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Re: [alsa-devel] Which project to choose?
by Raymond Yau 26 Jul '10

26 Jul '10

2010/7/14 Chris <cpollock(a)embarqmail.com> > On Wed, 2010-07-14 at 10:09 +0800, Raymond Yau wrote: > > > > > > > > > > You need to provide a full pulseaudio log and a test case > > which can > > > reproduce your problem > > > > > > open a new terminal > > > > > > pulseaudio -k ; pulseaudio -vvvvv > > > > Where should I post this Raymond? > > > > > > > > AFAIK , this mailing list has a limit of the message size > > > > > > BTW , > > > > did the program abort when you run alsa-time-test and pipe the output > > to "tail -n 500" > > > > > > > > http://thread.gmane.org/gmane.linux.alsa.devel/60371/focus=60520 > > > > Beware that the program generate a log of output > > Raymond, I've ran bzip2 on the output of pulseaudio -k ; pulseaudio > -vvvvv and it went from 129k to 9k. Should I fwd to you or post on the > list? > > -- > appl_ptr is also behind the hw_ptr in 0.9.21-0.1mdv2010.0 https://qa.mandriva.com/show_bug.cgi?id=56473#c15 Apr 12 15:21:43 localhost pulseaudio[13701]: alsa-util.c: appl_ptr : 28465776 Apr 12 15:21:43 localhost pulseaudio[13701]: alsa-util.c: hw_ptr : 28555040 and also appl_ptr is also behind the hw_ptr for snd_via82xx https://qa.mandriva.com/show_bug.cgi?id=56473#c0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: appl_ptr : 767919576 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: hw_ptr : 767990616

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Re: [alsa-devel] [Fwd: Re: [pulseaudio-discuss] pulseaudio debug mode]
by Raymond Yau 26 Jul '10

26 Jul '10

2010/7/26 Chris <cpollock(a)embarqmail.com> > Raymond, attached is a post I made today to the pulseaudio list. > > -- > Chris > KeyID 0xE372A7DA98E6705C > > > > ---------- 轉寄訊息 ---------- > From: Chris <cpollock(a)embarqmail.com> > To: pulseaudio-discuss(a)mail.0pointer.de > Date: Sun, 25 Jul 2010 12:15:02 -0500 > Subject: Re: [pulseaudio-discuss] pulseaudio debug mode > On Fri, 2010-07-23 at 08:58 +0100, Colin Guthrie wrote: > > 'Twas brillig, and Chris at 23/07/10 02:04 did gyre and gimble: > > > What is the best way to start PA for debugging and still have all the > > > usual clients running? > > > > If you mean having all the clients connect (e.g. applications with > > libcanberra support or similar for sound events), then there are > > basically two ways. > > > > The first is as Luke suggests. These clients will automatically > > reconnect to PA if they need to (provided you have a vaguely recent > > libcanberra), after it is restarted and run in debug mode. > > > > Alternatively you can simply set debug-level to "debug" in daemon.conf > > (in /etc/pulse or ~/.pulse), and then "grep pulseaudio /var/log/messages" > > > > Col > > > > Colin, link below is for debug output also some other output. Anything > look out of place that would cause the overruns? > > http://pastebin.com/ZWSWmXZt > ens1371 is ready to enter running state after PA server call snd_pcm_hw_params() The PA server only call snd_pcm_dump() at the probing phase to create working profile for the alsa sound cards, and snd_pcm_dump() indicate that appl_ptr is behind hw_ptr, but the playing thread is started after snd_pcm_dump() . This mean that some PA clients had connected and send audio to PA server before the first sink finished it setup since ens1371 is card 0 ( the fist sink device )

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Re: [alsa-devel] [LAD] announcing envy24control, mudita (*) edition.
by Niels Mayer 25 Jul '10

25 Jul '10

On Sun, Jul 25, 2010 at 12:21 PM, Paul Davis <paul(a)linuxaudiosystems.com> wrote: >> (2) All volumes are represented as decibels, including the 0 to -48dB > i don't own one, so its not of much concern to me, but dB<WHAT> ? The http://en.wikipedia.org/wiki/Decibel is a unitless measurement. :-) However, in this case there are a few different semi-implicit http://en.wikipedia.org/wiki/DBFS and other markings that I left off in the name of saving on screen-real-estate and reduced redundancy. Tooltips across-the board would be a very helpful addition to the envy24control 1.0.0 interface. On the other hand, these are exactly the same dB values as reported by ALSA's "amixer" and "alsamixer" so such complaints should go upstream. :-) Specifically: (1) The db markings in "Analog Volumes" correspond to analog levels that can be set for the DAC or ADC's. e.g. 0 -to- -63dB attenuation of the output of the analog DAC, and the +18 -to- -63dB attenuation/amplification of the analog ADC. These are the exact same dB readings reported by ALSA, e.g. "amixer -c M66": > Simple mixer control 'ADC',0 > Limits: 0 - 163 > Mono: 163 [100%] [18.00dB] > Simple mixer control 'ADC',1 > Limits: 0 - 163 > Mono: 163 [100%] [18.00dB] > Simple mixer control 'ADC',2 > Limits: 0 - 163 > Mono: 152 [93%] [12.50dB] > Simple mixer control 'ADC',3 > Limits: 0 - 163 > Mono: 152 [93%] [12.50dB] (2) The "0 to -48dB" peak metering levels are based on values from hardware peak metering (per http://alsa.cybermirror.org/manuals/icensemble/envy24.pdf ) > Peak data derived from the absolute value of 9 msb. 00h min - FFh max > volume. Reading the register resets the meter to 00h. When "0dBFS" reads on the peak meters in the new envy24control, it means 0dBFS relative to these 9MSB's. (see the red "0dBFS" labels in http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorInputs.… or http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorPCM.png ). Note that the signal would be able to go 7 or 15 LSB's higher before reaching actual full-scale peak. It would be more useful if there was a way for the hardware meters to detect actual clipping, e.g. value=256, whereas normally the meters report values from 0-255. For example "amixer -c M66 cget iface=PCM,name='Multi Track Peak',numid=45" returns: > type=INTEGER,access=r-------,values=22,min=0,max=255,step=0 > : values=0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,255,198,255,198 For proper metering, I suggest using Fons' excellent jkmeter with it's nice (and free) implementation of the K-System http://en.wikipedia.org/wiki/File:K-14_spectrafoo.jpg (caption: "digital audio meters in the Metric Halo application, called SpectraFoo"). (3) For the envy24's built in digital mixer, the 8PCMs + 2*SPDIF out, and up to 8 inputs + 2SPDIF inputs, all send to the digital mixer via 24dB attenuators. Seer http://nielsmayer.com/envy24control/envy24mixer-architecture.png for details. Thus all mixer input sliders have "0 -to- -144dB attenuation." (from http://alsa.cybermirror.org/manuals/icensemble/envy24.pdf ) This is what the above manual says about the Envy24's digital mixer: > 4.5.5 Multi-Track Digital Monitoring > > The Envy24 integrates a 36-bit resolution digital hardware mixer. The > width of the data path is strictly to ensure that during processing of > all the channels, under any condition, no resolution is lost. The > dynamic range of the end user system will be limited by the range of the > physical output devices used. In order to maintain identical gain to the > input stream (i.e. 0dB), the resulting 24-bit is not msb-aligned to the > 36-bit. The overflow bits correspond to the analog distortion due to > saturation. The user would need to reduce the overall attenuation of the > inputs to avoid clipping. Insertion of the digital mixer adds only a > single sample cycle delay with respect to the original data. This > extremely low latency all digital mixer provides monitoring > functionality and can replace a traditional external analog input > mixer. There are 20 independent audio data streams to mix and control > the volume. -- Niels http://nielsmayer.com

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[alsa-devel] ASoC update for 2.6.36
by Mark Brown 25 Jul '10

25 Jul '10

The following changes since commit 49e704279953b652ec6adc7f5dec412ad118a938: Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2010-07-21 15:01:07 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Kuninori Morimoto (1): ASoC: da7210: Add HeadPhone Playback Volume control sound/soc/codecs/da7210.c | 31 +++++++++++++++++++++++++++---- 1 files changed, 27 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH v3 0/1] Codec driver for TI WL1273 FM Radio.
by Matti J. Aaltonen 24 Jul '10

24 Jul '10

Hello all. Thank you for comments and thanks to Peter for advice on constraints and ALSA in general. New in this version: 1. Now constraints are used in the start-up function. It wasn't as big a deal as I thought... Also now the codec is always in full duplex mode. But it still refuses to do the wrong thing with FM as real full duplex is then impossible. 2. Fixed the exported mode support function. The name in now wl1273_get_format and it returns the audio interface format. 3. Removed the remaining obsolete functions. 4. Not actually new but... >> I didn't do this yet. But could you give me the git URL and then >> I'll resend? > git://git.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git I tried the corresponding http URL http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git but got fatal: http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git/info/refs not found: did you run git update-server-info on the server? Cheers, Matti Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec. sound/soc/codecs/wl1273.c | 598 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 43 ++++ 2 files changed, 641 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h

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[alsa-devel] [PATCH 6/6] pcmcia: remove cs_types.h
by Dominik Brodowski 23 Jul '10

23 Jul '10

Remove cs_types.h which is no longer needed: Most definitions aren't used at all, a few can be made away with, and two remaining definitions (typedefs, unfortunatley) may be moved to more specific places. CC: netdev(a)vger.kernel.org CC: linux-wireless(a)vger.kernel.org CC: linux-ide(a)vger.kernel.org CC: linux-usb(a)vger.kernel.org CC: laforge(a)gnumonks.org CC: linux-mtd(a)lists.infradead.org CC: linux-bluetooth(a)vger.kernel.org CC: alsa-devel(a)alsa-project.org CC: linux-serial(a)vger.kernel.org Signed-off-by: Dominik Brodowski <linux(a)dominikbrodowski.net> --- Documentation/pcmcia/driver-changes.txt | 5 +++ drivers/ata/pata_pcmcia.c | 1 - drivers/bluetooth/bluecard_cs.c | 1 - drivers/bluetooth/bt3c_cs.c | 1 - drivers/bluetooth/btuart_cs.c | 1 - drivers/bluetooth/dtl1_cs.c | 1 - drivers/char/pcmcia/cm4000_cs.c | 1 - drivers/char/pcmcia/cm4040_cs.c | 1 - drivers/char/pcmcia/ipwireless/main.h | 1 - drivers/char/pcmcia/ipwireless/tty.h | 1 - drivers/char/pcmcia/synclink_cs.c | 1 - drivers/ide/ide-cs.c | 1 - drivers/isdn/hardware/avm/avm_cs.c | 1 - drivers/isdn/hisax/avma1_cs.c | 1 - drivers/isdn/hisax/elsa_cs.c | 1 - drivers/isdn/hisax/sedlbauer_cs.c | 1 - drivers/isdn/hisax/teles_cs.c | 1 - drivers/mmc/host/sdricoh_cs.c | 1 - drivers/mtd/maps/pcmciamtd.c | 1 - drivers/net/pcmcia/3c574_cs.c | 1 - drivers/net/pcmcia/3c589_cs.c | 1 - drivers/net/pcmcia/axnet_cs.c | 1 - drivers/net/pcmcia/com20020_cs.c | 1 - drivers/net/pcmcia/fmvj18x_cs.c | 1 - drivers/net/pcmcia/ibmtr_cs.c | 1 - drivers/net/pcmcia/nmclan_cs.c | 1 - drivers/net/pcmcia/pcnet_cs.c | 5 +-- drivers/net/pcmcia/smc91c92_cs.c | 1 - drivers/net/pcmcia/xirc2ps_cs.c | 1 - drivers/net/wireless/airo_cs.c | 1 - drivers/net/wireless/atmel_cs.c | 1 - drivers/net/wireless/b43/pcmcia.c | 1 - drivers/net/wireless/hostap/hostap_cs.c | 3 +- drivers/net/wireless/libertas/if_cs.c | 1 - drivers/net/wireless/orinoco/orinoco_cs.c | 1 - drivers/net/wireless/orinoco/spectrum_cs.c | 1 - drivers/net/wireless/ray_cs.c | 1 - drivers/net/wireless/wl3501_cs.c | 10 +----- drivers/parport/parport_cs.c | 1 - drivers/pcmcia/au1000_generic.h | 1 - drivers/pcmcia/au1000_pb1x00.c | 2 - drivers/pcmcia/cistpl.c | 1 - drivers/pcmcia/cs.c | 1 - drivers/pcmcia/db1xxx_ss.c | 1 - drivers/pcmcia/ds.c | 1 - drivers/pcmcia/i82092.c | 1 - drivers/pcmcia/i82365.c | 1 - drivers/pcmcia/m32r_cfc.c | 1 - drivers/pcmcia/m32r_pcc.c | 1 - drivers/pcmcia/m8xx_pcmcia.c | 1 - drivers/pcmcia/pcmcia_cis.c | 1 - drivers/pcmcia/pcmcia_resource.c | 1 - drivers/pcmcia/pd6729.c | 1 - drivers/pcmcia/pxa2xx_base.c | 1 - drivers/pcmcia/rsrc_iodyn.c | 1 - drivers/pcmcia/rsrc_mgr.c | 1 - drivers/pcmcia/rsrc_nonstatic.c | 1 - drivers/pcmcia/sa1100_generic.c | 1 - drivers/pcmcia/soc_common.h | 1 - drivers/pcmcia/socket_sysfs.c | 1 - drivers/pcmcia/tcic.c | 1 - drivers/pcmcia/xxs1500_ss.c | 1 - drivers/pcmcia/yenta_socket.c | 1 - drivers/scsi/pcmcia/aha152x_stub.c | 1 - drivers/scsi/pcmcia/fdomain_stub.c | 1 - drivers/scsi/pcmcia/nsp_cs.c | 1 - drivers/scsi/pcmcia/qlogic_stub.c | 1 - drivers/scsi/pcmcia/sym53c500_cs.c | 1 - drivers/serial/serial_cs.c | 1 - drivers/ssb/main.c | 1 - drivers/ssb/pcmcia.c | 1 - drivers/ssb/scan.c | 1 - drivers/staging/comedi/drivers/cb_das16_cs.c | 1 - drivers/staging/comedi/drivers/das08_cs.c | 1 - drivers/staging/comedi/drivers/ni_daq_700.c | 1 - drivers/staging/comedi/drivers/ni_daq_dio24.c | 1 - drivers/staging/comedi/drivers/ni_labpc_cs.c | 1 - drivers/staging/comedi/drivers/ni_mio_cs.c | 1 - drivers/staging/comedi/drivers/quatech_daqp_cs.c | 1 - drivers/staging/wlags49_h2/wl_cs.c | 1 - drivers/staging/wlags49_h2/wl_internal.h | 1 - drivers/telephony/ixj_pcmcia.c | 1 - drivers/usb/host/sl811_cs.c | 5 +-- include/pcmcia/cistpl.h | 2 + include/pcmcia/cs.h | 10 +----- include/pcmcia/cs_types.h | 40 ---------------------- include/pcmcia/ds.h | 3 +- include/pcmcia/ss.h | 1 - sound/pcmcia/pdaudiocf/pdaudiocf.h | 1 - sound/pcmcia/vx/vxpocket.h | 1 - 90 files changed, 14 insertions(+), 151 deletions(-) delete mode 100644 include/pcmcia/cs_types.h diff --git a/Documentation/pcmcia/driver-changes.txt b/Documentation/pcmcia/driver-changes.txt index 61bc4e9..ff5f0be 100644 --- a/Documentation/pcmcia/driver-changes.txt +++ b/Documentation/pcmcia/driver-changes.txt @@ -1,4 +1,9 @@ This file details changes in 2.6 which affect PCMCIA card driver authors: +* No dev_info_t, no cs_types.h (as of 2.6.36) + dev_info_t and a few other typedefs are removed. No longer use them + in PCMCIA device drivers. Also, do not include pcmcia/cs_types.h, as + this file is gone. + * No dev_node_t (as of 2.6.35) There is no more need to fill out a "dev_node_t" structure. diff --git a/drivers/ata/pata_pcmcia.c b/drivers/ata/pata_pcmcia.c index 118c28e..3dcb2b1 100644 --- a/drivers/ata/pata_pcmcia.c +++ b/drivers/ata/pata_pcmcia.c @@ -34,7 +34,6 @@ #include <linux/ata.h> #include <linux/libata.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/bluetooth/bluecard_cs.c b/drivers/bluetooth/bluecard_cs.c index 6d34f40..eb085de 100644 --- a/drivers/bluetooth/bluecard_cs.c +++ b/drivers/bluetooth/bluecard_cs.c @@ -39,7 +39,6 @@ #include <linux/skbuff.h> #include <linux/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/bluetooth/bt3c_cs.c b/drivers/bluetooth/bt3c_cs.c index 21e05fd..457b603 100644 --- a/drivers/bluetooth/bt3c_cs.c +++ b/drivers/bluetooth/bt3c_cs.c @@ -45,7 +45,6 @@ #include <linux/device.h> #include <linux/firmware.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/bluetooth/btuart_cs.c b/drivers/bluetooth/btuart_cs.c index 4ed7288..e7e0a17 100644 --- a/drivers/bluetooth/btuart_cs.c +++ b/drivers/bluetooth/btuart_cs.c @@ -41,7 +41,6 @@ #include <asm/system.h> #include <asm/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/bluetooth/dtl1_cs.c b/drivers/bluetooth/dtl1_cs.c index ef044d5..7c94aad 100644 --- a/drivers/bluetooth/dtl1_cs.c +++ b/drivers/bluetooth/dtl1_cs.c @@ -41,7 +41,6 @@ #include <asm/system.h> #include <asm/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/char/pcmcia/cm4000_cs.c b/drivers/char/pcmcia/cm4000_cs.c index e7956ac..a8be2a7 100644 --- a/drivers/char/pcmcia/cm4000_cs.c +++ b/drivers/char/pcmcia/cm4000_cs.c @@ -34,7 +34,6 @@ #include <linux/uaccess.h> #include <linux/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/char/pcmcia/cm4040_cs.c b/drivers/char/pcmcia/cm4040_cs.c index c0775c8..44adae9 100644 --- a/drivers/char/pcmcia/cm4040_cs.c +++ b/drivers/char/pcmcia/cm4040_cs.c @@ -29,7 +29,6 @@ #include <asm/uaccess.h> #include <asm/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/char/pcmcia/ipwireless/main.h b/drivers/char/pcmcia/ipwireless/main.h index 96d0ef3..c207be8 100644 --- a/drivers/char/pcmcia/ipwireless/main.h +++ b/drivers/char/pcmcia/ipwireless/main.h @@ -21,7 +21,6 @@ #include <linux/sched.h> #include <linux/types.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/char/pcmcia/ipwireless/tty.h b/drivers/char/pcmcia/ipwireless/tty.h index 4da6c20..3e163d4 100644 --- a/drivers/char/pcmcia/ipwireless/tty.h +++ b/drivers/char/pcmcia/ipwireless/tty.h @@ -21,7 +21,6 @@ #include <linux/types.h> #include <linux/sched.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/char/pcmcia/synclink_cs.c b/drivers/char/pcmcia/synclink_cs.c index 308903e..522992e 100644 --- a/drivers/char/pcmcia/synclink_cs.c +++ b/drivers/char/pcmcia/synclink_cs.c @@ -70,7 +70,6 @@ #include <linux/workqueue.h> #include <linux/hdlc.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/ide/ide-cs.c b/drivers/ide/ide-cs.c index 0b7815d..27dbab8 100644 --- a/drivers/ide/ide-cs.c +++ b/drivers/ide/ide-cs.c @@ -43,7 +43,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/isdn/hardware/avm/avm_cs.c b/drivers/isdn/hardware/avm/avm_cs.c index f410d0e..e804a01 100644 --- a/drivers/isdn/hardware/avm/avm_cs.c +++ b/drivers/isdn/hardware/avm/avm_cs.c @@ -20,7 +20,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/isdn/hisax/avma1_cs.c b/drivers/isdn/hisax/avma1_cs.c index a80a761..49e141e 100644 --- a/drivers/isdn/hisax/avma1_cs.c +++ b/drivers/isdn/hisax/avma1_cs.c @@ -20,7 +20,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/isdn/hisax/elsa_cs.c b/drivers/isdn/hisax/elsa_cs.c index 218927e..425deea 100644 --- a/drivers/isdn/hisax/elsa_cs.c +++ b/drivers/isdn/hisax/elsa_cs.c @@ -46,7 +46,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/isdn/hisax/sedlbauer_cs.c b/drivers/isdn/hisax/sedlbauer_cs.c index 1f4feaa..5dbad96 100644 --- a/drivers/isdn/hisax/sedlbauer_cs.c +++ b/drivers/isdn/hisax/sedlbauer_cs.c @@ -46,7 +46,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/isdn/hisax/teles_cs.c b/drivers/isdn/hisax/teles_cs.c index 5771955..d3fb1b7 100644 --- a/drivers/isdn/hisax/teles_cs.c +++ b/drivers/isdn/hisax/teles_cs.c @@ -27,7 +27,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/mmc/host/sdricoh_cs.c b/drivers/mmc/host/sdricoh_cs.c index e7507af..7aa65bb 100644 --- a/drivers/mmc/host/sdricoh_cs.c +++ b/drivers/mmc/host/sdricoh_cs.c @@ -30,7 +30,6 @@ #include <linux/ioport.h> #include <linux/scatterlist.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/mtd/maps/pcmciamtd.c b/drivers/mtd/maps/pcmciamtd.c index e699e6a..7948816 100644 --- a/drivers/mtd/maps/pcmciamtd.c +++ b/drivers/mtd/maps/pcmciamtd.c @@ -16,7 +16,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/net/pcmcia/3c574_cs.c b/drivers/net/pcmcia/3c574_cs.c index 10ee106..e249b89 100644 --- a/drivers/net/pcmcia/3c574_cs.c +++ b/drivers/net/pcmcia/3c574_cs.c @@ -87,7 +87,6 @@ earlier 3Com products. #include <linux/bitops.h> #include <linux/mii.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/pcmcia/3c589_cs.c b/drivers/net/pcmcia/3c589_cs.c index ce63c37..b0772df 100644 --- a/drivers/net/pcmcia/3c589_cs.c +++ b/drivers/net/pcmcia/3c589_cs.c @@ -41,7 +41,6 @@ #include <linux/bitops.h> #include <linux/jiffies.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/pcmcia/axnet_cs.c b/drivers/net/pcmcia/axnet_cs.c index 33525bf..467fd4b 100644 --- a/drivers/net/pcmcia/axnet_cs.c +++ b/drivers/net/pcmcia/axnet_cs.c @@ -39,7 +39,6 @@ #include <linux/mii.h> #include "../8390.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/net/pcmcia/com20020_cs.c b/drivers/net/pcmcia/com20020_cs.c index 5643f94..99957af 100644 --- a/drivers/net/pcmcia/com20020_cs.c +++ b/drivers/net/pcmcia/com20020_cs.c @@ -43,7 +43,6 @@ #include <linux/arcdevice.h> #include <linux/com20020.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/net/pcmcia/fmvj18x_cs.c b/drivers/net/pcmcia/fmvj18x_cs.c index 7c27c50..95a991b 100644 --- a/drivers/net/pcmcia/fmvj18x_cs.c +++ b/drivers/net/pcmcia/fmvj18x_cs.c @@ -49,7 +49,6 @@ #include <linux/ioport.h> #include <linux/crc32.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/net/pcmcia/ibmtr_cs.c b/drivers/net/pcmcia/ibmtr_cs.c index 67ee985..c36dcd1 100644 --- a/drivers/net/pcmcia/ibmtr_cs.c +++ b/drivers/net/pcmcia/ibmtr_cs.c @@ -57,7 +57,6 @@ #include <linux/trdevice.h> #include <linux/ibmtr.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/net/pcmcia/nmclan_cs.c b/drivers/net/pcmcia/nmclan_cs.c index 9b63dec..c0eacfa 100644 --- a/drivers/net/pcmcia/nmclan_cs.c +++ b/drivers/net/pcmcia/nmclan_cs.c @@ -146,7 +146,6 @@ Include Files #include <linux/ioport.h> #include <linux/bitops.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cisreg.h> #include <pcmcia/cistpl.h> diff --git a/drivers/net/pcmcia/pcnet_cs.c b/drivers/net/pcmcia/pcnet_cs.c index bfdef72..db6dbda 100644 --- a/drivers/net/pcmcia/pcnet_cs.c +++ b/drivers/net/pcmcia/pcnet_cs.c @@ -42,7 +42,6 @@ #include <linux/mii.h> #include "../8390.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> @@ -113,8 +112,6 @@ static int setup_dma_config(struct pcmcia_device *link, int start_pg, static void pcnet_detach(struct pcmcia_device *p_dev); -static dev_info_t dev_info = "pcnet_cs"; - /*====================================================================*/ typedef struct hw_info_t { @@ -956,7 +953,7 @@ static int pcnet_open(struct net_device *dev) set_misc_reg(dev); outb_p(0xFF, nic_base + EN0_ISR); /* Clear bogus intr. */ - ret = request_irq(dev->irq, ei_irq_wrapper, IRQF_SHARED, dev_info, dev); + ret = request_irq(dev->irq, ei_irq_wrapper, IRQF_SHARED, dev->name, dev); if (ret) return ret; diff --git a/drivers/net/pcmcia/smc91c92_cs.c b/drivers/net/pcmcia/smc91c92_cs.c index 307cd17..88f503a 100644 --- a/drivers/net/pcmcia/smc91c92_cs.c +++ b/drivers/net/pcmcia/smc91c92_cs.c @@ -44,7 +44,6 @@ #include <linux/jiffies.h> #include <linux/firmware.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/pcmcia/xirc2ps_cs.c b/drivers/net/pcmcia/xirc2ps_cs.c index b6c3644..a7662f0 100644 --- a/drivers/net/pcmcia/xirc2ps_cs.c +++ b/drivers/net/pcmcia/xirc2ps_cs.c @@ -82,7 +82,6 @@ #include <linux/bitops.h> #include <linux/mii.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/airo_cs.c b/drivers/net/wireless/airo_cs.c index 33bdc6a..9389ba0 100644 --- a/drivers/net/wireless/airo_cs.c +++ b/drivers/net/wireless/airo_cs.c @@ -32,7 +32,6 @@ #include <linux/timer.h> #include <linux/netdevice.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/atmel_cs.c b/drivers/net/wireless/atmel_cs.c index c2746fc..91ee74a 100644 --- a/drivers/net/wireless/atmel_cs.c +++ b/drivers/net/wireless/atmel_cs.c @@ -42,7 +42,6 @@ #include <linux/moduleparam.h> #include <linux/device.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/b43/pcmcia.c b/drivers/net/wireless/b43/pcmcia.c index 0e99b63..f71bc78 100644 --- a/drivers/net/wireless/b43/pcmcia.c +++ b/drivers/net/wireless/b43/pcmcia.c @@ -26,7 +26,6 @@ #include <linux/ssb/ssb.h> #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/net/wireless/hostap/hostap_cs.c b/drivers/net/wireless/hostap/hostap_cs.c index 29b31a6..2f4b6d4 100644 --- a/drivers/net/wireless/hostap/hostap_cs.c +++ b/drivers/net/wireless/hostap/hostap_cs.c @@ -12,7 +12,6 @@ #include <linux/wireless.h> #include <net/iw_handler.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> @@ -23,7 +22,7 @@ #include "hostap_wlan.h" -static dev_info_t dev_info = "hostap_cs"; +static char *dev_info = "hostap_cs"; MODULE_AUTHOR("Jouni Malinen"); MODULE_DESCRIPTION("Support for Intersil Prism2-based 802.11 wireless LAN " diff --git a/drivers/net/wireless/libertas/if_cs.c b/drivers/net/wireless/libertas/if_cs.c index 08e4e39..1d3a7e0 100644 --- a/drivers/net/wireless/libertas/if_cs.c +++ b/drivers/net/wireless/libertas/if_cs.c @@ -28,7 +28,6 @@ #include <linux/firmware.h> #include <linux/netdevice.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/net/wireless/orinoco/orinoco_cs.c b/drivers/net/wireless/orinoco/orinoco_cs.c index b16d5db..41ca4f1 100644 --- a/drivers/net/wireless/orinoco/orinoco_cs.c +++ b/drivers/net/wireless/orinoco/orinoco_cs.c @@ -17,7 +17,6 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/orinoco/spectrum_cs.c b/drivers/net/wireless/orinoco/spectrum_cs.c index b51a9ad..cad30e4 100644 --- a/drivers/net/wireless/orinoco/spectrum_cs.c +++ b/drivers/net/wireless/orinoco/spectrum_cs.c @@ -25,7 +25,6 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/ray_cs.c b/drivers/net/wireless/ray_cs.c index abff893..165beb6 100644 --- a/drivers/net/wireless/ray_cs.c +++ b/drivers/net/wireless/ray_cs.c @@ -46,7 +46,6 @@ #include <linux/ethtool.h> #include <linux/ieee80211.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/wl3501_cs.c b/drivers/net/wireless/wl3501_cs.c index 376c6b9..35f431b 100644 --- a/drivers/net/wireless/wl3501_cs.c +++ b/drivers/net/wireless/wl3501_cs.c @@ -48,7 +48,6 @@ #include <net/iw_handler.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> @@ -89,13 +88,6 @@ static int wl3501_config(struct pcmcia_device *link); static void wl3501_release(struct pcmcia_device *link); -/* - * The dev_info variable is the "key" that is used to match up this - * device driver with appropriate cards, through the card configuration - * database. - */ -static dev_info_t wl3501_dev_info = "wl3501_cs"; - static const struct { int reg_domain; int min, max, deflt; @@ -1421,7 +1413,7 @@ static struct iw_statistics *wl3501_get_wireless_stats(struct net_device *dev) static void wl3501_get_drvinfo(struct net_device *dev, struct ethtool_drvinfo *info) { - strlcpy(info->driver, wl3501_dev_info, sizeof(info->driver)); + strlcpy(info->driver, "wl3501_cs", sizeof(info->driver)); } static const struct ethtool_ops ops = { diff --git a/drivers/parport/parport_cs.c b/drivers/parport/parport_cs.c index fd8cfe9..ee56fd6 100644 --- a/drivers/parport/parport_cs.c +++ b/drivers/parport/parport_cs.c @@ -48,7 +48,6 @@ #include <linux/parport.h> #include <linux/parport_pc.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/pcmcia/au1000_generic.h b/drivers/pcmcia/au1000_generic.h index a324d32..67530ce 100644 --- a/drivers/pcmcia/au1000_generic.h +++ b/drivers/pcmcia/au1000_generic.h @@ -23,7 +23,6 @@ /* include the world */ -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/au1000_pb1x00.c b/drivers/pcmcia/au1000_pb1x00.c index 5a979cb..807f2d7 100644 --- a/drivers/pcmcia/au1000_pb1x00.c +++ b/drivers/pcmcia/au1000_pb1x00.c @@ -31,11 +31,9 @@ #include <linux/proc_fs.h> #include <linux/types.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> -#include <pcmcia/bus_ops.h> #include <asm/io.h> #include <asm/irq.h> diff --git a/drivers/pcmcia/cistpl.c b/drivers/pcmcia/cistpl.c index 8844bc3..ba4a5ac 100644 --- a/drivers/pcmcia/cistpl.c +++ b/drivers/pcmcia/cistpl.c @@ -27,7 +27,6 @@ #include <asm/byteorder.h> #include <asm/unaligned.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cisreg.h> diff --git a/drivers/pcmcia/cs.c b/drivers/pcmcia/cs.c index efa30b8..2ec8ac9 100644 --- a/drivers/pcmcia/cs.c +++ b/drivers/pcmcia/cs.c @@ -32,7 +32,6 @@ #include <asm/system.h> #include <asm/irq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/db1xxx_ss.c b/drivers/pcmcia/db1xxx_ss.c index 0f4cc3f..27575e6 100644 --- a/drivers/pcmcia/db1xxx_ss.c +++ b/drivers/pcmcia/db1xxx_ss.c @@ -29,7 +29,6 @@ #include <linux/slab.h> #include <linux/spinlock.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <asm/mach-au1x00/au1000.h> diff --git a/drivers/pcmcia/ds.c b/drivers/pcmcia/ds.c index 73aeaea..56933eb 100644 --- a/drivers/pcmcia/ds.c +++ b/drivers/pcmcia/ds.c @@ -26,7 +26,6 @@ #include <linux/dma-mapping.h> #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/pcmcia/i82092.c b/drivers/pcmcia/i82092.c index 3003bb3..05d0879 100644 --- a/drivers/pcmcia/i82092.c +++ b/drivers/pcmcia/i82092.c @@ -15,7 +15,6 @@ #include <linux/interrupt.h> #include <linux/device.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/i82365.c b/drivers/pcmcia/i82365.c index 9e2a156..61746bd 100644 --- a/drivers/pcmcia/i82365.c +++ b/drivers/pcmcia/i82365.c @@ -50,7 +50,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/m32r_cfc.c b/drivers/pcmcia/m32r_cfc.c index 7e16ed8..24de499 100644 --- a/drivers/pcmcia/m32r_cfc.c +++ b/drivers/pcmcia/m32r_cfc.c @@ -26,7 +26,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/m32r_pcc.c b/drivers/pcmcia/m32r_pcc.c index 6c5c3f9..8e47238 100644 --- a/drivers/pcmcia/m32r_pcc.c +++ b/drivers/pcmcia/m32r_pcc.c @@ -27,7 +27,6 @@ #include <asm/system.h> #include <asm/addrspace.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/m8xx_pcmcia.c b/drivers/pcmcia/m8xx_pcmcia.c index 25e5e30..f2f90a7 100644 --- a/drivers/pcmcia/m8xx_pcmcia.c +++ b/drivers/pcmcia/m8xx_pcmcia.c @@ -59,7 +59,6 @@ #include <asm/irq.h> #include <asm/fs_pd.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> diff --git a/drivers/pcmcia/pcmcia_cis.c b/drivers/pcmcia/pcmcia_cis.c index 4a65eaf..0ac54da 100644 --- a/drivers/pcmcia/pcmcia_cis.c +++ b/drivers/pcmcia/pcmcia_cis.c @@ -19,7 +19,6 @@ #include <linux/kernel.h> #include <linux/netdevice.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cisreg.h> #include <pcmcia/cistpl.h> #include <pcmcia/ss.h> diff --git a/drivers/pcmcia/pcmcia_resource.c b/drivers/pcmcia/pcmcia_resource.c index a4cd9ad..2394de4 100644 --- a/drivers/pcmcia/pcmcia_resource.c +++ b/drivers/pcmcia/pcmcia_resource.c @@ -25,7 +25,6 @@ #include <asm/irq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/pd6729.c b/drivers/pcmcia/pd6729.c index b61a136..b8a869a 100644 --- a/drivers/pcmcia/pd6729.c +++ b/drivers/pcmcia/pd6729.c @@ -17,7 +17,6 @@ #include <linux/device.h> #include <linux/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/pxa2xx_base.c b/drivers/pcmcia/pxa2xx_base.c index df4532e..66c0225 100644 --- a/drivers/pcmcia/pxa2xx_base.c +++ b/drivers/pcmcia/pxa2xx_base.c @@ -32,7 +32,6 @@ #include <mach/pxa2xx-regs.h> #include <asm/mach-types.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/rsrc_iodyn.c b/drivers/pcmcia/rsrc_iodyn.c index 6ed7bf1..3b1dce2 100644 --- a/drivers/pcmcia/rsrc_iodyn.c +++ b/drivers/pcmcia/rsrc_iodyn.c @@ -16,7 +16,6 @@ #include <linux/module.h> #include <linux/kernel.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/rsrc_mgr.c b/drivers/pcmcia/rsrc_mgr.c index b12ecf7..b433a79 100644 --- a/drivers/pcmcia/rsrc_mgr.c +++ b/drivers/pcmcia/rsrc_mgr.c @@ -16,7 +16,6 @@ #include <linux/module.h> #include <linux/kernel.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/rsrc_nonstatic.c b/drivers/pcmcia/rsrc_nonstatic.c index 13245a2..0cca08f 100644 --- a/drivers/pcmcia/rsrc_nonstatic.c +++ b/drivers/pcmcia/rsrc_nonstatic.c @@ -28,7 +28,6 @@ #include <asm/irq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/sa1100_generic.c b/drivers/pcmcia/sa1100_generic.c index edbd8c4..e098514 100644 --- a/drivers/pcmcia/sa1100_generic.c +++ b/drivers/pcmcia/sa1100_generic.c @@ -35,7 +35,6 @@ #include <linux/slab.h> #include <linux/platform_device.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> diff --git a/drivers/pcmcia/soc_common.h b/drivers/pcmcia/soc_common.h index e40824c..3fba3a6 100644 --- a/drivers/pcmcia/soc_common.h +++ b/drivers/pcmcia/soc_common.h @@ -11,7 +11,6 @@ /* include the world */ #include <linux/cpufreq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/socket_sysfs.c b/drivers/pcmcia/socket_sysfs.c index 80e36bc..cb0d3ac 100644 --- a/drivers/pcmcia/socket_sysfs.c +++ b/drivers/pcmcia/socket_sysfs.c @@ -26,7 +26,6 @@ #include <asm/system.h> #include <asm/irq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/tcic.c b/drivers/pcmcia/tcic.c index 56004a1..be0d841 100644 --- a/drivers/pcmcia/tcic.c +++ b/drivers/pcmcia/tcic.c @@ -49,7 +49,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include "tcic.h" diff --git a/drivers/pcmcia/xxs1500_ss.c b/drivers/pcmcia/xxs1500_ss.c index 201ccfa..fa88c36 100644 --- a/drivers/pcmcia/xxs1500_ss.c +++ b/drivers/pcmcia/xxs1500_ss.c @@ -17,7 +17,6 @@ #include <linux/slab.h> #include <linux/spinlock.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/yenta_socket.c b/drivers/pcmcia/yenta_socket.c index f1d4137..414d9a6 100644 --- a/drivers/pcmcia/yenta_socket.c +++ b/drivers/pcmcia/yenta_socket.c @@ -19,7 +19,6 @@ #include <linux/io.h> #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/scsi/pcmcia/aha152x_stub.c b/drivers/scsi/pcmcia/aha152x_stub.c index 9d70aef..b07b53e 100644 --- a/drivers/scsi/pcmcia/aha152x_stub.c +++ b/drivers/scsi/pcmcia/aha152x_stub.c @@ -49,7 +49,6 @@ #include <scsi/scsi_host.h> #include "aha152x.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/scsi/pcmcia/fdomain_stub.c b/drivers/scsi/pcmcia/fdomain_stub.c index 21b1411..ee04897 100644 --- a/drivers/scsi/pcmcia/fdomain_stub.c +++ b/drivers/scsi/pcmcia/fdomain_stub.c @@ -46,7 +46,6 @@ #include <scsi/scsi_host.h> #include "fdomain.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/scsi/pcmcia/nsp_cs.c b/drivers/scsi/pcmcia/nsp_cs.c index 0f0e112..d414207 100644 --- a/drivers/scsi/pcmcia/nsp_cs.c +++ b/drivers/scsi/pcmcia/nsp_cs.c @@ -47,7 +47,6 @@ #include <scsi/scsi.h> #include <scsi/scsi_ioctl.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/scsi/pcmcia/qlogic_stub.c b/drivers/scsi/pcmcia/qlogic_stub.c index f0fc6ba..c1cf7f4 100644 --- a/drivers/scsi/pcmcia/qlogic_stub.c +++ b/drivers/scsi/pcmcia/qlogic_stub.c @@ -48,7 +48,6 @@ #include <scsi/scsi_host.h> #include "../qlogicfas408.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/scsi/pcmcia/sym53c500_cs.c b/drivers/scsi/pcmcia/sym53c500_cs.c index a511641..bd79e45 100644 --- a/drivers/scsi/pcmcia/sym53c500_cs.c +++ b/drivers/scsi/pcmcia/sym53c500_cs.c @@ -71,7 +71,6 @@ #include <scsi/scsi.h> #include <scsi/scsi_host.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/serial/serial_cs.c b/drivers/serial/serial_cs.c index ab17c08..2b99c7b 100644 --- a/drivers/serial/serial_cs.c +++ b/drivers/serial/serial_cs.c @@ -45,7 +45,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/ssb/main.c b/drivers/ssb/main.c index 51275aa..06f04b4 100644 --- a/drivers/ssb/main.c +++ b/drivers/ssb/main.c @@ -20,7 +20,6 @@ #include <linux/mmc/sdio_func.h> #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/ssb/pcmcia.c b/drivers/ssb/pcmcia.c index e72f404..2152030 100644 --- a/drivers/ssb/pcmcia.c +++ b/drivers/ssb/pcmcia.c @@ -13,7 +13,6 @@ #include <linux/io.h> #include <linux/etherdevice.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/ssb/scan.c b/drivers/ssb/scan.c index 0d6c028..9738cad 100644 --- a/drivers/ssb/scan.c +++ b/drivers/ssb/scan.c @@ -17,7 +17,6 @@ #include <linux/pci.h> #include <linux/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/staging/comedi/drivers/cb_das16_cs.c b/drivers/staging/comedi/drivers/cb_das16_cs.c index cfeb11f..acef29c 100644 --- a/drivers/staging/comedi/drivers/cb_das16_cs.c +++ b/drivers/staging/comedi/drivers/cb_das16_cs.c @@ -37,7 +37,6 @@ Status: experimental #include <linux/delay.h> #include <linux/pci.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/staging/comedi/drivers/das08_cs.c b/drivers/staging/comedi/drivers/das08_cs.c index 8761a6d..3eddb7c 100644 --- a/drivers/staging/comedi/drivers/das08_cs.c +++ b/drivers/staging/comedi/drivers/das08_cs.c @@ -48,7 +48,6 @@ Command support does not exist, but could be added for this board. #include "das08.h" /* pcmcia includes */ -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/staging/comedi/drivers/ni_daq_700.c b/drivers/staging/comedi/drivers/ni_daq_700.c index 6ec77bf..f3c4d2f 100644 --- a/drivers/staging/comedi/drivers/ni_daq_700.c +++ b/drivers/staging/comedi/drivers/ni_daq_700.c @@ -47,7 +47,6 @@ IRQ is assigned but not used. #include <linux/ioport.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/comedi/drivers/ni_daq_dio24.c b/drivers/staging/comedi/drivers/ni_daq_dio24.c index e4865b1..f0c4367 100644 --- a/drivers/staging/comedi/drivers/ni_daq_dio24.c +++ b/drivers/staging/comedi/drivers/ni_daq_dio24.c @@ -48,7 +48,6 @@ the PCMCIA interface. #include "8255.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/comedi/drivers/ni_labpc_cs.c b/drivers/staging/comedi/drivers/ni_labpc_cs.c index 163245e..1ee78f8 100644 --- a/drivers/staging/comedi/drivers/ni_labpc_cs.c +++ b/drivers/staging/comedi/drivers/ni_labpc_cs.c @@ -71,7 +71,6 @@ NI manuals: #include "comedi_fc.h" #include "ni_labpc.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/comedi/drivers/ni_mio_cs.c b/drivers/staging/comedi/drivers/ni_mio_cs.c index 3a46f0c..0bce220 100644 --- a/drivers/staging/comedi/drivers/ni_mio_cs.c +++ b/drivers/staging/comedi/drivers/ni_mio_cs.c @@ -48,7 +48,6 @@ See the notes in the ni_atmio.o driver. #include "ni_stc.h" #include "8255.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/staging/comedi/drivers/quatech_daqp_cs.c b/drivers/staging/comedi/drivers/quatech_daqp_cs.c index a91db6c..67c0fa6 100644 --- a/drivers/staging/comedi/drivers/quatech_daqp_cs.c +++ b/drivers/staging/comedi/drivers/quatech_daqp_cs.c @@ -50,7 +50,6 @@ Devices: [Quatech] DAQP-208 (daqp), DAQP-308 #include "../comedidev.h" #include <linux/semaphore.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/wlags49_h2/wl_cs.c b/drivers/staging/wlags49_h2/wl_cs.c index 10abd40..afe4575 100644 --- a/drivers/staging/wlags49_h2/wl_cs.c +++ b/drivers/staging/wlags49_h2/wl_cs.c @@ -83,7 +83,6 @@ #include <linux/if_arp.h> #include <linux/ioport.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/wlags49_h2/wl_internal.h b/drivers/staging/wlags49_h2/wl_internal.h index d9a0ad0..02f0a20 100644 --- a/drivers/staging/wlags49_h2/wl_internal.h +++ b/drivers/staging/wlags49_h2/wl_internal.h @@ -69,7 +69,6 @@ ******************************************************************************/ #include <linux/version.h> #ifdef BUS_PCMCIA -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/telephony/ixj_pcmcia.c b/drivers/telephony/ixj_pcmcia.c index 99cb224..f6c7e6f 100644 --- a/drivers/telephony/ixj_pcmcia.c +++ b/drivers/telephony/ixj_pcmcia.c @@ -8,7 +8,6 @@ #include <linux/errno.h> /* error codes */ #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/usb/host/sl811_cs.c b/drivers/usb/host/sl811_cs.c index 58cb73c..acb7e25 100644 --- a/drivers/usb/host/sl811_cs.c +++ b/drivers/usb/host/sl811_cs.c @@ -20,7 +20,6 @@ #include <linux/ioport.h> #include <linux/platform_device.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> @@ -43,8 +42,6 @@ MODULE_LICENSE("GPL"); /* VARIABLES */ /*====================================================================*/ -static const char driver_name[DEV_NAME_LEN] = "sl811_cs"; - typedef struct local_info_t { struct pcmcia_device *p_dev; } local_info_t; @@ -246,7 +243,7 @@ MODULE_DEVICE_TABLE(pcmcia, sl811_ids); static struct pcmcia_driver sl811_cs_driver = { .owner = THIS_MODULE, .drv = { - .name = (char *)driver_name, + .name = "sl811_cs", }, .probe = sl811_cs_probe, .remove = sl811_cs_detach, diff --git a/include/pcmcia/cistpl.h b/include/pcmcia/cistpl.h index cfdd5af..1c5088c 100644 --- a/include/pcmcia/cistpl.h +++ b/include/pcmcia/cistpl.h @@ -15,6 +15,8 @@ #ifndef _LINUX_CISTPL_H #define _LINUX_CISTPL_H +typedef unsigned char cisdata_t; + #define CISTPL_NULL 0x00 #define CISTPL_DEVICE 0x01 #define CISTPL_LONGLINK_CB 0x02 diff --git a/include/pcmcia/cs.h b/include/pcmcia/cs.h index c943c96..c78d9b1 100644 --- a/include/pcmcia/cs.h +++ b/include/pcmcia/cs.h @@ -43,14 +43,6 @@ typedef struct conf_reg_t { #define CV_COPY_VALUE 0x08 #define CV_EXT_STATUS 0x10 -/* For GetFirst/NextClient */ -typedef struct client_req_t { - socket_t Socket; - u_int Attributes; -} client_req_t; - -#define CLIENT_THIS_SOCKET 0x01 - /* ModifyConfiguration */ typedef struct modconf_t { u_int Attributes; @@ -133,7 +125,7 @@ typedef struct io_req_t { /* For GetMemPage, MapMemPage */ typedef struct memreq_t { u_int CardOffset; - page_t Page; + u_short Page; } memreq_t; /* For ModifyWindow */ diff --git a/include/pcmcia/cs_types.h b/include/pcmcia/cs_types.h deleted file mode 100644 index f5e3b83..0000000 --- a/include/pcmcia/cs_types.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * cs_types.h - * - * 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. - * - * The initial developer of the original code is David A. Hinds - * <dahinds(a)users.sourceforge.net>. Portions created by David A. Hinds - * are Copyright (C) 1999 David A. Hinds. All Rights Reserved. - * - * (C) 1999 David A. Hinds - */ - -#ifndef _LINUX_CS_TYPES_H -#define _LINUX_CS_TYPES_H - -#ifdef __KERNEL__ -#include <linux/types.h> -#else -#include <sys/types.h> -#endif - -typedef u_short socket_t; -typedef u_int event_t; -typedef u_char cisdata_t; -typedef u_short page_t; - -typedef unsigned long window_handle_t; - -struct region_t; -typedef struct region_t *memory_handle_t; - -#ifndef DEV_NAME_LEN -#define DEV_NAME_LEN 32 -#endif - -typedef char dev_info_t[DEV_NAME_LEN]; - -#endif /* _LINUX_CS_TYPES_H */ diff --git a/include/pcmcia/ds.h b/include/pcmcia/ds.h index 7d7721e..e614aa0 100644 --- a/include/pcmcia/ds.h +++ b/include/pcmcia/ds.h @@ -20,7 +20,6 @@ #include <linux/mod_devicetable.h> #endif -#include <pcmcia/cs_types.h> #include <pcmcia/device_id.h> #ifdef __KERNEL__ @@ -37,6 +36,8 @@ struct pcmcia_device; struct config_t; struct net_device; +typedef unsigned long window_handle_t; + /* dynamic device IDs for PCMCIA device drivers. See * Documentation/pcmcia/driver.txt for details. */ diff --git a/include/pcmcia/ss.h b/include/pcmcia/ss.h index aeac271..626b63c 100644 --- a/include/pcmcia/ss.h +++ b/include/pcmcia/ss.h @@ -19,7 +19,6 @@ #include <linux/sched.h> /* task_struct, completion */ #include <linux/mutex.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #ifdef CONFIG_CARDBUS #include <linux/pci.h> diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h index a0a7ec6..5cc3e45 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.h +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h @@ -24,7 +24,6 @@ #include <sound/pcm.h> #include <asm/io.h> #include <linux/interrupt.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/sound/pcmcia/vx/vxpocket.h b/sound/pcmcia/vx/vxpocket.h index ea4df16..d911066 100644 --- a/sound/pcmcia/vx/vxpocket.h +++ b/sound/pcmcia/vx/vxpocket.h @@ -23,7 +23,6 @@ #include <sound/vx_core.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> -- 1.7.0.4

