire-lib: move PCM substream constraint to AM824 layer ALSA: firewire-lib: add helper functions to set positions of data channels ALSA: firewire-lib: move MIDI trigger helper function to AM824 layer ALSA: firewire-lib: rename PCM format helper function ALSA: firewire-lib: rename macros with AM824 prefix ALSA: firewire-lib: complete AM824 data block processing layer sound/firewire/Makefile | 2 +- sound/firewire/amdtp-am824.c | 465 ++++++++++++++++++++++++++++ sound/firewire/amdtp-am824.h | 52 ++++ sound/firewire/{amdtp.c => amdtp-stream.c} | 377 ++++------------------ sound/firewire/{amdtp.h => amdtp-stream.h} | 116 +++---- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_midi.c | 16 +- sound/firewire/bebob/bebob_pcm.c | 14 +- sound/firewire/bebob/bebob_stream.c | 34 +- sound/firewire/dice/dice-midi.c | 12 +- sound/firewire/dice/dice-pcm.c | 12 +- sound/firewire/dice/dice-stream.c | 22 +- sound/firewire/dice/dice.h | 2 +- sound/firewire/fcp.c | 2 +- sound/firewire/fireworks/fireworks.c | 12 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_midi.c | 12 +- sound/firewire/fireworks/fireworks_pcm.c | 12 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/oxfw/oxfw-midi.c | 16 +- sound/firewire/oxfw/oxfw-pcm.c | 10 +- sound/firewire/oxfw/oxfw-stream.c | 11 +- sound/firewire/oxfw/oxfw.h | 2 +- 23 files changed, 730 insertions(+), 483 deletions(-) create mode 100644 sound/firewire/amdtp-am824.c create mode 100644 sound/firewire/amdtp-am824.h rename sound/firewire/{amdtp.c => amdtp-stream.c} (70%) rename sound/firewire/{amdtp.h => amdtp-stream.h} (78%) -- 2.1.4

2 19

[alsa-devel] [PATCH 1/1] ASoC: ad193x: add support to ad1934
by Cyrille Pitchen 29 Sep '15