4 3

[alsa-devel] [PATCH][v2] powerpc: rename immap_86xx.h to fsl_guts.h, and add 85xx support
by Timur Tabi 23 Jul '10

23 Jul '10

The immap_86xx.h header file only defines one data structure: the "global utilities" register set found on Freescale PowerPC SOCs. Rename this file to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS" register set on 85xx chips. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Liam, This patch is a powerpc-patch, but it's much simpler if you apply it to multi-component, since only the SSI audio drivers are affected. .../include/asm/{immap_86xx.h => fsl_guts.h} | 111 +++++++++++++------ sound/soc/fsl/fsl_ssi.c | 2 - sound/soc/fsl/mpc8610_hpcd.c | 10 +- 3 files changed, 81 insertions(+), 42 deletions(-) rename arch/powerpc/include/asm/{immap_86xx.h => fsl_guts.h} (66%) diff --git a/arch/powerpc/include/asm/immap_86xx.h b/arch/powerpc/include/asm/fsl_guts.h similarity index 66% rename from arch/powerpc/include/asm/immap_86xx.h rename to arch/powerpc/include/asm/fsl_guts.h index 0f165e5..bebd124 100644 --- a/arch/powerpc/include/asm/immap_86xx.h +++ b/arch/powerpc/include/asm/fsl_guts.h @@ -1,5 +1,5 @@ /** - * MPC86xx Internal Memory Map + * Freecale 85xx and 86xx Global Utilties register set * * Authors: Jeff Brown * Timur Tabi <timur(a)freescale.com> @@ -10,73 +10,112 @@ * 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 header file defines structures for various 86xx SOC devices that are - * used by multiple source files. */ -#ifndef __ASM_POWERPC_IMMAP_86XX_H__ -#define __ASM_POWERPC_IMMAP_86XX_H__ +#ifndef __ASM_POWERPC_FSL_GUTS_H__ +#define __ASM_POWERPC_FSL_GUTS_H__ #ifdef __KERNEL__ -/* Global Utility Registers */ -struct ccsr_guts { +/* + * These #ifdefs are safe because it's not possible to build a kernel that + * runs on e500 and e600 cores. + */ + +#if !defined(CONFIG_PPC_85xx) && !defined(CONFIG_PPC_86xx) +#error Only 85xx and 86xx SOCs are supported +#endif + +/** + * Global Utility Registers. + * + * Not all registers defined in this structure are available on all chips, so + * you are expected to know whether a given register actually exists on your + * chip before you access it. + * + * Also, some registers are similar on different chips but have slightly + * different names. In these cases, one name is chosen to avoid extraneous + * #ifdefs. + */ +#ifdef CONFIG_PPC_85xx +struct ccsr_guts_85xx { +#else +struct ccsr_guts_86xx { +#endif __be32 porpllsr; /* 0x.0000 - POR PLL Ratio Status Register */ __be32 porbmsr; /* 0x.0004 - POR Boot Mode Status Register */ __be32 porimpscr; /* 0x.0008 - POR I/O Impedance Status and Control Register */ __be32 pordevsr; /* 0x.000c - POR I/O Device Status Register */ __be32 pordbgmsr; /* 0x.0010 - POR Debug Mode Status Register */ - u8 res1[0x20 - 0x14]; + __be32 pordevsr2; /* 0x.0014 - POR device status register 2 */ + u8 res018[0x20 - 0x18]; __be32 porcir; /* 0x.0020 - POR Configuration Information Register */ - u8 res2[0x30 - 0x24]; + u8 res024[0x30 - 0x24]; __be32 gpiocr; /* 0x.0030 - GPIO Control Register */ - u8 res3[0x40 - 0x34]; + u8 res034[0x40 - 0x34]; __be32 gpoutdr; /* 0x.0040 - General-Purpose Output Data Register */ - u8 res4[0x50 - 0x44]; + u8 res044[0x50 - 0x44]; __be32 gpindr; /* 0x.0050 - General-Purpose Input Data Register */ - u8 res5[0x60 - 0x54]; + u8 res054[0x60 - 0x54]; __be32 pmuxcr; /* 0x.0060 - Alternate Function Signal Multiplex Control */ - u8 res6[0x70 - 0x64]; + __be32 pmuxcr2; /* 0x.0064 - Alternate function signal multiplex control 2 */ + __be32 dmuxcr; /* 0x.0068 - DMA Mux Control Register */ + u8 res06c[0x70 - 0x6c]; __be32 devdisr; /* 0x.0070 - Device Disable Control */ __be32 devdisr2; /* 0x.0074 - Device Disable Control 2 */ - u8 res7[0x80 - 0x78]; + u8 res078[0x7c - 0x78]; + __be32 pmjcr; /* 0x.007c - 4 Power Management Jog Control Register */ __be32 powmgtcsr; /* 0x.0080 - Power Management Status and Control Register */ - u8 res8[0x90 - 0x84]; + __be32 pmrccr; /* 0x.0084 - Power Management Reset Counter Configuration Register */ + __be32 pmpdccr; /* 0x.0088 - Power Management Power Down Counter Configuration Register */ + __be32 pmcdr; /* 0x.008c - 4Power management clock disable register */ __be32 mcpsumr; /* 0x.0090 - Machine Check Summary Register */ __be32 rstrscr; /* 0x.0094 - Reset Request Status and Control Register */ - u8 res9[0xA0 - 0x98]; + __be32 ectrstcr; /* 0x.0098 - Exception reset control register */ + __be32 autorstsr; /* 0x.009c - Automatic reset status register */ __be32 pvr; /* 0x.00a0 - Processor Version Register */ __be32 svr; /* 0x.00a4 - System Version Register */ - u8 res10[0xB0 - 0xA8]; + u8 res0a8[0xb0 - 0xa8]; __be32 rstcr; /* 0x.00b0 - Reset Control Register */ - u8 res11[0xC0 - 0xB4]; + u8 res0b4[0xc0 - 0xb4]; +#ifdef CONFIG_PPC_85xx + __be32 iovselsr; /* 0x.00c0 - I/O voltage select status register */ +#else __be32 elbcvselcr; /* 0x.00c0 - eLBC Voltage Select Ctrl Reg */ - u8 res12[0x800 - 0xC4]; +#endif + u8 res0c4[0x224 - 0xc4]; + __be32 iodelay1; /* 0x.0224 - IO delay control register 1 */ + __be32 iodelay2; /* 0x.0228 - IO delay control register 2 */ + u8 res22c[0x800 - 0x22c]; __be32 clkdvdr; /* 0x.0800 - Clock Divide Register */ - u8 res13[0x900 - 0x804]; + u8 res804[0x900 - 0x804]; __be32 ircr; /* 0x.0900 - Infrared Control Register */ - u8 res14[0x908 - 0x904]; + u8 res904[0x908 - 0x904]; __be32 dmacr; /* 0x.0908 - DMA Control Register */ - u8 res15[0x914 - 0x90C]; + u8 res90c[0x914 - 0x90c]; __be32 elbccr; /* 0x.0914 - eLBC Control Register */ - u8 res16[0xB20 - 0x918]; + u8 res918[0xb20 - 0x918]; __be32 ddr1clkdr; /* 0x.0b20 - DDR1 Clock Disable Register */ __be32 ddr2clkdr; /* 0x.0b24 - DDR2 Clock Disable Register */ __be32 ddrclkdr; /* 0x.0b28 - DDR Clock Disable Register */ - u8 res17[0xE00 - 0xB2C]; + u8 resb2c[0xe00 - 0xb2c]; __be32 clkocr; /* 0x.0e00 - Clock Out Select Register */ - u8 res18[0xE10 - 0xE04]; + u8 rese04[0xe10 - 0xe04]; __be32 ddrdllcr; /* 0x.0e10 - DDR DLL Control Register */ - u8 res19[0xE20 - 0xE14]; + u8 rese14[0xe20 - 0xe14]; __be32 lbcdllcr; /* 0x.0e20 - LBC DLL Control Register */ - u8 res20[0xF04 - 0xE24]; + __be32 cpfor; /* 0x.0e24 - L2 charge pump fuse override register */ + u8 rese28[0xf04 - 0xe28]; __be32 srds1cr0; /* 0x.0f04 - SerDes1 Control Register 0 */ __be32 srds1cr1; /* 0x.0f08 - SerDes1 Control Register 0 */ - u8 res21[0xF40 - 0xF0C]; - __be32 srds2cr0; /* 0x.0f40 - SerDes1 Control Register 0 */ - __be32 srds2cr1; /* 0x.0f44 - SerDes1 Control Register 0 */ + u8 resf0c[0xf2c - 0xf0c]; + __be32 itcr; /* 0x.0f2c - Internal transaction control register */ + u8 resf30[0xf40 - 0xf30]; + __be32 srds2cr0; /* 0x.0f40 - SerDes2 Control Register 0 */ + __be32 srds2cr1; /* 0x.0f44 - SerDes2 Control Register 0 */ } __attribute__ ((packed)); +#ifdef CONFIG_PPC_86xx + #define CCSR_GUTS_DMACR_DEV_SSI 0 /* DMA controller/channel set to SSI */ #define CCSR_GUTS_DMACR_DEV_IR 1 /* DMA controller/channel set to IR */ @@ -93,7 +132,7 @@ struct ccsr_guts { * ch: The channel on the DMA controller (0, 1, 2, or 3) * device: The device to set as the source (CCSR_GUTS_DMACR_DEV_xx) */ -static inline void guts_set_dmacr(struct ccsr_guts __iomem *guts, +static inline void guts_set_dmacr(struct ccsr_guts_86xx __iomem *guts, unsigned int co, unsigned int ch, unsigned int device) { unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); @@ -129,7 +168,7 @@ static inline void guts_set_dmacr(struct ccsr_guts __iomem *guts, * ch: The channel on the DMA controller (0, 1, 2, or 3) * value: the new value for the bit (0 or 1) */ -static inline void guts_set_pmuxcr_dma(struct ccsr_guts __iomem *guts, +static inline void guts_set_pmuxcr_dma(struct ccsr_guts_86xx __iomem *guts, unsigned int co, unsigned int ch, unsigned int value) { if ((ch == 0) || (ch == 3)) { @@ -152,5 +191,7 @@ static inline void guts_set_pmuxcr_dma(struct ccsr_guts __iomem *guts, #define CCSR_GUTS_CLKDVDR_SSICLK_MASK 0x000000FF #define CCSR_GUTS_CLKDVDR_SSICLK(x) ((x) & CCSR_GUTS_CLKDVDR_SSICLK_MASK) -#endif /* __ASM_POWERPC_IMMAP_86XX_H__ */ -#endif /* __KERNEL__ */ +#endif + +#endif +#endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 31e8e5c..48fc0bd 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -24,8 +24,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <asm/immap_86xx.h> - #include "fsl_ssi.h" /** diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 81ab639..69aeaeb 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -14,7 +14,7 @@ #include <linux/interrupt.h> #include <linux/of_device.h> #include <sound/soc.h> -#include <asm/immap_86xx.h> +#include <asm/fsl_guts.h> #include "../codecs/cs4270.h" #include "fsl_dma.h" @@ -54,9 +54,9 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; + struct ccsr_guts_86xx __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; @@ -139,9 +139,9 @@ static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; + struct ccsr_guts_86xx __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; -- 1.7.0.1

4 3

[alsa-devel] [PATCH] ASoC: multi-component - SAMSUNG - Fix compile error
by Seungwhan Youn 23 Jul '10

23 Jul '10

This patch fixes compile error on smdk_wm9713. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/s3c24xx/smdk_wm9713.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 849aac1..90108a7 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -48,7 +48,7 @@ static struct snd_soc_dai_link smdk_dai = { .stream_name = "AC97 PCM", .platform_name = "s3c24xx-pcm-audio", .cpu_dai_name = "s3c-ac97-dai", - .codec_dai_name = "wm9713-hifi" + .codec_dai_name = "wm9713-hifi", .codec_name = "wm9713-codec", }; -- 1.6.2.5

3 2

[alsa-devel] [PATCH] driver core: remove CONFIG_SYSFS_DEPRECATED
by Greg Kroah-Hartman 23 Jul '10

23 Jul '10

This is no longer needed by any userspace tools, so it's safe to remove. Cc: Jens Axboe <axboe(a)kernel.dk> Cc: Stephen Hemminger <shemminger(a)vyatta.com> Cc: "Eric W. Biederman" <ebiederm(a)xmission.com> Cc: Kay Sievers <kay.sievers(a)vrfy.org> Cc: Alan Stern <stern(a)rowland.harvard.edu> Cc: "James E.J. Bottomley" <James.Bottomley(a)suse.de> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: Alexey Kuznetsov <kuznet(a)ms2.inr.ac.ru> Cc: Randy Dunlap <randy.dunlap(a)oracle.com> Cc: Tejun Heo <tj(a)kernel.org> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Ingo Molnar <mingo(a)elte.hu> Cc: Peter Zijlstra <a.p.zijlstra(a)chello.nl> Cc: David Howells <dhowells(a)redhat.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)suse.de> --- block/genhd.c | 5 - drivers/base/bus.c | 22 ------ drivers/base/class.c | 25 ------- drivers/base/core.c | 164 ----------------------------------------------- drivers/scsi/hosts.c | 2 drivers/scsi/scsi_scan.c | 2 fs/partitions/check.c | 6 - include/sound/core.h | 6 - init/Kconfig | 36 ---------- sound/core/init.c | 11 --- 10 files changed, 7 insertions(+), 272 deletions(-) --- a/block/genhd.c +++ b/block/genhd.c @@ -22,9 +22,7 @@ #include "blk.h" static DEFINE_MUTEX(block_class_lock); -#ifndef CONFIG_SYSFS_DEPRECATED struct kobject *block_depr; -#endif /* for extended dynamic devt allocation, currently only one major is used */ #define MAX_EXT_DEVT (1 << MINORBITS) @@ -803,10 +801,9 @@ static int __init genhd_device_init(void register_blkdev(BLOCK_EXT_MAJOR, "blkext"); -#ifndef CONFIG_SYSFS_DEPRECATED /* create top-level block dir */ block_depr = kobject_create_and_add("block", NULL); -#endif + return 0; } --- a/drivers/base/bus.c +++ b/drivers/base/bus.c @@ -440,22 +440,6 @@ static void device_remove_attrs(struct b } } -#ifdef CONFIG_SYSFS_DEPRECATED -static int make_deprecated_bus_links(struct device *dev) -{ - return sysfs_create_link(&dev->kobj, - &dev->bus->p->subsys.kobj, "bus"); -} - -static void remove_deprecated_bus_links(struct device *dev) -{ - sysfs_remove_link(&dev->kobj, "bus"); -} -#else -static inline int make_deprecated_bus_links(struct device *dev) { return 0; } -static inline void remove_deprecated_bus_links(struct device *dev) { } -#endif - /** * bus_add_device - add device to bus * @dev: device being added @@ -482,15 +466,10 @@ int bus_add_device(struct device *dev) &dev->bus->p->subsys.kobj, "subsystem"); if (error) goto out_subsys; - error = make_deprecated_bus_links(dev); - if (error) - goto out_deprecated; klist_add_tail(&dev->p->knode_bus, &bus->p->klist_devices); } return 0; -out_deprecated: - sysfs_remove_link(&dev->kobj, "subsystem"); out_subsys: sysfs_remove_link(&bus->p->devices_kset->kobj, dev_name(dev)); out_id: @@ -530,7 +509,6 @@ void bus_remove_device(struct device *de { if (dev->bus) { sysfs_remove_link(&dev->kobj, "subsystem"); - remove_deprecated_bus_links(dev); sysfs_remove_link(&dev->bus->p->devices_kset->kobj, dev_name(dev)); device_remove_attrs(dev->bus, dev); --- a/drivers/base/class.c +++ b/drivers/base/class.c @@ -184,13 +184,7 @@ int __class_register(struct class *cls, if (!cls->dev_kobj) cls->dev_kobj = sysfs_dev_char_kobj; -#if defined(CONFIG_SYSFS_DEPRECATED) && defined(CONFIG_BLOCK) - /* let the block class directory show up in the root of sysfs */ - if (cls != &block_class) - cp->class_subsys.kobj.kset = class_kset; -#else cp->class_subsys.kobj.kset = class_kset; -#endif cp->class_subsys.kobj.ktype = &class_ktype; cp->class = cls; cls->p = cp; @@ -276,25 +270,6 @@ void class_destroy(struct class *cls) class_unregister(cls); } -#ifdef CONFIG_SYSFS_DEPRECATED -char *make_class_name(const char *name, struct kobject *kobj) -{ - char *class_name; - int size; - - size = strlen(name) + strlen(kobject_name(kobj)) + 2; - - class_name = kmalloc(size, GFP_KERNEL); - if (!class_name) - return NULL; - - strcpy(class_name, name); - strcat(class_name, ":"); - strcat(class_name, kobject_name(kobj)); - return class_name; -} -#endif - /** * class_dev_iter_init - initialize class device iterator * @iter: class iterator to initialize --- a/drivers/base/core.c +++ b/drivers/base/core.c @@ -203,37 +203,6 @@ static int dev_uevent(struct kset *kset, if (dev->driver) add_uevent_var(env, "DRIVER=%s", dev->driver->name); -#ifdef CONFIG_SYSFS_DEPRECATED - if (dev->class) { - struct device *parent = dev->parent; - - /* find first bus device in parent chain */ - while (parent && !parent->bus) - parent = parent->parent; - if (parent && parent->bus) { - const char *path; - - path = kobject_get_path(&parent->kobj, GFP_KERNEL); - if (path) { - add_uevent_var(env, "PHYSDEVPATH=%s", path); - kfree(path); - } - - add_uevent_var(env, "PHYSDEVBUS=%s", parent->bus->name); - - if (parent->driver) - add_uevent_var(env, "PHYSDEVDRIVER=%s", - parent->driver->name); - } - } else if (dev->bus) { - add_uevent_var(env, "PHYSDEVBUS=%s", dev->bus->name); - - if (dev->driver) - add_uevent_var(env, "PHYSDEVDRIVER=%s", - dev->driver->name); - } -#endif - /* have the bus specific function add its stuff */ if (dev->bus && dev->bus->uevent) { retval = dev->bus->uevent(dev, env); @@ -578,24 +547,6 @@ void device_initialize(struct device *de set_dev_node(dev, -1); } -#ifdef CONFIG_SYSFS_DEPRECATED -static struct kobject *get_device_parent(struct device *dev, - struct device *parent) -{ - /* class devices without a parent live in /sys/class/<classname>/ */ - if (dev->class && (!parent || parent->class != dev->class)) - return &dev->class->p->class_subsys.kobj; - /* all other devices keep their parent */ - else if (parent) - return &parent->kobj; - - return NULL; -} - -static inline void cleanup_device_parent(struct device *dev) {} -static inline void cleanup_glue_dir(struct device *dev, - struct kobject *glue_dir) {} -#else static struct kobject *virtual_device_parent(struct device *dev) { static struct kobject *virtual_dir = NULL; @@ -719,7 +670,6 @@ static void cleanup_device_parent(struct { cleanup_glue_dir(dev, dev->kobj.parent); } -#endif static void setup_parent(struct device *dev, struct device *parent) { @@ -742,53 +692,6 @@ static int device_add_class_symlinks(str if (error) goto out; -#ifdef CONFIG_SYSFS_DEPRECATED - /* stacked class devices need a symlink in the class directory */ - if (dev->kobj.parent != &dev->class->p->class_subsys.kobj && - device_is_not_partition(dev)) { - error = sysfs_create_link(&dev->class->p->class_subsys.kobj, - &dev->kobj, dev_name(dev)); - if (error) - goto out_subsys; - } - - if (dev->parent && device_is_not_partition(dev)) { - struct device *parent = dev->parent; - char *class_name; - - /* - * stacked class devices have the 'device' link - * pointing to the bus device instead of the parent - */ - while (parent->class && !parent->bus && parent->parent) - parent = parent->parent; - - error = sysfs_create_link(&dev->kobj, - &parent->kobj, - "device"); - if (error) - goto out_busid; - - class_name = make_class_name(dev->class->name, - &dev->kobj); - if (class_name) - error = sysfs_create_link(&dev->parent->kobj, - &dev->kobj, class_name); - kfree(class_name); - if (error) - goto out_device; - } - return 0; - -out_device: - if (dev->parent && device_is_not_partition(dev)) - sysfs_remove_link(&dev->kobj, "device"); -out_busid: - if (dev->kobj.parent != &dev->class->p->class_subsys.kobj && - device_is_not_partition(dev)) - sysfs_delete_link(&dev->class->p->class_subsys.kobj, &dev->kobj, - dev_name(dev)); -#else /* link in the class directory pointing to the device */ error = sysfs_create_link(&dev->class->p->class_subsys.kobj, &dev->kobj, dev_name(dev)); @@ -805,7 +708,6 @@ out_busid: out_busid: sysfs_delete_link(&dev->class->p->class_subsys.kobj, &dev->kobj, dev_name(dev)); -#endif out_subsys: sysfs_remove_link(&dev->kobj, "subsystem"); @@ -818,29 +720,10 @@ static void device_remove_class_symlinks if (!dev->class) return; -#ifdef CONFIG_SYSFS_DEPRECATED - if (dev->parent && device_is_not_partition(dev)) { - char *class_name; - - class_name = make_class_name(dev->class->name, &dev->kobj); - if (class_name) { - sysfs_remove_link(&dev->parent->kobj, class_name); - kfree(class_name); - } - sysfs_remove_link(&dev->kobj, "device"); - } - - if (dev->kobj.parent != &dev->class->p->class_subsys.kobj && - device_is_not_partition(dev)) - sysfs_delete_link(&dev->class->p->class_subsys.kobj, &dev->kobj, - dev_name(dev)); -#else if (dev->parent && device_is_not_partition(dev)) sysfs_remove_link(&dev->kobj, "device"); sysfs_delete_link(&dev->class->p->class_subsys.kobj, &dev->kobj, dev_name(dev)); -#endif - sysfs_remove_link(&dev->kobj, "subsystem"); } @@ -1613,41 +1496,23 @@ int device_rename(struct device *dev, ch pr_debug("device: '%s': %s: renaming to '%s'\n", dev_name(dev), __func__, new_name); -#ifdef CONFIG_SYSFS_DEPRECATED - if ((dev->class) && (dev->parent)) - old_class_name = make_class_name(dev->class->name, &dev->kobj); -#endif - old_device_name = kstrdup(dev_name(dev), GFP_KERNEL); if (!old_device_name) { error = -ENOMEM; goto out; } -#ifndef CONFIG_SYSFS_DEPRECATED if (dev->class) { error = sysfs_rename_link(&dev->class->p->class_subsys.kobj, &dev->kobj, old_device_name, new_name); if (error) goto out; } -#endif + error = kobject_rename(&dev->kobj, new_name); if (error) goto out; -#ifdef CONFIG_SYSFS_DEPRECATED - if (old_class_name) { - new_class_name = make_class_name(dev->class->name, &dev->kobj); - if (new_class_name) { - error = sysfs_rename_link(&dev->parent->kobj, - &dev->kobj, - old_class_name, - new_class_name); - } - } -#endif - out: put_device(dev); @@ -1664,40 +1529,13 @@ static int device_move_class_links(struc struct device *new_parent) { int error = 0; -#ifdef CONFIG_SYSFS_DEPRECATED - char *class_name; - class_name = make_class_name(dev->class->name, &dev->kobj); - if (!class_name) { - error = -ENOMEM; - goto out; - } - if (old_parent) { - sysfs_remove_link(&dev->kobj, "device"); - sysfs_remove_link(&old_parent->kobj, class_name); - } - if (new_parent) { - error = sysfs_create_link(&dev->kobj, &new_parent->kobj, - "device"); - if (error) - goto out; - error = sysfs_create_link(&new_parent->kobj, &dev->kobj, - class_name); - if (error) - sysfs_remove_link(&dev->kobj, "device"); - } else - error = 0; -out: - kfree(class_name); - return error; -#else if (old_parent) sysfs_remove_link(&dev->kobj, "device"); if (new_parent) error = sysfs_create_link(&dev->kobj, &new_parent->kobj, "device"); return error; -#endif } /** --- a/drivers/scsi/hosts.c +++ b/drivers/scsi/hosts.c @@ -406,9 +406,7 @@ struct Scsi_Host *scsi_host_alloc(struct device_initialize(&shost->shost_gendev); dev_set_name(&shost->shost_gendev, "host%d", shost->host_no); -#ifndef CONFIG_SYSFS_DEPRECATED shost->shost_gendev.bus = &scsi_bus_type; -#endif shost->shost_gendev.type = &scsi_host_type; device_initialize(&shost->shost_dev); --- a/drivers/scsi/scsi_scan.c +++ b/drivers/scsi/scsi_scan.c @@ -417,9 +417,7 @@ static struct scsi_target *scsi_alloc_ta starget->reap_ref = 1; dev->parent = get_device(parent); dev_set_name(dev, "target%d:%d:%d", shost->host_no, channel, id); -#ifndef CONFIG_SYSFS_DEPRECATED dev->bus = &scsi_bus_type; -#endif dev->type = &scsi_target_type; starget->id = id; starget->channel = channel; --- a/fs/partitions/check.c +++ b/fs/partitions/check.c @@ -500,14 +500,14 @@ void register_disk(struct gendisk *disk) if (device_add(ddev)) return; -#ifndef CONFIG_SYSFS_DEPRECATED + err = sysfs_create_link(block_depr, &ddev->kobj, kobject_name(&ddev->kobj)); if (err) { device_del(ddev); return; } -#endif + disk->part0.holder_dir = kobject_create_and_add("holders", &ddev->kobj); disk->slave_dir = kobject_create_and_add("slaves", &ddev->kobj); @@ -724,8 +724,6 @@ void del_gendisk(struct gendisk *disk) kobject_put(disk->part0.holder_dir); kobject_put(disk->slave_dir); disk->driverfs_dev = NULL; -#ifndef CONFIG_SYSFS_DEPRECATED sysfs_remove_link(block_depr, dev_name(disk_to_dev(disk))); -#endif device_del(disk_to_dev(disk)); } --- a/include/sound/core.h +++ b/include/sound/core.h @@ -133,9 +133,7 @@ struct snd_card { int free_on_last_close; /* free in context of file_release */ wait_queue_head_t shutdown_sleep; struct device *dev; /* device assigned to this card */ -#ifndef CONFIG_SYSFS_DEPRECATED struct device *card_dev; /* cardX object for sysfs */ -#endif #ifdef CONFIG_PM unsigned int power_state; /* power state */ @@ -196,11 +194,7 @@ struct snd_minor { /* return a device pointer linked to each sound device as a parent */ static inline struct device *snd_card_get_device_link(struct snd_card *card) { -#ifdef CONFIG_SYSFS_DEPRECATED - return card ? card->dev : NULL; -#else return card ? card->card_dev : NULL; -#endif } /* sound.c */ --- a/init/Kconfig +++ b/init/Kconfig @@ -651,42 +651,6 @@ endif # CGROUPS config MM_OWNER bool -config SYSFS_DEPRECATED - bool - -config SYSFS_DEPRECATED_V2 - bool "enable deprecated sysfs features to support old userspace tools" - depends on SYSFS - default n - select SYSFS_DEPRECATED - help - This option switches the layout of sysfs to the deprecated - version. Do not use it on recent distributions. - - The current sysfs layout features a unified device tree at - /sys/devices/, which is able to express a hierarchy between - class devices. If the deprecated option is set to Y, the - unified device tree is split into a bus device tree at - /sys/devices/ and several individual class device trees at - /sys/class/. The class and bus devices will be connected by - "<subsystem>:<name>" and the "device" links. The "block" - class devices, will not show up in /sys/class/block/. Some - subsystems will suppress the creation of some devices which - depend on the unified device tree. - - This option is not a pure compatibility option that can - be safely enabled on newer distributions. It will change the - layout of sysfs to the non-extensible deprecated version, - and disable some features, which can not be exported without - confusing older userspace tools. Since 2007/2008 all major - distributions do not enable this option, and ship no tools which - depend on the deprecated layout or this option. - - If you are using a new kernel on an older distribution, or use - older userspace tools, you might need to say Y here. Do not say Y, - if the original kernel, that came with your distribution, has - this option set to N. - config RELAY bool "Kernel->user space relay support (formerly relayfs)" help --- a/sound/core/init.c +++ b/sound/core/init.c @@ -395,12 +395,10 @@ int snd_card_disconnect(struct snd_card snd_printk(KERN_ERR "not all devices for card %i can be disconnected\n", card->number); snd_info_card_disconnect(card); -#ifndef CONFIG_SYSFS_DEPRECATED if (card->card_dev) { device_unregister(card->card_dev); card->card_dev = NULL; } -#endif #ifdef CONFIG_PM wake_up(&card->power_sleep); #endif @@ -573,7 +571,6 @@ void snd_card_set_id(struct snd_card *ca } EXPORT_SYMBOL(snd_card_set_id); -#ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) @@ -630,7 +627,6 @@ card_number_show_attr(struct device *dev static struct device_attribute card_number_attrs = __ATTR(number, S_IRUGO, card_number_show_attr, NULL); -#endif /* CONFIG_SYSFS_DEPRECATED */ /** * snd_card_register - register the soundcard @@ -649,7 +645,7 @@ int snd_card_register(struct snd_card *c if (snd_BUG_ON(!card)) return -EINVAL; -#ifndef CONFIG_SYSFS_DEPRECATED + if (!card->card_dev) { card->card_dev = device_create(sound_class, card->dev, MKDEV(0, 0), card, @@ -657,7 +653,7 @@ int snd_card_register(struct snd_card *c if (IS_ERR(card->card_dev)) card->card_dev = NULL; } -#endif + if ((err = snd_device_register_all(card)) < 0) return err; mutex_lock(&snd_card_mutex); @@ -674,7 +670,6 @@ int snd_card_register(struct snd_card *c if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER); #endif -#ifndef CONFIG_SYSFS_DEPRECATED if (card->card_dev) { err = device_create_file(card->card_dev, &card_id_attrs); if (err < 0) @@ -683,7 +678,7 @@ int snd_card_register(struct snd_card *c if (err < 0) return err; } -#endif + return 0; }

9 20

[alsa-devel] Which project to choose?
by Chris 23 Jul '10

23 Jul '10

I want to submit a bug report on the below. This issue was noted in Mandriva 2010 and is still present in 2010.1 pulseaudio[24553]: alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -161920 bytes (-917 ms). localhost pulseaudio[24553]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_ens1371'. Please report this issue to the ALSA developers. localhost pulseaudio[24553]: alsa-util.c: snd_pcm_dump(): localhost pulseaudio[24553]: alsa-util.c: Hardware PCM card 0 'Ensoniq AudioPCI' device 0 subdevice 0 localhost pulseaudio[24553]: alsa-util.c: Its setup is: localhost pulseaudio[24553]: alsa-util.c: stream : PLAYBACK localhost pulseaudio[24553]: alsa-util.c: access : MMAP_INTERLEAVED localhost pulseaudio[24553]: alsa-util.c: format : S16_LE localhost pulseaudio[24553]: alsa-util.c: subformat : STD localhost pulseaudio[24553]: alsa-util.c: channels : 2 localhost pulseaudio[24553]: alsa-util.c: rate : 44100 localhost pulseaudio[24553]: alsa-util.c: exact rate : 44101 (1445100000/32768) localhost pulseaudio[24553]: alsa-util.c: msbits : 16 localhost pulseaudio[24553]: alsa-util.c: buffer_size : 4408 localhost pulseaudio[24553]: alsa-util.c: period_size : 1102 localhost pulseaudio[24553]: alsa-util.c: period_time : 24988 localhost pulseaudio[24553]: alsa-util.c: tstamp_mode : ENABLE localhost pulseaudio[24553]: alsa-util.c: period_step : 1 localhost pulseaudio[24553]: alsa-util.c: avail_min : 1102 localhost pulseaudio[24553]: alsa-util.c: period_event : 1 localhost pulseaudio[24553]: alsa-util.c: start_threshold : -1 localhost pulseaudio[24553]: alsa-util.c: stop_threshold : 1155530752 localhost pulseaudio[24553]: alsa-util.c: silence_threshold: 0 localhost pulseaudio[24553]: alsa-util.c: silence_size : 0 localhost pulseaudio[24553]: alsa-util.c: boundary : 1155530752 localhost pulseaudio[24553]: alsa-util.c: appl_ptr : 735801008 localhost pulseaudio[24553]: alsa-util.c: hw_ptr : 735846184 localhost pulseaudio[24553]: ratelimit.c: 280 events suppressed localhost pulseaudio[24553]: asyncq.c: q overrun, queuing locally Because of the constant disk access when this is happening my system becomes almost unusable. -- Chris KeyID 0xE372A7DA98E6705C

4 15

[alsa-devel] open sound device from asound.conf
by Peter Lukac 23 Jul '10

23 Jul '10

hello all maybe trivial question... I want open sound device from /etc/asound.conf with API snd_pcm_open() but unsuccessfull. As name i'm used: "plughw:0,0" "plughw:0,1" "plughw:0,2" "plughw:1,0" "plughw:1,1" "plughw:1,2" ... i'm try alternative name with "hw:X,Y" "hw:0,0" "hw:0,1" ... but in this method i'm open only HW device. I'm get follow error: ALSA lib pcm_hw.c:1429:(_snd_pcm_hw_open) Invalid value for card how can i open my "virtual" device in asound.conf here is asound.conf: pcm.!default { type plug slave.pcm "output" } pcm.output { type dmix ipc_key 1024 slave { pcm "hw:0,0" channels 2 rate 44100 format S16_LE } } pcm.dsp0 { type plug slave.pcm "output" } ...exists some universal method how can i open all HW device and "virtual" devices? thanks

3 4

[alsa-devel] [PATCH v2] ASoC: da7210: Add HeadPhone Playback Volume control
by Kuninori Morimoto 23 Jul '10

23 Jul '10

HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 o remove default HeadPhone Playback Volume sound/soc/codecs/da7210.c | 31 +++++++++++++++++++++++++++---- 1 files changed, 27 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a83aa18..3e42d1e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -23,6 +23,7 @@ #include <sound/pcm_params.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "da7210.h" @@ -136,6 +137,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -218,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -647,6 +667,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: -- 1.7.0.4

3 2

[alsa-devel] [PATCH] ASoC: multi-component - Fix Core AC97 set device name
by Seungwhan Youn 23 Jul '10

23 Jul '10

AC97 driver needs to make kobject with id which from codec. But in multi-component, this id was removed, so this patch fix this same as 'sound/pci/ac97/ac97_codec.c'. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 23dad36..d074b2b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -310,7 +310,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", - codec->id, 0, codec->name); + codec->ac97->bus->card->number, 0, codec->name); err = device_register(&codec->ac97->dev); if (err < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); -- 1.6.2.5

2 1

Re: [alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Viliam Kubis 22 Jul '10

22 Jul '10

I have tried the patch on clean (unpatched) patch_via.c. dmesg prints out ***PATCH 08/07/10 as expected, but the surround does not work. Surround and cen/lfe controls are also not created for alsamixer without other patches. What does work is specifying -Dsurround40 for speaker-test but there is no sound from rear speakers. If you want, I can post the whole patched patch_via.c somewhere or we can work on new patches starting with this one. On Thu, 08 Jul 2010 08:33:34 +0200, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> Viliam Kubis wrote: >>> Hello, for quite some time the "Smart 5.1" function in alsa is not >>> working >>> properly when motherboard has only 3 audio jacks. >>> >>> speaker-test -c 4 -t wav -Dsurround40 >>> >>> Channels count (4) not available for playbacks: Invalid argument >> >> Please try the patch below. >> >> (This function uses 0x12 and 0x13 as the NIDs for the center/LFE and >> side DACs. Shouldn't these be 0x24 and 0x25, as per the datasheet?) >> >> --- a/sound/pci/hda/patch_via.c >> +++ b/sound/pci/hda/patch_via.c >> @@ -2265,6 +2265,13 @@ static int vt1708_auto_fill_dac_nids(str >> } >> } >> + /* for Smart 5.1, line/mic inputs double as output pins */ >> + if (cfg->line_outs == 1) { >> + spec->multiout.num_dacs = 3; >> + spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; >> + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x12; >> + } >> + >> return 0; >> } >> > -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

3 3

[alsa-devel] [PATCH] powerpc: rename immap_86xx.h to fsl_guts.h, and add 85xx support
by Timur Tabi 22 Jul '10

22 Jul '10

The immap_86xx.h header file only defines one data structure: the "global utilities" register set found on Freescale PowerPC SOCs. Rename this file to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS" register set on 85xx chips. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Liam, This patch is a powerpc-patch, but it's much simpler if you apply it to multi-component, since only the SSI audio drivers are affected. Kumar Gala will ack this patch. .../include/asm/{immap_86xx.h => fsl_guts.h} | 106 +++++++++++++------- sound/soc/fsl/fsl_ssi.c | 2 - sound/soc/fsl/mpc8610_hpcd.c | 10 +- 3 files changed, 76 insertions(+), 42 deletions(-) rename arch/powerpc/include/asm/{immap_86xx.h => fsl_guts.h} (67%) diff --git a/arch/powerpc/include/asm/immap_86xx.h b/arch/powerpc/include/asm/fsl_guts.h similarity index 67% rename from arch/powerpc/include/asm/immap_86xx.h rename to arch/powerpc/include/asm/fsl_guts.h index 0f165e5..4827900 100644 --- a/arch/powerpc/include/asm/immap_86xx.h +++ b/arch/powerpc/include/asm/fsl_guts.h @@ -1,5 +1,5 @@ /** - * MPC86xx Internal Memory Map + * Freecale 85xx and 86xx Global Utilties register set * * Authors: Jeff Brown * Timur Tabi <timur(a)freescale.com> @@ -10,73 +10,107 @@ * 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 header file defines structures for various 86xx SOC devices that are - * used by multiple source files. */ -#ifndef __ASM_POWERPC_IMMAP_86XX_H__ -#define __ASM_POWERPC_IMMAP_86XX_H__ +#ifndef __ASM_POWERPC_FSL_GUTS_H__ +#define __ASM_POWERPC_FSL_GUTS_H__ #ifdef __KERNEL__ -/* Global Utility Registers */ -struct ccsr_guts { +#if !defined(CONFIG_PPC_85xx) && !defined(CONFIG_PPC_86xx) +#error Only 85xx and 86xx SOCs are supported +#endif + +/** + * Global Utility Registers. + * + * Not all registers defined in this structure are available on all chips, so + * you are expected to know whether a given register actually exists on your + * chip before you access it. + * + * Also, some registers are similar on different chips but have slightly + * different names. In these cases, one name is chosen to avoid extraneous + * #ifdefs. + */ +#ifdef CONFIG_PPC_85xx +struct ccsr_guts_85xx { +#else +struct ccsr_guts_86xx { +#endif __be32 porpllsr; /* 0x.0000 - POR PLL Ratio Status Register */ __be32 porbmsr; /* 0x.0004 - POR Boot Mode Status Register */ __be32 porimpscr; /* 0x.0008 - POR I/O Impedance Status and Control Register */ __be32 pordevsr; /* 0x.000c - POR I/O Device Status Register */ __be32 pordbgmsr; /* 0x.0010 - POR Debug Mode Status Register */ - u8 res1[0x20 - 0x14]; + __be32 pordevsr2; /* 0x.0014 - POR device status register 2 */ + u8 res018[0x20 - 0x18]; __be32 porcir; /* 0x.0020 - POR Configuration Information Register */ - u8 res2[0x30 - 0x24]; + u8 res024[0x30 - 0x24]; __be32 gpiocr; /* 0x.0030 - GPIO Control Register */ - u8 res3[0x40 - 0x34]; + u8 res034[0x40 - 0x34]; __be32 gpoutdr; /* 0x.0040 - General-Purpose Output Data Register */ - u8 res4[0x50 - 0x44]; + u8 res044[0x50 - 0x44]; __be32 gpindr; /* 0x.0050 - General-Purpose Input Data Register */ - u8 res5[0x60 - 0x54]; + u8 res054[0x60 - 0x54]; __be32 pmuxcr; /* 0x.0060 - Alternate Function Signal Multiplex Control */ - u8 res6[0x70 - 0x64]; + __be32 pmuxcr2; /* 0x.0064 - Alternate function signal multiplex control 2 */ + __be32 dmuxcr; /* 0x.0068 - DMA Mux Control Register */ + u8 res06c[0x70 - 0x6c]; __be32 devdisr; /* 0x.0070 - Device Disable Control */ __be32 devdisr2; /* 0x.0074 - Device Disable Control 2 */ - u8 res7[0x80 - 0x78]; + u8 res078[0x7c - 0x78]; + __be32 pmjcr; /* 0x.007c - 4 Power Management Jog Control Register */ __be32 powmgtcsr; /* 0x.0080 - Power Management Status and Control Register */ - u8 res8[0x90 - 0x84]; + __be32 pmrccr; /* 0x.0084 - Power Management Reset Counter Configuration Register */ + __be32 pmpdccr; /* 0x.0088 - Power Management Power Down Counter Configuration Register */ + __be32 pmcdr; /* 0x.008c - 4Power management clock disable register */ __be32 mcpsumr; /* 0x.0090 - Machine Check Summary Register */ __be32 rstrscr; /* 0x.0094 - Reset Request Status and Control Register */ - u8 res9[0xA0 - 0x98]; + __be32 ectrstcr; /* 0x.0098 - Exception reset control register */ + __be32 autorstsr; /* 0x.009c - Automatic reset status register */ __be32 pvr; /* 0x.00a0 - Processor Version Register */ __be32 svr; /* 0x.00a4 - System Version Register */ - u8 res10[0xB0 - 0xA8]; + u8 res0a8[0xb0 - 0xa8]; __be32 rstcr; /* 0x.00b0 - Reset Control Register */ - u8 res11[0xC0 - 0xB4]; + u8 res0b4[0xc0 - 0xb4]; +#ifdef CONFIG_PPC_85xx + __be32 iovselsr; /* 0x.00c0 - I/O voltage select status register */ +#else __be32 elbcvselcr; /* 0x.00c0 - eLBC Voltage Select Ctrl Reg */ - u8 res12[0x800 - 0xC4]; +#endif + u8 res0c4[0x224 - 0xc4]; + __be32 iodelay1; /* 0x.0224 - IO delay control register 1 */ + __be32 iodelay2; /* 0x.0228 - IO delay control register 2 */ + u8 res22c[0x800 - 0x22c]; __be32 clkdvdr; /* 0x.0800 - Clock Divide Register */ - u8 res13[0x900 - 0x804]; + u8 res804[0x900 - 0x804]; __be32 ircr; /* 0x.0900 - Infrared Control Register */ - u8 res14[0x908 - 0x904]; + u8 res904[0x908 - 0x904]; __be32 dmacr; /* 0x.0908 - DMA Control Register */ - u8 res15[0x914 - 0x90C]; + u8 res90c[0x914 - 0x90c]; __be32 elbccr; /* 0x.0914 - eLBC Control Register */ - u8 res16[0xB20 - 0x918]; + u8 res918[0xb20 - 0x918]; __be32 ddr1clkdr; /* 0x.0b20 - DDR1 Clock Disable Register */ __be32 ddr2clkdr; /* 0x.0b24 - DDR2 Clock Disable Register */ __be32 ddrclkdr; /* 0x.0b28 - DDR Clock Disable Register */ - u8 res17[0xE00 - 0xB2C]; + u8 resb2c[0xe00 - 0xb2c]; __be32 clkocr; /* 0x.0e00 - Clock Out Select Register */ - u8 res18[0xE10 - 0xE04]; + u8 rese04[0xe10 - 0xe04]; __be32 ddrdllcr; /* 0x.0e10 - DDR DLL Control Register */ - u8 res19[0xE20 - 0xE14]; + u8 rese14[0xe20 - 0xe14]; __be32 lbcdllcr; /* 0x.0e20 - LBC DLL Control Register */ - u8 res20[0xF04 - 0xE24]; + __be32 cpfor; /* 0x.0e24 - L2 charge pump fuse override register */ + u8 rese28[0xf04 - 0xe28]; __be32 srds1cr0; /* 0x.0f04 - SerDes1 Control Register 0 */ __be32 srds1cr1; /* 0x.0f08 - SerDes1 Control Register 0 */ - u8 res21[0xF40 - 0xF0C]; - __be32 srds2cr0; /* 0x.0f40 - SerDes1 Control Register 0 */ - __be32 srds2cr1; /* 0x.0f44 - SerDes1 Control Register 0 */ + u8 resf0c[0xf2c - 0xf0c]; + __be32 itcr; /* 0x.0f2c - Internal transaction control register */ + u8 resf30[0xf40 - 0xf30]; + __be32 srds2cr0; /* 0x.0f40 - SerDes2 Control Register 0 */ + __be32 srds2cr1; /* 0x.0f44 - SerDes2 Control Register 0 */ } __attribute__ ((packed)); +#ifdef CONFIG_PPC_86xx + #define CCSR_GUTS_DMACR_DEV_SSI 0 /* DMA controller/channel set to SSI */ #define CCSR_GUTS_DMACR_DEV_IR 1 /* DMA controller/channel set to IR */ @@ -93,7 +127,7 @@ struct ccsr_guts { * ch: The channel on the DMA controller (0, 1, 2, or 3) * device: The device to set as the source (CCSR_GUTS_DMACR_DEV_xx) */ -static inline void guts_set_dmacr(struct ccsr_guts __iomem *guts, +static inline void guts_set_dmacr(struct ccsr_guts_86xx __iomem *guts, unsigned int co, unsigned int ch, unsigned int device) { unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); @@ -129,7 +163,7 @@ static inline void guts_set_dmacr(struct ccsr_guts __iomem *guts, * ch: The channel on the DMA controller (0, 1, 2, or 3) * value: the new value for the bit (0 or 1) */ -static inline void guts_set_pmuxcr_dma(struct ccsr_guts __iomem *guts, +static inline void guts_set_pmuxcr_dma(struct ccsr_guts_86xx __iomem *guts, unsigned int co, unsigned int ch, unsigned int value) { if ((ch == 0) || (ch == 3)) { @@ -152,5 +186,7 @@ static inline void guts_set_pmuxcr_dma(struct ccsr_guts __iomem *guts, #define CCSR_GUTS_CLKDVDR_SSICLK_MASK 0x000000FF #define CCSR_GUTS_CLKDVDR_SSICLK(x) ((x) & CCSR_GUTS_CLKDVDR_SSICLK_MASK) -#endif /* __ASM_POWERPC_IMMAP_86XX_H__ */ -#endif /* __KERNEL__ */ +#endif + +#endif +#endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 31e8e5c..48fc0bd 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -24,8 +24,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <asm/immap_86xx.h> - #include "fsl_ssi.h" /** diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 81ab639..69aeaeb 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -14,7 +14,7 @@ #include <linux/interrupt.h> #include <linux/of_device.h> #include <sound/soc.h> -#include <asm/immap_86xx.h> +#include <asm/fsl_guts.h> #include "../codecs/cs4270.h" #include "fsl_dma.h" @@ -54,9 +54,9 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; + struct ccsr_guts_86xx __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; @@ -139,9 +139,9 @@ static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; + struct ccsr_guts_86xx __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; -- 1.7.0.1

3 3

[alsa-devel] [PATCH v2 0/1] *** SUBJECT HERE ***
by Matti J. Aaltonen 22 Jul '10

22 Jul '10

Hello. And thank you for all the comments. Comments to comments: >> +#undef DEBUG > > This shouldn't be here. Removed. >> + wl1273->mode = ucontrol->value.integer.value[0]; > > You're not doing any validation of the supplied value here. Validation added. > Don't put your enums in arrays, it just makes things harder to maintain > since you have to reference an index into an array rather than a named > value. Yes using one-element arrays is kind of meaningless, so arrays removed. > My major comment about this driver is that it'd probably make sense to > redo it on top of Liam's multi-component branch, though it shouldn't be > a pressing concern for merge. I didn't do this yet. But could you give me the git URL and then I'll resend? > Wouldn't it be nicer to do this stuff with the ALSA constraint APIs > rather than futzing with the constant data? I principle I of course agree. But it seems - also discussed with the local ALSA specialist - that using static constraints does not work here. HALF_DUPLEX is not one of the SNDRV_PCM_HW_PARAM_'s... or something like that. As a ALSA non-specialist I'd be willing to this leave as it is... >> +static int wl1273_set_dai_fmt(struct snd_soc_dai *codec_dai, >> + unsigned int fmt) >> +{ >> + return 0; >> +} > >Remove unused functions. I tried leaving this out but then the codec stopped working. I guess I should mention that my test environment is 2.6.32. I'm only able to test that the codec compiles under 2.6.35. Anyway I left the function in. >> +enum wl1273_mode wl1273_get_codec_mode(struct snd_soc_codec *codec) >> +{ >> + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); >> + return wl1273->mode; >> +} >> +EXPORT_SYMBOL_GPL(wl1273_get_codec_mode); > > Why is this being exported? This is because the soc_card driver needs to know the codec mode and accessing the internals of the codec struct is ugly. >> +static int wl1273_soc_suspend(struct platform_device *pdev, pm_message_t state) >> +{ >> + return 0; >> +} >> + >> +static int wl1273_soc_resume(struct platform_device *pdev) >> +{ >> + return 0; >> +} > > Remove unused functions. Dropping these caused no problems, so removed... Cheers, Matti A. Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec. sound/soc/codecs/wl1273.c | 586 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 42 ++++ 2 files changed, 628 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h

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[alsa-devel] ASoC: multi-component updates from RFC
by Liam Girdwood 22 Jul '10

22 Jul '10

I've just pushed some updates to multi-component that came out of the RFC discussion. The main update is the use of dev_name (or DAI name for devices with multiple DAIs) to identify ASoC components. This replaces the current method of using a struct codec|dai|platform pointer and ID to define a DAI link. e.g. we can now define a DAI link as follows :- static struct snd_soc_dai_link sdp3430_dai[] = { { .name = "TWL4030 I2S", .stream_name = "TWL4030 Audio", .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }, { .name = "TWL4030 PCM", .stream_name = "TWL4030 Voice", .cpu_dai_name = "omap-mcbsp-dai.2", .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", .init = sdp3430_twl4030_voice_init, .ops = &sdp3430_voice_ops, }, }; This now means we now no longer have to export all the CODEC | DAI | platform structure symbols around and can keep them static. We also manage to delete a few headers as a result of this change. Since we are now using dev_name or DAI name (for single devices with multiple DAIs) I've taken the opportunity to standardise the ASoC component naming. It's currently very ad-hoc and platform specific, so I've now added some useful prefixes to help. e.g. So we now have :- "-codec" prefix for codecs, e.g. "wm8731-codec" "-dai" prefix for CPU DAI's e.g. "pxa-i2s-dai" "-hifi", -"voice", etc for CODEC DAIs e.g. "wm8731-hifi" "-pcm-audio" for platform DMA, e.g. "pxa-pcm-audio" for PXA DMA (although most platform DMA controllers used this anyway). This naming standard will also come in handy for device tree too. The only other change I've made is fixing the twl4030 and DAC33 codec probe and remove (as spotted by Peter). I've also received and applied numerous patches from Jarkko, Timur, Ryan, Chanwoo, Joonyoug, Kyungmin, Jassi and Seungwhan. The multi-component code can be found here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git topic/multi-component As usual, please give this some testing (especially ASoC platform maintainers) and if all is OK we can upstream some time next week (or the week after). Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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Re: [alsa-devel] Which project to choose?
by Raymond Yau 22 Jul '10

22 Jul '10

2010/7/22 Chris <cpollock(a)embarqmail.com> > On Thu, 2010-07-22 at 11:03 +0800, Raymond Yau wrote: > > > > > > 2010/7/22 Chris <cpollock(a)embarqmail.com> > > > > > > > > > > AFAIK, the most buggy PA client is pavucontrol because > > > > > > 1) it has those vumeters which need to get audio data from > > the > > > playback and capture devices of the sound card , > > > 2) when you move the slider to left ( over 100%) and this > > force PA to > > > provide software gain > > > > > > You should notice that the new gnome-volume-control is > > actually a > > > simplified version of pavucontrol > > > > > > Hmm, I just pulled up PulseAudio Volume Control and see no > > vumeters, I > > can't figure out though why the drive thrashing at various > > times whether > > I have a stream playing or not. > > > > > > Did you click the input device tab page of the gnome volume control ? > > > > The vumeter is "Input level" above the volume control slider ? > > > > You should noticed that the following message appear in the pulseaudio > > log > > > > I: source-output.c : Created output 0 "Peak detect" on alsa_input > > > > I.source-output:c application.id = "org.gnome.volume.control" > > I.source-output:c media.name = "Peak detect" > > I.source-output:c application.name = "GNOME Volume control dialog" > > > > when you using mplayer to play music and open pavucontrol to > > > > pavucontrol sometime abort suddenly when you aborte mplayer but this > > is not 100% reproducible > > > > pavucontrol > > ** ( pavucontrol:3232) : DEBUG: failed to initialize device manager > > extension: No such extension > > ** (pavucontrol:3056) : DEBUG: Ignoring sink-input due to it being > > designated as an event and thuse handled by the event widget > > > > > > The point is after the complete intergration of pulseaudio in to the > > system ( libcanberra, mixer-applet , ... ) > > > > I think "pulseaudio -k; pulseaudio -vvvv" is no longer the correct way > > to debug since there are lots of PA clients running at the desktop > > > > You have to ask the mandriva maintainer how to increase the debug > > level of PA server and have to perform a logout/login to debug the PA > > server > > Thanks Raymond, so I should contact Colin and go from there? I'm not > sure what to even tell him though. I thank you for all your help. > > Chris > There are over 15 PA clients running when you are using gnome desktop running "pulseaudio -k " to shutdown PA server , this kill all the PA clients restart PA server by "pulseaudio -vvvv" , only one or two PA clients (e.g. consolekit, .. ) can connected back to PA server This command "pulseaudio -k;pulseaudio -vvvv" cannot provide a normal environment to debug PA server since most of PA clients are killed

1 0

[alsa-devel] [PATCH] ASoC: multi-component: SAMSUNG: Fix compile error when registering a DAI
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch modify some code which register a DAI with ASoC core according to modified registration API of DAI. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/aquila_wm8994.c | 3 +-- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c index 69c41eb..007ed3a 100644 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -268,8 +268,7 @@ static int __init aquila_init(void) return -ENOMEM; /* register voice DAI here */ - ret = snd_soc_register_dai(&aquila_snd_device->dev, - 0, &voice_dai); + ret = snd_soc_register_dai(&aquila_snd_device->dev, &voice_dai); if (ret) return ret; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index f4fbc0e..b3866d5 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -750,7 +750,7 @@ int s3c_i2sv2_register_dai(struct device *dev, int id, drv->suspend = s3c2412_i2s_suspend; drv->resume = s3c2412_i2s_resume; - return snd_soc_register_dai(dev, id, drv); + return snd_soc_register_dai(dev, drv); } EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); -- 1.6.3.3

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[alsa-devel] multi-component: determine which DAI is active?
by Timur Tabi 22 Jul '10

22 Jul '10

Liam, In order to get fsl_dma.c to work with the WM8776, which has separate DAIs for playback and capture, I've had to do this: if (pcm->streams[0].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); if (ret) { dev_err(card->dev, "can't allocate playback dma buffer\n"); return ret; } } if (pcm->streams[1].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); dev_err(card->dev, "can't allocate capture dma buffer\n"); return ret; } } That is, I need to check if the 'substream' pointer has been initialized. This looks ugly to me. In an earlier version of ASoC, I would have done this: if (dai->playback.channels_min) { ... if (dai->capture.channels_min) { ... But dai->playback and dai->capture no longer exist. How should I handle this?

3 2

[alsa-devel] [PATCH] ASoC: multi-component - Add Goni sound driver
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch add sound support for the Goni board based on S5PV210. The Goni board is based on Samsung SoC(S5PV210) and include WM8994 codec over I2S to support sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Goni board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Goni board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 ++ sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/goni_wm8994.c | 298 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 309 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/goni_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index c09b402..1cdc37b 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -140,3 +140,12 @@ config SND_S5PC110_SOC_AQUILA_WM8994 help Say Y if you want to add support for SoC audio on aquila with the WM8994. + +config SND_S5PV210_SOC_GONI_WM8994 + tristate "SoC I2S Audio support for GONI - WM8994" + depends on SND_S3C24XX_SOC && MACH_GONI + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on goni + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 02dd12c..47ed6d7 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -31,6 +31,7 @@ snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-aquila-wm8994-objs := aquila_wm8994.o +snd-soc-goni-wm8994-objs := goni_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -45,3 +46,4 @@ obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o +obj-$(CONFIG_SND_S5PV210_SOC_GONI_WM8994) += snd-soc-goni-wm8994.o diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c new file mode 100644 index 0000000..20bcaab --- /dev/null +++ b/sound/soc/s3c24xx/goni_wm8994.c @@ -0,0 +1,298 @@ +/* + * goni_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.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/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +static struct snd_soc_card goni; +static struct platform_device *goni_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget goni_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Left Spk", NULL), + SND_SOC_DAPM_SPK("Ext Right Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route goni_dapm_routes[] = { + {"Ext Left Spk", NULL, "SPKOUTLP"}, + {"Ext Left Spk", NULL, "SPKOUTLN"}, + + {"Ext Right Spk", NULL, "SPKOUTRP"}, + {"Ext Right Spk", NULL, "SPKOUTRN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add goni specific widgets */ + snd_soc_dapm_new_controls(codec, goni_dapm_widgets, + ARRAY_SIZE(goni_dapm_widgets)); + + /* set up goni specific audio routes */ + snd_soc_dapm_add_routes(codec, goni_dapm_routes, + ARRAY_SIZE(goni_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&goni, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int goni_hifi_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; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops goni_hifi_ops = { + .hw_params = goni_hifi_hw_params, +}; + +static int goni_voice_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; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "goni-voice-dai", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops goni_voice_ops = { + .hw_params = goni_voice_hw_params, +}; + +static struct snd_soc_dai_link goni_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_name = "s3c64xx-i2s-v4-dai", + .codec_dai_name = "wm8994-hifi", + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "wm8994-codec", + .init = goni_wm8994_init, + .ops = &goni_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_name = "goni-voice-dai", + .codec_dai_name = "wm8994-voice", + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "wm8994-codec", + .ops = &goni_voice_ops, +}, +}; + +static struct snd_soc_card goni = { + .name = "goni", + .dai_link = goni_dai, + .num_links = ARRAY_SIZE(goni_dai), +}; + +static int __init goni_init(void) +{ + int ret; + + if (!machine_is_goni()) + return -ENODEV; + + goni_snd_device = platform_device_alloc("soc-audio", -1); + if (!goni_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&goni_snd_device->dev, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(goni_snd_device, &goni); + ret = platform_device_add(goni_snd_device); + + if (ret) + platform_device_put(goni_snd_device); + + return ret; +} + +static void __exit goni_exit(void) +{ + platform_device_unregister(goni_snd_device); +} + +module_init(goni_init); +module_exit(goni_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

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[alsa-devel] [PATCH] ASoC: multi-component: SAMSUNG: Fix wrong field name on Aquila board
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch modify the wrong field name on Aquila board. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/aquila_wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c index f6f0e60..69c41eb 100644 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -242,7 +242,7 @@ static struct snd_soc_dai_link aquila_dai[] = { }, { .name = "WM8994 Voice", .stream_name = "Voice", - .cpu_dai_drv = "aquila-voice-dai", + .cpu_dai_name = "aquila-voice-dai", .codec_dai_name = "wm8994-voice", .platform_name = "s3c24xx-pcm-audio", .codec_name = "wm8994-codec", -- 1.6.3.3

2 1

[alsa-devel] problems about converting sample rates when using ALSA-LIB
by longer 22 Jul '10

22 Jul '10

Hi all, I have a problem when using ALSA lib . The problem is : My ALSA driver only support a special sample rate , so ALSA lib must convert sample rates. But My driver don't get any audio data. When tracing the code of ALSA lib (alsa-lib-1.0.23), I found that : It mmap a address from my driver , but writing audio data to another address instead of that mmaped address. I think this is why my driver don't get any audio data . But I don't know why it don't write audio data to that mmaped address. Can you help me ? Any advices would help me a lot. Thank you, longer.zhou

1 0

[alsa-devel] problems about converting sample rate when using ALSA lib
by longer zhou (周龙) 22 Jul '10

22 Jul '10

Hi all, I have a problem when using ALSA lib . The problem is : My ALSA driver only support a special sample rate , so ALSA lib must convert sample rates. But My driver don't get any audio data. When tracing the code of ALSA lib (alsa-driver-1.0.23), I found that : It mmap a address from my driver , but writting audio data to another address instead of that mmaped address. I think this is why my driver don't get any audio data .But I don't know why it don't write audo data to that mmaped address. Can you help me ? Any advices would help me a lot. Thank you, longer.zhou

1 0

[alsa-devel] add support for Codec: Conexant ID 5068 (not supported by ALSA yet)
by ccamacho＠linuxmail.org 22 Jul '10

22 Jul '10

Hi all, The codec (Conexant ID 5068) required by my sound card (Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)) is not present yet in alsa, I would highly appreciate if you guys could add support for it. You guys might find the information collected by the alsa-info.sh script on http://www.alsa-project.org/db/?f=8f8001d873d5c10c293eac1c94f2ede0b73cd865 Thanks a lot in advance for looking into this. Cheers, Cristopher

1 0

[alsa-devel] Small memory leak in snd_pcm_ladspa_init()
by John Lindgren 22 Jul '10

22 Jul '10

Hello, Valgrind shows up a small memory leak in libasound when using a LADSPA plugin. Sorry, I don't have a patch this time. ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,976 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BEFD: snd_pcm_ladspa_init (pcm_ladspa.c:744) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,977 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BF25: snd_pcm_ladspa_init (pcm_ladspa.c:746) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ... ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,980 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BEFD: snd_pcm_ladspa_init (pcm_ladspa.c:744) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== by 0x1B03B2DC: snd_pcm_recover (pcm.c:7246) ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,981 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BF25: snd_pcm_ladspa_init (pcm_ladspa.c:746) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== by 0x1B03B2DC: snd_pcm_recover (pcm.c:7246) ... and so on. -- John

2 2

[alsa-devel] add support for Codec: Conexant ID 5068 (not present in ALSA yet)
by ccamacho＠linuxmail.org 21 Jul '10

21 Jul '10

Hi all, The codec (Conexant ID 5068) required by my sound card (Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)) is not present yet in alsa, I would highly appreciate if you guys could add support for it. You guys might find the information collected by the alsa-info.sh script on http://www.alsa-project.org/db/?f=8f8001d873d5c10c293eac1c94f2ede0b73cd865 Thanks a lot in advance for looking into this. Cheers, Cristopher

1 0

[alsa-devel] Compiler errors in alsa-lib-1.0.22/test/*.c
by John Simpson 21 Jul '10

21 Jul '10

Howdy, I've been trying to use the alsa-lib API, but I haven't been able to get anything to compile. ================= ~/alsa/alsa-lib-1.0.22/test$ for f in *.c; do echo; cc $f 2>&1 | head -n2; done /tmp/ccNjLWLK.o: In function `dump_event_filter': client_event_filter.c:(.text+0x23): undefined reference to `snd_seq_client_info_event_filter_check' /tmp/ccNLDqdU.o: In function `main': control.c:(.text+0x1c): undefined reference to `snd_ctl_card_info_sizeof' /tmp/ccBehb51.o: In function `setparams_stream': latency.c:(.text+0x24): undefined reference to `snd_pcm_hw_params_any' /usr/lib/gcc/x86_64-linux-gnu/4.4.3/../../../../lib/crt1.o: In function `_start': (.text+0x20): undefined reference to `main' midiloop.c: In function ‘main’: midiloop.c:172: warning: format ‘%i’ expects type ‘int’, but argument 2 has type ‘size_t’ namehint.c: In function ‘main’: namehint.c:12: warning: passing argument 3 of ‘snd_device_name_hint’ from incompatible pointer type oldapi.c: In function ‘main’: oldapi.c:40: warning: assignment from incompatible pointer type /tmp/ccmB528p.o: In function `generate_sine': pcm.c:(.text+0x119): undefined reference to `snd_pcm_format_width' /tmp/ccChgWYC.o: In function `main': pcm_min.c:(.text+0x4d): undefined reference to `snd_pcm_open' /tmp/cc9FiONG.o: In function `write_ev': playmidi1.c:(.text+0x1b65): undefined reference to `snd_seq_event_output' /tmp/ccQkntBX.o: In function `main': queue_timer.c:(.text+0x1d0): undefined reference to `snd_seq_open' /tmp/ccmIMCZ4.o: In function `main': rawmidi.c:(.text+0x4e3): undefined reference to `snd_rawmidi_open' /tmp/cc3g8KGc.o: In function `event_decoder_start_timer': seq.c:(.text+0x4e2): undefined reference to `snd_seq_control_queue' seq-decoder.c:6: error: ‘SND_SEQ_EVENT_SYSTEM’ undeclared here (not in a function) seq-decoder.c:6: error: array index in initializer not of integer type seq-sender.c:68: error: expected ‘)’ before ‘*’ token seq-sender.c:100: error: expected ‘)’ before ‘*’ token /tmp/ccxyZfhg.o: In function `show_status': timer.c:(.text+0x1d): undefined reference to `snd_timer_status_sizeof' ================= Did I miss something? TIA -- John Wesley Simpson SwaJime's Cove℠ www.swajime.com

3 11

[alsa-devel] [PATCH 0/3] Removing dead code
by Christian Dietrich 21 Jul '10

21 Jul '10

Hi all! As part of the VAMOS[0] research project at the University of Erlangen we are looking at multiple integrity errors in linux' configuration system. I've been running a check on the sound/ sourcetree for config Items not defined in Kconfig and found 3 such cases. Sourcecode blocks depending on these Items are not reachable from a vanilla kernel -- dead code. I've seen such dead blocks made on purpose e.g. while integrating new features into the kernel but generally they're just useless. Each of the patches in this patchset removes on such dead config Item, I'd be glad if you consider applying them. I've been doing deeper analysis of such issues before and can do so again but I'm not so sure they were fastly usefull. I build the patches against a vanilla kernel in order to try if the kernel compiles with this patches Please keep me informed of this patch getting confirmed / merged so we can keep track of it. Regards Christian Dietrich [0] http://vamos1.informatik.uni-erlangen.de/ Christian Dietrich (3): sound/oss: Remove dead CONFIG_SOFTOSS* sound/soc/atmel: Remove dead CONFIG_AT32_ENHANCED_PORTMUX sound/isa/gus: Remove dead CONFIG_SND_DEBUG_ROM sound/isa/gus/interwave.c | 8 -------- sound/oss/vidc.c | 3 --- sound/soc/atmel/playpaq_wm8510.c | 9 --------- 3 files changed, 0 insertions(+), 20 deletions(-)

3 6

[alsa-devel] [GIT PULL] ASoC updates for 2.6.36
by Liam Girdwood 21 Jul '10

21 Jul '10

The following changes since commit 66b47fdb851924956b6e4696fb43a3496ae2c462: Mark Brown (1): ASoC: Implement WM8994 OPCLK support are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.36 Jorge Eduardo Candelaria (1): ASoC: TWL6040: Correct widget handling for drivers Peter Ujfalusi (3): ASoC: TWL4030: Fix for digital loopback gain range ASoC: TWL4030: DAC power optimization ASoC: TWL4030: Add configurable delay after digimic enable sound/soc/codecs/twl4030.c | 47 ++++++++++++++++++++++++++---------- sound/soc/codecs/twl4030.h | 1 + sound/soc/codecs/twl6040.c | 56 ++++++++++++++++++++++---------------------- 3 files changed, 63 insertions(+), 41 deletions(-)

2 1

[alsa-devel] mmotm 2010-07-19 - more pm_qos woes - audio this time
by Valdis.Kletnieks＠vt.edu 21 Jul '10

21 Jul '10

On Mon, 19 Jul 2010 16:38:09 PDT, akpm(a)linux-foundation.org said: > The mm-of-the-moment snapshot 2010-07-19-16-37 has been uploaded to > > http://userweb.kernel.org/~akpm/mmotm/ I've accumulated about 1,000 of these overnight. Seems to be consistently landing in pairs based on the timestamps. [42875.544006] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42875.544006] Hardware name: Latitude E6500 [42875.544006] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42875.544006] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42875.544006] Call Trace: [42875.544006] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42875.544006] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42875.544006] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42875.544006] [<ffffffff812052c7>] plist_add+0x1b/0xb1 [42875.544006] [<ffffffff81057b2e>] update_target+0x94/0xd0 [42875.544006] [<ffffffff81057d54>] pm_qos_add_request+0x79/0x7b [42875.544006] [<ffffffff8141d199>] snd_pcm_hw_params+0x25a/0x28a [42875.544006] [<ffffffff8141da29>] snd_pcm_common_ioctl1+0x23b/0x980 [42875.544006] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42875.544006] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42875.544006] [<ffffffff810d9178>] ? fd_install+0x36/0xd5 [42875.544006] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42875.544006] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42875.544006] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42875.544006] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42875.544006] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42875.544006] ---[ end trace 23d666d936caf587 ]--- [42875.544006] ------------[ cut here ]------------ [42875.544006] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42875.544006] Hardware name: Latitude E6500 [42875.544006] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42875.544006] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42875.544006] Call Trace: [42875.544006] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42875.544006] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42875.544006] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42875.544006] [<ffffffff81205354>] plist_add+0xa8/0xb1 [42875.544006] [<ffffffff81057b2e>] update_target+0x94/0xd0 [42875.544006] [<ffffffff81057d54>] pm_qos_add_request+0x79/0x7b [42875.544006] [<ffffffff8141d199>] snd_pcm_hw_params+0x25a/0x28a [42875.544006] [<ffffffff8141da29>] snd_pcm_common_ioctl1+0x23b/0x980 [42875.544006] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42875.544006] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42875.544006] [<ffffffff810d9178>] ? fd_install+0x36/0xd5 [42875.544006] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42875.544006] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42875.544006] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42875.544006] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42875.544006] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42875.544006] ---[ end trace 23d666d936caf588 ]--- [42882.427310] ------------[ cut here ]------------ [42882.428016] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42882.428016] Hardware name: Latitude E6500 [42882.428016] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42882.428016] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42882.428016] Call Trace: [42882.428016] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42882.428016] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42882.428016] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42882.428016] [<ffffffff8120537c>] plist_del+0x1f/0x8b [42882.428016] [<ffffffff81057b21>] update_target+0x87/0xd0 [42882.428016] [<ffffffff81057bb6>] pm_qos_remove_request+0x4c/0x5c [42882.428016] [<ffffffff8141dad9>] snd_pcm_common_ioctl1+0x2eb/0x980 [42882.428016] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42882.428016] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42882.428016] [<ffffffff8109589d>] ? trace_preempt_on+0x15/0x28 [42882.428016] [<ffffffff810c727e>] ? unmap_region+0xbd/0xcc [42882.428016] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42882.428016] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42882.428016] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42882.428016] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42882.428016] [<ffffffff810c746a>] ? unlink_file_vma+0x2d/0x4d [42882.428016] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42882.428016] ---[ end trace 23d666d936caf589 ]--- [42882.428016] ------------[ cut here ]------------ [42882.428016] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42882.428016] Hardware name: Latitude E6500 [42882.428016] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42882.428016] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42882.428016] Call Trace: [42882.428016] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42882.428016] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42882.428016] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42882.428016] [<ffffffff812053df>] plist_del+0x82/0x8b [42882.428016] [<ffffffff81057b21>] update_target+0x87/0xd0 [42882.428016] [<ffffffff81057bb6>] pm_qos_remove_request+0x4c/0x5c [42882.428016] [<ffffffff8141dad9>] snd_pcm_common_ioctl1+0x2eb/0x980 [42882.428016] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42882.428016] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42882.428016] [<ffffffff8109589d>] ? trace_preempt_on+0x15/0x28 [42882.428016] [<ffffffff810c727e>] ? unmap_region+0xbd/0xcc [42882.428016] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42882.428016] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42882.428016] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42882.428016] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42882.428016] [<ffffffff810c746a>] ? unlink_file_vma+0x2d/0x4d [42882.428016] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42882.428016] ---[ end trace 23d666d936caf58a ]---

2 1

[alsa-devel] Missing config option for SFFSDR_FPGA
by Christian Dietrich 21 Jul '10

21 Jul '10

Hi all! As part of the VAMOS[0] research project at the University of Erlangen we are looking at multiple integrity errors in linux' configuration system. I've been running a check on the sound/ sourcetree for config Items not defined in Kconfig and found such a case. Sourcecode blocks depending on these Items are not reachable from a vanilla kernel -- dead code. I've seen such dead blocks made on purpose e.g. while integrating new features into the kernel but generally they're just useless. There are two ifdef blocks, which do a check on CONFIG_SFFSDR_FPGA, but this flag isn't defined anywhere. There is an select in sound/soc/davinci[1], but Kconfig doesn't define the symbol, if there isn't a config option for it. So this blocks can never be selected. I don't think this is intended, so i didn't wrote a patch. Please just notice it. Regards Christian Dietrich [0] http://vamos1.informatik.uni-erlangen.de/ [1] sound/soc/davinci/Kconfig: select SFFSDR_FPGA sound/soc/davinci/davinci-sffsdr.c:#ifdef CONFIG_SFFSDR_FPGA sound/soc/davinci/davinci-sffsdr.c:#ifndef CONFIG_SFFSDR_FPGA sound/soc/davinci/davinci-sffsdr.c:#ifndef CONFIG_SFFSDR_FPGA -- (λ x . x x) (λ x . x x) -- See how beatiful the lambda is No documentation is better than bad documentation -- Das Ausdrucken dieser Mail wird urheberrechtlich verfolgt.

3 3

[alsa-devel] [PATCH] ASoC: TWL4030: Add configurable delay after digimic enable
by Peter Ujfalusi 21 Jul '10

21 Jul '10

When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, I have been looking after this problem for a year now (with low priority), and finally I did figured out what is caused the glitch at the beginning of the recording via the digimic interface. Adding platform dependent delay after enabling the digimic interface fixes the issue I am tracking, in other HW setups with different components the delay might not be needed. Peter sound/soc/codecs/twl4030.c | 25 +++++++++++++++++++++---- sound/soc/codecs/twl4030.h | 1 + 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6fd6d0b..bd557c2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -143,6 +143,9 @@ struct twl4030_priv { u8 earpiece_enabled; u8 predrivel_enabled, predriver_enabled; u8 carkitl_enabled, carkitr_enabled; + + /* Delay needed after enabling the digimic interface */ + unsigned int digimic_delay; }; /* @@ -312,6 +315,8 @@ static void twl4030_init_chip(struct platform_device *pdev) if (!setup) return; + twl4030->digimic_delay = setup->digimic_delay; + /* Configuration for headset ramp delay from setup data */ if (setup->sysclk != twl4030->sysclk) dev_warn(codec->dev, @@ -855,6 +860,16 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, return 0; } +static int digimic_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); + + if (twl4030->digimic_delay) + mdelay(twl4030->digimic_delay); + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -1439,10 +1454,12 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_PGA("ADC Physical Right", TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic0 Enable", - TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic1 Enable", - TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("Digimic0 Enable", + TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Digimic1 Enable", + TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 788e3d1..6c57430 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -41,6 +41,7 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; + unsigned int digimic_delay; /* in ms */ unsigned int sysclk; unsigned int offset_cncl_path; unsigned int check_defaults:1; -- 1.7.1.1

3 2

[alsa-devel] [PATCHv2] ASoC: multi-component - Add Aquila sound driver
by Chanwoo Choi 21 Jul '10

21 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2S to support sound. This uses the I2Sv4 driver compatible with I2Sv5 to run sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 301 +++++++++++++++++++++++++++++++++++++ 3 files changed, 312 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..c09b402 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..375f4c0 --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,301 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.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/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define WM8994_DAI_HIFI 0 +#define WM8994_DAI_VOICE 1 + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_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; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_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; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "Voice", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], + .codec_dai_id = WM8994_DAI_HIFI, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_drv = &voice_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], + .codec_dai_id = WM8994_DAI_VOICE, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + aquila_snd_device = platform_device_alloc("soc-audio", -1); + if (!aquila_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&aquila_snd_device->dev, + 0, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(aquila_snd_device, &aquila); + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

2 1

[alsa-devel] [PATCH] Memory leak in namehint.c
by John Lindgren 21 Jul '10

21 Jul '10

get_card_name() can be called more than once on the same list, so it must free the previous list->cardname before replacing it. Signed-off-by: John Lindgren <john.lindgren(a)tds.net> --- src/control/namehint.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/src/control/namehint.c b/src/control/namehint.c index 78572d8..fc1cb27 100644 --- a/src/control/namehint.c +++ b/src/control/namehint.c @@ -471,6 +471,8 @@ static int get_card_name(struct hint_list *list, int card) char scard[16], *s; int err; + if (list->cardname) + free(list->cardname); err = snd_card_get_name(card, &list->cardname); if (err <= 0) return 0;

2 2

[alsa-devel] Question for ALSA I/O Plugin
by Raul Xiong 21 Jul '10

21 Jul '10

Hi all, I know that there are two kinds of ALSA external Plugin: filter-type and I/O type. Now I have an I/O Plugin named phone. Can it be a slave of a internal Plugin? E.g. can I use this way: aplay -Dplug:phone wav. Thanks!