29 Sep '15

The AD1934 codec has no DAC feature. Hence it register mapping is slightly different from the register mapping of other members of the AD193x family. Some ASoC controls and widgets are related to the DAC feature so are not relevant in the case of an AD1934 codec. Signed-off-by: Cyrille Pitchen <cyrille.pitchen(a)atmel.com> --- Hi all, This patch adds support to the AD1934 codec, which is DAC only. Best Regards, Cyrille --- sound/soc/codecs/ad193x-i2c.c | 8 +-- sound/soc/codecs/ad193x-spi.c | 16 +++++- sound/soc/codecs/ad193x.c | 121 ++++++++++++++++++++++++++++++++---------- sound/soc/codecs/ad193x.h | 9 +++- 4 files changed, 120 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c index df3a1a415825..8aa50d35662f 100644 --- a/sound/soc/codecs/ad193x-i2c.c +++ b/sound/soc/codecs/ad193x-i2c.c @@ -15,8 +15,8 @@ #include "ad193x.h" static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, + { "ad1936", AD193x }, + { "ad1937", AD193x }, { } }; MODULE_DEVICE_TABLE(i2c, ad193x_id); @@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client, config.val_bits = 8; config.reg_bits = 8; - return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); + return ad193x_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 390cef9b9dc2..364e82112e8c 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -16,6 +16,7 @@ static int ad193x_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct regmap_config config; config = ad193x_regmap_config; @@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi) config.read_flag_mask = 0x09; config.write_flag_mask = 0x08; - return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_spi_remove(struct spi_device *spi) @@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id ad193x_spi_id[] = { + { "ad193x", AD193X }, + { "ad1933", AD1933 }, + { "ad1934", AD1934 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, + { "ad1938", AD193X }, + { } +}; +MODULE_DEVICE_TABLE(spi, ad193x_spi_id); + static struct spi_driver ad193x_spi_driver = { .driver = { .name = "ad193x", @@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = { }, .probe = ad193x_spi_probe, .remove = ad193x_spi_remove, + .id_table = ad193x_spi_id, }; module_spi_driver(ad193x_spi_driver); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 17c953595660..80ffecf98673 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,6 +23,7 @@ /* codec private data */ struct ad193x_priv { struct regmap *regmap; + enum ad193x_type type; int sysclk; }; @@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, - AD193X_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, - AD193X_ADCR2_MUTE, 1, 1), - /* DAC switch control */ SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, AD193X_DACR1_MUTE, 1, 1), @@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, AD193X_DACR4_MUTE, 1, 1), + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = { + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, AD193X_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_INPUT("ADC1IN"), SND_SOC_DAPM_INPUT("ADC2IN"), }; @@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, { "DAC Output", NULL, "DAC" }, { "DAC Output", NULL, "VMID" }, - { "ADC", NULL, "SYSCLK" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC Output" }, { "DAC2OUT", NULL, "DAC Output" }, { "DAC3OUT", NULL, "DAC Output" }, { "DAC4OUT", NULL, "DAC Output" }, + { "SYSCLK", NULL, "PLL_PWR" }, +}; + +static const struct snd_soc_dapm_route adc_audio_paths[] = { + { "ADC", NULL, "SYSCLK" }, + { "ADC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, - { "SYSCLK", NULL, "PLL_PWR" }, }; +static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) +{ + switch (ad193x->type) { + case AD1933: + case AD1934: + return false; + default: + break; + } + + return true; +} + /* * DAI ops entries */ @@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: - return -EINVAL; + if (ad193x_has_adc(ad193x)) + return -EINVAL; + break; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_SERFMT_MASK, adc_serfmt); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_FMT_MASK, adc_fmt); + if (ad193x_has_adc(ad193x)) { + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + } regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, dac_fmt); @@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_WORD_LEN_MASK, word_len); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -326,6 +354,7 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + int num, ret; /* default setting for ad193x */ @@ -335,14 +364,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* dac in tdm mode */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* high-pass filter enable */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + } + /* pll input: mclki/xi */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* add adc controls */ + num = ARRAY_SIZE(ad193x_adc_snd_controls); + ret = snd_soc_add_codec_controls(codec, + ad193x_adc_snd_controls, + num); + if (ret) + return ret; + + /* add adc widgets */ + num = ARRAY_SIZE(ad193x_adc_widgets); + ret = snd_soc_dapm_new_controls(&codec->dapm, + ad193x_adc_widgets, + num); + if (ret) + return ret; + + /* add adc routes */ + num = ARRAY_SIZE(adc_audio_paths); + ret = snd_soc_dapm_add_routes(&codec->dapm, + adc_audio_paths, + num); + if (ret) + return ret; + } + return 0; } @@ -367,7 +428,8 @@ const struct regmap_config ad193x_regmap_config = { }; EXPORT_SYMBOL_GPL(ad193x_regmap_config); -int ad193x_probe(struct device *dev, struct regmap *regmap) +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type) { struct ad193x_priv *ad193x; @@ -379,6 +441,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; ad193x->regmap = regmap; + ad193x->type = type; dev_set_drvdata(dev, ad193x); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index ab9a998f15be..8b1e65f928d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -13,8 +13,15 @@ struct device; +enum ad193x_type { + AD193X, + AD1933, + AD1934, +}; + extern const struct regmap_config ad193x_regmap_config; -int ad193x_probe(struct device *dev, struct regmap *regmap); +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type); #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -- 1.8.2.2

2 2

[alsa-devel] [PATCH v2 1/1] ASoC: ad193x: add support to ad1934
by Cyrille Pitchen 29 Sep '15