1 0

[alsa-devel] snd-hda-intel acl269 toshiba r700 speakers work, headphones do not
by Thomas Hoffmann 21 Jul '10

21 Jul '10

Dear Alsa developers, Sound works from the built in laptop speakers. When headphones are plugged in, the speakers properly mute, but there is no sound coming from the headphones. I've tried some of the options to the alsa-base.conf file to no avail; generally they made it so sound came out of the speakers always, even when headphones were plugged in. I previously sent an e-mail to this list server: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5064 where they suggested I e-mail you. My system info is available from http://www.alsa-project.org/db/?f=f9aaf6e43b669cf2c4b2215781a414b12843e2a3 Best, Tom

3 3

[alsa-devel] [PATCH 1/3] ASoC: multi-component - Add Aquila sound driver
by Chanwoo Choi 21 Jul '10

21 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2S to support sound. This uses the I2Sv4 driver compatible with I2Sv5 to run sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 300 +++++++++++++++++++++++++++++++++++++ 3 files changed, 311 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..c09b402 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..6ff5068 --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,300 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.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/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define WM8994_DAI_HIFI 0 +#define WM8994_DAI_VOICE 1 + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_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; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_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; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "Voice", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], + .codec_dai_id = WM8994_DAI_HIFI, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_drv = &voice_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], + .codec_dai_id = WM8994_DAI_VOICE, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&aquila_snd_device->dev, + 0, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(aquila_snd_device, &aquila); + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

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[alsa-devel] Audio dropouts with pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32).
by VDR User 20 Jul '10

20 Jul '10

Hi. I've been experiencing audio drops on dvb streams that use ac3 audio (h264+ac3 hdtv streams) and occasionally mp2 audio (sdtv channels) . When the ac3 audio drops out, it usually recovers after a few seconds though sometimes it never does. When mp2 drops out, it almost never recovers. Retuning the channel always fixes the drop out. When this happens there's usually no interruption in the video so I question whether stream errors are causing the problem. When a dropout occurs, I always see the following in my xine log: pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). I (and others apparently) have been experiencing this problem for quite some time. I have read about and tried adding notsc to no kernel boot line, which has had no affect. Here's a clip from the most recent drop outs: audio jump, diff=-90720 pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). audio_out: inserting 21713 0-frames to fill a gap of 40723 pts video_out: throwing away image with pts 176336725 because it's too old (diff : 10842). video_out: throwing away image with pts 176339939 because it's too old (diff : 16618). video jump 200 frames delivered, 0 frames skipped, 2 frames discarded video jump audio jump, diff=-103680 pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). audio_out: inserting 21233 0-frames to fill a gap of 39823 pts video_out: throwing away image with pts 183068800 because it's too old (diff : 6699). video jump video_out: throwing away image with pts 183071835 because it's too old (diff : 12728). pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). 200 frames delivered, 0 frames skipped, 2 frames discarded Some system specs are: debian testing kernel 2.6.34.1 alsa-base 1.0.23+dfsg-1 alsa-utils 1.0.23-2 alsa drivers compiled from today's snapshot (2010-07-20). in /etc/modprobe.d/alsa-base.conf: options snd-hda-intel enable_msi=1 probe_mask=0xffff,0xfff2 lsmod: snd_hda_codec_nvhdmi 9750 1 snd_hda_intel 14764 0 snd_hda_codec 48881 2 snd_hda_codec_nvhdmi,snd_hda_intel snd_hwdep 3566 1 snd_hda_codec snd_pcm 44172 2 snd_hda_intel,snd_hda_codec snd_timer 11991 1 snd_pcm snd 28664 6 snd_hda_codec_nvhdmi,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer soundcore 548 1 snd snd_page_alloc 4669 2 snd_hda_intel,snd_pcm I'd greatly appreciate any help in getting this problem worked out and hopefully solved once and for all. Thanks in advance for _anything_. Best regards, Derek

1 0

[alsa-devel] [PATCH] wm8960: fix compile error
by Axel Lin 20 Jul '10

20 Jul '10

This patch fixes compile error: sound/soc/codecs/wm8960.c: In function 'wm8960_hw_params': sound/soc/codecs/wm8960.c:511: error: 'codec' undeclared (first use in this function) sound/soc/codecs/wm8960.c:511: error: (Each undeclared identifier is reported only once sound/soc/codecs/wm8960.c:511: error: for each function it appears in.) make[3]: *** [sound/soc/codecs/wm8960.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch is against Liam's multi-component branch. Axel sound/soc/codecs/wm8960.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e6df6e3..599c559 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -508,8 +508,8 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); struct snd_soc_codec *codec = rtd->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; int i; -- 1.5.4.3

2 3

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 20 Jul '10

20 Jul '10

The following changes since commit 0c74a939d84730818bc9bf1efa5ad7184bb2f248: ASoC: au1x: fix section mismatch in psc-i2s.c (2010-07-13 12:39:14 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Axel Lin (2): ASoC: wm8727: add a missing return in wm8727_platform_probe ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register Chanwoo Choi (2): ASoC: Select wm_hubs automatically for WM8994 ASoC:Support Samsung SoC(S5P) in I2Sv2 Eric Bénard (1): ASoC: eukrea-tlv320: add support for cpuimx35sd Kulikov Vasiliy (3): ASoC: davinci: check kzalloc() result (typo) ASoC: imx: check kzalloc() result and fix memory leak ASoC: imx: check kzalloc() result and fix memory leak Kuninori Morimoto (1): ASoC: fsi: Add specified ID for soc-audio Mark Brown (8): Merge branch 'for-2.6.35' into for-2.6.36 ASoC: Handle read failures in codec_reg ASoC: Remove duplicate AUX definition from WM8776 Merge branch 'for-2.6.35' into for-2.6.36 ASoC: Explicitly disable DC servo on WM hubs headphone powerdown ASoC: Implement WM8994 AIF1ADC2 paths Merge branch 'for-2.6.35' into for-2.6.36 ASoC: Unconditionally enable WM8994 AIF1ADC TDM mode Sekhar Nori (1): ASoC: davinci: let platform data define edma queue numbers arch/arm/mach-davinci/board-da830-evm.c | 2 +- arch/arm/mach-davinci/board-da850-evm.c | 2 +- arch/arm/mach-davinci/board-dm646x-evm.c | 4 +- arch/arm/mach-davinci/include/mach/asp.h | 3 +- include/sound/sh_fsi.h | 3 ++ sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/wm8727.c | 2 + sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8994.c | 32 ++++++++++++++++++++++++++++++ sound/soc/codecs/wm_hubs.c | 2 + sound/soc/davinci/davinci-i2s.c | 10 +++++++++ sound/soc/davinci/davinci-mcasp.c | 6 +++- sound/soc/davinci/davinci-pcm.c | 5 ++- sound/soc/davinci/davinci-pcm.h | 3 +- sound/soc/davinci/davinci-vcif.c | 2 +- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 +- sound/soc/imx/imx-pcm-dma-mx2.c | 6 ++++- sound/soc/imx/imx-pcm-fiq.c | 6 ++++- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 +- sound/soc/sh/fsi-ak4642.c | 4 +- sound/soc/sh/fsi-da7210.c | 4 +- sound/soc/soc-core.c | 22 +++++++++++++++---- 24 files changed, 103 insertions(+), 29 deletions(-)

2 1

[alsa-devel] [PATCH RFC 0/1] ASoC: WL1273 FM Radio codes
by Matti J. Aaltonen 20 Jul '10

20 Jul '10

Hello. This is the ALSA part of our WL1273 FM Radio driver. I've sent the complete driver to the linux-media list for review. But as people on that list are not in general ALSA specialists I'm sending the codec part here. So all comments are welcome. Thank you, Matti A. Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec. sound/soc/codecs/wl1273.c | 593 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 42 ++++ 2 files changed, 635 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h

2 3

[alsa-devel] ASoC update for 2.6.35
by Mark Brown 20 Jul '10

20 Jul '10

The following changes since commit 637727838a5e82bc9285ab078a793eaae590bacb: ASoC: fsi: fixup wrong value setting order of TDM (2010-07-13 12:26:26 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Axel Lin (2): ASoC: wm8727: add a missing return in wm8727_platform_probe ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register Chanwoo Choi (1): ASoC: Select wm_hubs automatically for WM8994 Mark Brown (1): ASoC: Remove duplicate AUX definition from WM8776 Another series of small, driver-specific fixes. sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/wm8727.c | 2 ++ sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8988.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH - alsa-lib 0/3] Fixes for snd_tlv_convert_from_dB
by Peter Ujfalusi 20 Jul '10

20 Jul '10

Hello, One cosmetic fix, and two real fixes for alsa-lib's tlv handling. There were two separate issue with the snd_tlv_convert_from_dB function: 1. When user asked for out of range dB value, and the control had SND_CTL_TLVT_DB_RANGE, the control's full range was not checked against the requested value, which resulted failure. 2. When there are 'holes' in the SND_CTL_TLVT_DB_RANGE array, snd_tlv_convert_from_dB will fail to find the apropriate raw value. For issue #2, I have sent patches for the drivers I maintain, but Mark Brown suggested to fix the issue in alsa-lib instead of converting the driver's non-overlapping to overlapping mapping: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-July/029483.html So the following series for alsa-lib, will fix the SND_CTL_TLVT_DB_RANGE handling. --- Peter Ujfalusi (3): tlv: Check out of range dB with SND_CTL_TLVT_DB_RANGE tlv: Handle 'holes' in SND_CTL_TLVT_DB_RANGE array tlv: Remove tailing tab after snd_ctl_get_dB_range function src/control/tlv.c | 17 +++++++++++++++-- 1 files changed, 15 insertions(+), 2 deletions(-)

4 18

[alsa-devel] [PATCH 0/12] sound/alsa/soc/codec: fix memory leak and resource relaim in error path
by Axel Lin 20 Jul '10

20 Jul '10

This serial of patches fixes memory leak and resource relaim in error path [PATCH 1/12] ad1836: fix a memory leak if another ad1836 is registered [PATCH 2/12] ak4642: fix a memory leak if failed to initialise AK4642 [PATCH 3/12] da7210: fix a memory leak if failed to initialise da7210 audio codec [PATCH 4/12] wm8523: fix resource reclaim in wm8523_register error path [PATCH 5/12] wm8711: fix a memory leak if another WM8711 is registered [PATCH 6/12] wm8904: fix resource reclaim in wm8904_register error path [PATCH 7/12] wm8940: fix resource reclaim in wm8940_register error path [PATCH 8/12] wm8955: fix resource reclaim in wm8955_register error path [PATCH 9/12] wm8961: fix resource reclaim in wm8961_register error path [PATCH 10/12] wm8974: fix a memory leak if another WM8974 is registered [PATCH 11/12] wm8978: fix a memory leak if another WM8978 is registered [PATCH 12/12] wm9081: fix resource reclaim in wm9081_register error path Regards, Axel

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Re: [alsa-devel] [PATCH v2] asoc: davinci: let platform data define edma queue numbers
by Mark Brown 20 Jul '10

20 Jul '10

On Mon, Jul 19, 2010 at 12:31:16PM +0530, Sekhar Nori wrote: > Currently the EDMA queue to be used by for servicing ASP through > internal RAM is fixed to EDMAQ_0 and that to service internal RAM > from external RAM is fixed to EDMAQ_1. Applied, thanks. It's "ASoC", BTW.

1 0

[alsa-devel] [PATCH 2/3] ASoC: multi-component - Support Samsung SoC(S5P) in I2Sv2
by Chanwoo Choi 20 Jul '10

20 Jul '10

This patch modify I2Sv2 driver to support Samsung SoC(S5PV210). Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 6575410..f4fbc0e 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,7 +32,8 @@ #undef S3C_IIS_V2_SUPPORTED -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \ + || defined(CONFIG_CPU_S5PV210) #define S3C_IIS_V2_SUPPORTED #endif -- 1.6.3.3

3 3

[alsa-devel] [PATCH 3/3] ASoC: multi-component - Enable WM8993/4 common code to support WM8994
by Chanwoo Choi 20 Jul '10

20 Jul '10

This patch enable WM8993/4 common code which is used in WM8994 driver. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/codecs/Kconfig | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea1f5ed..17e28eb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -85,8 +85,8 @@ config SND_SOC_ALL_CODECS config SND_SOC_WM_HUBS tristate - default y if SND_SOC_WM8993=y - default m if SND_SOC_WM8993=m + default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y + default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m config SND_SOC_AC97_CODEC tristate -- 1.6.3.3

3 2

[alsa-devel] [PATCH] ASoC: da7210: Add HeadPhone Playback Volume control
by Kuninori Morimoto 20 Jul '10

20 Jul '10

HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. This patch add HeadPhone Playback Volume 40 as default, because DA7210 doesn't have default volume. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/da7210.c | 37 +++++++++++++++++++++++++++++++++---- 1 files changed, 33 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 75af2d6..f0af2c8 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -145,6 +145,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -227,10 +250,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -572,6 +591,13 @@ static int da7210_init(struct da7210_priv *da7210) /* Activate all enabled subsystem */ da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + /* + * HeadPhone PlayBack Volume 40 is given here + * because DA7210 doesn't have default volume. + */ + snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); + snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); + return ret; init_err: @@ -656,6 +682,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: -- 1.7.0.4

2 2

[alsa-devel] [PATCH v2 - alsa-lib 0/3] Fixes for snd_tlv_convert_from_dB
by Peter Ujfalusi 20 Jul '10

20 Jul '10

Hello, Changes since the initial version: - In patch one the range check has been converted to be more intuitive Intro part from the first version: One cosmetic fix, and two real fixes for alsa-lib's tlv handling. There were two separate issue with the snd_tlv_convert_from_dB function: 1. When user asked for out of range dB value, and the control had SND_CTL_TLVT_DB_RANGE, the control's full range was not checked against the requested value, which resulted failure. 2. When there are 'holes' in the SND_CTL_TLVT_DB_RANGE array, snd_tlv_convert_from_dB will fail to find the apropriate raw value. For issue #2, I have sent patches for the drivers I maintain, but Mark Brown suggested to fix the issue in alsa-lib instead of converting the driver's non-overlapping to overlapping mapping: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-July/029483.html So the following series for alsa-lib, will fix the SND_CTL_TLVT_DB_RANGE handling. --- Peter Ujfalusi (3): tlv: Check out of range dB with SND_CTL_TLVT_DB_RANGE tlv: Handle 'holes' in SND_CTL_TLVT_DB_RANGE array tlv: Remove tailing tab after snd_ctl_get_dB_range function src/control/tlv.c | 20 ++++++++++++++++++-- 1 files changed, 18 insertions(+), 2 deletions(-)

1 3

[alsa-devel] [PATCH 0/3] ASoC: twl4030/tpa6130a2: DB_RANGE mapping fixes
by Peter Ujfalusi 20 Jul '10

20 Jul '10

Hello, There were 'holes' in the DB_RANGE and SCALE_ITEM mapping, which results failure, when user space requests dB value in these holes, since alsa-lib will fail to find a range, and can not round the requested dB to raw value. Fixing the drivers I'm maintaining to correctly fill the tlv struct. Peter --- Peter Ujfalusi (3): ASoC: TWL4030: Fix DB_RANGE tlv mapping for digital loopback ASoC: tpa6130a2: Fix DB_RANGE mapping for tpa6130a2 chip ASoC: tpa6130a2: Fix DB_RANGE mapping for tpa6140a2 chip sound/soc/codecs/tpa6130a2.c | 22 ++++++++++++++-------- sound/soc/codecs/twl4030.c | 4 ++-- 2 files changed, 16 insertions(+), 10 deletions(-)

3 15

[alsa-devel] [PATCH v2] ASoC: da7210: Add HeadPhone Playback Volume control
by Kuninori Morimoto 20 Jul '10

20 Jul '10

HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. This patch add HeadPhone Playback Volume 40 as default, because DA7210 doesn't have default volume. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 o based on ASoC's git tree (add #include <sound/tlv.h>) sound/soc/codecs/da7210.c | 38 ++++++++++++++++++++++++++++++++++---- 1 files changed, 34 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a83aa18..ec292cc 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -23,6 +23,7 @@ #include <sound/pcm_params.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "da7210.h" @@ -136,6 +137,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -218,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -563,6 +583,13 @@ static int da7210_init(struct da7210_priv *da7210) /* Activate all enabled subsystem */ da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + /* + * HeadPhone PlayBack Volume 40 is given here + * because DA7210 doesn't have default volume. + */ + snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); + snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); + return ret; init_err: @@ -647,6 +674,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: -- 1.7.0.4

1 0

[alsa-devel] [PATCH] Add conexant quirk for AMD based Lenovo G series machines
by Jerone Young 19 Jul '10

19 Jul '10

This is a follow on patch adds support for AMD based Lenovo G series machines, such as the Lenovo G555. Signed-off-by: Jerone Young <jerone.young(a)canonical.com> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index c49030b..774edb9 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -3000,6 +3000,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), {} };

2 2

[alsa-devel] DMA Physical Addr Issue.
by Adish Kuvelker 19 Jul '10

19 Jul '10

Hello, I have allocated the DMA memory using the function "snd_pcm_lib_preallocate_pages_for_all" passing SNDRV_DMA_TYPE_DEV and "card->dev" as inputs to the function. Later, in the hw_params call out function I allocate pages to the requested DMA memory region using the function "snd_pcm_lib_malloc_pages". I play a wav file "Noise.wav" from command line by giving the following command ./aplay --file-type wav -f dat Noise.wav And, when I print the DMA region physical address in the "prepare" call out through "dma_addr" member of the runtime structure I get a weird decimal value of "-1036562432" as the physical address. Can anyone point where I might be going wrong in the whole DMA memory allocation? Regards; Aadish

2 1

[alsa-devel] [PATCH 12/15] pci: riptide: check kzalloc() result
by Kulikov Vasiliy 19 Jul '10

19 Jul '10

If kzalloc() fails exit with -ENOMEM. Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/pci/riptide/riptide.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 59d7996..f64fb7d 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1615,7 +1615,10 @@ static int snd_riptide_playback_open(struct snd_pcm_substream *substream) chip->playback_substream[sub_num] = substream; runtime->hw = snd_riptide_playback; + data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; data->paths = lbus_play_paths[sub_num]; data->id = play_ids[sub_num]; data->source = play_sources[sub_num]; @@ -1635,7 +1638,10 @@ static int snd_riptide_capture_open(struct snd_pcm_substream *substream) chip->capture_substream = substream; runtime->hw = snd_riptide_capture; + data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; data->paths = lbus_rec_path; data->id = PADC; data->source = ACLNK2PADC; -- 1.7.0.4

2 1

[alsa-devel] [PATCH 11/15] pci: echoaudio: check kmalloc() result
by Kulikov Vasiliy 19 Jul '10

19 Jul '10

If kmalloc() fails exit with -ENOMEM. Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/pci/echoaudio/echoaudio.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 668a5ec..20763dd 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2250,6 +2250,8 @@ static int snd_echo_resume(struct pci_dev *pci) DE_INIT(("resume start\n")); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + if (commpage_bak == NULL) + return -ENOMEM; commpage = chip->comm_page; memcpy(commpage_bak, commpage, sizeof(struct comm_page)); -- 1.7.0.4

3 2

[alsa-devel] XRUN handling
by melwyn lobo 19 Jul '10

19 Jul '10

Hi all, I have noticed when an Xrun occurs, the bytes between the hw_ptr and appl_ptr are not copied when both hw and appl pointers are reset in snd_pcm_prepare(). Consider the following parameters: buffer_size : 22352 period_size : 1016 period_time : 23038 tstamp_mode : NONE period_step : 1 avail_min : 1016 period_event : 0 start_threshold : 22352 stop_threshold : 22352 silence_threshold: 0 silence_size : 0 In case snd_pcm_playback_avail() is less than stop threshold but greater than 0, then these bytes are missing after Xrun prepare and start stage. Is my analysis correct? If so how do we ensure correct behaviour ie byte accuracy after Xrun. Thanks, M.

4 11

[alsa-devel] [PATCH 2/2]ASoC: multi-component - Fix S5P compile error
by Chanwoo Choi 19 Jul '10

19 Jul '10

This patch fix compile error on using I2S and WM8994 common code. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/codecs/Kconfig | 4 ++-- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 ++- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea1f5ed..55bc2be 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -85,8 +85,8 @@ config SND_SOC_ALL_CODECS config SND_SOC_WM_HUBS tristate - default y if SND_SOC_WM8993=y - default m if SND_SOC_WM8993=m + default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y + default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m config SND_SOC_AC97_CODEC tristate diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 6575410..f4fbc0e 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,7 +32,8 @@ #undef S3C_IIS_V2_SUPPORTED -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \ + || defined(CONFIG_CPU_S5PV210) #define S3C_IIS_V2_SUPPORTED #endif

2 2

[alsa-devel] [PATCH 1/2]ASoC: multi-component - Add Aquila sound driver
by Chanwoo Choi 19 Jul '10

19 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2S to support sound. This uses the I2Sv4 driver compatible with I2Sv5 to run sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 300 +++++++++++++++++++++++++++++++++++++ 3 files changed, 311 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..c09b402 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..6ff5068 --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,300 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.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/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define WM8994_DAI_HIFI 0 +#define WM8994_DAI_VOICE 1 + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_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; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_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; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "Voice", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], + .codec_dai_id = WM8994_DAI_HIFI, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_drv = &voice_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], + .codec_dai_id = WM8994_DAI_VOICE, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&aquila_snd_device->dev, + 0, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(aquila_snd_device, &aquila); + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

1 0

[alsa-devel] Clicking and stuttering audio when playing two or more files
by Adam Rosenberg 19 Jul '10

19 Jul '10

I have audio output perfectly from my second CS42448 CODEC now, but only when one file is playing. When I attempt to play two files, one or both of the files play with clicking and stuttering. I am also noticing that the PCM prepare operation is occurring more than once. What could be causing this problem? HW settings are: buffer_bytes_max = 32768 period_bytes min = 8 max = 32768 periods min = 1 max = 4096 PCM runtime is reporting: frames per period = 1114 periods = 2 total frames = 2228 frame bytes = 8 Thank you, Adam

2 1

[alsa-devel] [PATCH] Add conexant quirk for Lenovo G series support
by Jerone Young 19 Jul '10

19 Jul '10

This patch adds quirk add support for Lenovo G series laptops. Launchpad bug can be found here: https://bugs.launchpad.net/ubuntu/+source/linux-backports-modules-2.6.32/+b… Signed-off-by: Jerone Young <jerone.young(a)canonical.com> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index 54f7419..c49030b 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -2999,6 +2999,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), {} };

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[alsa-devel] [PATCH] asoc: davinci: let platform data define edma queue numbers
by Sekhar Nori 17 Jul '10

17 Jul '10

Currently the EDMA queue to be used by for servicing ASP through internal RAM is fixed to EDMAQ_0 and that to service internal RAM from external RAM is fixed to EDMAQ_1. This may not be the desirable configuration on all platforms. For example, on DM365, queue 0 has large fifo size and is more suitable for video transfers. Having audio and video transfers on the same queue may lead to starvation on audio side. platform data as defined currently passes a queue number to the driver but that remains unused inside the driver. Fix this by defining one queue each for ASP and RAM transfers in the platform data and using it inside the driver. Since EDMAQ_0 maps to 0, thats the queue that will be used if the asp queue number is not initialized. None of the platforms currently utilize ping-pong transfers through internal RAM so that functionality remains unchanged too. This patch has been tested on DM644x and OMAP-L138 EVMs. Signed-off-by: Sekhar Nori <nsekhar(a)ti.com> --- This patch applies to latest of Linus's tree. arch/arm/mach-davinci/board-da830-evm.c | 2 +- arch/arm/mach-davinci/board-da850-evm.c | 2 +- arch/arm/mach-davinci/board-dm646x-evm.c | 4 ++-- arch/arm/mach-davinci/include/mach/asp.h | 3 ++- sound/soc/davinci/davinci-i2s.c | 10 ++++++++++ sound/soc/davinci/davinci-mcasp.c | 6 ++++-- sound/soc/davinci/davinci-pcm.c | 5 +++-- sound/soc/davinci/davinci-pcm.h | 3 ++- 8 files changed, 25 insertions(+), 10 deletions(-) diff --git a/arch/arm/mach-davinci/board-da830-evm.c b/arch/arm/mach-davinci/board-da830-evm.c index 212d970..bc384d3 100644 --- a/arch/arm/mach-davinci/board-da830-evm.c +++ b/arch/arm/mach-davinci/board-da830-evm.c @@ -208,7 +208,7 @@ static struct snd_platform_data da830_evm_snd_data = { .num_serializer = ARRAY_SIZE(da830_iis_serializer_direction), .tdm_slots = 2, .serial_dir = da830_iis_serializer_direction, - .eventq_no = EVENTQ_0, + .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_2, .txnumevt = 1, .rxnumevt = 1, diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index 2ec3095..d4ec18d 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -342,7 +342,7 @@ static struct snd_platform_data da850_evm_snd_data = { .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), .tdm_slots = 2, .serial_dir = da850_iis_serializer_direction, - .eventq_no = EVENTQ_1, + .asp_chan_q = EVENTQ_1, .version = MCASP_VERSION_2, .txnumevt = 1, .rxnumevt = 1, diff --git a/arch/arm/mach-davinci/board-dm646x-evm.c b/arch/arm/mach-davinci/board-dm646x-evm.c index 6d88893..87521f2 100644 --- a/arch/arm/mach-davinci/board-dm646x-evm.c +++ b/arch/arm/mach-davinci/board-dm646x-evm.c @@ -323,7 +323,7 @@ static struct snd_platform_data dm646x_evm_snd_data[] = { .num_serializer = ARRAY_SIZE(dm646x_iis_serializer_direction), .tdm_slots = 2, .serial_dir = dm646x_iis_serializer_direction, - .eventq_no = EVENTQ_0, + .asp_chan_q = EVENTQ_0, }, { .tx_dma_offset = 0x400, @@ -332,7 +332,7 @@ static struct snd_platform_data dm646x_evm_snd_data[] = { .num_serializer = ARRAY_SIZE(dm646x_dit_serializer_direction), .tdm_slots = 32, .serial_dir = dm646x_dit_serializer_direction, - .eventq_no = EVENTQ_0, + .asp_chan_q = EVENTQ_0, }, }; diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..5d0a4c7 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -52,7 +52,8 @@ struct snd_platform_data { u32 tx_dma_offset; u32 rx_dma_offset; - enum dma_event_q eventq_no; /* event queue number */ + enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ + enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..4c1a7d2 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -526,6 +526,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; + enum dma_event_q asp_chan_q = EVENTQ_0; + enum dma_event_q ram_chan_q = EVENTQ_1; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -552,7 +554,15 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + asp_chan_q = pdata->asp_chan_q; + ram_chan_q = pdata->ram_chan_q; } + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; + dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d395509..b247208 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -890,7 +890,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->rxnumevt = pdata->rxnumevt; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; - dma_data->eventq_no = pdata->eventq_no; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + io_v2p(dev->base)); @@ -904,7 +905,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; - dma_data->eventq_no = pdata->eventq_no; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + io_v2p(dev->base)); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2dc406f..f026ac5 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -381,7 +381,7 @@ static int request_ping_pong(struct snd_pcm_substream *substream, /* Request ram master channel */ link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, davinci_pcm_dma_irq, substream, - EVENTQ_1); + prtd->params->ram_chan_q); if (link < 0) goto exit1; @@ -477,7 +477,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) /* Request asp master DMA channel */ link = prtd->asp_channel = edma_alloc_channel(params->channel, - davinci_pcm_dma_irq, substream, EVENTQ_0); + davinci_pcm_dma_irq, substream, + prtd->params->asp_chan_q); if (link < 0) goto exit1; diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0764944..b799a02 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -21,7 +21,8 @@ struct davinci_pcm_dma_params { unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ unsigned sram_size; - enum dma_event_q eventq_no; /* event queue number */ + enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ + enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; -- 1.6.2.4

3 2

[alsa-devel] [PATCH] ASoC: fsi: Add specified ID for soc-audio
by Kuninori Morimoto 17 Jul '10

17 Jul '10

Specified ID is necessary, when some codecs are used with FSI. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark. Thank you about your advice. it solved my issue. This patch is v2 of "ASoC: FSI - codecs settings" include/sound/sh_fsi.h | 3 +++ sound/soc/sh/fsi-ak4642.c | 4 ++-- sound/soc/sh/fsi-da7210.c | 4 ++-- 3 files changed, 7 insertions(+), 4 deletions(-) diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index c022736..6463fd6 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -12,6 +12,9 @@ * published by the Free Software Foundation. */ +#define FSI_PORT_A 0 +#define FSI_PORT_B 1 + /* flags format * 0xABCDEEFF diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index be01854..2c61ed2 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -38,7 +38,7 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) static struct snd_soc_dai_link fsi_dai_link = { .name = "AK4642", .stream_name = "AK4642", - .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .cpu_dai = &fsi_soc_dai[FSI_PORT_A], .codec_dai = &ak4642_dai, .init = fsi_ak4642_dai_init, .ops = NULL, @@ -62,7 +62,7 @@ static int __init fsi_ak4642_init(void) { int ret = -ENOMEM; - fsi_snd_device = platform_device_alloc("soc-audio", -1); + fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A); if (!fsi_snd_device) goto out; diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index 33b4d17..5774449 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -33,7 +33,7 @@ static int fsi_da7210_init(struct snd_soc_codec *codec) static struct snd_soc_dai_link fsi_da7210_dai = { .name = "DA7210", .stream_name = "DA7210", - .cpu_dai = &fsi_soc_dai[1], /* FSI B */ + .cpu_dai = &fsi_soc_dai[FSI_PORT_B], .codec_dai = &da7210_dai, .init = fsi_da7210_init, }; @@ -56,7 +56,7 @@ static int __init fsi_da7210_sound_init(void) { int ret; - fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); + fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); if (!fsi_da7210_snd_device) return -ENOMEM; -- 1.7.0.4

3 2

[alsa-devel] [PATCH 13/15] sound/soc: davinci: check kzalloc() result (typo)
by Kulikov Vasiliy 17 Jul '10

17 Jul '10

The code checks 'davinci_vc' after kzalloc() and do not checks 'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that it is a typo (autocompletion?). Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/soc/davinci/davinci-vcif.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9aa980d..4867853 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -203,7 +203,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) int ret; davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); - if (!davinci_vc) { + if (!davinci_vcif_dev) { dev_dbg(&pdev->dev, "could not allocate memory for private data\n"); return -ENOMEM; -- 1.7.0.4

3 2

[alsa-devel] [PATCH v2] ASoC: TWL6040: Correct widget handling for drivers
by Jorge Eduardo Candelaria 17 Jul '10

17 Jul '10

In order to reduce pop-noise at powering up/down of the DACs and Drivers, these components have to be handled in a specific sequence. Headset, Handsfree, and Earphone drivers are now registered as PGA components to ensure DACs are enabled first. Also, add a delay to leave time for DACs to settle before continuing power up/down sequence. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 56 ++++++++++++++++++++++---------------------- 1 files changed, 28 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 715caef..345d86a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -360,6 +360,13 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) return 0; } +static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + msleep(1); + return 0; +} + static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -371,6 +378,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, else priv->non_lp--; + msleep(1); + return 0; } @@ -471,20 +480,6 @@ static const struct snd_kcontrol_new hfdacl_switch_controls = static const struct snd_kcontrol_new hfdacr_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); -/* Headset driver switches */ -static const struct snd_kcontrol_new hsl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); - -static const struct snd_kcontrol_new hsr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); - -/* Handsfree driver switches */ -static const struct snd_kcontrol_new hfl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); - -static const struct snd_kcontrol_new hfr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); - static const struct snd_kcontrol_new ep_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); @@ -548,10 +543,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0), /* DACs */ - SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0), - SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", TWL6040_REG_HFLCTL, 0, 0, twl6040_power_mode_event, @@ -571,18 +570,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HFDAC Right Playback", SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Left Driver", - SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Right Driver", - SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), - SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", - SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + /* Analog playback drivers */ + SND_SOC_DAPM_PGA_E("Handsfree Left Driver", + TWL6040_REG_HFLCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", - SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + SND_SOC_DAPM_PGA_E("Handsfree Right Driver", + TWL6040_REG_HFRCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA("Headset Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0), SND_SOC_DAPM_SWITCH_E("Earphone Driver", SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, twl6040_power_mode_event, @@ -616,8 +616,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSDAC Left Playback", "Switch", "HSDAC Left"}, {"HSDAC Right Playback", "Switch", "HSDAC Right"}, - {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, - {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + {"Headset Left Driver", NULL, "HSDAC Left Playback"}, + {"Headset Right Driver", NULL, "HSDAC Right Playback"}, {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, -- 1.7.0.4

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[alsa-devel] [PATCHv2 0/4] ASoC: multi-component - Samsung S3C Platform
by Seungwhan Youn 17 Jul '10

17 Jul '10

Hi, This is a second version of patch series adding support for multi component model on Samsung S3C Platform. This patch series can be apply upon "ASoC: multi-component - Fix S3C compile error" in the first patch of V1 that already applied by Liam. Changes since V1: - Support for GPIO, DMA and SFR settings pull from platform device resource. - Modify description of add WM9713 platform device on smdk board. The following patch series contains: [PATCHv2 1/4] ASoC: S3C64XX: Move GPIO settings to platform [PATCHv2 2/4] ASoC: S3C64XX: Move DMA settings to platform [PATCHv2 3/4] ASoC: multi-component - S3C - Get original functionality [PATCHv2 4/4] ASoC: multi-component - S3C - Add WM9713 platform device Thanks, Claude

3 7

[alsa-devel] [PATCH] asoc: multi-component: mpc8610: fix typo in mpc8610_hpcd_data
by Timur Tabi 17 Jul '10

17 Jul '10

The dma_id[] variable in struct mpc8610_hpcd_data is a zero-length array, when it should be of length 2. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Liam, please squash this into your repository. sound/soc/fsl/mpc8610_hpcd.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 81ab639..8665cc2 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -37,7 +37,7 @@ struct mpc8610_hpcd_data { unsigned int cpu_clk_direction; unsigned int clk_frequency; unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ - unsigned int dma_id[0]; /* 0 = DMA1, 1 = DMA2, etc */ + unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ char dai_name[32]; }; -- 1.7.0.1

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Re: [alsa-devel] [PlanetCCRMA] [LAD] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 17 Jul '10

17 Jul '10

On Fri, Jul 16, 2010 at 5:27 AM, Geoff King <gsking1(a)gmail.com> wrote: > Niels, Thank you for taking the time to work on improving Envy24Ctl. > I tried your patch for the peaks and it seemed to work fine for me. I > can't comment on the code (as I'm much more musician than programmer), > but had no problem patching and installing. Looking forward to trying > this new patch and whatever else you come up with. Geoff Geoff -- thank you for trying out the patch! (And, OT, thanks for working w/ Rui to help fix http://sourceforge.net/tracker/?func=detail&atid=733076&aid=3021645&group_i… :: "help fix qtractor crash on bus changing/configuration (3021645)" -- Envy24 and multichannel qtractor users, yes, it's time for a "svn up" ). FYI, to try out the new levelmeters "easy", if you're running Fedora x86_64, use the "envy24control" binary directly, or drop the levelmeters.c file into your build: http://nielsmayer.com/npm/Efficient-Meters-Envy24Control.tgz ( for details http://nielsmayer.com/npm/Efficient-Meters-Envy24Control.README ) ........ On a completely different "note" the envy 24 manual ( http://alsa.cybermirror.org/manuals/icensemble/envy24.pdf ) has interesting info on the envy24 digital mixer that I've snapshotted: Diagram: http://nielsmayer.com/npm/envy24mixer-architecture.png Note the way it truncates in the mixer: the more inputs you "mix" at once, the fewer bits each input source gets, and it's not clear what kind of dither, or it's a straight truncate of 24... ultimately the envy24 seems oriented towards producing a 16 bit, and not 24 bit master (which makes sense given that the chip is well over a decade old and HD audio production for "prosumer" was rare): .......... 4.5.5 Multi-Track Digital Monitoring The Envy24 integrates a 36-bit resolution digital hardware mixer. The width of the data path is strictly to ensure that during processing of all the channels, under any condition, no resolution is lost. The dynamic range of the end user system will be limited by the range of the physical output devices used. In order to maintain identical gain to the input stream (i.e. 0dB), the resulting 24-bit is not msb-aligned to the 36-bit. The overflow bits correspond to the analog distortion due to saturation. The user would need to reduce the overall attenuation of the inputs to avoid clipping. Insertion of the digital mixer adds only a single sample cycle delay with respect to the original data. This extremely low latency all digital mixer provides monitoring functionality and can replace a traditional external analog input mixer. There are 20 independent audio data streams to mix and control the volume. ... .............. --Niels http://nielsmayer.com

1 0

[alsa-devel] [PATCH 15/15] sound/soc: imx: check kzalloc() result and fix memory leak
by Kulikov Vasiliy 16 Jul '10

16 Jul '10

If kzalloc() fails we must exit with -ENOMEM. Also we must free allocated runtime->private_data on error as it would be lost on next call to snd_imx_open(). Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/soc/imx/imx-pcm-fiq.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 6b518e0..b2bf272 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -192,6 +192,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream) int ret; iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + if (iprtd == NULL) + return -ENOMEM; runtime->private_data = iprtd; iprtd->substream = substream; @@ -202,8 +204,10 @@ static int snd_imx_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) + if (ret < 0) { + kfree(iprtd); return ret; + } snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); return 0; -- 1.7.0.4

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[alsa-devel] [PATCH 14/15] sound/soc: imx: check kzalloc() result and fix memory leak
by Kulikov Vasiliy 16 Jul '10

16 Jul '10

If kzalloc() fails we must exit with -ENOMEM. Also we must free allocated runtime->private_data on error as it would be lost on next call to snd_imx_open(). Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/soc/imx/imx-pcm-dma-mx2.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 05f19c9..0a595da 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -292,12 +292,16 @@ static int snd_imx_open(struct snd_pcm_substream *substream) int ret; iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + if (iprtd == NULL) + return -ENOMEM; runtime->private_data = iprtd; ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) + if (ret < 0) { + kfree(iprtd); return ret; + } snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); return 0; -- 1.7.0.4

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[alsa-devel] [PATCH] twl6040: fix wrong kfree in twl6040_remove and twl6040_codec_remove
by Axel Lin 16 Jul '10

16 Jul '10

Memory allocation part: We allocate a memory in twl6040_codec_probe(): priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); twl6040_codec = codec = &priv->codec; Memory release part: In twl6040_codec_remove() we should kfree(priv) instead of kfree(twl6040_codec). In twl6040_remove(), no need and should not kfree(codec). Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/twl6040.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index af36346..9d4a85f 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1068,7 +1068,6 @@ static int twl6040_remove(struct platform_device *pdev) twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - kfree(codec); return 0; } @@ -1214,7 +1213,7 @@ static int __devexit twl6040_codec_remove(struct platform_device *pdev) snd_soc_unregister_dai(&twl6040_dai); snd_soc_unregister_codec(twl6040_codec); - kfree(twl6040_codec); + kfree(priv); twl6040_codec = NULL; return 0; -- 1.5.4.3

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[alsa-devel] ASoC: Davinci Voice codec compliant to arago kernel for dm365
by Raffaele Recalcati 16 Jul '10

16 Jul '10

Hi Miguel, I've used successfully the patches below on top of 2.6.34 Kevin Hilman kernel for davinci http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git Now I need to backport your patches to arago kernel http://arago-project.org/git/projects/linux-davinci.git that means 2.6.32-rc1 I use able adapt your patches to it, but I guess that there are many changes from 2.6.32-rc1 to 2.6.34 ... Can you tell me what is the best thing to do? I'm afraid you'll tell me that I need to rewrite completely your patches to be really compliant to 2.6.32-rc1. Here you are a not working log: aplay tone1KHzMono_8000.wav Playing WAVE 'tone1KHzMono_8000.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono davinci_pcm_prepare - 203 davinci_pcm_enqueue_dma - 77: period=2000,dma_offset=0,dma_pos=-2044854272 davinci_pcm: audio_set_dma_params_play channel = 79 dma_ptr = 861e0000 period_size=7d0 davinci_pcm_enqueue_dma - 84: data_type=0 davinci_pcm_enqueue_dma - 77: period=2000,dma_offset=2000,dma_pos=-2044852272 davinci_pcm: audio_set_dma_params_play channel = 79 dma_ptr = 861e07d0 period_size=7d0 davinci_pcm_enqueue_dma - 84: data_type=0 aplay: pcm_write:1528: write error: Input/output error Thanks, Raffaele --------------- Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Date: Thu Mar 11 09:32:21 2010 -0600 MFD: DaVinci Voice Codec This is the MFD driver for the DaVinci Voice codec, it has two clients: * Voice codec interface * Voice codec CQ93VC Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> commit 94a532f7156582ec7459f21baa4de6665773974a Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Date: Thu Mar 11 09:32:59 2010 -0600 ASoC: DaVinci: CQ93VC Voice Codec Currently the DM365 is the only SoC that includes this Voice Codec. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> commit 80a6609eeb0995732318c7ab672767e39a111f3b Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Date: Thu Mar 11 09:32:42 2010 -0600 ASoC: DaVinci: Voice Codec Interface This patch adds the support for the interface needed by the DaVinci Voice Codec CQ93VC. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> -- www.opensurf.it

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[alsa-devel] HPI version 4.04.01
by linux＠audioscience.com 16 Jul '10

16 Jul '10

HPI stable driver 4.04.01 has just been released. Only very minor patches since the last beta version. Please also update alsa-firmware/asihpi with firmware from http://audioscience.com/internet/download/drivers/released/v4/04/01/asihpi-…

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[alsa-devel] Accessing snd_soc_read/snd_soc_write directly for diagnostics
by Mike Crowe 16 Jul '10

16 Jul '10

Hi folks, I know this is a strange request, however, for diagnostic purposes in our embedded system, we would like to talk directly to the codec registers of our sound chip, bypassing alsa. I have the driver compiled into the kernel (not loaded as a module). Is it possible to create a program which can talk to the snd_soc_read and snd_soc_write functions directly? I'm not sure if this is an option, but if it is, I'd appreciate a pointer to something similar to use as a guide. TIA Mike

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Re: [alsa-devel] [LAD] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 16 Jul '10

16 Jul '10

On Tue, Jul 13, 2010 at 3:46 PM, Tim E. Real <termtech(a)rogers.com> wrote: > But nothing was ever done about the slider markings. > Having them go from 0 to 164 is not really helpful. > It's not clear where the 0dB mark is. > At the very least a 0dB mark should be added. > Uh, do you feel like taking a stab at this? > IIRC it would be a bit tricky because you have to look up > the corresponding values in the IC data sheets, I think. This makes sense... I was thinking of a simple ruler with a 0dB label at 127 and a + dB label above and a - dB label below. Some thoughts on further usability improvements to envy24control: (1) In "alsamixer" the snd-ice1712-card's input levels are represented by (versus envy24control's value): 100% (163) -> ADC [dB gain: 18.00] 78% (127) -> ADC [dB gain: 0.00] 1% (001) -> ADC [dB gain: -63.00] 0% (000) -> ADC [dB gain: mute] Mixers like envy24control, kmix, alsamixergui, gst-mixer, etc, are a lot less useful than they could be because they just present sliders. At least envy24control gives numeric values, -- which is essential if you want to know stereo channels are balenced, or what unity-gain levels are for inputs. In "alsamixer", the snd-ice1712-card's output levels are represented by (versus envy24control value): 100% (127) -> DAC [dB gain: 0.00] 1% (001) -> DAC [dB gain: -63.00] 0% (000) -> DAC [dB gain: mute] As this is a separate issue from the meters, it makes sense to do the changes to the "Analog Volume" panel of envy24control as a separate patch. (2) The "0-96" sliders, feeding the digital mixer, need better labeling and mappings too: The "96" actually being "-96dB" and only valid for 16 bit depths. On the other hand, alsamixer is showing the 24 bit dB values, even when 16 bits are being used: 100% (0) -> H/W Multi [dB gain: 0.00, ...] 1% (95) -> H/W Multi [dB gain: -142.50,...] 0% (96) -> H/W Multi [dB gain: -144.00,...] Worse than the labelling, but associated -- the scaling: Right now the digital mixer's input sliders all have linear response -- which wastes the lower half of the slider and makes setting levels difficult because all the "action" is near the top. If you use an external MIDI controller for the mixer, you can use this existing option: -M, --midienhanced Use an enhanced mapping from midi controller to db slider However this does not apply to the GUI controller, which stays linear. Clearly the GUIs digital mixer controls also need a logarithmic scale (perhaps as default settings); matching that, each mixer input's meters also need to have their scale expressed "logarithmically" so as to provide, for example 50% of the display resolution for 0 to -20dB, and remaining 50% for -20dB to -96dB/-144dB. Alongside the numbers indicating the 0 to -96db L or R gain on each mixer input channel, I would also add the current peak value held in the meters in dB, which would be based on hardware metering level "0xFF" representing 0db, and hardware meter level "0x01" representing all but the last of the 9 MSB's being 0 (per http://alsa.cybermirror.org/manuals/icensemble/envy24.pdf: "Peak data derived from the absolute value of 9 msb. 00h * min - FFh max volume. Reading the register resets the meter to 00h." Other missing features: (3) Due to the hardware metering employed by envy24control, the peak meters don't actually detect "clipping" ; rather, they just detect "full scale" as 255. Is there a hardware clipped indicator on the ice1712? It doesn't make sense to turn on a big red "clipped" light for 0dB output, so the current peak-meters are "subtle," and just indicate that a peak occurred at the given level... And color-code the peak towards red as it reaches 255. (4) The "IEC958 Input Status" isn't presented, even though it's available and working, e.g from the command-line,. with SPDIF input going to the Delta66 soundcard: ..... gnulem-176-~> amixer -c M66 cget numid=50,iface=MIXER,name='Delta IEC958 Input Status' numid=50,iface=MIXER,name='Delta IEC958 Input Status' ; type=BOOLEAN,access=r-------,values=1 : values=on ..... And without any SPDIF input: ..... gnulem-177-~> amixer -c M66 cget numid=50,iface=MIXER,name='Delta IEC958 Input Status' numid=50,iface=MIXER,name='Delta IEC958 Input Status' ; type=BOOLEAN,access=r-------,values=1 : values=off Perhaps this isn't available in envy24control because it has differing levels of support across envy24-based cards? I only have a Delta66's and a Terratec Dmx6Fire 24/96 for testing, and both support the SPDIF input status. -- Niels http://nielsmayer.com PS: ALSA tip of the day. To read ice1712's hardware level-meter values from the command line: "amixer -c M66 cget iface=PCM,name='Multi Track Peak',numid=45" type=INTEGER,access=r-------,values=22,min=0,max=255,step=0 : values=0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,255,198,255,198

1 0

[alsa-devel] [PATCH] uda134x: properly free allocated memory if uda134x_soc_probe fail with unknown codec type
by Axel Lin 16 Jul '10

16 Jul '10

If uda134x_soc_probe fail with unknown codec type, it should properly free allocated memory before return -EINVAL. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This case won't be executed in current implementation because there is a model checking in the begin of uda134x_soc_probe. But since the code is there, I think it's good to make it correct. Otherwise, we can simply remove the default case. sound/soc/codecs/uda134x.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 28aac53..9bff3c6 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -566,9 +566,10 @@ static int uda134x_soc_probe(struct platform_device *pdev) ARRAY_SIZE(uda1345_snd_controls)); break; default: + ret = -EINVAL; printk(KERN_ERR "%s unknown codec type: %d", __func__, pd->model); - return -EINVAL; + break; } if (ret < 0) { -- 1.5.4.3

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[alsa-devel] [PATCH 8/8] sound: asihpi: fix sign bug
by Kulikov Vasiliy 15 Jul '10

15 Jul '10

bytes_per_sec is unsigned, so if snd_pcm_format_width() return error we would not see it. Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/pci/asihpi/asihpi.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 91218f7..c80b0b8 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -460,6 +460,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); int err; u16 format; + int width; unsigned int bytes_per_sec; print_hwparams(params); @@ -512,9 +513,10 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, dpcm->hpi_buffer_attached); } bytes_per_sec = params_rate(params) * params_channels(params); - bytes_per_sec *= snd_pcm_format_width(params_format(params)); + width = snd_pcm_format_width(params_format(params)); + bytes_per_sec *= width; bytes_per_sec /= 8; - if (bytes_per_sec <= 0) + if (width < 0 || bytes_per_sec == 0) return -EINVAL; dpcm->bytes_per_sec = bytes_per_sec; -- 1.7.0.4

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[alsa-devel] ALSA Help?
by Matthew Hudson 15 Jul '10

15 Jul '10

Hello, I posted similar messages to the alsa-user mail list and to ubuntu forums ( http://ubuntuforums.org/showthread.php?t=1528442 ), but haven't been able to get help. Could any of you help with this? So, I've got a great Acer Aspire 7736. Seems to have a RealTek ALC888 card & Intel HDA. I've upgraded Alsa. I've used the Drivers at the RealTek site. I've checked alsamixer over and over and over. Yet, currently I cannot get sound through the headphone jack. Originally, I had problems w/ headphone/speakers playing simlutaneously. Somewhere in my adventures I fixed that, but then only one app would use sound during a login (UGH). So, I kept poking...as all good linux users do...now I'm afraid I've totally wonked it. I'll put all pertinent alsa-info up here for anyone who will help!! http://www.alsa-project.org/db/?f=58850a92a9ea0edbf28a8d014a875cdbd81b621d I guess I'm bright enough to really mess it up, but dumb enough not to have a clue how to fix it. It has NEVER been able to recognize a headphone plugged in the jack automatically. I hope someone knows... -- Soli Deo Gloria! Matt Hudson "There is a kind of happiness and wonder that makes you serious." C.S. Lewis

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[alsa-devel] [PATCH 1/3] ASoC: multi-component - Fix S3C compile error
by Seungwhan Youn 15 Jul '10

15 Jul '10

This patch fix compile error on using S3C I2S. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index ad92e10..5017e31 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -56,11 +56,11 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, switch (clk_id) { case S3C64XX_CLKSRC_PCLK: - iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX; + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_MUX: - iismod |= S3C64XX_IISMOD_IMS_SYSMUX; + iismod |= S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_CDCLK: @@ -90,7 +90,7 @@ struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(dai); u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) + if (iismod & S3C2412_IISMOD_IMS_SYSMUX) return i2s->iis_cclk; else return i2s->iis_pclk; -- 1.6.2.5

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Re: [alsa-devel] [PATCH 1/2] Kconfig: SND_AC97_POWER_SAVE description improvement
by Jiri Kosina 15 Jul '10

15 Jul '10

On Wed, 14 Jul 2010, Michael Witten wrote: > The description has been expanded to explain the time-out > value provided by the power_save module parameter. Michael, thanks a lot for the patch. I'd rather let it go through Takashi still anyway (added to CC). Thanks. > > Signed-off-by: Michael Witten <mfwitten(a)gmail.com> > --- > sound/drivers/Kconfig | 24 +++++++++++++++++++++--- > 1 files changed, 21 insertions(+), 3 deletions(-) > > diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig > index 84714a6..3264600 100644 > --- a/sound/drivers/Kconfig > +++ b/sound/drivers/Kconfig > @@ -170,9 +170,25 @@ config SND_AC97_POWER_SAVE > AC97 codecs. In this mode, the power-mode is dynamically > controlled at each open/close. > > - The mode is activated by passing power_save=1 option to > - snd-ac97-codec driver. You can toggle it dynamically over > - sysfs, too. > + The mode is activated by passing 'power_save=X' to the > + snd-ac97-codec driver module, where 'X' is the time-out > + value, a nonnegative integer that specifies how many > + seconds of idle time the driver must count before it may > + put the AC97 into power-save mode; a value of 0 (zero) > + disables the use of this power-save mode. > + > + After the snd-ac97-codec driver module has been loaded, > + the 'power_save' parameter can be set via sysfs as follows: > + > + echo 10 > /sys/module/snd_ac97_codec/parameters/power_save > + > + In this case, the time-out is set to 10 seconds; setting > + the time-out to 1 second (the minimum activation value) > + isn't recommended because many applications try to reopen > + the device frequently. A value of 10 seconds would be a > + good choice for normal operations. > + > + See Documentation/sound/alsa/powersave.txt for more details. > > config SND_AC97_POWER_SAVE_DEFAULT > int "Default time-out for AC97 power-save mode" > @@ -182,4 +198,6 @@ config SND_AC97_POWER_SAVE_DEFAULT > The default time-out value in seconds for AC97 automatic > power-save mode. 0 means to disable the power-save mode. > > + See SND_AC97_POWER_SAVE for more details. > + > endif # SND_DRIVERS > -- > 1.7.2.rc1.9.g5b527 > -- Jiri Kosina SUSE Labs, Novell Inc.

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[alsa-devel] [RFC][PATCH 0/3] ASoC: FSI - codecs settings
by Kuninori Morimoto 15 Jul '10

15 Jul '10

Dear Mark, Liam These patches are FSI - codecs settings Kuninori Morimoto (3): ASoC: ak4642: make sure snd_soc_codec value for ak4642 ASoC: da7210: make sure snd_soc_codec value for da7210 ASoC: Add fsi-codec support But I'm not sure this type of fix is correct or not. So, please check and teach me. Now, FSI driver has 2 connections like below. fsi - ak4642 fsi - da7210 In future, I will send "fsi - HDMI" pair. On AP4 board, I would like to use "fsi - ak4642" and "fsi - HDMI" in same time. To use it, I guess I should modify snd_soc_card :: dai_link correct ? I expected this is correct, I tried to use "fsi - ak4642" and "fsi - da7210" in same time for test. But when I used "soc_codec_dev_ak4642" for snd_soc_device :: codec_dev, da7210 didn't work. because i2c address of ak4642 is used for da7210. Same things happen on "soc_codec_dev_da7210" - ak4642. My patches solve above issue for me. But I'm not sure this is correct way. Best regards -- Kuninori Morimoto

2 6

[alsa-devel] controlling access to the sound card
by Jason Butler 14 Jul '10

14 Jul '10

I have three applications that want to use the sound card, two audio stream players, and a voip phone. Is it possible to use alsa so that if a call comes in on the phone alsa will disconnect the audio players, give exclusive access to the voip phone, and then when the phone is done reconnect the audio players to the sound card. Can this be done using alsa and if so how? Could it be done using just the alsa command line tools and config files and if so how?

1 0

[alsa-devel] [PATCH] wm8727: add a missing return in wm8727_platform_probe
by Axel Lin 14 Jul '10

14 Jul '10

otherwise the error path will always be executed. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8727.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 1072621..9d1df26 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -127,6 +127,8 @@ static __devinit int wm8727_platform_probe(struct platform_device *pdev) goto err_codec; } + return 0; + err_codec: snd_soc_unregister_codec(codec); err: -- 1.7.0.4

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Re: [alsa-devel] [PATCH 4/4] ASoC: DaVinci: pcm, fix underrun by using sram
by troy.kisky＠boundarydevices.com 14 Jul '10

14 Jul '10

BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; } On Tue 13/07/10 6:00 AM , "Nori, Sekhar" nsekhar(a)ti.com sent: Hi Troy, On Tue, Nov 17, 2009 at 05:22:34, Troy Kisky wrote: > Fix underruns by using dma to copy 1st to sram > in a ping/pong buffer style and then copying from > the sram to the ASP. This also has the advantage > of tolerating very long interrupt latency on dma > completion. > > Signed-off-by: Troy Kisky [...] > +static int request_ping_pong(struct snd_pcm_substream *substream, > + struct davinci_runtime_data *prtd, > + struct snd_dma_buffer *iram_dma) > +{ > + dma_addr_t asp_src_ping; > + dma_addr_t asp_dst_ping; > + int link; > + struct davinci_pcm_dma_params *dma_data = prtd->params; > + > + /* Request ram master channel */ > + link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, > + davinci_pcm_dma_irq, substream, > + EVENTQ_1); What is the reason for choosing EVENTQ_1 for this channel? EVENTQ_0 is already being used for ASP channel. I imagine it will be much easier to tune the queue usage in the system if all of audio data was using the same queue. I am working on a patch which lets platform specify the event queues for audio DMA. I am not sure if I really need to make a provision for two different queues to be specified - that's why I ask. Thanks, Sekhar The reason is so that the IRAM data can be fetched and used while EVENTQ_1 fetches the next buffer of data from sdram into IRAM. Troy

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[alsa-devel] [PATCH] ASoC: Handle read failures in codec_reg
by Mark Brown 14 Jul '10

14 Jul '10

When a device is powered down volatile registers can't be read so attempts to display codec_reg will show error values, and obviously it is also possible for there to be hardware errors too. Check for errors from reads and display them more clearly when formatting codec_reg. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 22 +++++++++++++++++----- 1 files changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96f9a8e..844ae82 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static int run_delayed_work(struct delayed_work *dwork) /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { - int i, step = 1, count = 0; + int ret, i, step = 1, count = 0; if (!codec->reg_cache_size) return 0; @@ -101,12 +101,24 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (count >= PAGE_SIZE - 1) break; - if (codec->display_register) + if (codec->display_register) { count += codec->display_register(codec, buf + count, PAGE_SIZE - count, i); - else - count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); + } else { + /* If the read fails it's almost certainly due to + * the register being volatile and the device being + * powered off. + */ + ret = codec->read(codec, i); + if (ret >= 0) + count += snprintf(buf + count, + PAGE_SIZE - count, + "%4x", ret); + else + count += snprintf(buf + count, + PAGE_SIZE - count, + "<no data: %d>", ret); + } if (count >= PAGE_SIZE - 1) break; -- 1.7.1

2 1

[alsa-devel] [PATCH] Asoc Davinci Voicecodec: Added support based on copy_from_user instead of DMA
by Raffaele Recalcati 14 Jul '10

14 Jul '10

From: Davide Bonfanti <davide.bonfanti(a)bticino.it> Since DM365 has the same DMA event for McBSP and Voicecodec this two peripherals cannot be used at the same time. This driver implements Voicecodec without the use of a DMA but with a copy_from_user. There's a buffer of BUF_SIZE (tested with 2KB) bytes in the driver that is filled with davinci_pcm_copy. When pcm is running, a TIMER interrupt is activated each 1.36ms in order to fill HW FIFO (dm_vc_irq). BUG: It happens sometimes that the peripheral stops working so there's a trap. This driver was tested with timer1 and timer4. The measures made using a GPIO and scope give a time for copy_from_user about 8/10us (triggered by alsa layer) while TIMER interrupt takes 1.5us. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and tested on bmx board. Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> --- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-pcm-copyfromuser.c | 333 ++++++++++++++++++++++++++ sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/soc-core.c | 31 ++- 4 files changed, 354 insertions(+), 12 deletions(-) create mode 100644 sound/soc/davinci/davinci-pcm-copyfromuser.c diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a93679d..7d6a9a1 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,5 +1,6 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o +snd-soc-davinci-objs += davinci-pcm-copyfromuser.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o diff --git a/sound/soc/davinci/davinci-pcm-copyfromuser.c b/sound/soc/davinci/davinci-pcm-copyfromuser.c new file mode 100644 index 0000000..df37873 --- /dev/null +++ b/sound/soc/davinci/davinci-pcm-copyfromuser.c @@ -0,0 +1,333 @@ +/* + * + * Copyright (C) 2010 Bticino S.p.a + * Author: Davide Bonfanti <davide.bonfanti(a)bticino.it> + * + * Contributors: + * Raffaele Recalcati <raffaele.recalcati(a)bticino.it> + * + * 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 version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; 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/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/kernel.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/gpio.h> + +#include "davinci-pcm.h" + +/* Timer register offsets */ +#define PID12 0x00 +#define TIM12 0x10 +#define TIM34 0x14 +#define PRD12 0x18 +#define PRD34 0x1c +#define TCR 0x20 +#define TGCR 0x24 + +/* Timer register bitfields */ +#define TCR_ENAMODE_DISABLE 0x0 +#define TCR_ENAMODE_ONESHOT 0x1 +#define TCR_ENAMODE_PERIODIC 0x2 +#define TCR_ENAMODE_MASK 0x3 + +#define TGCR_TIMMODE_SHIFT 2 +#define TGCR_TIMMODE_64BIT_GP 0x0 +#define TGCR_TIMMODE_32BIT_UNCHAINED 0x1 +#define TGCR_TIMMODE_64BIT_WDOG 0x2 +#define TGCR_TIMMODE_32BIT_CHAINED 0x3 + +#define TGCR_TIM12RS_SHIFT 0 +#define TGCR_TIM34RS_SHIFT 1 +#define TGCR_RESET 0x0 +#define TGCR_UNRESET 0x1 +#define TGCR_RESET_MASK 0x3 + +#define TIMER1_BASE 0x01C21800 +#define TIMER4_BASE 0x01C23800 + +/* choose the timer to use */ +#define TIMER_BASE TIMER4_BASE +/* driver buffer dimension */ +#define BUF_SIZE 2048 + +#if (TIMER_BASE == TIMER1_BASE) +#define TINT IRQ_TINT1_TINT12 +#define TIMER "timer1" +#elif (TIMER_BASE == TIMER4_BASE) +#define TINT IRQ_PWMINT2 +#define TIMER "timer4" +#endif + +int pointer_sub; +u16 local_buffer[BUF_SIZE/2]; + +static struct snd_pcm_hardware pcm_hardware_playback = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000), + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = BUF_SIZE, + .period_bytes_min = 512, + .period_bytes_max = 512, + .periods_min = BUF_SIZE / 512, + .periods_max = BUF_SIZE / 512, + .fifo_size = 0, +}; + +/* + * How this driver works... + * + * Since DM365 have the same DMA event for I2S and Voicecodec this two + * peripheralscannot be used at the same time. + * This driver implements voicecodec without the use of a DMA but with + * a copy_from_user. + * There's a buffer of BUF_SIZE bytes in the driver that is filled with + * davinci_pcm_copy. + * When pcm is running, a TIMER interrupt is activated each 1.36ms in + * order to fill HW FIFO (dm_vc_irq). + * It happens that the peripheral stop working so there's a trap... + * Driver was proved with timer1 and timer4. if timer2 is used the device + * resets during power-up. Timer3 have a structure a bit different and, + * in fact, doesn't work. + * Measures using a GPIO and scope give a time for copy_from_user about + * 8/10us while interrupt takes 1.5us + */ + +static snd_pcm_uframes_t +davinci_pcm_pointer(struct snd_pcm_substream *substream) +{ + return pointer_sub; +} + +static int davinci_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd; + struct snd_pcm_hardware *ppcm; + int ret = 0; + struct davinci_pcm_dma_params *pa; + struct davinci_pcm_dma_params *params; + struct clk *clk; + + pointer_sub = 0; + +/* + * ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + * &pcm_hardware_playback : &pcm_hardware_capture; + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ppcm = &pcm_hardware_playback; + else + return -ENODEV; + + snd_soc_set_runtime_hwparams(substream, ppcm); + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + __raw_writel(0x0, IO_ADDRESS(0x01D0C008)); + __raw_writel(0x7400, IO_ADDRESS(0x01D0C004)); + clk = clk_get(NULL, TIMER); + if (!IS_ERR(clk)) + clk_enable(clk); + return 0; +} + +static int davinci_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct clk *clk; + + clk = clk_get(NULL, TIMER); + if (!IS_ERR(clk)) + clk_disable(clk); + return 0; +} + +static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int davinci_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + if (copy_from_user(local_buffer + hwoff, buf, + frames_to_bytes(runtime, frames))) { + printk(KERN_ERR "ERROR COPY_FROM_USER\n"); + return -EFAULT; + } + return 0; +} + +static struct snd_pcm_ops davinci_pcm_ops = { + .open = davinci_pcm_open, + .close = davinci_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = davinci_pcm_hw_params, + .hw_free = davinci_pcm_hw_free, + .pointer = davinci_pcm_pointer, + .copy = davinci_pcm_copy, +}; + +static u64 davinci_pcm_dmamask = 0xffffffff; + +static irqreturn_t dm_vc_irq(int irq, void *sbst) +{ + struct snd_pcm_substream *substream = + (struct snd_pcm_substream *) sbst; + int fifo, diff, per_size, buf_size; + static int last_ptr; + + if (substream->runtime && substream->runtime->status && + snd_pcm_running(substream)) { + fifo = __raw_readl(IO_ADDRESS(0x01D0C028)); + fifo = (fifo & 0x1f00) >> 8; + if (fifo == 15) { + /* peripheral blocked! restart */ + __raw_writel(0, IO_ADDRESS(0x01D0C004)); + __raw_writel(0x5400, IO_ADDRESS(0x01D0C004)); + } + buf_size = substream->runtime->buffer_size; + per_size = substream->runtime->period_size; + for (; fifo < 0x10; fifo++) { + __raw_writew(local_buffer[pointer_sub++], + IO_ADDRESS(0x01D0C024)); + pointer_sub %= buf_size; + do { + diff = __raw_readl(IO_ADDRESS(0x01D0C00C)); + } while (!(diff & 0x8)); + } + if (last_ptr >= pointer_sub) + diff = buf_size + pointer_sub - last_ptr; + else + diff = pointer_sub - last_ptr; + if (diff >= per_size) { + snd_pcm_period_elapsed(substream); + last_ptr += per_size; + if (last_ptr >= buf_size) + last_ptr -= buf_size; + } + } else + last_ptr = 0; + return IRQ_HANDLED; +} + +static int davinci_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int tcr, tgcr; + struct snd_dma_buffer *buf; + struct snd_pcm_substream *substream; + struct clk *clk; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &davinci_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + buf = &substream->dma_buffer; + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->bytes = pcm_hardware_playback.buffer_bytes_max; + } + clk = clk_get(NULL, TIMER); + if (IS_ERR(clk)) + printk(KERN_ERR "ERROR getting timer\n"); + + /* Disabled, Internal clock source */ + __raw_writel(0, IO_ADDRESS(TIMER_BASE + TCR)); + + /* reset both timers, no pre-scaler for timer34 */ + tgcr = 0; + __raw_writel(tgcr, IO_ADDRESS(TIMER_BASE + TGCR)); + + /* Set both timers to unchained 32-bit */ + tgcr = TGCR_TIMMODE_32BIT_UNCHAINED << TGCR_TIMMODE_SHIFT; + __raw_writel(tgcr, IO_ADDRESS(TIMER_BASE + TGCR)); + + /* Unreset timers */ + tgcr |= (TGCR_UNRESET << TGCR_TIM12RS_SHIFT); + __raw_writel(tgcr, IO_ADDRESS(TIMER_BASE + TGCR)); + + /* Init both counters to zero */ + __raw_writel(0, IO_ADDRESS(TIMER_BASE + TIM12)); + tcr = __raw_readl(IO_ADDRESS(TIMER_BASE + TCR)); + + /* disable timer */ + tcr &= ~(TCR_ENAMODE_MASK << 6); + __raw_writel(tcr, IO_ADDRESS(TIMER_BASE + TCR)); + + /* reset counter to zero, set new period */ + __raw_writel(0, IO_ADDRESS(TIMER_BASE + TIM12)); + __raw_writel(0x8000, IO_ADDRESS(TIMER_BASE + PRD12)); + + /* Set enable mode */ + tcr |= TCR_ENAMODE_PERIODIC << 6; + __raw_writel(tcr, IO_ADDRESS(TIMER_BASE + TCR)); + +#if (TIMER_BASE == TIMER4_BASE) + __raw_writel(__raw_readl(IO_ADDRESS(0x01c40018)) | + 0x1000, IO_ADDRESS(0x01c40018)); +#endif + if (request_irq(TINT, dm_vc_irq, IRQF_TIMER, "VOICE-TOUT", substream)) + printk(KERN_ERR "%s ERROR requesting Interrupt\n", __func__); + return 0; +} + +struct snd_soc_platform davinci_soc_platform_copy = { + .name = "davinci-audio-copy", + .pcm_ops = &davinci_pcm_ops, + .pcm_new = davinci_pcm_new, +}; EXPORT_SYMBOL_GPL(davinci_soc_platform_copy); + +static int __init davinci_soc_copy_platform_init(void) +{ + return snd_soc_register_platform(&davinci_soc_platform_copy); +} +module_init(davinci_soc_copy_platform_init); + +static void __exit davinci_soc_copy_platform_exit(void) +{ + snd_soc_unregister_platform(&davinci_soc_platform_copy); +} +module_exit(davinci_soc_copy_platform_exit); + +MODULE_AUTHOR("bticino s.p.a."); +MODULE_DESCRIPTION("TI DAVINCI PCM copy_from_user module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0764944..cb7c2aa 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -29,5 +29,6 @@ struct davinci_pcm_dma_params { extern struct snd_soc_platform davinci_soc_platform; +extern struct snd_soc_platform davinci_soc_platform_copy; #endif diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad7f952..696e8b3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -801,7 +801,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } return 0; } - +#if 0 /* ASoC PCM operations */ static struct snd_pcm_ops soc_pcm_ops = { .open = soc_pcm_open, @@ -811,7 +811,7 @@ static struct snd_pcm_ops soc_pcm_ops = { .prepare = soc_pcm_prepare, .trigger = soc_pcm_trigger, }; - +#endif #ifdef CONFIG_PM /* powers down audio subsystem for suspend */ static int soc_suspend(struct device *dev) @@ -1306,6 +1306,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + struct snd_pcm_ops *soc_pcm_ops; rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); if (rtd == NULL) @@ -1335,19 +1336,25 @@ static int soc_new_pcm(struct snd_soc_device *socdev, dai_link->pcm = pcm; pcm->private_data = rtd; - soc_pcm_ops.mmap = platform->pcm_ops->mmap; - soc_pcm_ops.pointer = platform->pcm_ops->pointer; - soc_pcm_ops.ioctl = platform->pcm_ops->ioctl; - soc_pcm_ops.copy = platform->pcm_ops->copy; - soc_pcm_ops.silence = platform->pcm_ops->silence; - soc_pcm_ops.ack = platform->pcm_ops->ack; - soc_pcm_ops.page = platform->pcm_ops->page; + soc_pcm_ops = kmalloc(sizeof(struct snd_pcm_ops), GFP_KERNEL); + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_codec_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->mmap = platform->pcm_ops->mmap; + soc_pcm_ops->pointer = platform->pcm_ops->pointer; + soc_pcm_ops->ioctl = platform->pcm_ops->ioctl; + soc_pcm_ops->copy = platform->pcm_ops->copy; + soc_pcm_ops->silence = platform->pcm_ops->silence; + soc_pcm_ops->ack = platform->pcm_ops->ack; + soc_pcm_ops->page = platform->pcm_ops->page; if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); - + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); ret = platform->pcm_new(codec->card, codec_dai, pcm); if (ret < 0) { -- 1.7.0.4

2 1

Re: [alsa-devel] [LAD] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 14 Jul '10

14 Jul '10

On Tue, Jul 13, 2010 at 1:19 AM, <fons(a)kokkinizita.net> wrote: >On Mon, Jul 12, 2010 at 06:18:29PM -0700, Niels Mayer wrote: >> Speaking of making my brain hurt -- I really would not have done the >> meters as done in envy24control. I really hate the concept of >> emulating the look of real gear down to the level that you have to >> "quantize" to individual LED's... > > That is indeed quite absurd. Yes, the meters in envy24control -- http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=envy24control/… -- are absurd! What's even more absurd, is to have the benefits of the ice1712's hardware metering squandered by drawing thousands of rectangles per second (each individual "LED" on each meter is a separate x draw-rectangle call). The existing envy24control meters are an abomination of inefficiency! Especially when used with a remote X display. As a second patch (coming soon), I've rewritten the meters in a more sensible fashion, drawing a single rectangle to represent the instantaneous level (2-3 X-primitive draws per meter total and one blit, versus hundreds of draws and a blit per value change per meter in the original code). I'm also being a little more careful about not needing to redraw the entire meter for each change -- using the monotonicity of the audio signal being represented to optimize the way you draw it; not redrawing if nothing has changed, etc. And since the hardware metering already quantizes to 256 levels -- there's no point in re-quantizing those to individual "virtual LEDs" as is done w/ the current envy24control. The result so far are promising: Here's 'top' output (*) using the new envy24control. Note the load is lower, and 'envy24control' and 'X' both use much less CPU. In fact running this meter lowers the system load. whereas running the old envy24control raises it (and makes the fans run faster!). top - 21:30:13 up 11:30, 9 users, load average: 0.02, 0.11, 0.06 2664 root 20 0 544m 134m 34m S 2.6 3.4 26:03.51 X 9404 npm 20 0 192m 9340 6908 S 0.7 0.2 0:01.72 envy24control The new metering gives 2.6% CPU for X, and 0.7% CPU for the new-metered envy24control; Versus 6.3% CPU for X and 2.7% for the old metering in envy24control (the one from my first patch): top - 21:25:57 up 11:25, 9 users, load average: 0.10, 0.09, 0.03 2664 root 20 0 544m 134m 34m S 6.3 3.4 25:50.67 X 9398 npm 20 0 192m 9368 6940 S 2.7 0.2 0:16.81 envy24control Now granted this is not a scientific test, and "top" is not exactly the best way to gauge performance. But it does suggest that the computational complexity caused by the bad metering algorithm directly translates to higher-power consumption, higher loads, and poorer performance. Compared to the existing envy24control metering -- It would probably me more efficient to actually compute RMS levels off the audio stream -- or even hook up jkmeter through jack -- instead of using the hardware metering data through these computationally wasteful level meters. -- Niels http://nielsmayer.com

2 2

[alsa-devel] [PATCH] wm8988: remove a redundant snd_soc_unregister_codec call in wm8988_register
by Axel Lin 14 Jul '10

14 Jul '10

snd_soc_unregister_codec is called twice if snd_soc_register_dai fail. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8988.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 0417dae..19ad590 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -885,7 +885,6 @@ static int wm8988_register(struct wm8988_priv *wm8988, ret = snd_soc_register_dai(&wm8988_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); goto err_codec; } -- 1.7.0.4

1 0

[alsa-devel] ASoC updates for-2.6.36
by Mark Brown 13 Jul '10

13 Jul '10

The following changes since commit 66b47fdb851924956b6e4696fb43a3496ae2c462: ASoC: Implement WM8994 OPCLK support (2010-07-09 08:50:12 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Arnaud Patard (2): ASoC: Fix kirkwood i2s mono playback ASoC: kirkwood-i2s: Handle mute/unmute playback/record Joe Perches (2): ASoC: Remove unnecessary casts of private_data ASoC: Remove unnecessary casts of private_data Kuninori Morimoto (8): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM ASoC: fsi: modify format area definition on flags ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode Manuel Lauss (1): ASoC: au1x: fix section mismatch in psc-i2s.c Mark Brown (1): Merge branch 'for-2.6.35' into for-2.6.36 include/sound/sh_fsi.h | 44 +++++++- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 6 +- sound/soc/blackfin/bf5xx-tdm.c | 6 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 27 ++++-- sound/soc/kirkwood/kirkwood.h | 3 + sound/soc/sh/fsi.c | 213 ++++++++++++++++++++++++++----------- 8 files changed, 219 insertions(+), 84 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 13 Jul '10

13 Jul '10

The following changes since commit 1c5474a65bf15a4cb162dfff86d6d0b5a08a740c: Linux 2.6.35-rc5 (2010-07-12 14:55:33 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Kuninori Morimoto (2): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM Small, targetted fixes for a specific platform. sound/soc/sh/fsi.c | 27 +++++++++++---------------- 1 files changed, 11 insertions(+), 16 deletions(-)

2 1

[alsa-devel] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 13 Jul '10

13 Jul '10

I've been long annoyed by alsa-tools' envy24control (*) lack of peak-level indication in it's metering, and the non-implementation of its "Reset Peaks" button: https://bugzilla.redhat.com/show_bug.cgi?id=602903 I've fixed that annoyance. Now, a small white line appears above the highest level on each meter, and can be reset via "Reset Peaks" button. Original source: http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=envy24control/… Patch: http://nielsmayer.com/npm/602903.patch To run this version of envy24control grab the most recent stable release ( ftp://ftp.alsa-project.org/pub/tools/alsa-tools-1.0.23.tar.bz2 ) or git pull from trunk of the "alsa-tools" project ( http://alsa-project.org/main/index.php/GIT_Server ). After unpacking and assuming you've got the patch in ~/602903.patch do: >> cd alsa-tools-1.0.23 >> cat ~/602903.patch | patch -p1 it should give message "patching file envy24control/levelmeters.c" ... follow the directions to compile alsa-tools.... Can whoever's in charge of these things commit this change "upstream" and get it to propagate to the distros? Alternately, give me commit access and I'll do it myself, if people agree this patch is worth having. Next steps: (1) Think about building-in meter ballistics from Fons Adriaensen's jkmeter ( http://www.kokkinizita.net/linuxaudio/downloads/jkmeter-0.4.0.tar.bz2 ). Which helps implent Bob Katz' "K System" ( http://www.digido.com/level-practices-part-2-includes-the-k-system.html ). Although it probably most makes sense to implement such metering on the main "digital mixer" meter of envy24control, if it's not too inefficient, I'd want it available on all meters, despite the warning from Fons ( http://old.nabble.com/First-release-of-jkmeter-td18798950.html ). > This is the type of meter you want for live recording, > mixing and mastering. It probably makes no sense to > use it on all tracks of a DAW, where keeping digital > level within limits is the main purpose of metering. (2) Fix https://bugzilla.redhat.com/show_bug.cgi?id=602900 which causes 'envy24control' to crash when given "-D" argument. and behave incorrectly when given a correct non-numeric name for an audio device: e.g. >> envy24control --card M66 >> invalid card type (driver is USB-Audio) >> envy24control --card M66 --device M66 >> Segmentation fault (core dumped) (3) Apply more reasonable defaults and automatically size to correct dimensions. I use, for example: >> envy24control --card 2 --outputs 4 --input 4 --pcm_output 8 --view_spdif_playback --midichannel 1 --midienhanced --window_width 1275 --tall_eq_mixer_heights 1 & (4) Add midi output, in addition to midi input, which would monitor the hardware peak metering from the ice1712 ("amixer -c M66 cget iface=PCM,name='Multi Track Peak',numid=45") and output a MIDI note-on on the corresponding MIDI channel whenever a specified threshold value was exceeded. This could be used to implement "automatic gain control" in association with the midi-controllable ADC gain provided by envy24control. Niels http://nielsmayer.com PS: (*) http://alsa.opensrc.org/index.php/Envy24control Cards supported by this fix: * M Audio Delta 1010 * M Audio Delta 1010LT * M Audio Delta DiO 2496 * M Audio Delta 66 * M Audio Delta 44 * M Audio Delta 410 * M Audio Audiophile 2496 * TerraTec EWS 88MT * TerraTec EWS 88D * TerraTec EWX 24/96 * TerraTec DMX 6Fire * TerraTec Phase 88 * Hoontech SoundTrack DSP 24 * Hoontech SoundTrack DSP 24 Value * Hoontech SoundTrack DSP 24 Media 7.1 * Event Electronics, EZ8 * Digigram VX442 * Lionstracs, Mediastaton * Terrasoniq TS 88

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[alsa-devel] [PATCH] ASoC: au1x: fix section mismatch in psc-i2s.c
by Manuel Lauss 13 Jul '10

13 Jul '10

Annotate platform probe callback with __devinit instead of plain __init. Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- sound/soc/au1x/psc-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 495be6e..24454c9 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -300,7 +300,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = { }; EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev) +static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) { struct resource *r; unsigned long sel; -- 1.7.1.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 2

[alsa-devel] [PATCH 34/36] sound/soc/blackfin: Remove unnecessary casts of private_data
by Joe Perches 13 Jul '10

13 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/blackfin/bf5xx-ac97.c | 6 ++---- sound/soc/blackfin/bf5xx-tdm.c | 6 ++---- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 523b7fc..c0eba51 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -255,8 +255,7 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) @@ -271,8 +270,7 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) static int bf5xx_ac97_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 4b36012..24c1426 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -210,8 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; if (!dai->active) return 0; @@ -225,8 +224,7 @@ static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) static int bf5xx_tdm_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; if (!dai->active) return 0; -- 1.7.1.337.g6068.dirty

3 2

[alsa-devel] [patch 0/2] Kirkwood audio fixes
by arnaud.patard＠rtp-net.org 13 Jul '10

13 Jul '10

Hi, While working on the t5325 audio support, I found two small buglets in the kirkwood ASoC support. The first one is aiming at fixing mono audio playback and should fix mono playback for all platforms. The second is adding support for muting/unmuting kirkwood audio. It's mainly usefull for the t5325 which has a bootloader playing with this (hardware default is not muted). Arnaud

3 5

[alsa-devel] [PATCH 1/6 v3] ASoC: fsi: modify format area definition on flags
by Kuninori Morimoto 13 Jul '10

13 Jul '10

There is no necessity that each bit in this area has the meaning. This patch modify it to sequence number Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 o don't use << 0 shift 2/6 - 6/6 doesn't depend on 1/6 patch include/sound/sh_fsi.h | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index c022736..3ecdb96 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -55,12 +55,12 @@ #define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) #define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) -#define SH_FSI_FMT_MONO (1 << 0) -#define SH_FSI_FMT_MONO_DELAY (1 << 1) -#define SH_FSI_FMT_PCM (1 << 2) -#define SH_FSI_FMT_I2S (1 << 3) -#define SH_FSI_FMT_TDM (1 << 4) -#define SH_FSI_FMT_TDM_DELAY (1 << 5) +#define SH_FSI_FMT_MONO 0 +#define SH_FSI_FMT_MONO_DELAY 1 +#define SH_FSI_FMT_PCM 2 +#define SH_FSI_FMT_I2S 3 +#define SH_FSI_FMT_TDM 4 +#define SH_FSI_FMT_TDM_DELAY 5 #define SH_FSI_IFMT_TDM_CH(x) \ (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) -- 1.7.0.4

3 2

[alsa-devel] [PATCH 0/2] ASoC: fsi: fixup patches
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These are v2 bug fix patches for FSI driver. Kuninori Morimoto (2): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM v1 -> v2 It is bug fix only Best regards -- Kuninori Morimoto

3 4

[alsa-devel] [PATCH 1/4] ASoC: DaVinci: remove requirement that dma_params is 1st in structure
by Troy Kisky 13 Jul '10

13 Jul '10

Remove requirement that dma_params is 1st in the structures davinci_audio_dev and davinci_mcbsp_dev. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 6 +----- sound/soc/davinci/davinci-mcasp.c | 1 + sound/soc/davinci/davinci-mcasp.h | 5 ----- sound/soc/davinci/davinci-pcm.c | 7 ++++--- 4 files changed, 6 insertions(+), 13 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae7070..9e69a4e 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -98,11 +98,6 @@ enum { }; struct davinci_mcbsp_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -547,6 +542,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; + davinci_i2s_dai.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a..5023915 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -907,6 +907,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9d179cc..582c924 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,11 +39,6 @@ enum { }; struct davinci_audio_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index c73a915..c2c2e85 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -238,10 +238,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; - struct davinci_pcm_dma_params *params = &pa[substream->stream]; - if (!params) + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *params; + if (!pa) return -ENODEV; + params = &pa[substream->stream]; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ -- 1.5.6.3

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[alsa-devel] [PATCH] ASoC: TWL4030: Fix for digital loopback gain range
by Peter Ujfalusi 13 Jul '10

13 Jul '10

When the gain is configured using dB value it was not possible to use -24dB since the loopback got muted instead of -24dB. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 8d36bfa..d7f0048 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -540,10 +540,11 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); -/* Digital bypass gain, 0 mutes the bypass */ +/* Digital bypass gain, mute instead of -30dB */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { - TLV_DB_RANGE_HEAD(2), - 0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1), + TLV_DB_RANGE_HEAD(3), + 0, 1, TLV_DB_SCALE_ITEM(-3000, 600, 1), + 2, 3, TLV_DB_SCALE_ITEM(-2400, 0, 0), 4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0), }; -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: TWL4030: DAC power optimization
by Peter Ujfalusi 13 Jul '10

13 Jul '10

Restructure the DAPM connections in order to enable only the needed DAC (out of four in twl4030 series). I need to keep the 'AIF Enable' supply connected to the L2/R2 digital path, since the digital loopback needs AIF and APLL running. If no valid route available, than none of the DAC will be powered, but the AIF and APLL is going to be enabled. Furthermore, if only one audio path have valid route, than only the corresponding DAC will be powered. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 15 +++++++++------ 1 files changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index d7f0048..6fd6d0b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1460,8 +1460,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Supply for the digital part (APLL) */ {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, - {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, - {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"DAC Left1", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"DAC Left2", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, @@ -1532,10 +1535,10 @@ static const struct snd_soc_dapm_route intercon[] = { /* outputs */ /* Must be always connected (for AIF and APLL) */ - {"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"}, + {"Virtual HiFi OUT", NULL, "DAC Left1"}, + {"Virtual HiFi OUT", NULL, "DAC Right1"}, + {"Virtual HiFi OUT", NULL, "DAC Left2"}, + {"Virtual HiFi OUT", NULL, "DAC Right2"}, /* Must be always connected (for APLL) */ {"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"}, /* Physical outputs */ -- 1.7.1.1