29 Sep '15

The AD1934 codec has no DAC feature. Hence it register mapping is slightly different from the register mapping of other members of the AD193x family. Some ASoC controls and widgets are related to the DAC feature so are not relevant in the case of an AD1934 codec. Signed-off-by: Cyrille Pitchen <cyrille.pitchen(a)atmel.com> --- Hi all, This patch adds support to the AD1934 codec, which is DAC only. Best Regards, Cyrille ChangeLog v2: - replace AD193x by AD193X in ad193x-i2c.c - add ad193x_ prefix to adc_audio_paths --- sound/soc/codecs/ad193x-i2c.c | 8 +-- sound/soc/codecs/ad193x-spi.c | 16 +++++- sound/soc/codecs/ad193x.c | 121 ++++++++++++++++++++++++++++++++---------- sound/soc/codecs/ad193x.h | 9 +++- 4 files changed, 120 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c index df3a1a415825..171313664bc8 100644 --- a/sound/soc/codecs/ad193x-i2c.c +++ b/sound/soc/codecs/ad193x-i2c.c @@ -15,8 +15,8 @@ #include "ad193x.h" static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, { } }; MODULE_DEVICE_TABLE(i2c, ad193x_id); @@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client, config.val_bits = 8; config.reg_bits = 8; - return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); + return ad193x_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 390cef9b9dc2..364e82112e8c 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -16,6 +16,7 @@ static int ad193x_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct regmap_config config; config = ad193x_regmap_config; @@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi) config.read_flag_mask = 0x09; config.write_flag_mask = 0x08; - return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_spi_remove(struct spi_device *spi) @@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id ad193x_spi_id[] = { + { "ad193x", AD193X }, + { "ad1933", AD1933 }, + { "ad1934", AD1934 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, + { "ad1938", AD193X }, + { } +}; +MODULE_DEVICE_TABLE(spi, ad193x_spi_id); + static struct spi_driver ad193x_spi_driver = { .driver = { .name = "ad193x", @@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = { }, .probe = ad193x_spi_probe, .remove = ad193x_spi_remove, + .id_table = ad193x_spi_id, }; module_spi_driver(ad193x_spi_driver); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 17c953595660..fa0a053bda58 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,6 +23,7 @@ /* codec private data */ struct ad193x_priv { struct regmap *regmap; + enum ad193x_type type; int sysclk; }; @@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, - AD193X_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, - AD193X_ADCR2_MUTE, 1, 1), - /* DAC switch control */ SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, AD193X_DACR1_MUTE, 1, 1), @@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, AD193X_DACR4_MUTE, 1, 1), + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = { + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, AD193X_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_INPUT("ADC1IN"), SND_SOC_DAPM_INPUT("ADC2IN"), }; @@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, { "DAC Output", NULL, "DAC" }, { "DAC Output", NULL, "VMID" }, - { "ADC", NULL, "SYSCLK" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC Output" }, { "DAC2OUT", NULL, "DAC Output" }, { "DAC3OUT", NULL, "DAC Output" }, { "DAC4OUT", NULL, "DAC Output" }, + { "SYSCLK", NULL, "PLL_PWR" }, +}; + +static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = { + { "ADC", NULL, "SYSCLK" }, + { "ADC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, - { "SYSCLK", NULL, "PLL_PWR" }, }; +static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) +{ + switch (ad193x->type) { + case AD1933: + case AD1934: + return false; + default: + break; + } + + return true; +} + /* * DAI ops entries */ @@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: - return -EINVAL; + if (ad193x_has_adc(ad193x)) + return -EINVAL; + break; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_SERFMT_MASK, adc_serfmt); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_FMT_MASK, adc_fmt); + if (ad193x_has_adc(ad193x)) { + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + } regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, dac_fmt); @@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_WORD_LEN_MASK, word_len); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -326,6 +354,7 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + int num, ret; /* default setting for ad193x */ @@ -335,14 +364,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* dac in tdm mode */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* high-pass filter enable */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + } + /* pll input: mclki/xi */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* add adc controls */ + num = ARRAY_SIZE(ad193x_adc_snd_controls); + ret = snd_soc_add_codec_controls(codec, + ad193x_adc_snd_controls, + num); + if (ret) + return ret; + + /* add adc widgets */ + num = ARRAY_SIZE(ad193x_adc_widgets); + ret = snd_soc_dapm_new_controls(&codec->dapm, + ad193x_adc_widgets, + num); + if (ret) + return ret; + + /* add adc routes */ + num = ARRAY_SIZE(ad193x_adc_audio_paths); + ret = snd_soc_dapm_add_routes(&codec->dapm, + ad193x_adc_audio_paths, + num); + if (ret) + return ret; + } + return 0; } @@ -367,7 +428,8 @@ const struct regmap_config ad193x_regmap_config = { }; EXPORT_SYMBOL_GPL(ad193x_regmap_config); -int ad193x_probe(struct device *dev, struct regmap *regmap) +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type) { struct ad193x_priv *ad193x; @@ -379,6 +441,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; ad193x->regmap = regmap; + ad193x->type = type; dev_set_drvdata(dev, ad193x); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index ab9a998f15be..8b1e65f928d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -13,8 +13,15 @@ struct device; +enum ad193x_type { + AD193X, + AD1933, + AD1934, +}; + extern const struct regmap_config ad193x_regmap_config; -int ad193x_probe(struct device *dev, struct regmap *regmap); +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type); #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -- 1.8.2.2