3 2

[alsa-devel] [PATCH 0/8] ASoC: fsi: fixup patches
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These patches are fixup for FSI driver Kuninori Morimoto (8): ASoC: fsi: modify format area definition on flags ASoC: fsi: fixup clock inversion operation ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: modify wrong value setting order of TDM ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode Some of these are not so important for now. But it is used from "FSI - HDMI" patches which I will send in future

2 13

[alsa-devel] issue with TWL4030
by Grazvydas Ignotas 13 Jul '10

13 Jul '10

Hi, there seems to be a problem with TWL4030 hw_param programming, which is easy to reproduce over OSS emulation with this simple program: int oss_dev = open("/dev/dsp", O_WRONLY); int stereo = 1, bits = 16, rate = 44100; ioctl(oss_dev, SNDCTL_DSP_SETFMT, &bits); ioctl(oss_dev, SNDCTL_DSP_SPEED, &rate); ioctl(oss_dev, SNDCTL_DSP_STEREO, &stereo); write(oss_dev, somebuff, sizeof(somebuff)); The above program ends up playing at 8kHz, instead of expected 44kHz. I did some call tracing and got this: [ 1092.246032] soc_pcm_hw_params, rate 8000 [ 1092.250122] twl4030_hw_params [ 1092.253234] set rate 8000 [ 1092.256103] configured 1 [ 1092.259368] soc_pcm_hw_params, rate 8000 [ 1092.263427] twl4030_hw_params [ 1092.266845] soc_pcm_hw_params, rate 44100 [ 1092.270965] twl4030_hw_params [ 1094.459106] twl4030_shutdown [ 1094.462036] configured 0 So it looks like twl4030->configured is the culprit of this, removing it solves the problem, but I guess that will break something else. Any ideas how to properly fix this? Gražvydas

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[alsa-devel] [PATCH 0/6] ASoC: fsi: modify patch
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These are v2 patches for FSI driver almost all are same as v1 Kuninori Morimoto (6): ASoC: fsi: modify format area definition on flags ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode v1 -> v2 o modiry straight start number as 0 o based on bug fix patches Best regards -- Kuninori Morimoto

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[alsa-devel] [PATCH 0/3] further BKL removal
by Arnd Bergmann 13 Jul '10

13 Jul '10

With the ioctl, block, tty, vfs and llseek series on their way into linux-next, these three patches are attacking the hardest remaining issues. If we get these ready for 2.6.36, the kernel should be almost usable with the BKL disabled. In all three cases, I'm cheating a bit, because the BKL still remains lurking in the dark corners of the three subsystems (i810/i830 for drm, OSS for sound and lockd for fs/locks.c). Arnd Cc: Christoph Hellwig <hch(a)lst.de> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Frederic Weisbecker <fweisbec(a)gmail.com> Cc: Ingo Molnar <mingo(a)redhat.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: "J. Bruce Fields" <bfields(a)fieldses.org> Cc: John Kacur <jkacur(a)redhat.com> Cc: Matthew Wilcox <willy(a)linux.intel.com> Cc: Miklos Szeredi <mszeredi(a)suse.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Trond Myklebust <trond.myklebust(a)fys.uio.no> Cc: alsa-devel(a)alsa-project.org Cc: linux-fsdevel(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Arnd Bergmann (2): drm: kill BKL from common code sound: push BKL into open functions Matthew Wilcox (1): Remove BKL from fs/locks.c arch/um/drivers/hostaudio_kern.c | 6 ++ drivers/gpu/drm/drm_drv.c | 4 +- drivers/gpu/drm/drm_fops.c | 23 ++++---- drivers/gpu/drm/i810/i810_dma.c | 44 +++++++++----- drivers/gpu/drm/i810/i810_drv.c | 2 +- drivers/gpu/drm/i810/i810_drv.h | 1 + drivers/gpu/drm/i830/i830_dma.c | 42 +++++++++---- drivers/gpu/drm/i830/i830_drv.c | 2 +- drivers/gpu/drm/i830/i830_drv.h | 1 + fs/Kconfig | 4 + fs/locks.c | 116 ++++++++++++++++++++++-------------- include/drm/drmP.h | 2 +- sound/core/hwdep.c | 14 +++- sound/core/oss/mixer_oss.c | 19 ++++--- sound/oss/au1550_ac97.c | 26 +++++--- sound/oss/dmasound/dmasound_core.c | 28 +++++++-- sound/oss/msnd_pinnacle.c | 10 ++- sound/oss/sh_dac_audio.c | 9 ++- sound/oss/soundcard.c | 20 ++++--- sound/oss/swarm_cs4297a.c | 17 +++++- sound/oss/vwsnd.c | 8 +++ sound/sound_core.c | 6 +-- 22 files changed, 267 insertions(+), 137 deletions(-)

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[alsa-devel] [PATCH 36/36] sound/core: Remove unnecessary casts of private_data
by Joe Perches 12 Jul '10

12 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/core/oss/mixer_oss.c | 22 +++++++++++----------- sound/core/pcm.c | 3 +-- sound/core/timer.c | 4 ++-- 3 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index f50ebf2..86afb13 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -77,7 +77,7 @@ static int snd_mixer_oss_release(struct inode *inode, struct file *file) struct snd_mixer_oss_file *fmixer; if (file->private_data) { - fmixer = (struct snd_mixer_oss_file *) file->private_data; + fmixer = file->private_data; module_put(fmixer->card->module); snd_card_file_remove(fmixer->card, file); kfree(fmixer); @@ -368,7 +368,7 @@ static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int static long snd_mixer_oss_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - return snd_mixer_oss_ioctl1((struct snd_mixer_oss_file *) file->private_data, cmd, arg); + return snd_mixer_oss_ioctl1(file->private_data, cmd, arg); } int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg) @@ -582,7 +582,7 @@ static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *left, int *right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; *left = *right = 100; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { @@ -691,7 +691,7 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int left, int right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); @@ -740,7 +740,7 @@ static int snd_mixer_oss_get_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -753,7 +753,7 @@ static int snd_mixer_oss_get_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -766,7 +766,7 @@ static int snd_mixer_oss_put_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], active, active, 0); return 0; @@ -776,7 +776,7 @@ static int snd_mixer_oss_put_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], active, active, 1); return 0; @@ -813,7 +813,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -861,7 +861,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -925,7 +925,7 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl static void snd_mixer_oss_slot_free(struct snd_mixer_oss_slot *chn) { - struct slot *p = (struct slot *)chn->private_data; + struct slot *p = chn->private_data; if (p) { if (p->allocated && p->assigned) { kfree(p->assigned->name); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cbe815d..de5a0eb 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -358,8 +358,7 @@ static void snd_pcm_stream_proc_info_read(struct snd_info_entry *entry, static void snd_pcm_substream_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - snd_pcm_proc_info_read((struct snd_pcm_substream *)entry->private_data, - buffer); + snd_pcm_proc_info_read(entry->private_data, buffer); } static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, diff --git a/sound/core/timer.c b/sound/core/timer.c index 13afb60..5ab1014 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -962,7 +962,7 @@ static int snd_timer_s_start(struct snd_timer * timer) struct snd_timer_system_private *priv; unsigned long njiff; - priv = (struct snd_timer_system_private *) timer->private_data; + priv = timer->private_data; njiff = (priv->last_jiffies = jiffies); if (priv->correction > timer->sticks - 1) { priv->correction -= timer->sticks - 1; @@ -981,7 +981,7 @@ static int snd_timer_s_stop(struct snd_timer * timer) struct snd_timer_system_private *priv; unsigned long jiff; - priv = (struct snd_timer_system_private *) timer->private_data; + priv = timer->private_data; del_timer(&priv->tlist); jiff = jiffies; if (time_before(jiff, priv->last_expires)) -- 1.7.1.337.g6068.dirty

1 0

[alsa-devel] [PATCH 35/36] sound/soc/davinci: Remove unnecessary casts of private_data
by Joe Perches 12 Jul '10

12 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/davinci/davinci-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2dc406f..def454e 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -800,7 +800,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; - iram_dma = (struct snd_dma_buffer *)buf->private_data; + iram_dma = buf->private_data; if (iram_dma) { sram_free(iram_dma->area, iram_dma->bytes); kfree(iram_dma); -- 1.7.1.337.g6068.dirty

1 0

[alsa-devel] How does struct snd_pcm_hardware relate to the actual hardware
by Adam Rosenberg 12 Jul '10

12 Jul '10

I am using DMA to transfer data to SPORT1 on my Blackfin BF537. I am using a CS42448 CODEC that has 8 DACs. I am using TDM mode and have set the following: DMA Buffer Size For Each SPORT Data Line (Rx or Tx) = 131072 bytes DMA Periods = 8 static struct snd_pcm_hardware snd_pcm_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME), .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, .rate_max = 48000, .buffer_bytes_max = 32768, .channels_min = 2, .channels_max = 2, .period_bytes_min = 8, .period_bytes_max = 4096, .periods_min = 8, .periods_max = 8, }; Everytime I receive a DMA interrupt from the SPORT driver I call snd_pcm_period_elapsed for all of the substreams that are currently active. In the PCM Copy callback I calculate the position within the DMA buffer to write to based on the TDM slot for the channel and the position passed into the function. The audio is not being placed into the DMA buffer properly. The audio plays faster than it should and has a constant stutter. I have read the "writing an ALSA driver" guide multiple times but it does not explain the relationship between struct snd_pcm_hardware, snd_pcm_period_elapsed, and the copy callback clearly enough for me to understand what I am doing wrong. Please explain how these interact in more detail to help me fix this problem. I can also send the full source code to the CS42448 driver I am working on if more details are needed. Thank you, Adam

2 2

[alsa-devel] [PATCH] ASoC: SAMSUNG: Add Aquila sound driver
by Chanwoo Choi 12 Jul '10

12 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2Sv5 to support sound. As there is currently no driver for the I2Sv5, the code to use I2Sv5 is deleted on this patch. After I2Sv5 driver is posted in a few weeks, I will post incremental patch which some codes for using I2Sv5. The owner of I2Sv5 said that he will post I2Sv5 driver after a few weeks. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 298 +++++++++++++++++++++++++++++++++++++ 3 files changed, 308 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..0c5d8a5 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,11 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..706087f --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,298 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.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/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_codec *codec) +{ + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_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->dai->codec_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai voice_dai = { + .name = "Voice", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + /* After I2Sv5 driver is posted, cpu_dai will be enabled. */ + /* .cpu_dai = &s5p_i2s_dai[0], */ + .codec_dai = &wm8994_dai[0], + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai = &voice_dai, + .codec_dai = &wm8994_dai[1], + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .platform = &s3c24xx_soc_platform, + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static struct snd_soc_device aquila_snd_devdata = { + .card = &aquila, + .codec_dev = &soc_codec_dev_wm8994, +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&voice_dai); + if (ret) + return ret; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + platform_set_drvdata(aquila_snd_device, &aquila_snd_devdata); + aquila_snd_devdata.dev = &aquila_snd_device->dev; + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

4 12

[alsa-devel] Help required regarding Capture & palyback code for ALSA
by Reena Chauhan 12 Jul '10

12 Jul '10

Hii I m new to ALSA programming .. i have follow the Guide "Introduction to Sound Programming with ALSA" to make code for capture & playback.when i have compiled the code for capture i dont know from where to give the input. in the Guide it is written that "If you have a microphone connected to your sound card, use a mixer program to set the recording source and level. Alternatively, you can run a CD player program and set the recording source to CD. Try running Listing 4 and redirecting the output to a file" i dont know how to use the mixer program .. but I dont know how to do it to capture some input given by me & playback it .. thanx & Regrads -- Reena Chauhan Scientist-C Application & Baseband Platform, SBSG ,DEAL,Raipur Road Dehradun-248001

1 0

Re: [alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Raymond Yau 10 Jul '10

10 Jul '10

>> It might be. It is important not to mix VT1708 and VT1708S. For VT1708S, the following DACs are used in code: /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: spec->multiout.dac_nids[i] = 0x25; break; } >> So 0x12 is not used for any of the surround outputs. I think your testing config is not compiled with smart 5.1 requirement Refer to smart51_get() , this function exepct red jack and blue jack are configued to Output But you still plugged in Mic into the pink jack according to your photo. Your speakers have to plugged in both pink and blue jacks

2 1

[alsa-devel] Kirkwood SoC - AVB support questions ..?
by errordeveloper＠gmail.com 10 Jul '10

10 Jul '10

Hello dear ALSA developer list members, I have a question regarding hardware features of Marvell Kirkwood chip. I is known that the NIC core on the chip includes support for Ethernet AV Bridging, which is still a draft standard ..but i suppose that Marvell released this chip with quite functional AVB implementation. grepping the sk98 driver code actually shows numerous lines with AVB as well as PTP, so i suppose they have implemented IOCTL in the sk98 driver ..i'm still doubtfull about Linux kernel PTP implementation ..is there now in some form or not? Did any of you have a look at AVB ? how do you think the driver for it could fit into Linux Netdev and ALSA? How are the CobraNet and EtherSound drivers designed, has anyone used those cards ?? (one was from DigiGram and another from Audio Science IIRC) Regards, -- Ilya Dmitrichenko

2 2

[alsa-devel] Envy24HT (VT1721) 192kHz digital out capable?
by Val Krutov 09 Jul '10

09 Jul '10

Could anybody get 192kHz out of VT1721 on spdif output working correctly? If yes please name your card. I have two cards on VT1721 chip, Audiotrak HD2 Advance DE and Chaintech AV-710. Both can work properly only up to 96kHz. The 192kHz samples play at 96kHz according to my receiver display sample rate input but alsamixer reports 192kHz which is incorrect. I believe my receiver' indication rather than alsamixer. Here is my register settings output for Chaintech AV-710 at playng 192kHz sample and everything there looks right to me but my receiver reports it is 96kHz. I just modified the card's ID and others eeprom settings as otherwise spdif can work only at 44.1 and 48 kHz. ICEnsemble ICE1724 at 0x2c40, irq 26 EEPROM: Subvendor : 0x12142419 Size : 28 bytes Version : 1 System Config : 0x2 ACLink : 0x82 I2S : 0x78 S/PDIF : 0xc1 GPIO direction : 0xff GPIO mask : 0xff GPIO state : 0xff Extra #18 : 0x1 Extra #19 : 0x0 Extra #20 : 0x0 Extra #21 : 0x0 Extra #22 : 0x0 Extra #23 : 0x0 Extra #24 : 0x0 Extra #25 : 0x0 Extra #26 : 0x0 Extra #27 : 0x0 Registers: PSDOUT03 : 0x00000000 CCS00 : 0x00 CCS01 : 0xa0 CCS02 : 0x00 CCS03 : 0x00 CCS04 : 0x02 CCS05 : 0x82 CCS06 : 0x78 CCS07 : 0xc1 CCS08 : 0x00 CCS09 : 0x00 CCS0a : 0x00 CCS0b : 0x00 CCS0c : 0x00 CCS0d : 0x0a CCS0e : 0x00 CCS0f : 0x00 CCS10 : 0xa0 CCS11 : 0x1b CCS12 : 0x00 CCS13 : 0x80 CCS14 : 0x00 CCS15 : 0xaa CCS16 : 0xff CCS17 : 0x00 CCS18 : 0xff CCS19 : 0x00 CCS1a : 0x00 CCS1b : 0x00 CCS1c : 0x00 CCS1d : 0x00 CCS1e : 0x2a CCS1f : 0x00 MT00 : 0x00 MT01 : 0x0e MT02 : 0x08 MT03 : 0x08 MT04 : 0x00 MT05 : 0x00 MT06 : 0x00 MT07 : 0x00 MT08 : 0x00 MT09 : 0x00 MT0a : 0x00 MT0b : 0x00 MT0c : 0x00 MT0d : 0x00 MT0e : 0x00 MT0f : 0x00 MT10 : 0x00 MT11 : 0x00 MT12 : 0x00 MT13 : 0x00 MT14 : 0x00 MT15 : 0x00 MT16 : 0x00 MT17 : 0x00 MT18 : 0x80 MT19 : 0x00 MT1a : 0x00 MT1b : 0x00 MT1c : 0x00 MT1d : 0x00 MT1e : 0x00 MT1f : 0x00 MT20 : 0x00 MT21 : 0x00 MT22 : 0x00 MT23 : 0x00 MT24 : 0x00 MT25 : 0x00 MT26 : 0x00 MT27 : 0x00 MT28 : 0x00 MT29 : 0x00 MT2a : 0x00 MT2b : 0x00 MT2c : 0x00 MT2d : 0x00 MT2e : 0x00 MT2f : 0x00 MT30 : 0x00 MT31 : 0x00 MT32 : 0x00 MT33 : 0x00 MT34 : 0x00 MT35 : 0x00 MT36 : 0x00 MT37 : 0x00 MT38 : 0x00 MT39 : 0x00 MT3a : 0x00 MT3b : 0x00 MT3c : 0x00 MT3d : 0x60 MT3e : 0x00 MT3f : 0x00 MT40 : 0x80 MT41 : 0x48 MT42 : 0x98 MT43 : 0x32 MT44 : 0xdf MT45 : 0xed MT46 : 0xff MT47 : 0x3f MT48 : 0x00 MT49 : 0x00 MT4a : 0x00 MT4b : 0x00 MT4c : 0x00 MT4d : 0x00 MT4e : 0x00 MT4f : 0x00 MT50 : 0x00 MT51 : 0x00 MT52 : 0x00 MT53 : 0x00 MT54 : 0x00 MT55 : 0x00 MT56 : 0x00 MT57 : 0x00 MT58 : 0x00 MT59 : 0x00 MT5a : 0x00 MT5b : 0x00 MT5c : 0x00 MT5d : 0x00 MT5e : 0x00 MT5f : 0x00 MT60 : 0x00 MT61 : 0x00 MT62 : 0x00 MT63 : 0x00 MT64 : 0x00 MT65 : 0x00 MT66 : 0x00 MT67 : 0x00 MT68 : 0x00 MT69 : 0x00 MT6a : 0x00 MT6b : 0x00 MT6c : 0x00 MT6d : 0x00 MT6e : 0x00 MT6f : 0x00 MT70 : 0x00 MT71 : 0x00 MT72 : 0x00 MT73 : 0x00 MT74 : 0x00 MT75 : 0x00 MT76 : 0x00 MT77 : 0x00 Valera

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Re: [alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Viliam Kubis 09 Jul '10

09 Jul '10

It might be. It is important not to mix VT1708 and VT1708S. For VT1708S, the following DACs are used in code: /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: spec->multiout.dac_nids[i] = 0x25; break; } So 0x12 is not used for any of the surround outputs. On Fri, 09 Jul 2010 03:32:13 +0200, Raymond Yau <superquad.vortex2(a)gmail.com> wrote: >> 2010/7/8 Clemens Ladisch <clemens(a)ladisch.de> >> >>> Viliam Kubis wrote: >>> > Hello, for quite some time the "Smart 5.1" function in alsa is not >>> working >>> > properly when motherboard has only 3 audio jacks. >>> > >>> > speaker-test -c 4 -t wav -Dsurround40 >>> > >>> > Channels count (4) not available for playbacks: Invalid argument >>> >>> Please try the patch below. >>> >>> (This function uses 0x12 and 0x13 as the NIDs for the center/LFE and >>> side DACs. Shouldn't these be 0x24 and 0x25, as per the datasheet?) >>> >> >> Node 0x12 seem to be for digital out since it connected to node 0x20 >> >> I guess we still lack the info about the retasking of pink and blue >> jack to >> the surround output >> >> >> Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital >> Control: name="IEC958 Playback Con Mask", index=0, device=0 >> Control: name="IEC958 Playback Pro Mask", index=0, device=0 >> Control: name="IEC958 Playback Default", index=0, device=0 >> Control: name="IEC958 Playback Switch", index=0, device=0 >> Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 >> Device: name="VT1708S Digital", type="SPDIF", device=1 >> Converter: stream=5, channel=0 >> Digital: Enabled >> Digital category: 0x0 >> PCM: >> rates [0x5e0]: 44100 48000 88200 96000 192000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> >> >> Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital >> Pincap 0x00000010: OUT >> Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel >> Conn = ATAPI, Color = Black >> DefAssociation = 0xf, Sequence = 0x0 >> Misc = NO_PRESENCE >> Pin-ctls: 0x40: OUT >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Connection: 1 >> 0x12 >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

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Re: [alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Viliam Kubis 09 Jul '10

09 Jul '10

Hello, I have patched the vt1708s_auto_fill_dac_nids function instead. I have left values 0x11 and 0x12 for surround and cen/lfe, althrough it might be 0x11 and 0x24 for vt1708S as originally specified in vt1708s_auto_fill_dac_nids switch statement. The whole function looks like this: static int vt1708S_auto_fill_dac_nids(struct via_spec *spec, const struct auto_pin_cfg *cfg) { int i; hda_nid_t nid; spec->multiout.num_dacs = cfg->line_outs; spec->multiout.dac_nids = spec->private_dac_nids; for (i = 0; i < 4; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: printk(KERN_INFO "***PATCH CREATING SURROUND DAC_NID"); spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: spec->multiout.dac_nids[i] = 0x25; break; } } } if (cfg->line_outs == 1) { printk(KERN_INFO "***PATCH 08/07/10"); spec->multiout.num_dacs = 3; spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x12; } return 0; } The patch is in effect since the following is visible in dmesg after the driver is loaded: ***PATCH 08/07/10 ***PATCH CREATING Surround CONTROL ***PATCH HDA max_channels = 6 ***PATCH playback stream channels_max = 6 Smart 5.1 Patch installed NEW ***PATCH Setting 64 mode [out=64] for pin index 0 nid 26 ***PATCH Setting 64 mode [out=64] for pin index 1 nid 30 ***PATCH Setting 64 mode [out=64] for pin index 2 nid 27 printk(KERN_INFO "***PATCH CREATING SURROUND DAC_NID"); This line does not print since cfg->line_out_pins[AUTO_SEQ_SURROUND] is not filled at that time, but we fix it in the new if block. Surround still does not work (still the same, the rear speakers are soundless). On Thu, 08 Jul 2010 08:33:34 +0200, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> _auto_fill_dac_nids > -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 09 Jul '10

09 Jul '10

The following changes since commit db059c0f6edec0b1deba665e5853f4ed829003b7: ASoC: Automatically manage ALC coefficients for WM8960 (2010-07-06 08:46:10 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Mark Brown (2): ASoC: Include WM8994 GPIO and interrupt registers in codec_reg ASoC: Implement WM8994 OPCLK support Maurus Cuelenaere (1): ASoC: Invert speaker enabling behaviour in SmartQ sound driver Raffaele Recalcati (3): ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S) ASoC: DaVinci: Added selection of clk input pin for McBSP ASoC: DaVinci: More accurate continuous serial clock for McBSP (I2S) arch/arm/mach-davinci/include/mach/asp.h | 48 +++++++++ sound/soc/codecs/wm8994.c | 43 +++++++++ sound/soc/codecs/wm8994.h | 3 + sound/soc/davinci/davinci-i2s.c | 153 +++++++++++++++++++++++++++--- sound/soc/davinci/davinci-i2s.h | 5 + sound/soc/s3c24xx/smartq_wm8987.c | 1 - 6 files changed, 238 insertions(+), 15 deletions(-)

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[alsa-devel] Question about SuperH HDMI sound
by Kuninori Morimoto 09 Jul '10

09 Jul '10

Dear Mark Cc: Guennadi In these days, Guennadi's "SuperH HDMI support patches" which support HDMI video output were posted to ML. And I created HDMI sound support for it in my local environment. It works and I would like to post this patch. Thus, I would like to ask you about HDMI "sound" support file position. HDMI "video" support file is under "drivers/video/sh_mobile_hdmi.c" Should I add sound support in it ? or under "sound/soc/codecs" ? Best regards -- Kuninori Morimoto

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[alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Viliam Kubis 09 Jul '10

09 Jul '10

Hello, for quite some time the "Smart 5.1" function in alsa is not working properly when motherboard has only 3 audio jacks. Me and Raymond on alsa-bugtracker have been trying to solve the issue for a long time but were unsuccessful. Please take a look at this alsa bug ticket: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4895 It contains A LOT of info on the problem + steps that we tried, which included patching the code. Synthom : NO "Surround" , "Center" and "LFE" Control and switch with VT1708S on a motherboard with 3 audio jacks Also NO surround sound. speaker-test -c 4 -t wav -Dsurround40 Channels count (4) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument Please advise if you can. Thank you Best Regards Viliam Kubis -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

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[alsa-devel] wm8776_dai vs. cs4270_dai
by Timur Tabi 09 Jul '10

09 Jul '10

Mark, In wm8776.c, the DAI structure is defined as an array of two elements, one for playback and one for capture: struct snd_soc_dai_driver wm8776_dai[] = { { .name = "WM8776 Playback", .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = WM8776_RATES, .formats = WM8776_FORMATS, }, .ops = &wm8776_dac_ops, }, { .name = "WM8776 Capture", .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = WM8776_RATES, .formats = WM8776_FORMATS, }, .ops = &wm8776_adc_ops, }, }; In my CS4270 driver, I have these combined into one structure: struct snd_soc_dai_driver cs4270_dai = { .name = "cs4270", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 4000, .rate_max = 216000, .formats = CS4270_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 4000, .rate_max = 216000, .formats = CS4270_FORMATS, }, .ops = &cs4270_dai_ops, }; Should the CS4270 driver be changed to use two structures?

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[alsa-devel] Problems running "amixer"
by Adish Kuvelker 08 Jul '10

08 Jul '10

Hello, I have cross-compiled alsa-utils (alsa-utils-1.0.13) and alsa-lib (alsa-lib-1.0.13). But when I ran the "amixer" I found some errors. I am pasting the same for your reference. # ./amixer ALSA lib control.c:910:(snd_ctl_open_noupdate) Invalid CTL default amixer: Mixer attach default error: No such file or directory The snddevices script creates a node "/dev/snd" and the device number of these nodes matches with the ones registered in "/proc/asound". Can anyone point me to the reason why the above mentioned error occurs? If any more information is required over the way my sound nodes are registered with the system I shall post it. Regards; Aadish

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[alsa-devel] Still unhappy with removal of DXS controls
by Soeren D. Schulze 08 Jul '10

08 Jul '10

[Sorry if this is a duplicate -- I was too stupid to subscribe the first time and my mail hasn't been moderated yet, apparently.] Me again... Referring to the issue I described on 2010-03-28 [http://www.spinics.net/lists/alsa-devel/msg32731.html], the situation hasn't changed since then. In short, I *really* need to lower the DXS volumes in order to play sound properly. If I don't do it, I get massive distortion. It's not a matter of the output stage, but of the input stage. Lowering PCM or Master doesn't help anything, but using softvol in mplayer does (I need to get down to 20%, so I effectively hear 14-bit audio, nice...). Back in March, Clemens told me to use hwmixvolume. It's nice, I can now access what I could previously access in alsamixer. But when I close the audio stream and start a new one, the volume is reset to 100% -- blowing my ears with loud and distorted sound. So my solution for now is: stick with Linux 2.6.30... Please, even if the solution with the DXS controls in alsamixer didn't comply with the ALSA policies, *it did work*! The current one *does not*. Maybe make the maximum (or default?) DXS volume available as a mixer control or as a module option or something -- whatever you consider the best solution. But the current state is just not acceptable. Thanks Sören

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Add Aquila sound driver
by 최찬우 08 Jul '10

08 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2Sv5 to support sound. As there is currently no driver for the I2Sv5, the code to use I2Sv5 is deleted on this patch. After I2Sv5 driver is posted in a few weeks, I will post incremental patch which some codes for using I2Sv5. The owner of I2Sv5 said that he will post I2Sv5 driver after a few weeks. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 306 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..0c5d8a5 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,11 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..5caeedd --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,296 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.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/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +extern struct snd_soc_platform s3c24xx_soc_platform; +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_kcontrol_new aquila_controls[] = { +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, +}; + +static int aquila_wm8994_init(struct snd_soc_codec *codec) +{ + int ret; + unsigned int reg; + + /* add aquila specific kcontorls */ + ret = snd_soc_add_controls(codec, aquila_controls, + ARRAY_SIZE(aquila_controls)); + + if (ret < 0) + return ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_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->dai->codec_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai voice_dai = { + .name = "Voice", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + /* After I2Sv5 driver is posted, cpu_dai will is enabled. */ + /* .cpu_dai = &s5p_i2s_dai[0], */ + .codec_dai = &wm8994_dai[0], + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai = &voice_dai, + .codec_dai = &wm8994_dai[1], + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .platform = &s3c24xx_soc_platform, + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static struct snd_soc_device aquila_snd_devdata = { + .card = &aquila, + .codec_dev = &soc_codec_dev_wm8994, +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&voice_dai); + if (ret) + return ret; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + platform_set_drvdata(aquila_snd_device, &aquila_snd_devdata); + aquila_snd_devdata.dev = &aquila_snd_device->dev; + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(C110)"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

2 2

[alsa-devel] [PATCH] alsa-utils: Allow building in subdirectory.
by Thierry Reding 07 Jul '10

07 Jul '10

This patch allows building alsa-utils in a separate subdirectory. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- alsactl/Makefile.am | 2 ++ alsamixer/Makefile.am | 2 +- 2 files changed, 3 insertions(+), 1 deletions(-) diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index 359f73a..945a54c 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -1,5 +1,7 @@ SUBDIRS = init +INCLUDES = -I$(top_srcdir)/include + sbin_PROGRAMS=alsactl man_MANS=alsactl.1 if USE_XMLTO diff --git a/alsamixer/Makefile.am b/alsamixer/Makefile.am index 1de47c6..d01b1e4 100644 --- a/alsamixer/Makefile.am +++ b/alsamixer/Makefile.am @@ -1,4 +1,5 @@ AM_CFLAGS = @CURSES_CFLAGS@ -DCURSESINC="@CURSESINC@" +INCLUDES = -I$(top_srcdir)/include LDADD = @CURSESLIB@ bin_PROGRAMS = alsamixer @@ -18,7 +19,6 @@ alsamixer_SOURCES = card_select.c card_select.h \ widget.c widget.h man_MANS = alsamixer.1 EXTRA_DIST = alsamixer.1 -alsamixer_INCLUDES = -I$(top_srcdir)/include #LDFLAGS = -static #CFLAGS += -g -Wall -- 1.7.1

1 0

[PATCH - Fix Fast Track Ultra series 1/1] Fix 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix Homann 07 Jul '10

07 Jul '10

Signed-off-by: Felix Homann <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..4c1dc12 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2080), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.1 --------------090102060507020204090408--

1 0

[PATCH - Fix Fast Track Ultra series 1/1] Fix 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix Homann 07 Jul '10

07 Jul '10

Signed-off-by: Felix Homann <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..4c1dc12 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2080), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.1 --------------020109060502090203040503--

1 0

Re: [alsa-devel] [PATCH 1/3] ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)
by Mark Brown 06 Jul '10

06 Jul '10

On Tue, Jul 06, 2010 at 05:00:24PM +0530, Sudhakar Rajashekhara wrote: > > Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> > Acked-by: Sudhakar Rajashekhara <sudhakar.raj(a)ti.com> Applied all, thanks.

1 0

[alsa-devel] [PATCH 2/3] ASoC: Add SmartQ sound driver
by Maurus Cuelenaere 06 Jul '10

06 Jul '10

This adds sound support for the SmartQ board. The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750 driver is used for driving the WM8987, as they are register compatible. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/soc/s3c24xx/Kconfig | 6 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smartq_wm8987.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 304 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smartq_wm8987.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2a7cc22..4f46ba0 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -125,3 +125,9 @@ config SND_SOC_SMDK_WM9713 select SND_S3C_SOC_AC97 help Sat Y if you want to add support for SoC audio on the SMDK. + +config SND_S3C64XX_SOC_SMARTQ + tristate "SoC I2S Audio support for SmartQ board" + depends on SND_S3C24XX_SOC && MACH_SMARTQ + select SND_S3C64XX_SOC_I2S + select SND_SOC_WM8750 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 81d8dc5..50172c3 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -29,6 +29,7 @@ snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o +snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -41,3 +42,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o +obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c new file mode 100644 index 0000000..c90ef96 --- /dev/null +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -0,0 +1,296 @@ +/* sound/soc/s3c24xx/smartq_wm8987.c + * + * Copyright 2010 Maurus Cuelenaere <mcuelenaere(a)gmail.com> + * + * Based on smdk6410_wm8987.c + * Copyright 2007 Wolfson Microelectronics PLC. - linux(a)wolfsonmicro.com + * Graeme Gregory - graeme.gregory(a)wolfsonmicro.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/gpio.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#include "../codecs/wm8750.h" + +/* + * WM8987 is register compatible with WM8750, so using that as base driver. + */ + +static struct snd_soc_card snd_soc_smartq; + +static int smartq_hifi_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret; + + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c_i2sv2_get_clock(cpu_dai)); + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + clk = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SmartQ WM8987 HiFi DAI operations. + */ +static struct snd_soc_ops smartq_hifi_ops = { + .hw_params = smartq_hifi_hw_params, +}; + +static struct snd_soc_jack smartq_jack; + +static struct snd_soc_jack_pin smartq_jack_pins[] = { + /* Disable speaker when headphone is plugged in */ + { + .pin = "Internal Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = true, + }, +}; + +static struct snd_soc_jack_gpio smartq_jack_gpios[] = { + { + .gpio = S3C64XX_GPL(12), + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 200, + }, +}; + +static const struct snd_kcontrol_new wm8987_smartq_controls[] = { + SOC_DAPM_PIN_SWITCH("Internal Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), +}; + +static int smartq_speaker_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8987_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Internal Speaker", smartq_speaker_event), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LOUT2"}, + {"Headphone Jack", NULL, "ROUT2"}, + + {"Internal Speaker", NULL, "LOUT2"}, + {"Internal Speaker", NULL, "ROUT2"}, + + {"Mic Bias", NULL, "Internal Mic"}, + {"LINPUT2", NULL, "Mic Bias"}, +}; + +static int smartq_wm8987_init(struct snd_soc_codec *codec) +{ + int err = 0; + + /* Add SmartQ specific widgets */ + snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets, + ARRAY_SIZE(wm8987_dapm_widgets)); + + /* add SmartQ specific controls */ + err = snd_soc_add_controls(codec, wm8987_smartq_controls, + ARRAY_SIZE(wm8987_smartq_controls)); + + if (err < 0) + return err; + + /* setup SmartQ specific audio path */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "LINPUT1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "ROUT1"); + + /* set endpoints to default off mode */ + snd_soc_dapm_enable_pin(codec, "Internal Speaker"); + snd_soc_dapm_enable_pin(codec, "Internal Mic"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + err = snd_soc_dapm_sync(codec); + if (err) + return err; + + /* Headphone jack detection */ + err = snd_soc_jack_new(&snd_soc_smartq, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack); + if (err) + return err; + + err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), + smartq_jack_pins); + if (err) + return err; + + err = snd_soc_jack_add_gpios(&smartq_jack, + ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + return err; +} + +static struct snd_soc_dai_link smartq_dai[] = { + { + .name = "wm8987", + .stream_name = "SmartQ Hi-Fi", + .cpu_dai = &s3c64xx_i2s_dai[0], + .codec_dai = &wm8750_dai, + .init = smartq_wm8987_init, + .ops = &smartq_hifi_ops, + }, +}; + +static struct snd_soc_card snd_soc_smartq = { + .name = "SmartQ", + .platform = &s3c24xx_soc_platform, + .dai_link = smartq_dai, + .num_links = ARRAY_SIZE(smartq_dai), +}; + +static struct snd_soc_device smartq_snd_devdata = { + .card = &snd_soc_smartq, + .codec_dev = &soc_codec_dev_wm8750, +}; + +static struct platform_device *smartq_snd_device; + +static int __init smartq_init(void) +{ + int ret; + + if (!machine_is_smartq7() && !machine_is_smartq5()) { + pr_info("Only SmartQ is supported by this ASoC driver\n"); + return -ENODEV; + } + + smartq_snd_device = platform_device_alloc("soc-audio", -1); + if (!smartq_snd_device) + return -ENOMEM; + + platform_set_drvdata(smartq_snd_device, &smartq_snd_devdata); + smartq_snd_devdata.dev = &smartq_snd_device->dev; + + ret = platform_device_add(smartq_snd_device); + if (ret) { + platform_device_put(smartq_snd_device); + return ret; + } + + /* Initialise GPIOs used by amplifiers */ + ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown"); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n"); + goto err_unregister_device; + } + + /* Disable amplifiers */ + ret = gpio_direction_output(S3C64XX_GPK(12), 1); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n"); + goto err_free_gpio_amp_shut; + } + + return 0; + +err_free_gpio_amp_shut: + gpio_free(S3C64XX_GPK(12)); +err_unregister_device: + platform_device_unregister(smartq_snd_device); + + return ret; +} + +static void __exit smartq_exit(void) +{ + snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + platform_device_unregister(smartq_snd_device); +} + +module_init(smartq_init); +module_exit(smartq_exit); + +/* Module information */ +MODULE_AUTHOR("Maurus Cuelenaere <mcuelenaere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC SmartQ WM8987"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

3 9

[alsa-devel] [PATCH 1/3] ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)
by Raffaele Recalcati 06 Jul '10

06 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added two clocking options for dm365 McBSP peripheral when used with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock from external pin and generates frame sync). A slave clock management can be important when the external codec needs the system clock and the bit clock synchronized (tested with uda1345). This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- sound/soc/davinci/davinci-i2s.c | 110 +++++++++++++++++++++++++++++++++++--- sound/soc/davinci/davinci-i2s.h | 5 ++ 2 files changed, 106 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..c8f038c 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -26,6 +26,7 @@ #include <mach/asp.h> #include "davinci-pcm.h" +#include "davinci-i2s.h" /* @@ -68,16 +69,21 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE BIT(31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE BIT(31) #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM BIT(29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,6 +150,9 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + unsigned int fmt; + int clk_div; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -254,10 +263,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = cpu_dai->private_data; unsigned int pcr; unsigned int srgr; + /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + dev->fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -372,6 +383,18 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + + if (div_id != DAVINCI_MCBSP_CLKGDV) + return -ENODEV; + + dev->clk_div = div; + return 0; +} + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,8 +403,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; @@ -396,12 +419,47 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + freq = clk_get_rate(dev->clk); + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFS: + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = dev->clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Clock and frame sync given from external sources */ + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + break; + default: + return -EINVAL; + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,12 +484,41 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + break; + default: + return -EINVAL; + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + break; + default: + return -EINVAL; + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -442,6 +529,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,6 +591,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h index 241648c..0b1e77b 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/davinci/davinci-i2s.h @@ -12,6 +12,11 @@ #ifndef _DAVINCI_I2S_H #define _DAVINCI_I2S_H +/* McBSP dividers */ +enum davinci_mcbsp_div { + DAVINCI_MCBSP_CLKGDV, /* Sample rate generator divider */ +}; + extern struct snd_soc_dai davinci_i2s_dai; #endif -- 1.7.0.4

3 2

[alsa-devel] [PATCH 3/3] ASoC: DaVinci: More accurate continuous serial clock for McBSP (I2S)
by Raffaele Recalcati 06 Jul '10

06 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> i2s_accurate_sck switch can be used to have a better approximate sampling frequency. The clock is an externally visible bit clock and it is named i2s continuous serial clock (I2S_SCK). The trade off is between more accurate clock (fast clock) and less accurate clock (slow clock). The waveform will be not symmetric. Probably it is possible to get a better algorithm for calculating the divider, trying to keep a slower clock as possible. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 33 ++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s.c | 24 +++++++++++++++++---- 2 files changed, 52 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 0847d21..b12c69e 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -73,6 +73,39 @@ struct snd_platform_data { */ int clk_input_pin; + /* + * This flag works when both clock and FS are outputs for the cpu + * and makes clock more accurate (FS is not symmetrical and the + * clock is very fast. + * The clock becoming faster is named + * i2s continuous serial clock (I2S_SCK) and it is an externally + * visible bit clock. + * + * first line : WordSelect + * second line : ContinuousSerialClock + * third line: SerialData + * + * SYMMETRICAL APPROACH: + * _______________________ LEFT + * _| RIGHT |______________________| + * _ _ _ _ _ _ _ _ + * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_ + * _ _ _ _ _ _ _ _ + * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_ + * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ + * + * ACCURATE CLOCK APPROACH: + * ______________ LEFT + * _| RIGHT |_______________________________| + * _ _ _ _ _ _ _ _ _ + * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| | + * _ _ _ _ dummy cycles + * _/ \_ ... _/ \_/ \_ ... _/ \__________________ + * \_/ \_/ \_/ \_/ + * + */ + bool i2s_accurate_sck; + /* McASP specific fields */ int tdm_slots; u8 op_mode; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ba5644b..b251bc9 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -155,6 +155,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; int clk_input_pin; + bool i2s_accurate_sck; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -447,11 +448,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SRGR_CLKSM; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); - /* symmetric waveforms */ - clk_div = freq / (mcbsp_word_length * 16) / - params->rate_num * params->rate_den; - srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * - 16 - 1); + if (dev->i2s_accurate_sck) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } clk_div &= 0xFF; srgr |= clk_div; break; @@ -662,6 +675,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; dev->clk_input_pin = pdata->clk_input_pin; + dev->i2s_accurate_sck = pdata->i2s_accurate_sck; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { -- 1.7.0.4

1 0

[alsa-devel] [PATCH 2/3] ASoC: DaVinci: Added selection of clk input pin for McBSP
by Raffaele Recalcati 06 Jul '10

06 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> When McBSP peripheral gets the clock from an external pin, there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR and MCBSP_CLKS. evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different hardware connection and I use MCBSP_CLKS, so I have added this possibility. This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm) Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 15 +++++++++++++++ sound/soc/davinci/davinci-i2s.c | 29 ++++++++++++++++++++++++----- 2 files changed, 39 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..0847d21 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,16 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * If McBSP peripheral gets the clock from an external pin, + * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR + * and MCBSP_CLKS. + * Depending on different hardware connections it is possible + * to use this setting to change the behaviour of McBSP + * driver. The dm365_clk_input_pin enum is available for dm365 + */ + int clk_input_pin; + /* McASP specific fields */ int tdm_slots; u8 op_mode; @@ -78,6 +88,11 @@ enum { MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; +enum dm365_clk_input_pin { + MCBSP_CLKR = 0, /* DM365 */ + MCBSP_CLKS, +}; + #define INACTIVE_MODE 0 #define TX_MODE 1 #define RX_MODE 2 diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c8f038c..ba5644b 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -122,6 +122,7 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { }; struct davinci_mcbsp_dev { + struct device *dev; struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -153,6 +154,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; + int clk_input_pin; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -279,11 +281,26 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: - /* McBSP CLKR pin is the input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + /* + * Selection of the clock input pin that is the + * input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + switch (dev->clk_input_pin) { + case MCBSP_CLKS: + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case MCBSP_CLKR: + pcr |= DAVINCI_MCBSP_PCR_SCLKME; + break; + default: + dev_err(dev->dev, "bad clk_input_pin\n"); + return -EINVAL; + } + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -644,6 +661,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { @@ -676,6 +694,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; + dev->dev = &pdev->dev; davinci_i2s_dai.private_data = dev; davinci_i2s_dai.capture.dma_data = dev->dma_params; -- 1.7.0.4

1 0

[PATCH - Fix Fast Track Ultra series: 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix 06 Jul '10

06 Jul '10

Signed-off-by: Felix <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..af9c2a1 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2081), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.0.4 --------------090100050500090907030802--

1 0

[PATCH - Fix Fast Track Ultra series: 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix 06 Jul '10

06 Jul '10

Signed-off-by: Felix <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..af9c2a1 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2081), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.0.4 --------------040009020804090202030406--

1 0

[alsa-devel] Resent HPI 4.03.36 updated summaries
by linux＠audioscience.com 06 Jul '10

06 Jul '10

Same patches, new summaries.