1 0

[alsa-devel] [PATCH] ALSA: pcm: Avoid double hw_free calls at releasing a stream
by Takashi Iwai 29 Sep '15

29 Sep '15

snd_pcm_release_substream() always calls hw_free op when the stream was opened. This is superfluous in most cases because it's been already released via explicit hw_free ioctl. Although this double call is usually OK as this callback should be written to be called multiple times, it's better to avoid superfluous calls. Reported-by: Vinod Koul <vinod.koul(a)intel.com> Tested-by: Jeeja Kp <jeeja.kp(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4863af5e7b26..139887011ba2 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2227,7 +2227,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->ops->hw_free != NULL) + if (substream->ops->hw_free && + substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) substream->ops->hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; -- 2.5.3

1 0

[alsa-devel] [RESEND PATCH v2 0/2] ASoC: atmel-classd: add driver for Atmel CLASSD
by Songjun Wu 29 Sep '15

29 Sep '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Changes in v2: - Change the "Mono", "Swap" and "Deemphasis" controls to "Mono Switch", "Swap Switch" and "Deemphasis Switch". - Merge "EQ Bass", "EQ Medium" and "EQ Treble" controls into one "EQ" control. - Change the "single-ended" and "differential" to "Single ended", "Differential". - Merge separate left and right controls into one single stereo control. - Add a warning if the user trys to specify an invalid value in function "atmel_classd_codec_probe". - Use the "dev_get_regmap()" in function "atmel_classd_codec_get_remap". - Check for errors when invoking the "clk_prepare_enable". - Remove the "dev_info" in function "atmel_classd_probe". - Add some code to create a sound card in function "atmel_classd_probe". - Remove the DT node "Sound" and the related code. - Delete the device node "Sound" in dts. Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier SoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 52 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 680 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 ++++ 5 files changed, 863 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

4 5

[alsa-devel] [PATCH V4 1/2] ACPI / EC: Fix broken 64bit big-endian users of 'global_lock'
by Viresh Kumar 28 Sep '15

28 Sep '15

global_lock is defined as an unsigned long and accessing only its lower 32 bits from sysfs is incorrect, as we need to consider other 32 bits for big endian 64 bit systems. There are no such platforms yet, but the code needs to be robust for such a case. Fix that by passing a local variable to debugfs_create_bool() and assigning its value to global_lock later. Signed-off-by: Viresh Kumar <viresh.kumar(a)linaro.org> --- V3->V4: - Create a local variable instead of changing type of global_lock (Rafael) - Drop the stable tag - BCC'd a lot of people (rather than cc'ing them) to make sure - the series reaches them - mailing lists do not block the patchset due to long cc list - and we don't spam the BCC'd people for every reply --- drivers/acpi/ec_sys.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/drivers/acpi/ec_sys.c b/drivers/acpi/ec_sys.c index b4c216bab22b..b44b91331a56 100644 --- a/drivers/acpi/ec_sys.c +++ b/drivers/acpi/ec_sys.c @@ -110,6 +110,7 @@ static int acpi_ec_add_debugfs(struct acpi_ec *ec, unsigned int ec_device_count) struct dentry *dev_dir; char name[64]; umode_t mode = 0400; + u32 val; if (ec_device_count == 0) { acpi_ec_debugfs_dir = debugfs_create_dir("ec", NULL); @@ -127,10 +128,11 @@ static int acpi_ec_add_debugfs(struct acpi_ec *ec, unsigned int ec_device_count) if (!debugfs_create_x32("gpe", 0444, dev_dir, (u32 *)&first_ec->gpe)) goto error; - if (!debugfs_create_bool("use_global_lock", 0444, dev_dir, - (u32 *)&first_ec->global_lock)) + if (!debugfs_create_bool("use_global_lock", 0444, dev_dir, &val)) goto error; + first_ec->global_lock = val; + if (write_support) mode = 0600; if (!debugfs_create_file("io", mode, dev_dir, ec, &acpi_ec_io_ops)) -- 2.4.0