2 7

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 06 Jul '10

06 Jul '10

The following changes since commit ce93a3702832121d517ad348817929f22fcce47c: ASoC: Add SmartQ sound driver (2010-07-04 18:04:12 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Mark Brown (4): ASoC: Fix sorting of Makefile and Kconfig ASoC: Remove current WM8960 deemphasis control ASoC: Automatically manage DAC deemphasis rate for WM8960 ASoC: Automatically manage ALC coefficients for WM8960 sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/codecs/wm8960.c | 99 +++++++++++++++++++++++++++++++++++++++++---- 3 files changed, 93 insertions(+), 10 deletions(-)

2 1

[alsa-devel] [PATCH] Updated alsa-kernel asihpi sources so patches are no longer required.
by linux＠audioscience.com 06 Jul '10

06 Jul '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- .gitignore | 2 -- pci/asihpi/Makefile | 3 --- pci/asihpi/hpidspcd.c | 2 ++ pci/asihpi/hpidspcd.patch | 21 --------------------- pci/asihpi/hpios.c | 2 ++ pci/asihpi/hpios.patch | 35 ----------------------------------- 6 files changed, 4 insertions(+), 61 deletions(-) create mode 100644 pci/asihpi/hpidspcd.c delete mode 100644 pci/asihpi/hpidspcd.patch create mode 100644 pci/asihpi/hpios.c delete mode 100644 pci/asihpi/hpios.patch diff --git a/.gitignore b/.gitignore index 204b952..db642b2 100644 --- a/.gitignore +++ b/.gitignore @@ -75,8 +75,6 @@ pci/ac97/ac97_bus.c pci/ac97/ac97_codec.c pci/ad1889.c pci/ali5451/ali5451.c -pci/asihpi/hpidspcd.c -pci/asihpi/hpios.c pci/atiixp.c pci/atiixp_modem.c pci/au88x0/au88x0.c diff --git a/pci/asihpi/Makefile b/pci/asihpi/Makefile index 95ee449..50ab325 100644 --- a/pci/asihpi/Makefile +++ b/pci/asihpi/Makefile @@ -5,7 +5,6 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf -clean-files := hpidspcd.c hpios.c EXTRA_CFLAGS = -I$(SND_TOPDIR)/alsa-kernel/pci/asihpi @@ -13,5 +12,3 @@ include $(SND_TOPDIR)/alsa-kernel/pci/asihpi/Makefile include $(SND_TOPDIR)/Rules.make -hpidspcd.c: hpidspcd.patch $(SND_TOPDIR)/alsa-kernel/pci/asihpi/hpidspcd.c -hpios.c: hpios.patch $(SND_TOPDIR)/alsa-kernel/pci/asihpi/hpios.c diff --git a/pci/asihpi/hpidspcd.c b/pci/asihpi/hpidspcd.c new file mode 100644 index 0000000..9b7d80a --- /dev/null +++ b/pci/asihpi/hpidspcd.c @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/pci/asihpi/hpidspcd.c" diff --git a/pci/asihpi/hpidspcd.patch b/pci/asihpi/hpidspcd.patch deleted file mode 100644 index 7831639..0000000 --- a/pci/asihpi/hpidspcd.patch +++ /dev/null @@ -1,21 +0,0 @@ ---- ../../alsa-kernel/pci/asihpi/hpidspcd.c 2010-04-21 18:36:16.613453111 +0200 -+++ hpidspcd.c 2010-04-21 18:41:26.104970371 +0200 -@@ -1,3 +1,4 @@ -+#include "adriver.h" - /***********************************************************************/ - /*! - -@@ -73,8 +74,13 @@ - sprintf(fw_name, "asihpi/dsp%04x.bin", adapter); - HPI_DEBUG_LOG(INFO, "requesting firmware for %s\n", fw_name); - -+#if (LINUX_VERSION_CODE < KERNEL_VERSION(2 , 5 , 0)) -+ err = request_firmware(&ps_firmware, fw_name, -+ ps_dsp_code->ps_dev->slot_name); -+#else - err = request_firmware(&ps_firmware, fw_name, - &ps_dsp_code->ps_dev->dev); -+#endif - if (err != 0) { - HPI_DEBUG_LOG(ERROR, "%d, request_firmware failed for %s\n", - err, fw_name); diff --git a/pci/asihpi/hpios.c b/pci/asihpi/hpios.c new file mode 100644 index 0000000..3f151a3 --- /dev/null +++ b/pci/asihpi/hpios.c @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/pci/asihpi/hpios.c" diff --git a/pci/asihpi/hpios.patch b/pci/asihpi/hpios.patch deleted file mode 100644 index 1af0f7f..0000000 --- a/pci/asihpi/hpios.patch +++ /dev/null @@ -1,35 +0,0 @@ ---- ../../alsa-kernel/pci/asihpi/hpios.c 2010-04-21 18:36:16.617452068 +0200 -+++ hpios.c 2010-04-21 18:40:32.700700812 +0200 -@@ -1,3 +1,4 @@ -+#include "adriver.h" - /****************************************************************************** - - AudioScience HPI driver -@@ -26,6 +27,7 @@ - #include <linux/delay.h> - #include <linux/sched.h> - -+#if LINUX_VERSION_CODE >= KERNEL_VERSION(2 , 6 , 14) - void hpios_delay_micro_seconds(u32 num_micro_sec) - { - if ((usecs_to_jiffies(num_micro_sec) > 1) && !in_interrupt()) { -@@ -38,6 +40,19 @@ - mdelay(num_micro_sec / 1000); - - } -+#else -+void hpios_delay_micro_seconds(u32 num_micro_sec) -+{ -+ if ((num_micro_sec / 1000 >= 1000000 / HZ) && !in_interrupt()) { -+ set_current_state(TASK_UNINTERRUPTIBLE); -+ schedule_timeout((HZ * num_micro_sec + (HZ - 1)) / 1000000); -+ } else if (num_micro_sec <= 2000) -+ udelay(num_micro_sec); -+ else -+ mdelay(num_micro_sec / 1000); -+ -+} -+#endif - - void hpios_locked_mem_init(void) - { -- 1.7.0.4

3 3

Re: [alsa-devel] Help with dummy.c (where/how to write?)
by Smilen Dimitrov 06 Jul '10

06 Jul '10

Hi Eliot, Thanks so much for your response: >> >> I am trying to work on dummy.c: >> >> http://lxr.linux.no/#linux+v2.6.32/sound/drivers/dummy.c >> >> as an exercise, and I'd like to modify it so it behaves as an >> 'oscillator'; [snip] >> > > You are better to look at the loopback driver in alsa-driver repo: > > http://git.alsa-project.org/?p=alsa-driver.git;a=blob_plain;f=drivers/aloop… > > You would put your generation code in clear_capture_buf() > Fantastic info - that is exactly what I needed :) (Don't know why your response hasn't shown in the alsa-devel archives yet, hopefully my reply will bump it up there) Cheers!

1 0

[alsa-devel] [RFC 00/16] ASoC: multi-component - CODECs
by Liam Girdwood 06 Jul '10

06 Jul '10

This is the second part of the multi-component series and focuses on the changes required to support ASoC multi-component on CODEC drivers. I've split the CODEC driver patches into smaller chunks and tried to group (where possible) similar CODECs together. All the patches will be rebased together at upstream time so we don't break bisect. Most CODEC drivers were extremely straight forward with changes limited to their probe() and remove() sections. Some CODECs did require a little extra work and this is marked in their commit messages. Thanks Liam

5 24

[alsa-devel] [RFC 0/5] ASoC multi-component support : core
by Liam Girdwood 06 Jul '10

06 Jul '10

ASoC is now coming up to it's 5th birthday and is beginning to show it's age a little when it comes to supporting the next generation of smart phones and multimedia devices with it's current single CODEC and single DMA platform design. Historically, ASoC is designed around an embedded sound card with a single CODEC and a single platform DMA driver (with multiple DAIs). This design works great for most of the current audio hardware in production, however we are now seeing hardware designs for the next generation of devices that can have many CODECs, DMAs, DSPs and a host of other audio peripherals connected to the audio subsystem. So we now need ASoC to support new audio hardware architectures with multiple audio CODECs, DMA engines, DSPs, DAIs etc. This patch series allows an ASoC sound card to support multiple components (CODECs, DAI and DMA) and is the first part in a series of work (other parts coming later in the year from Mark and I) designed to enhance ASoC for modern mobile audio hardware. This patch series provides the following important features:- 1) Allow sound cards to have multiple CODECs and platform DMA controllers. 2) Allow CODEC drivers to support more than one CODEC device. 3) Allow all CODEC, platform DMA and DAI drivers to have platform data. This is achieved by: - 1) Separating the component struct driver data from the component struct device data. 2) Making all components kernel devices. (A side effect of this change is that we can now store all our component private data in struct device private data (like everyone else) and there is also now less pointer indirection for components.) The change mainly affects only the probe() and remove() sections of component drivers and the component enumeration section of the core. The core also now handles setting up the CODEC data structs too (meaning CODEC drivers dont have to do this). All component PCM, DAPM and Kcontrol operations remain unchanged. I've so far tested the new multi-component enumeration on the TI OMAP and Marvell PXA platforms with AC97, MFD and I2C based CODECs. I do however need ASoC platform maintainers to test on their systems (as I only have a subset of the hardware). Some ASOC platforms do need a little extra checking/fixups too :- o Samsung platform: required the most work to untangle the components. o i.MX platform: required work to de-couple components. o Injenic J4740 platform: I do not have a toolchain to even build this arch so it will also need someone to build it too. If you find your platform does not work then the most likely reason is that I'm missing a platform_device registration for your CODEC, DAI, etc. device in your arch board.c or device.c files. Please let me know or send a patch :) All the multi-component code is in my multi-component branch here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git This RFC post contains the changes to ASoC core only. I'll post the multi-component CODEC and platform changes as separate RFCs in order not to flood the list (as there are 38 patches in all). This patch series will eventually be rebased into a single patch for upstream so we dont break bisect. The subsequent patches for CODECs and platforms in this series can be found in the git branch above. The intention is to upstream this for 2.6.36 Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

5 14

[alsa-devel] Help with dummy.c (where/how to write?)
by Smilen Dimitrov 05 Jul '10

05 Jul '10

Hi all, I am trying to work on dummy.c: http://lxr.linux.no/#linux+v2.6.32/sound/drivers/dummy.c as an exercise, and I'd like to modify it so it behaves as an 'oscillator'; that is, when I choose 'dummy' as an input in say Audacity, and press Record - I'd like to be able to write something to the audio (pcm) buffer of the driver directly from the driver itself (say a single sample), and be able to observe those samples recorded in Audacity. So far, by adding some debug statements, the only thing I can figure, is that when 'Record' in Audacity is pressed, 'dummy_hrtimer_callback' is called continuously - but I cannot tell what functions to use to modify / write to output buffers (not even sure what the output buffers there are, I guess dpcm->substream->private_data). Could anyone point to me if there is a similar example existing already, or if not. provide me with some pointers on how to do this? Thanks in advance, Cheers!

1 0

[alsa-devel] [PATCH] ALSA: pcm_lib: avoid timing jitter in snd_pcm_read/write()
by David Dillow 05 Jul '10

05 Jul '10

When using poll() to wait for the next period -- or avail_min samples -- one gets a consistent delay for each system call that is usually just a little short of the selected period time. However, When using snd_pcm_read/write(), one gets a jittery delay that alternates between less than a millisecond and approximately two period times. This is caused by snd_pcm_lib_{read,write}1() transferring any available samples to the user's buffer and adjusting the application pointer prior to sleeping to the end of the current period. When the next period interrupt occurs, there is then less than avail_min samples remaining to be transferred in the period, so we end up sleeping until a second period occurs. This is solved by using runtime->twake as the number of samples needed for a wakeup in addition to selecting the proper wait queue to wake in snd_pcm_update_state(). This requires twake to be non-zero when used by snd_pcm_lib_{read,write}1() even if avail_min is zero. Signed-off-by: Dave Dillow <dave(a)thedillows.org> --- Hans, you may want this one as well for the SIS7019, as it gives more margin to deal with captured data on alternating calls into the stack. diff --git a/include/sound/pcm.h b/include/sound/pcm.h index dd76cde..83c6fa6 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -313,7 +313,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - unsigned int twake: 1; /* do transfer (!poll) wakeup */ + snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e9d98be..bcf95d3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -287,8 +287,11 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) - wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); + if (runtime->twake) { + if (avail >= runtime->twake) + wake_up(&runtime->tsleep); + } else if (avail >= runtime->control->avail_min) + wake_up(&runtime->sleep); return 0; } @@ -1707,7 +1710,7 @@ EXPORT_SYMBOL(snd_pcm_period_elapsed); * The available space is stored on availp. When err = 0 and avail = 0 * on the capture stream, it indicates the stream is in DRAINING state. */ -static int wait_for_avail_min(struct snd_pcm_substream *substream, +static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t *availp) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1757,7 +1760,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, avail = snd_pcm_playback_avail(runtime); else avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->control->avail_min) + if (avail >= runtime->twake) break; } _endloop: @@ -1820,7 +1823,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1833,7 +1836,9 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; } @@ -2042,7 +2047,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2060,7 +2065,9 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; if (!avail)

5 12

[alsa-devel] [PATCH] ASoC: DaVinci: More accurate calculation for clock divider for McBSP (I2S)
by Raffaele Recalcati 05 Jul '10

05 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> i2s_accurate_sck switch can be used to have a better approximate sampling frequency. The trade off is between more accurate clock (fast clock) and less accurate clock (slow clock). The waveform will be not symmetric. Probably it is possible to get a better algorithm for calculating the divider, trying to keep a slower clock as possible. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 33 ++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s.c | 28 ++++++++++++++++++++----- 2 files changed, 55 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 0847d21..5149abe 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -73,6 +73,39 @@ struct snd_platform_data { */ int clk_input_pin; + /* + * This flag works when both clock and FS are outputs for the cpu + * and makes clock more accurate (FS is not symmetrical and the + * clock is very fast. + * The clock becoming faster is named + * i2s continuous serial clocl (I2S_SCK) and it is an externally + * visible bit clock. + * + * first line : WordSelect + * second line : ContinuousSerialClock + * third line: SerialData + * + * SYMMETRICAL APPROACH: + * _______________________ LEFT + * _| RIGHT |______________________| + * _ _ _ _ _ _ _ _ + * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_ + * _ _ _ _ _ _ _ _ + * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_ + * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ + * + * ACCURATE CLOCK APPROACH: + * ______________ LEFT + * _| RIGHT |_______________________________| + * _ _ _ _ _ _ _ _ _ + * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| | + * _ _ _ _ dummy cycles + * _/ \_ ... _/ \_/ \_ ... _/ \__________________ + * \_/ \_/ \_/ \_/ + * + */ + bool i2s_accurate_sck; + /* McASP specific fields */ int tdm_slots; u8 op_mode; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 17f594f..e6dcd81 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -122,6 +122,7 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { }; struct davinci_mcbsp_dev { + struct device *dev; struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -154,6 +155,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; int clk_input_pin; + bool i2s_accurate_sck; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -296,7 +298,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, pcr |= DAVINCI_MCBSP_PCR_SCLKME; break; default: - dev_err(&pdev->dev, "bad clk_input_pin\n"); + dev_err(dev->dev, "bad clk_input_pin\n"); return -EINVAL; } @@ -447,11 +449,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SRGR_CLKSM; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); - /* symmetric waveforms */ - clk_div = freq / (mcbsp_word_length * 16) / - params->rate_num * params->rate_den; - srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * - 16 - 1); + if (dev->i2s_accurate_sck) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } clk_div &= 0xFF; srgr |= clk_div; break; @@ -662,6 +676,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; dev->clk_input_pin = pdata->clk_input_pin; + dev->i2s_accurate_sck = pdata->i2s_accurate_sck; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { @@ -694,6 +709,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; + dev->dev = &pdev->dev; davinci_i2s_dai.private_data = dev; davinci_i2s_dai.capture.dma_data = dev->dma_params; -- 1.7.0.4

1 0

Re: [alsa-devel] ALSA: hda - Don't check capture source mixer if no ADC is available
by Dan Carpenter 05 Jul '10

05 Jul '10

Commit fbe618f2168 "ALSA: hda - Don't check capture source mixer if no ADC is available" added support for when there is no ADC available and spec->adc_nids is NULL. If spec->adc_nids is NULL, can it cause a problem later when we call snd_hda_add_nid()? sound/pci/hda/patch_realtek.c +2643 alc_build_controls(96) error: we previously assumed 'spec->adc_nids' could be null. 2635 if (spec->cap_mixer) { 2636 const char *kname = kctl ? kctl->id.name : NULL; 2637 for (knew = spec->cap_mixer; knew->name; knew++) { 2638 if (kname && strcmp(knew->name, kname) == 0) 2639 continue; 2640 kctl = snd_hda_find_mixer_ctl(codec, knew->name); 2641 for (i = 0; kctl && i < kctl->count; i++) { 2642 err = snd_hda_add_nid(codec, kctl, i, 2643 spec->adc_nids[i]); ^^^^^^^^^^^^^^^^^ 2644 if (err < 0) 2645 return err; 2646 } 2647 } 2648 } regards, dan carpenter

2 1

[alsa-devel] linux vs c99 fixed width types
by Eliot Blennerhassett 05 Jul '10

05 Jul '10

Hi Takashi, & list it is quite likely that we will be moving our own codebase to use the C99 fixed width types rather than our own e.g. uint32_t vs HW32 etc. Currently HW32 is translated into u32 before I submit patches to you. Should I do the same for the C99 types, or let them pass through unaltered? regards -- Eliot Blennerhassett AudioScience Inc.

3 2

[alsa-devel] Update asihpi to hpi version 4.03.36
by linux＠audioscience.com 05 Jul '10

05 Jul '10

2 7

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 05 Jul '10

05 Jul '10

The following changes since commit e827e32efc313e1a061c81007e96fff439ccb802: Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2010-06-24 11:11:41 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Eric Bénard (1): ASoC: clean i.MX Kconfig Kuninori Morimoto (1): ASoC: ak4642: Add Digital Playback Volume control Mark Brown (1): Merge remote branch 'takashi/topic/asoc' into for-2.6.36 Vladimir Zapolskiy (4): ASoC: uda134x: replace a macro with a value in platform struct. ASoC: uda134x: fix bias level setup on initialization ASoC: uda134x: add DATA011 register found in codecs family ASoC: uda134x: correct bias level setup for codecs family include/sound/uda134x.h | 12 ++++++++ sound/soc/codecs/ak4642.c | 24 ++++++++++++++-- sound/soc/codecs/uda134x.c | 64 +++++++++++++++++++++++++++++-------------- sound/soc/codecs/uda134x.h | 5 ++- sound/soc/imx/Kconfig | 14 ++++----- 5 files changed, 85 insertions(+), 34 deletions(-)

2 2

[alsa-devel] PC Beeps not working with HDA Intel module
by Shérab 05 Jul '10

05 Jul '10

Dear all, The problem is on a computer with an HDA Intel sound card, Intel G45 DEVCTG chip and appears will all version s of the kernel. Running e.g. beep 440 produces a beep whose audible frequency is very low, definitely not 440 Hz. However, it is worth notiing that grub is able to produce a beep of the right frequency at boot time. Can this problem be fixed ? Should you need any additional bit of information or testing, please ask. Many thanks in advance for your help, Shérab.

2 1

[alsa-devel] Full duplex code sample?
by robert 05 Jul '10

05 Jul '10

Dear All, Is there an example code on how to program a full duplex audio transmission? I did search the internet but it is full of Paul Davis advising to use Jack instead of ALSA. I think I would rather use ALSA for my purposes. Cheers, Robert

2 1

[alsa-devel] [RFC][PATCH] snd-usb: improving support for CM6206
by Adrian Pardini 05 Jul '10

05 Jul '10

Hi all, I'm working in my spare time to get better support for this chip. Right now it works but some features present in the datasheet are missing. This patch gives the option to select the source for the "headphone" jack. I have some concerns with the way I accomplished this and I'd like if someone could kick me in the right direction. In particular with the caching of some register values. Added another field to usb_mixer_interface, maybe it's not the best place. Also, according to the datasheet, in order to read back the register I must issue a request as done in snd_usb_cm106_write_int_reg() and the result will come in the HID input report. For this it doesn't matter but I'm planning on making some controls for spdif and I'll need it. Where should I tap to get that information?. To be done (I need help): * Mapping of streams: Front is ok. Rear_L jack is assigned to "Center", Rear_R jack gets "Speaker Woofer". "Center/Bass" jack is managed with the mixer "Speaker". Traced the circuit and the connections are fine, how do I switch them from ALSA? (nb: not with a ttable, from the driver) To be done (I think I'll somehow get this done on my own): * Input levels are reported with more resolution than the real. Add another case to build_feature_ctl(). Not very important now. * Engaging of the "mute" controls: now I can't mute left or right channel without muting everything. Have to play more with mixer_maps. Kind regards and thanks a lot for your time. Signed off by: Adrian Pardini <adrian.pardini(a)solar.org.ar> --- diff --git a/alsa-kernel/usb/mixer.h b/alsa-kernel/usb/mixer.h index 26c636c..291ba4b 100644 --- a/alsa-kernel/usb/mixer.h +++ b/alsa-kernel/usb/mixer.h @@ -22,6 +22,7 @@ struct usb_mixer_interface { u8 audigy2nx_leds[3]; u8 xonar_u1_status; + u16 cm6206reg2; }; #define MAX_CHANNELS 10 /* max logical channels */ diff --git a/usb/mixer_quirks.c b/usb/mixer_quirks.c index 794cdbd..8adeef0 100644 --- a/usb/mixer_quirks.c +++ b/usb/mixer_quirks.c @@ -360,6 +360,129 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } +/* + * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely + * documented in the device's data sheet. + */ +int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) +{ + u8 buf[4]; + buf[0] = 0x20; + buf[1] = value & 0xff; + buf[2] = (value >> 8) & 0xff; + buf[3] = reg; + return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, + USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, + 0, 0, &buf, 4, 1000); +} + +static int snd_cm6206_headphone_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = kcontrol->private_value; + return 0; +} + +static int snd_cm6206_headphone_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + struct usb_device *dev = mixer->chip->dev; + + u16 reg2 = mixer->cm6206reg2; + u16 idx = kcontrol->private_value = ucontrol->value.integer.value[0]; + + reg2 = (u16)( (((1<<7) | (idx<<5))<<8) | (reg2 & 0x1800) ); + + snd_usb_cm106_write_int_reg(dev, 2, reg2); + mixer->cm6206reg2 = reg2; + return 1; +} + +static int snd_cm6206_headphone_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = kcontrol->private_value & (1<<11); + ucontrol->value.integer.value[1] = kcontrol->private_value & (1<<12); + return 0; +} + +static int snd_cm6206_headphone_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + struct usb_device *dev = mixer->chip->dev; + + u16 reg2 = mixer->cm6206reg2; + u16 mute; + + mute = ucontrol->value.integer.value[1] ? (1<<12) : 0; + mute |= ucontrol->value.integer.value[0] ? (1<<11) : 0; + kcontrol->private_value = mute; + + reg2 = (u16)(mute | (reg2 & ~0x1800)); + + snd_usb_cm106_write_int_reg(dev, 2, reg2); + mixer->cm6206reg2 = reg2; + return 1; +} + +static int snd_cm6206_headphone_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const names[] = { + "Side", + "Surround", + "Center", + "Front" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, names[uinfo->value.enumerated.item]); + return 0; +} + +#define snd_cm6206_headphone_mute_info snd_ctl_boolean_stereo_info + +static struct snd_kcontrol_new cm6206_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "Headphone Source", + .info = snd_cm6206_headphone_source_info, + .get = snd_cm6206_headphone_source_get, + .put = snd_cm6206_headphone_source_put, + .private_value = 3 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "Headphone", + .info = snd_cm6206_headphone_mute_info, + .get = snd_cm6206_headphone_mute_get, + .put = snd_cm6206_headphone_mute_put, + .private_value = 24 + } +}; + +static int snd_cm6206_controls_create(struct usb_mixer_interface *mixer) +{ + int i, err; + + for (i=0; i < ARRAY_SIZE(cm6206_controls); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&cm6206_controls[i], mixer)); + if (err < 0) + return err; + } + mixer->cm6206reg2 = 0xf800; /* Headphone source=front, no mute. */ + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err; @@ -385,6 +508,11 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) return err; } + if (mixer->chip->usb_id == USB_ID(0x0d8c, 0x0102)) { + err = snd_cm6206_controls_create(mixer); + if (err < 0) + return err; + } return 0; } @@ -415,4 +543,3 @@ void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, break; } } - diff --git a/alsa-kernel/usb/mixer_quirks.h b/alsa-kernel/usb/mixer_quirks.h index bdbfab0..bc5d577 100644 --- a/alsa-kernel/usb/mixer_quirks.h +++ b/alsa-kernel/usb/mixer_quirks.h @@ -9,5 +9,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid); +int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value); + #endif /* SND_USB_MIXER_QUIRKS_H */ diff --git a/usb/quirks.c b/usb/quirks.c index bf97ccd..ab57d58 100644 --- a/usb/quirks.c +++ b/usb/quirks.c @@ -376,22 +376,6 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) return 0; } -/* - * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely - * documented in the device's data sheet. - */ -static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) -{ - u8 buf[4]; - buf[0] = 0x20; - buf[1] = value & 0xff; - buf[2] = (value >> 8) & 0xff; - buf[3] = reg; - return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, - USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, - 0, 0, &buf, 4, 1000); -} - static int snd_usb_cm106_boot_quirk(struct usb_device *dev) { /* -- Adrian. http://elesquinazotango.com.ar http://www.noalcodigodescioli.blogspot.com/

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Re: [alsa-devel] [PATCH 2/6] sound/oss: convert to unlocked_ioctl
by Sam Ravnborg 04 Jul '10

04 Jul '10

> > diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c > index c1070e3..3547450 100644 > --- a/sound/oss/au1550_ac97.c > +++ b/sound/oss/au1550_ac97.c > @@ -824,22 +824,26 @@ mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd, > return codec->mixer_ioctl(codec, cmd, arg); > } > > -static int > -au1550_ioctl_mixdev(struct inode *inode, struct file *file, > - unsigned int cmd, unsigned long arg) > +static long > +au1550_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) > { > struct au1550_state *s = (struct au1550_state *)file->private_data; > struct ac97_codec *codec = s->codec; > + int ret; > + > + lock_kernel(); > + ret = mixdev_ioctl(codec, cmd, arg); > + unlock_kernel(); > > - return mixdev_ioctl(codec, cmd, arg); > + return ret; > } General comment... In several places you declare a local variable of type "int", but the function return a long. I know that mixdev_ioctl() return an int so nothing is lost but it just looks wrong. > diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c > index 3f3c3f7..b5464fb 100644 > --- a/sound/oss/dmasound/dmasound_core.c > +++ b/sound/oss/dmasound/dmasound_core.c > @@ -337,8 +337,8 @@ static int mixer_release(struct inode *inode, struct file *file) > unlock_kernel(); > return 0; > } > -static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd, > - u_long arg) > + > +static long mixer_ioctl(struct file *file, u_int cmd, u_long arg) > { > if (_SIOC_DIR(cmd) & _SIOC_WRITE) > mixer.modify_counter++; > @@ -362,11 +362,22 @@ static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd, > return -EINVAL; > } > > +static long mixer_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) > +{ > + int ret; > + > + lock_kernel(); > + ret = mixer_ioctl(file, cmd, arg); > + unlock_kernel(); > + > + return ret; > +} Here it looks like a potential but. mixer_ioctl() return a long which is stored in an int and then we return a long. > -static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd, > - u_long arg) > +static long sq_ioctl(struct file *file, u_int cmd, u_long arg) > { > int val, result; > u_long fmt; > @@ -1114,18 +1124,29 @@ static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd, > return IOCTL_OUT(arg,val); > > default: > - return mixer_ioctl(inode, file, cmd, arg); > + return mixer_ioctl(file, cmd, arg); > } > return -EINVAL; > } > > +static long sq_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) > +{ > + int ret; > + > + lock_kernel(); > + ret = sq_ioctl(file, cmd, arg); > + unlock_kernel(); > + > + return ret; > +} ditto: long -> int -> long Sam

2 1

[alsa-devel] ESI DuaFire Vendor Information
by Florian Scholz 04 Jul '10

04 Jul '10

Hi I like to implement a driver for the ESI DuaFire Interface and need the information from ESI. Andreas Preuss from ESI wrote me (translated): thanks you for your message. every information that you need for the development of a firewire audio driver are public. the information are accessable for the alsa-project and maybe the ffado project.

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[alsa-devel] [PATCH 1/3] ASoC: codec: Add WM8987 device id to WM8750 driver
by Maurus Cuelenaere 04 Jul '10

04 Jul '10

The WM8987 codec is register compatible with the WM8750, so just add it to the SPI and I²C device table. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/soc/codecs/wm8750.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 9407e19..e2c05e3 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -884,6 +884,7 @@ static int wm8750_i2c_remove(struct i2c_client *client) static const struct i2c_device_id wm8750_i2c_id[] = { { "wm8750", 0 }, + { "wm8987", 0 }, /* WM8987 is register compatible with WM8750 */ { } }; MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); @@ -925,14 +926,22 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id wm8750_spi_id[] = { + { "wm8750", 0 }, + { "wm8987", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, wm8750_spi_id); + static struct spi_driver wm8750_spi_driver = { .driver = { - .name = "wm8750", + .name = "WM8750 SPI Codec", .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = wm8750_spi_probe, .remove = __devexit_p(wm8750_spi_remove), + .id_table = wm8750_spi_id, }; #endif -- 1.7.0.4

2 3

[alsa-devel] Question about snd_pcm_sw_params_set_start_threshold
by Christian Gmeiner 03 Jul '10

03 Jul '10

Hi all, I am running into some troubles with my linux plugin for the vdr. I want to play audio in sync of video. My current solution is to buffer about 500ms of video on the output device (pts based FIFO). I can play audio with alsa, but sometimes I am not in sync with the video. So I want to set start threshold to 0 to disable automatic playback start. Instead I want to buffer audio and start playback manual, if pts of video is near current scr clock of output device. What are my options to manually start playback? thanks, -- Christian Gmeiner, MSc

1 0

[alsa-devel] [PATCH] sis7019: fix capture issues with multiple periods per buffer
by David Dillow 03 Jul '10

03 Jul '10

When using a timing voice to clock out periods during capture, the driver would slowly loose synchronization and never catch up, eventually reaching a point where it no longer generated interrupts. To avoid this situation, the virtual period clocking was changed to shorten the next timing period when our timing voice falls too far behind the capture voice. In addition, the first virtual period for the timing voice was slightly too short, causing the timing voice to initially be ahead of the capture voice. While tracking down this problem, I noticed that the expected sample offset was being incorrectly initialized, causing an overrun to be incorrectly reported when the timing voice happened to be perfectly synchronized. Reported-by: Hans Schou <linux(a)schou.dk> Signed-off-by: Dave Dillow <dave(a)thedillows.org> --- This has survived overnight testing at 44.1 kHz/16 bit/mono without issue. diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 9cc1b5a..614ff6e 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -264,11 +264,13 @@ static void sis_update_voice(struct voice *voice) * if using small periods. * * If we're less than 9 samples behind, we're on target. + * Otherwise, shorten the next vperiod by the amount we've + * been delayed. */ if (sync > -9) voice->vperiod = voice->sync_period_size + 1; else - voice->vperiod = voice->sync_period_size - 4; + voice->vperiod = voice->sync_period_size + sync + 10; if (voice->vperiod < voice->buffer_size) { sis_update_sso(voice, voice->vperiod); @@ -736,7 +738,7 @@ static void sis_prepare_timing_voice(struct voice *voice, period_size = buffer_size; /* Initially, we want to interrupt just a bit behind the end of - * the period we're clocking out. 10 samples seems to give a good + * the period we're clocking out. 12 samples seems to give a good * delay. * * We want to spread our interrupts throughout the virtual period, @@ -747,7 +749,7 @@ static void sis_prepare_timing_voice(struct voice *voice, * * This is all moot if we don't need to use virtual periods. */ - vperiod = runtime->period_size + 10; + vperiod = runtime->period_size + 12; if (vperiod > period_size) { u16 tail = vperiod % period_size; u16 quarter_period = period_size / 4; @@ -776,7 +778,7 @@ static void sis_prepare_timing_voice(struct voice *voice, */ timing->flags |= VOICE_SYNC_TIMING; timing->sync_base = voice->ctrl_base; - timing->sync_cso = runtime->period_size - 1; + timing->sync_cso = runtime->period_size; timing->sync_period_size = runtime->period_size; timing->sync_buffer_size = runtime->buffer_size; timing->period_size = period_size;

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Re: [alsa-devel] [PATCH] - arecord - remove "front" and related devices from the output of list-pcm option of arecord
by Jaroslav Kysela 03 Jul '10

03 Jul '10

On Thu, 1 Jul 2010, Raymond Yau wrote: > Replace "front" device by "hw" device and remove "rear", "center_lfe" , > "side" , "surroundxx" from the output of "arecord -L" since "front" device > cannot be used for capturing and the front device is for playback only . NAK. Fix alsa-lib/configuration to return proper "IOID". Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 5

[alsa-devel] [PATCH 3/3] ARM: S3C64XX: Add audio support to SmartQ
by Maurus Cuelenaere 03 Jul '10

03 Jul '10

This adds the I²C board information for the WM8987 used in the SmartQ as audio codec and adds the I²C/I²S platform drivers. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- arch/arm/mach-s3c64xx/mach-smartq.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s3c64xx/mach-smartq.c b/arch/arm/mach-s3c64xx/mach-smartq.c index 028d080..5d8673b 100644 --- a/arch/arm/mach-s3c64xx/mach-smartq.c +++ b/arch/arm/mach-s3c64xx/mach-smartq.c @@ -199,6 +199,9 @@ static struct platform_device smartq_lcd_power_device = { .dev.platform_data = &smartq_lcd_power_data, }; +static struct i2c_board_info smartq_i2c_devs[] __initdata = { + { I2C_BOARD_INFO("wm8987", 0x1a), }, +}; static struct platform_device *smartq_devices[] __initdata = { &s3c_device_hsmmc1, /* Init iNAND first, ... */ @@ -213,6 +216,7 @@ static struct platform_device *smartq_devices[] __initdata = { &s3c_device_timer[1], &s3c_device_ts, &s3c_device_usb_hsotg, + &s3c64xx_device_iis0, &smartq_backlight_device, &smartq_lcd_power_device, &smartq_usb_otg_vbus_dev, @@ -354,6 +358,9 @@ void __init smartq_machine_init(void) s3c_sdhci2_set_platdata(&smartq_internal_hsmmc_pdata); s3c24xx_ts_set_platdata(&smartq_touchscreen_pdata); + i2c_register_board_info(0, smartq_i2c_devs, + ARRAY_SIZE(smartq_i2c_devs)); + WARN_ON(smartq_power_off_init()); WARN_ON(smartq_usb_host_init()); WARN_ON(smartq_usb_otg_init()); -- 1.7.0.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: Add SmartQ sound driver
by Maurus Cuelenaere 03 Jul '10

03 Jul '10

This adds sound support for the SmartQ board. The hardware consists of a S3C6410 coupled with a WM8987 over I²S. As there's no driver for the WM8987 currently, this uses the WM8750 driver (which should be register compatible). Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/soc/s3c24xx/Kconfig | 6 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smartq_wm8987.c | 300 +++++++++++++++++++++++++++++++++++++ 3 files changed, 308 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smartq_wm8987.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2a7cc22..4f46ba0 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -125,3 +125,9 @@ config SND_SOC_SMDK_WM9713 select SND_S3C_SOC_AC97 help Sat Y if you want to add support for SoC audio on the SMDK. + +config SND_S3C64XX_SOC_SMARTQ + tristate "SoC I2S Audio support for SmartQ board" + depends on SND_S3C24XX_SOC && MACH_SMARTQ + select SND_S3C64XX_SOC_I2S + select SND_SOC_WM8750 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 81d8dc5..50172c3 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -29,6 +29,7 @@ snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o +snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -41,3 +42,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o +obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c new file mode 100644 index 0000000..ef6183f --- /dev/null +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -0,0 +1,300 @@ +/* sound/soc/s3c24xx/smartq_wm8987.c + * + * Copyright 2010 Maurus Cuelenaere <mcuelenaere(a)gmail.com> + * + * Based on smdk6410_wm8987.c + * Copyright 2007 Wolfson Microelectronics PLC. - linux(a)wolfsonmicro.com + * Graeme Gregory - graeme.gregory(a)wolfsonmicro.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/gpio.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#include "../codecs/wm8750.h" + +/* + * According to http://www.wolfsonmicro.com/products/WM8987 the WM8987 is + * register compatible with the WM8750, so I'm currently using that one as + * there's no WM8987 codec driver. + */ + +static struct snd_soc_card snd_soc_smartq; + +static int smartq_hifi_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret; + + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c_i2sv2_get_clock(cpu_dai)); + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + clk = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SmartQ WM8987 HiFi DAI operations. + */ +static struct snd_soc_ops smartq_hifi_ops = { + .hw_params = smartq_hifi_hw_params, +}; + +static struct snd_soc_jack smartq_jack; + +static struct snd_soc_jack_pin smartq_jack_pins[] = { + /* Disable speaker when headphone is plugged in */ + { + .pin = "Internal Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = true, + }, +}; + +static struct snd_soc_jack_gpio smartq_jack_gpios[] = { + { + .gpio = S3C64XX_GPL(12), + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 200, + }, +}; + +static const struct snd_kcontrol_new wm8987_smartq_controls[] = { + SOC_DAPM_PIN_SWITCH("Internal Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), +}; + +static int smartq_speaker_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8987_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Internal Speaker", smartq_speaker_event), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LOUT2"}, + {"Headphone Jack", NULL, "ROUT2"}, + + {"Internal Speaker", NULL, "LOUT2"}, + {"Internal Speaker", NULL, "ROUT2"}, + + {"Mic Bias", NULL, "Internal Mic"}, + {"LINPUT2", NULL, "Mic Bias"}, +}; + +static int smartq_wm8987_init(struct snd_soc_codec *codec) +{ + int err = 0; + + /* Add SmartQ specific widgets */ + snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets, + ARRAY_SIZE(wm8987_dapm_widgets)); + + /* add SmartQ specific controls */ + err = snd_soc_add_controls(codec, wm8987_smartq_controls, + ARRAY_SIZE(wm8987_smartq_controls)); + + if (err < 0) + return err; + + /* setup SmartQ specific audio path */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "LINPUT1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "ROUT1"); + + /* set endpoints to default off mode */ + snd_soc_dapm_enable_pin(codec, "Internal Speaker"); + snd_soc_dapm_enable_pin(codec, "Internal Mic"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + err = snd_soc_dapm_sync(codec); + if (err) + return err; + + /* Headphone jack detection */ + err = snd_soc_jack_new(&snd_soc_smartq, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack); + if (err) + return err; + + err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), + smartq_jack_pins); + if (err) + return err; + + err = snd_soc_jack_add_gpios(&smartq_jack, + ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + return err; +} + +static struct snd_soc_dai_link smartq_dai[] = { + { + .name = "wm8987", + .stream_name = "SmartQ Hi-Fi", + .cpu_dai = &s3c64xx_i2s_dai[0], + .codec_dai = &wm8750_dai, + .init = smartq_wm8987_init, + .ops = &smartq_hifi_ops, + }, +}; + +static struct snd_soc_card snd_soc_smartq = { + .name = "SmartQ", + .platform = &s3c24xx_soc_platform, + .dai_link = smartq_dai, + .num_links = ARRAY_SIZE(smartq_dai), +}; + +static struct snd_soc_device smartq_snd_devdata = { + .card = &snd_soc_smartq, + .codec_dev = &soc_codec_dev_wm8750, +}; + +static struct platform_device *smartq_snd_device; + +static int __init smartq_init(void) +{ + int ret; + + if (!machine_is_smartq7() && !machine_is_smartq5()) { + pr_info("Only SmartQ is supported by this ASoC driver\n"); + return -ENODEV; + } + + smartq_snd_device = platform_device_alloc("soc-audio", -1); + if (!smartq_snd_device) + return -ENOMEM; + + platform_set_drvdata(smartq_snd_device, &smartq_snd_devdata); + smartq_snd_devdata.dev = &smartq_snd_device->dev; + + ret = platform_device_add(smartq_snd_device); + if (ret) { + platform_device_put(smartq_snd_device); + return ret; + } + + /* Initialise GPIOs used by amplifiers */ + ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown"); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n"); + goto err_unregister_device; + } + + /* Disable amplifiers */ + ret = gpio_direction_output(S3C64XX_GPK(12), 1); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n"); + goto err_free_gpio_amp_shut; + } + + dev_info(&smartq_snd_device->dev, "SmartQ WM8987 audio driver inited\n"); + + return 0; + +err_free_gpio_amp_shut: + gpio_free(S3C64XX_GPK(12)); +err_unregister_device: + platform_device_unregister(smartq_snd_device); + + return ret; +} + +static void __exit smartq_exit(void) +{ + snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + platform_device_unregister(smartq_snd_device); +} + +module_init(smartq_init); +module_exit(smartq_exit); + +/* Module information */ +MODULE_AUTHOR("Maurus Cuelenaere"); +MODULE_DESCRIPTION("ALSA SoC SmartQ WM8987"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

2 2

[alsa-devel] [PATCH 2/2] ARM: S3C64XX: Add audio support to SmartQ
by Maurus Cuelenaere 03 Jul '10

03 Jul '10

This adds the I²C board information for the WM8987 used in the SmartQ as audio codec and adds the I²C/I²S platform drivers. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- arch/arm/mach-s3c64xx/mach-smartq.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s3c64xx/mach-smartq.c b/arch/arm/mach-s3c64xx/mach-smartq.c index 028d080..ee77255 100644 --- a/arch/arm/mach-s3c64xx/mach-smartq.c +++ b/arch/arm/mach-s3c64xx/mach-smartq.c @@ -199,6 +199,9 @@ static struct platform_device smartq_lcd_power_device = { .dev.platform_data = &smartq_lcd_power_data, }; +static struct i2c_board_info smartq_i2c_devs[] __initdata = { + { I2C_BOARD_INFO("wm8750", 0x1a), }, +}; static struct platform_device *smartq_devices[] __initdata = { &s3c_device_hsmmc1, /* Init iNAND first, ... */ @@ -213,6 +216,7 @@ static struct platform_device *smartq_devices[] __initdata = { &s3c_device_timer[1], &s3c_device_ts, &s3c_device_usb_hsotg, + &s3c64xx_device_iis0, &smartq_backlight_device, &smartq_lcd_power_device, &smartq_usb_otg_vbus_dev, @@ -354,6 +358,9 @@ void __init smartq_machine_init(void) s3c_sdhci2_set_platdata(&smartq_internal_hsmmc_pdata); s3c24xx_ts_set_platdata(&smartq_touchscreen_pdata); + i2c_register_board_info(0, smartq_i2c_devs, + ARRAY_SIZE(smartq_i2c_devs)); + WARN_ON(smartq_power_off_init()); WARN_ON(smartq_usb_host_init()); WARN_ON(smartq_usb_otg_init()); -- 1.7.0.4

2 1

[alsa-devel] alsamixer-qt4 version 0.3 released
by Sebastian H. 01 Jul '10

01 Jul '10

Hello again The new version 0.3.0 of alsamixer-qt4 is available. Check it out at http://xwmw.org/alsamixer-qt4/ The files are at sourceforge http://sourceforge.net/projects/alsamixer-qt4/files/ Regards, Sebastian H.