8 27

[alsa-devel] [patch] ALSA: usb-audio: harmless underflow in snd_audigy2nx_led_put()
by Dan Carpenter 28 Sep '15

28 Sep '15

We want to verify that "value" is either zero or one, so we test if it is greater than one. Unfortunately, this is a signed int so it could also be negative. I think this is harmless but it introduces a static checker warning. Let's make "value" unsigned. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 337c317..4602a78 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -340,7 +340,7 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); struct usb_mixer_interface *mixer = list->mixer; int index = kcontrol->private_value & 0xff; - int value = ucontrol->value.integer.value[0]; + unsigned int value = ucontrol->value.integer.value[0]; int old_value = kcontrol->private_value >> 8; int err;

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix typos in snd_hdac_regmap_*() documents
by Takashi Iwai 28 Sep '15

28 Sep '15

Fixes the wrong reference names to regmap amp functions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_regmap.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index df705908480a..2767c55a641e 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -67,7 +67,7 @@ int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, * @reg: verb to write * @val: value to write * - * For writing an amp value, use snd_hda_regmap_amp_update(). + * For writing an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, @@ -85,7 +85,7 @@ snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, * @mask: bit mask to update * @val: value to update * - * For updating an amp value, use snd_hda_regmap_amp_update(). + * For updating an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid, -- 2.5.3

1 0

Re: [alsa-devel] [PATCH RFC v4 3/8] ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders
by Arnaud Pouliquen 28 Sep '15

28 Sep '15

few questions/remarks BR, Arnaud > +static void hdmi_codec_abort(struct device *dev) > +{ > + struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); > + > + dev_dbg(dev, "%s()\n", __func__); > + > + mutex_lock(&hcp->current_stream_lock); > + if (hcp->current_stream && hcp->current_stream->runtime && > + snd_pcm_running(hcp->current_stream)) { > + dev_info(dev, "HDMI audio playback aborted\n"); > + snd_pcm_stream_lock_irq(hcp->current_stream); > + snd_pcm_stop(hcp->current_stream, SNDRV_PCM_STATE_DISCONNECTED); > + snd_pcm_stream_unlock_irq(hcp->current_stream); > + } > + mutex_unlock(&hcp->current_stream_lock); > +} Does driver should stop the stream in case of unplug? This could generate unexpected behavior at application level. Perhaps should be better if this part is managed in DRM driver. if HDMI master, I2S bus should be maintained ON to consume the audio stream until application action. > + > +static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); > + struct hdmi_codec_params hp = { > + .iec = { > + .status = { 0 }, > + .subcode = { 0 }, > + .pad = 0, > + .dig_subframe = { 0 }, > + } > + }; > + int ret; > + > + dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__, > + params_width(params), params_rate(params), > + params_channels(params)); > + > + ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, > + sizeof(hp.iec.status)); Tell me if i am wrong, but in case of SPDIF format, IEC status is managed by cpu_dai not by the codec. To manage IEC61937 a control should be needed to update IEC status bits... > + if (ret < 0) { > + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", > + ret); > + return ret; > + } > + > + ret = hdmi_codec_new_stream(substream, dai); > + if (ret) > + return ret; > + > + hdmi_audio_infoframe_init(&hp.cea); > + hp.cea.channels = params_channels(params); > + hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; > + hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; > + hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; > + > + hp.sample_width = params_width(params); > + hp.sample_rate = params_rate(params); > + hp.channels = params_channels(params); > + > + return hcp->hcd.ops->hw_params(dai->dev->parent, > &hcp->daifmt[dai->id], > + &hp); > +} > +

4 3