1 0

[alsa-devel] [PATCH v3] ASoC: ak4642: Add Digital Playback Volume control
by Kuninori Morimoto 01 Jul '10

01 Jul '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 o remove "default volume" o remove explain from commit log sound/soc/codecs/ak4642.c | 24 +++++++++++++++++++++--- 1 files changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7528a54..b41599d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -36,6 +36,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -111,6 +112,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -204,7 +222,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -214,8 +231,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -548,6 +563,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -- 1.7.0.4

2 1

[alsa-devel] wrong decibel data?
by Nicolo' Chieffo 01 Jul '10

01 Jul '10

Hello, I have a laptop with an intel hda audio chipset (STAC92xx) My problem is that when I put pulseaudio volume to 14% (or less) I can't hear any audio coming from speakers or headphones: in fact the alsamixer volume is set to 0% I already filed a bug to pulseaudio [1] in which the developers say that this is a common problem which is usually caused by the ALSA audio driver which reports a wrong decibel value. [1] http://pulseaudio.org/ticket/580

9 95

[alsa-devel] [PATCH 3/3] ASoC: DaVinci: Added fast clock timing for McBSP (I2S)
by Raffaele Recalcati 01 Jul '10

01 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> i2s_fast_clock switch can be used to have a better approximate frequency. The waveform will be not symmetric. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++++ sound/soc/davinci/davinci-i2s.c | 24 +++++++++++++++++++----- 2 files changed, 26 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 314570d..2d8f8af 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,13 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not symmetrical, but sampling + * frequency is better approximated + */ + bool i2s_fast_clock; + /* To be used when cpu gets clock from external pin */ int clk_input_pin; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index e478be9..0af5bae 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -150,6 +150,7 @@ struct davinci_mcbsp_dev { */ unsigned enable_channel_combine:1; + bool i2s_fast_clock; unsigned int fmt; int clk_div; int clk_input_pin; @@ -443,11 +444,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SRGR_CLKSM; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); - /* symmetric waveforms */ - clk_div = freq / (mcbsp_word_length * 16) / - params->rate_num * params->rate_den; - srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * - 16 - 1); + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } clk_div &= 0xFF; srgr |= clk_div; break; @@ -643,6 +656,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->i2s_fast_clock = pdata->i2s_fast_clock; dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); -- 1.7.0.4

2 2

[alsa-devel] [PATCH 2/3] ASoC: DaVinci: Added selection of clk input pin for McBSP
by Raffaele Recalcati 01 Jul '10

01 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> When McBSP peripheral gets the clock from an external pin, there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR and MCBSP_CLKS. evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different hardware connection and I use MCBSP_CLKS, so I have added this possibility. This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm) Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 8 ++++++++ sound/soc/davinci/davinci-i2s.c | 28 +++++++++++++++++++++++----- 2 files changed, 31 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..314570d 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,9 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* To be used when cpu gets clock from external pin */ + int clk_input_pin; + /* McASP specific fields */ int tdm_slots; u8 op_mode; @@ -78,6 +81,11 @@ enum { MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; +enum { + MCBSP_CLKR = 0, /* DM365 */ + MCBSP_CLKS, +}; + #define INACTIVE_MODE 0 #define TX_MODE 1 #define RX_MODE 2 diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 319e8ca..e478be9 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -152,6 +152,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; + int clk_input_pin; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -278,11 +279,26 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: - /* McBSP CLKR pin is the input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + /* + * Selection of the clock input pin that is the + * input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + switch (dev->clk_input_pin) { + case MCBSP_CLKS: + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case MCBSP_CLKR: + pcr |= DAVINCI_MCBSP_PCR_SCLKME; + break; + default: + printk(KERN_ERR "%s:bad clk_input_pin\n", __func__); + return -EINVAL; + } + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -436,6 +452,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, srgr |= clk_div; break; case SND_SOC_DAIFMT_CBM_CFS: + /* Clock given on CLKS */ srgr = DAVINCI_MCBSP_SRGR_FSGM; clk_div = dev->clk_div - 1; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); @@ -626,6 +643,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)
by Raffaele Recalcati 01 Jul '10

01 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added two clocking options for dm365 McBSP peripheral when used with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock from external pin and generates frame sync). A slave clock management can be important when the external codec needs the system clock and the bit clock synchronized (tested with uda1345). This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- sound/soc/davinci/davinci-i2s.c | 92 +++++++++++++++++++++++++++++++++++---- 1 files changed, 83 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..319e8ca 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -68,16 +68,21 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE (1 << 31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE (1 << 31) #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM (1 << 29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,6 +149,9 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + unsigned int fmt; + int clk_div; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -254,10 +262,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = cpu_dai->private_data; unsigned int pcr; unsigned int srgr; + /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + dev->fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -372,6 +382,16 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + int srgr; + + dev->clk_div = div; + return 0; +} + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,8 +400,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; @@ -396,12 +416,44 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + freq = clk_get_rate(dev->clk); + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFS: + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = dev->clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + default: + /* Clock and frame sync given from external sources */ + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,12 +478,29 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -442,6 +511,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,6 +573,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; -- 1.7.0.4

2 1

[alsa-devel] [RFC 00/15] ASoC: multi-component - Platform Drivers
by Liam Girdwood 01 Jul '10

01 Jul '10

This is the third and final part of the ASoC multi-component series and focuses on the SoC Platform support. I've split the patches by SoC architecture and again all will be rebased later to avoid breaking git bisect. Most platform drivers were quite straight forward with most changes in the probe() and remove() areas. However, the Samsung and i.MX platform did require some extra work to decouple some of the components. Thanks Liam

5 22

[alsa-devel] [PATCH] ASoC: Multi-component - Fix tlv320aic23 hw_write/hw_read functions
by Ryan Mallon 01 Jul '10

01 Jul '10

Commit 5ce5888769c4a453bb5a852e156ab4ac72822c40 removed the initialisations of the codec->hw_write/read functions for the tlv320aic23. Fix this by adding the initialisations in tlv320aic23_probe. Signed-off-by: Ryan Mallon <ryan(a)bluewatersys.com> --- diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index fbc95c5..fe89521 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -637,6 +637,8 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); codec->control_data = aic23->control_data; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = NULL; /* Reset codec */ tlv320aic23_write(codec, TLV320AIC23_RESET, 0);

2 1

[alsa-devel] [RFC 3/5] ASoC multi-component support : core
by Liam Girdwood 01 Jul '10

01 Jul '10

This patch adds multi-component support to the ASoC core and reflects the new ASoC driver/device structure model defined in soc.h and soc-dai.h. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/soc-core.c | 1595 +++++++++++++++++++++++++++++--------------------- 1 files changed, 914 insertions(+), 681 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3deb226..280c2f0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -39,6 +39,8 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> +#define NAME_SIZE 32 + static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); @@ -54,6 +56,7 @@ static LIST_HEAD(codec_list); static int snd_soc_register_card(struct snd_soc_card *card); static int snd_soc_unregister_card(struct snd_soc_card *card); +static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -88,27 +91,27 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { int i, step = 1, count = 0; - if (!codec->reg_cache_size) + if (!codec->driver->reg_cache_size) return 0; - if (codec->reg_cache_step) - step = codec->reg_cache_step; + if (codec->driver->reg_cache_step) + step = codec->driver->reg_cache_step; count += sprintf(buf, "%s registers\n", codec->name); - for (i = 0; i < codec->reg_cache_size; i += step) { - if (codec->readable_register && !codec->readable_register(i)) + for (i = 0; i < codec->driver->reg_cache_size; i += step) { + if (codec->driver->readable_register && !codec->driver->readable_register(i)) continue; count += sprintf(buf + count, "%2x: ", i); if (count >= PAGE_SIZE - 1) break; - if (codec->display_register) - count += codec->display_register(codec, buf + count, + if (codec->driver->display_register) + count += codec->driver->display_register(codec, buf + count, PAGE_SIZE - count, i); else count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); + "%4x", codec->driver->read(codec, i)); if (count >= PAGE_SIZE - 1) break; @@ -127,8 +130,8 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) static ssize_t codec_reg_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); - return soc_codec_reg_show(devdata->card->codec, buf); + struct snd_soc_codec *codec = dev_get_drvdata(dev); + return soc_codec_reg_show(codec, buf); } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); @@ -136,20 +139,20 @@ static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); static ssize_t pmdown_time_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_soc_device *socdev = dev_get_drvdata(dev); - struct snd_soc_card *card = socdev->card; + struct snd_soc_pcm_runtime *rtd = + container_of(dev, struct snd_soc_pcm_runtime, dev); - return sprintf(buf, "%ld\n", card->pmdown_time); + return sprintf(buf, "%ld\n", rtd->pmdown_time); } static ssize_t pmdown_time_set(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { - struct snd_soc_device *socdev = dev_get_drvdata(dev); - struct snd_soc_card *card = socdev->card; + struct snd_soc_pcm_runtime *rtd = + container_of(dev, struct snd_soc_pcm_runtime, dev); - strict_strtol(buf, 10, &card->pmdown_time); + strict_strtol(buf, 10, &rtd->pmdown_time); return count; } @@ -193,19 +196,19 @@ static ssize_t codec_reg_write_file(struct file *file, return -EFAULT; buf[buf_size] = 0; - if (codec->reg_cache_step) - step = codec->reg_cache_step; + if (codec->driver->reg_cache_step) + step = codec->driver->reg_cache_step; while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->reg_cache_size) || (reg % step)) + if ((reg >= codec->driver->reg_cache_size) || (reg % step)) return -EINVAL; while (*start == ' ') start++; if (strict_strtoul(start, 16, &value)) return -EINVAL; - codec->write(codec, reg, value); + codec->driver->write(codec, reg, value); return buf_size; } @@ -295,7 +298,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", - codec->card->number, 0, codec->name); + codec->id, 0, codec->name); err = device_register(&codec->ac97->dev); if (err < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); @@ -309,24 +312,21 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates || - machine->symmetric_rates) { - dev_dbg(card->dev, "Symmetry forces %dHz rate\n", - machine->rate); + if (codec_dai->driver->symmetric_rates || cpu_dai->driver->symmetric_rates || + rtd->dai_link->symmetric_rates) { + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", + rtd->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - machine->rate, - machine->rate); + rtd->rate, + rtd->rate); if (ret < 0) { - dev_err(card->dev, + dev_err(&rtd->dev, "Unable to apply rate symmetry constraint: %d\n", ret); return ret; } @@ -343,48 +343,47 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) static int soc_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; mutex_lock(&pcm_mutex); /* startup the audio subsystem */ - if (cpu_dai->ops->startup) { - ret = cpu_dai->ops->startup(substream, cpu_dai); + if (cpu_dai->driver->ops->startup) { + ret = cpu_dai->driver->ops->startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); + cpu_dai->driver->name); goto out; } } - if (platform->pcm_ops->open) { - ret = platform->pcm_ops->open(substream); + if (platform->driver->ops->open) { + ret = platform->driver->ops->open(substream); if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + printk(KERN_ERR "asoc: can't open platform %s\n", platform->driver->name); goto platform_err; } } - if (codec_dai->ops->startup) { - ret = codec_dai->ops->startup(substream, codec_dai); + if (codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); + codec_dai->driver->name); goto codec_dai_err; } } - if (machine->ops && machine->ops->startup) { - ret = machine->ops->startup(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + ret = rtd->dai_link->ops->startup(substream); if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", machine->name); + printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); goto machine_err; } } @@ -392,66 +391,66 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAI's are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = - max(codec_dai->playback.rate_min, - cpu_dai->playback.rate_min); + max(codec_dai_drv->playback.rate_min, + cpu_dai_drv->playback.rate_min); runtime->hw.rate_max = - min(codec_dai->playback.rate_max, - cpu_dai->playback.rate_max); + min(codec_dai_drv->playback.rate_max, + cpu_dai_drv->playback.rate_max); runtime->hw.channels_min = - max(codec_dai->playback.channels_min, - cpu_dai->playback.channels_min); + max(codec_dai_drv->playback.channels_min, + cpu_dai_drv->playback.channels_min); runtime->hw.channels_max = - min(codec_dai->playback.channels_max, - cpu_dai->playback.channels_max); + min(codec_dai_drv->playback.channels_max, + cpu_dai_drv->playback.channels_max); runtime->hw.formats = - codec_dai->playback.formats & cpu_dai->playback.formats; + codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; runtime->hw.rates = - codec_dai->playback.rates & cpu_dai->playback.rates; - if (codec_dai->playback.rates + codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; + if (codec_dai_drv->playback.rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai->playback.rates; - if (cpu_dai->playback.rates + runtime->hw.rates |= cpu_dai_drv->playback.rates; + if (cpu_dai_drv->playback.rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai->playback.rates; + runtime->hw.rates |= codec_dai_drv->playback.rates; } else { runtime->hw.rate_min = - max(codec_dai->capture.rate_min, - cpu_dai->capture.rate_min); + max(codec_dai_drv->capture.rate_min, + cpu_dai_drv->capture.rate_min); runtime->hw.rate_max = - min(codec_dai->capture.rate_max, - cpu_dai->capture.rate_max); + min(codec_dai_drv->capture.rate_max, + cpu_dai_drv->capture.rate_max); runtime->hw.channels_min = - max(codec_dai->capture.channels_min, - cpu_dai->capture.channels_min); + max(codec_dai_drv->capture.channels_min, + cpu_dai_drv->capture.channels_min); runtime->hw.channels_max = - min(codec_dai->capture.channels_max, - cpu_dai->capture.channels_max); + min(codec_dai_drv->capture.channels_max, + cpu_dai_drv->capture.channels_max); runtime->hw.formats = - codec_dai->capture.formats & cpu_dai->capture.formats; + codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; runtime->hw.rates = - codec_dai->capture.rates & cpu_dai->capture.rates; - if (codec_dai->capture.rates + codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; + if (codec_dai_drv->capture.rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai->capture.rates; - if (cpu_dai->capture.rates + runtime->hw.rates |= cpu_dai_drv->capture.rates; + if (cpu_dai_drv->capture.rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai->capture.rates; + runtime->hw.rates |= codec_dai_drv->capture.rates; } snd_pcm_limit_hw_rates(runtime); if (!runtime->hw.rates) { printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); + codec_dai->driver->name, cpu_dai->driver->name); goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); + codec_dai->driver->name, cpu_dai->driver->name); goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max) { printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); + codec_dai->driver->name, cpu_dai->driver->name); goto config_err; } @@ -462,7 +461,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); + pr_debug("asoc: %s <-> %s info:\n", + codec_dai->driver->name, cpu_dai->driver->name); pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -470,33 +470,33 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rate_max); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback.active++; - codec_dai->playback.active++; + cpu_dai->playback_active++; + codec_dai->playback_active++; } else { - cpu_dai->capture.active++; - codec_dai->capture.active++; + cpu_dai->capture_active++; + codec_dai->capture_active++; } cpu_dai->active++; codec_dai->active++; - card->codec->active++; + rtd->codec->active++; mutex_unlock(&pcm_mutex); return 0; config_err: - if (machine->ops && machine->ops->shutdown) - machine->ops->shutdown(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); machine_err: - if (codec_dai->ops->shutdown) - codec_dai->ops->shutdown(substream, codec_dai); + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); codec_dai_err: - if (platform->pcm_ops->close) - platform->pcm_ops->close(substream); + if (platform->driver->ops->close) + platform->driver->ops->close(substream); platform_err: - if (cpu_dai->ops->shutdown) - cpu_dai->ops->shutdown(substream, cpu_dai); + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: mutex_unlock(&pcm_mutex); return ret; @@ -509,29 +509,25 @@ out: */ static void close_delayed_work(struct work_struct *work) { - struct snd_soc_card *card = container_of(work, struct snd_soc_card, - delayed_work.work); - struct snd_soc_codec *codec = card->codec; - struct snd_soc_dai *codec_dai; - int i; + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); + struct snd_soc_dai *codec_dai = rtd->codec_dai; mutex_lock(&pcm_mutex); - for (i = 0; i < codec->num_dai; i++) { - codec_dai = &codec->dai[i]; - - pr_debug("pop wq checking: %s status: %s waiting: %s\n", - codec_dai->playback.stream_name, - codec_dai->playback.active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); - - /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(codec, - codec_dai->playback.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } + + pr_debug("pop wq checking: %s status: %s waiting: %s\n", + codec_dai->driver->playback.stream_name, + codec_dai->playback_active ? "active" : "inactive", + codec_dai->pop_wait ? "yes" : "no"); + + /* are we waiting on this codec DAI stream */ + if (codec_dai->pop_wait == 1) { + codec_dai->pop_wait = 0; + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_STOP); } + mutex_unlock(&pcm_mutex); } @@ -543,22 +539,19 @@ static void close_delayed_work(struct work_struct *work) static int soc_codec_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = card->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; mutex_lock(&pcm_mutex); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback.active--; - codec_dai->playback.active--; + cpu_dai->playback_active--; + codec_dai->playback_active--; } else { - cpu_dai->capture.active--; - codec_dai->capture.active--; + cpu_dai->capture_active--; + codec_dai->capture_active--; } cpu_dai->active--; @@ -571,27 +564,28 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_digital_mute(codec_dai, 1); - if (cpu_dai->ops->shutdown) - cpu_dai->ops->shutdown(substream, cpu_dai); + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); - if (codec_dai->ops->shutdown) - codec_dai->ops->shutdown(substream, codec_dai); + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); - if (machine->ops && machine->ops->shutdown) - machine->ops->shutdown(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); - if (platform->pcm_ops->close) - platform->pcm_ops->close(substream); + if (platform->driver->ops->close) + platform->driver->ops->close(substream); + cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* start delayed pop wq here for playback streams */ codec_dai->pop_wait = 1; - schedule_delayed_work(&card->delayed_work, - msecs_to_jiffies(card->pmdown_time)); + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } else { /* capture streams can be powered down now */ - snd_soc_dapm_stream_event(codec, - codec_dai->capture.stream_name, + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, SND_SOC_DAPM_STREAM_STOP); } @@ -607,43 +601,39 @@ static int soc_codec_close(struct snd_pcm_substream *substream) static int soc_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = card->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); - if (machine->ops && machine->ops->prepare) { - ret = machine->ops->prepare(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + ret = rtd->dai_link->ops->prepare(substream); if (ret < 0) { printk(KERN_ERR "asoc: machine prepare error\n"); goto out; } } - if (platform->pcm_ops->prepare) { - ret = platform->pcm_ops->prepare(substream); + if (platform->driver->ops->prepare) { + ret = platform->driver->ops->prepare(substream); if (ret < 0) { printk(KERN_ERR "asoc: platform prepare error\n"); goto out; } } - if (codec_dai->ops->prepare) { - ret = codec_dai->ops->prepare(substream, codec_dai); + if (codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); goto out; } } - if (cpu_dai->ops->prepare) { - ret = cpu_dai->ops->prepare(substream, cpu_dai); + if (cpu_dai->driver->ops->prepare) { + ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: cpu DAI prepare error\n"); goto out; @@ -654,16 +644,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && codec_dai->pop_wait) { codec_dai->pop_wait = 0; - cancel_delayed_work(&card->delayed_work); + cancel_delayed_work(&rtd->delayed_work); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(codec, - codec_dai->playback.stream_name, + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, SND_SOC_DAPM_STREAM_START); else - snd_soc_dapm_stream_event(codec, - codec_dai->capture.stream_name, + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, SND_SOC_DAPM_STREAM_START); snd_soc_dai_digital_mute(codec_dai, 0); @@ -682,68 +672,65 @@ static int soc_pcm_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_device *socdev = rtd->socdev; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); - if (machine->ops && machine->ops->hw_params) { - ret = machine->ops->hw_params(substream, params); + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { printk(KERN_ERR "asoc: machine hw_params failed\n"); goto out; } } - if (codec_dai->ops->hw_params) { - ret = codec_dai->ops->hw_params(substream, params, codec_dai); + if (codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); + codec_dai->driver->name); goto codec_err; } } - if (cpu_dai->ops->hw_params) { - ret = cpu_dai->ops->hw_params(substream, params, cpu_dai); + if (cpu_dai->driver->ops->hw_params) { + ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); + cpu_dai->driver->name); goto interface_err; } } - if (platform->pcm_ops->hw_params) { - ret = platform->pcm_ops->hw_params(substream, params); + if (platform->driver->ops->hw_params) { + ret = platform->driver->ops->hw_params(substream, params); if (ret < 0) { printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); + platform->driver->name); goto platform_err; } } - machine->rate = params_rate(params); + rtd->rate = params_rate(params); out: mutex_unlock(&pcm_mutex); return ret; platform_err: - if (cpu_dai->ops->hw_free) - cpu_dai->ops->hw_free(substream, cpu_dai); + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->ops->hw_free) - codec_dai->ops->hw_free(substream, codec_dai); + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); codec_err: - if (machine->ops && machine->ops->hw_free) - machine->ops->hw_free(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); mutex_unlock(&pcm_mutex); return ret; @@ -755,13 +742,10 @@ codec_err: static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = card->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; mutex_lock(&pcm_mutex); @@ -770,19 +754,19 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(codec_dai, 1); /* free any machine hw params */ - if (machine->ops && machine->ops->hw_free) - machine->ops->hw_free(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); /* free any DMA resources */ - if (platform->pcm_ops->hw_free) - platform->pcm_ops->hw_free(substream); + if (platform->driver->ops->hw_free) + platform->driver->ops->hw_free(substream); /* now free hw params for the DAI's */ - if (codec_dai->ops->hw_free) - codec_dai->ops->hw_free(substream, codec_dai); + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); - if (cpu_dai->ops->hw_free) - cpu_dai->ops->hw_free(substream, cpu_dai); + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; @@ -791,28 +775,25 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card= socdev->card; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->ops->trigger) { - ret = codec_dai->ops->trigger(substream, cmd, codec_dai); + if (codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } - if (platform->pcm_ops->trigger) { - ret = platform->pcm_ops->trigger(substream, cmd); + if (platform->driver->ops->trigger) { + ret = platform->driver->ops->trigger(substream, cmd); if (ret < 0) return ret; } - if (cpu_dai->ops->trigger) { - ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai); + if (cpu_dai->driver->ops->trigger) { + ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; } @@ -827,27 +808,24 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t offset = 0; snd_pcm_sframes_t delay = 0; - if (platform->pcm_ops->pointer) - offset = platform->pcm_ops->pointer(substream); + if (platform->driver->ops->pointer) + offset = platform->driver->ops->pointer(substream); - if (cpu_dai->ops->delay) - delay += cpu_dai->ops->delay(substream, cpu_dai); + if (cpu_dai->driver->ops->delay) + delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - if (codec_dai->ops->delay) - delay += codec_dai->ops->delay(substream, codec_dai); + if (codec_dai->driver->ops->delay) + delay += codec_dai->driver->ops->delay(substream, codec_dai); - if (platform->delay) - delay += platform->delay(substream, codec_dai); + if (platform->driver->delay) + delay += platform->driver->delay(substream, codec_dai); runtime->delay = delay; @@ -870,104 +848,111 @@ static struct snd_pcm_ops soc_pcm_ops = { static int soc_suspend(struct device *dev) { struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - struct snd_soc_codec *codec = card->codec; + struct snd_soc_card *card = platform_get_drvdata(pdev); int i; /* If the initialization of this soc device failed, there is no codec * associated with it. Just bail out in this case. */ - if (!codec) + if (list_empty(&card->codec_dev_list)) return 0; /* Due to the resume being scheduled into a workqueue we could * suspend before that's finished - wait for it to complete. */ - snd_power_lock(codec->card); - snd_power_wait(codec->card, SNDRV_CTL_POWER_D0); - snd_power_unlock(codec->card); + snd_power_lock(card->snd_card); + snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0); + snd_power_unlock(card->snd_card); /* we're going to block userspace touching us until resume completes */ - snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); + snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot); /* mute any active DAC's */ - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *dai = card->rtd[i].codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (dai->ops->digital_mute && dai->playback.active) - dai->ops->digital_mute(dai, 1); + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 1); } /* suspend all pcms */ - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].ignore_suspend) + for (i = 0; i < card->num_rtd; i++) { + if (card->rtd[i].dai_link->ignore_suspend) continue; - snd_pcm_suspend_all(card->dai_link[i].pcm); + snd_pcm_suspend_all(card->rtd[i].pcm); } if (card->suspend_pre) card->suspend_pre(pdev, PMSG_SUSPEND); - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_platform *platform = card->rtd[i].platform; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->suspend && !cpu_dai->ac97_control) - cpu_dai->suspend(cpu_dai); - if (platform->suspend) - platform->suspend(&card->dai_link[i]); + if (cpu_dai->driver->suspend && !cpu_dai->driver->ac97_control) + cpu_dai->driver->suspend(cpu_dai); + if (platform->driver->suspend && !platform->suspended) { + platform->driver->suspend(cpu_dai); + platform->suspended = 1; + } } /* close any waiting streams and save state */ - run_delayed_work(&card->delayed_work); - codec->suspend_bias_level = codec->bias_level; + for (i = 0; i < card->num_rtd; i++) { + run_delayed_work(&card->rtd[i].delayed_work); + card->rtd[i].codec->suspend_bias_level = card->rtd[i].codec->bias_level; + } - for (i = 0; i < codec->num_dai; i++) { - char *stream = codec->dai[i].playback.stream_name; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai_driver *driver = card->rtd[i].codec_dai->driver; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (stream != NULL) - snd_soc_dapm_stream_event(codec, stream, + if (driver->playback.stream_name != NULL) + snd_soc_dapm_stream_event(&card->rtd[i], driver->playback.stream_name, SND_SOC_DAPM_STREAM_SUSPEND); - stream = codec->dai[i].capture.stream_name; - if (stream != NULL) - snd_soc_dapm_stream_event(codec, stream, + + if (driver->capture.stream_name != NULL) + snd_soc_dapm_stream_event(&card->rtd[i], driver->capture.stream_name, SND_SOC_DAPM_STREAM_SUSPEND); } - /* If there are paths active then the CODEC will be held with - * bias _ON and should not be suspended. */ - if (codec_dev->suspend) { - switch (codec->bias_level) { - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - codec_dev->suspend(pdev, PMSG_SUSPEND); - break; - default: - dev_dbg(socdev->dev, "CODEC is on over suspend\n"); - break; + /* suspend all CODECs */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_codec *codec = card->rtd[i].codec; + /* If there are paths active then the CODEC will be held with + * bias _ON and should not be suspended. */ + if (!codec->suspended && codec->driver->suspend) { + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + codec->driver->suspend(codec, PMSG_SUSPEND); + codec->suspended = 1; + break; + default: + dev_dbg(codec->dev, "CODEC is on over suspend\n"); + break; + } } } - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->suspend && cpu_dai->ac97_control) - cpu_dai->suspend(cpu_dai); + if (cpu_dai->driver->suspend && cpu_dai->driver->ac97_control) + cpu_dai->driver->suspend(cpu_dai); } if (card->suspend_post) @@ -981,127 +966,127 @@ static int soc_suspend(struct device *dev) */ static void soc_resume_deferred(struct work_struct *work) { - struct snd_soc_card *card = container_of(work, - struct snd_soc_card, - deferred_resume_work); - struct snd_soc_device *socdev = card->socdev; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - struct snd_soc_codec *codec = card->codec; - struct platform_device *pdev = to_platform_device(socdev->dev); + struct snd_soc_card *card = + container_of(work, struct snd_soc_card, deferred_resume_work); + struct platform_device *pdev = to_platform_device(card->dev); int i; /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us */ - dev_dbg(socdev->dev, "starting resume work\n"); + dev_dbg(card->dev, "starting resume work\n"); /* Bring us up into D2 so that DAPM starts enabling things */ - snd_power_change_state(codec->card, SNDRV_CTL_POWER_D2); + snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D2); if (card->resume_pre) card->resume_pre(pdev); - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + /* resume AC97 DAIs */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->resume && cpu_dai->ac97_control) - cpu_dai->resume(cpu_dai); - } - - /* If the CODEC was idle over suspend then it will have been - * left with bias OFF or STANDBY and suspended so we must now - * resume. Otherwise the suspend was suppressed. - */ - if (codec_dev->resume) { - switch (codec->bias_level) { - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - codec_dev->resume(pdev); - break; - default: - dev_dbg(socdev->dev, "CODEC was on over suspend\n"); - break; + if (cpu_dai->driver->resume && cpu_dai->driver->ac97_control) + cpu_dai->driver->resume(cpu_dai); + } + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_codec *codec = card->rtd[i].codec; + /* If the CODEC was idle over suspend then it will have been + * left with bias OFF or STANDBY and suspended so we must now + * resume. Otherwise the suspend was suppressed. + */ + if (codec->driver->resume && codec->suspended) { + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + codec->driver->resume(codec); + codec->suspended = 0; + break; + default: + dev_dbg(codec->dev, "CODEC was on over suspend\n"); + break; + } } } - for (i = 0; i < codec->num_dai; i++) { - char *stream = codec->dai[i].playback.stream_name; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai_driver *driver = card->rtd[i].codec_dai->driver; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (stream != NULL) - snd_soc_dapm_stream_event(codec, stream, + if (driver->playback.stream_name != NULL) + snd_soc_dapm_stream_event(&card->rtd[i], driver->playback.stream_name, SND_SOC_DAPM_STREAM_RESUME); - stream = codec->dai[i].capture.stream_name; - if (stream != NULL) - snd_soc_dapm_stream_event(codec, stream, + + if (driver->capture.stream_name != NULL) + snd_soc_dapm_stream_event(&card->rtd[i], driver->capture.stream_name, SND_SOC_DAPM_STREAM_RESUME); } /* unmute any active DACs */ - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *dai = card->rtd[i].codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (dai->ops->digital_mute && dai->playback.active) - dai->ops->digital_mute(dai, 0); + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 0); } - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_platform *platform = card->rtd[i].platform; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->resume && !cpu_dai->ac97_control) - cpu_dai->resume(cpu_dai); - if (platform->resume) - platform->resume(&card->dai_link[i]); + if (cpu_dai->driver->resume && !cpu_dai->driver->ac97_control) + cpu_dai->driver->resume(cpu_dai); + if (platform->driver->resume && platform->suspended) { + platform->driver->resume(cpu_dai); + platform->suspended = 0; + } } if (card->resume_post) card->resume_post(pdev); - dev_dbg(socdev->dev, "resume work completed\n"); + dev_dbg(card->dev, "resume work completed\n"); /* userspace can access us now we are back as we were before */ - snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); + snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); } /* powers up audio subsystem after a suspend */ static int soc_resume(struct device *dev) { struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; - - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (!card->codec) - return 0; + struct snd_soc_card *card = platform_get_drvdata(pdev); + int i; /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume * due to I/O costs and anti-pop so handle them out of line. */ - if (cpu_dai->ac97_control) { - dev_dbg(socdev->dev, "Resuming AC97 immediately\n"); - soc_resume_deferred(&card->deferred_resume_work); - } else { - dev_dbg(socdev->dev, "Scheduling resume work\n"); - if (!schedule_work(&card->deferred_resume_work)) - dev_err(socdev->dev, "resume work item may be lost\n"); + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + if (cpu_dai->driver->ac97_control) { + dev_dbg(dev, "Resuming AC97 immediately\n"); + soc_resume_deferred(&card->deferred_resume_work); + } else { + dev_dbg(dev, "Scheduling resume work\n"); + if (!schedule_work(&card->deferred_resume_work)) + dev_err(dev, "resume work item may be lost\n"); + } } return 0; @@ -1114,198 +1099,408 @@ static int soc_resume(struct device *dev) static struct snd_soc_dai_ops null_dai_ops = { }; -static void snd_soc_instantiate_card(struct snd_soc_card *card) +static void soc_bind_dai_link(struct snd_soc_card *card, int num) { - struct platform_device *pdev = container_of(card->dev, - struct platform_device, - dev); - struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev; + struct snd_soc_dai_link *dai_link = &card->dai_link[num]; + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec; struct snd_soc_platform *platform; - struct snd_soc_dai *dai; - int i, found, ret, ac97; + struct snd_soc_dai *codec_dai, *cpu_dai; + int devices = 0; - if (card->instantiated) + if (rtd->complete) return; - found = 0; - list_for_each_entry(platform, &platform_list, list) - if (card->platform == platform) { - found = 1; - break; + /* do we already have the CPU DAI for this link ? */ + if (rtd->cpu_dai) { + devices++; + goto find_codec_dai; + } + /* no, then find CPU DAI from registered DAIs*/ + list_for_each_entry(cpu_dai, &dai_list, list) { + if ((cpu_dai->id == dai_link->cpu_dai_id || cpu_dai->id == -1) && + cpu_dai->driver == dai_link->cpu_dai_drv) { + rtd->cpu_dai = cpu_dai; + devices++; + goto find_codec_dai; } - if (!found) { - dev_dbg(card->dev, "Platform %s not registered\n", - card->platform->name); - return; } - - ac97 = 0; - for (i = 0; i < card->num_links; i++) { - found = 0; - list_for_each_entry(dai, &dai_list, list) - if (card->dai_link[i].cpu_dai == dai) { - found = 1; - break; - } - if (!found) { - dev_dbg(card->dev, "DAI %s not registered\n", - card->dai_link[i].cpu_dai->name); - return; + dev_dbg(card->dev, "CPU DAI %s not registered\n", + dai_link->cpu_dai_drv->name); + +find_codec_dai: + /* do we already have the CODEC DAI for this link ? */ + if (rtd->codec_dai) { + devices++; + goto find_codec; + } + /* no, then find CPU DAI from registered DAIs*/ + list_for_each_entry(codec_dai, &dai_list, list) { + if ((codec_dai->id == dai_link->codec_dai_id || codec_dai->id == -1) && + codec_dai->driver == dai_link->codec_dai_drv) { + rtd->codec_dai = codec_dai; + devices++; + goto find_codec; + } + } + dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dai_link->codec_dai_drv->name); + +find_codec: + /* do we already have the CODEC for this link ? */ + if (rtd->codec) { + devices++; + goto find_platform; + } + /* no, then find CODEC from registered CODECs*/ + list_for_each_entry(codec, &codec_list, list) { + if ((codec->id == dai_link->codec_id || codec->id == -1) && + codec->driver == dai_link->codec_drv) { + rtd->codec = codec; + devices++; + goto find_platform; } + } + dev_dbg(card->dev, "CODEC %s not registered\n", + dai_link->codec_drv->name); + +find_platform: + /* do we already have the CODEC DAI for this link ? */ + if (rtd->platform) { + devices++; + goto out; + } + /* no, then find CPU DAI from registered DAIs*/ + list_for_each_entry(platform, &platform_list, list) { + if ((platform->id == dai_link->platform_id || platform->id == -1) && + platform->driver == dai_link->platform_drv) { + rtd->platform = platform; + devices++; + goto out; + } + } - if (card->dai_link[i].cpu_dai->ac97_control) - ac97 = 1; + dev_dbg(card->dev, "platform %s not registered\n", + dai_link->platform_drv->name); + return; + +out: + /* mark rtd as complete if we found all 4 of our client devices */ + if (devices == 4) { + rtd->complete = 1; + card->num_rtd++; } +} - for (i = 0; i < card->num_links; i++) { - if (!card->dai_link[i].codec_dai->ops) - card->dai_link[i].codec_dai->ops = &null_dai_ops; +static void soc_remove_dai_link(struct snd_soc_card *card, int num) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; + int err; + + /* unregister the rtd device */ + if (rtd->dev_registered) { + device_remove_file(&rtd->dev, &dev_attr_pmdown_time); + device_unregister(&rtd->dev); + rtd->dev_registered = 0; } - /* If we have AC97 in the system then don't wait for the - * codec. This will need revisiting if we have to handle - * systems with mixed AC97 and non-AC97 parts. Only check for - * DAIs currently; we can't do this per link since some AC97 - * codecs have non-AC97 DAIs. - */ - if (!ac97) - for (i = 0; i < card->num_links; i++) { - found = 0; - list_for_each_entry(dai, &dai_list, list) - if (card->dai_link[i].codec_dai == dai) { - found = 1; - break; - } - if (!found) { - dev_dbg(card->dev, "DAI %s not registered\n", - card->dai_link[i].codec_dai->name); - return; - } + /* remove the CODEC DAI */ + if (codec_dai && codec_dai->probed) { + if (codec_dai->driver->remove) { + err = codec_dai->driver->remove(codec_dai); + if (err < 0) + printk(KERN_ERR "asoc: failed to remove %s\n", codec_dai->name); } + codec_dai->probed = 0; + list_del(&codec_dai->card_list); + } - /* Note that we do not current check for codec components */ + /* remove the platform */ + if (platform && platform->probed) { + if (platform->driver->remove) { + err = platform->driver->remove(platform); + if (err < 0) + printk(KERN_ERR "asoc: failed to remove %s\n", platform->name); + } + platform->probed = 0; + list_del(&platform->card_list); + } - dev_dbg(card->dev, "All components present, instantiating\n"); + /* remove the CODEC */ + if (codec && codec->probed) { + if (codec->driver->remove) { + err = codec->driver->remove(codec); + if (err < 0) + printk(KERN_ERR "asoc: failed to remove %s\n", codec->name); + } - /* Found everything, bring it up */ - card->pmdown_time = pmdown_time; + /* Make sure all DAPM widgets are freed */ + snd_soc_dapm_free(codec); - if (card->probe) { - ret = card->probe(pdev); - if (ret < 0) - return; + soc_cleanup_codec_debugfs(codec); + device_remove_file(codec->dev, &dev_attr_codec_reg); + codec->probed = 0; + list_del(&codec->card_list); } - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->probe) { - ret = cpu_dai->probe(pdev, cpu_dai); + /* remove the cpu_dai */ + if (cpu_dai && cpu_dai->probed) { + if (cpu_dai->driver->remove) { + err = cpu_dai->driver->remove(cpu_dai); + if (err < 0) + printk(KERN_ERR "asoc: failed to remove %s\n", cpu_dai->name); + } + cpu_dai->probed = 0; + list_del(&cpu_dai->card_list); + } +} + +static int soc_probe_dai_link(struct snd_soc_card *card, int num) +{ + struct snd_soc_dai_link *dai_link = &card->dai_link[num]; + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; + int ret; + + dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num); + + /* config components */ + codec->snd_card = card->snd_card; + codec_dai->codec = codec; + codec->card = card; + cpu_dai->platform = platform; + rtd->card = card; + rtd->dev.parent = card->dev; + + /* set default power off timeout */ + rtd->pmdown_time = pmdown_time; + + /* probe the cpu_dai */ + if (!cpu_dai->probed) { + if (cpu_dai->driver->probe) { + ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) - goto cpu_dai_err; + return ret; } + cpu_dai->probed = 1; + /* mark cpu_dai as probed and add to card cpu_dai list */ + list_add(&cpu_dai->card_list, &card->dai_dev_list); } - if (codec_dev->probe) { - ret = codec_dev->probe(pdev); + /* probe the CODEC */ + if (!codec->probed) { + if (codec->driver->probe) { + ret = codec->driver->probe(codec); + if (ret < 0) + return ret; + } + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codec); + + /* mark codec as probed and add to card codec list */ + codec->probed = 1; + list_add(&codec->card_list, &card->codec_dev_list); + + /* add DAPM sysfs entries for this codec */ + ret = snd_soc_dapm_sys_add(codec->dev); if (ret < 0) - goto cpu_dai_err; - } - codec = card->codec; + printk(KERN_WARNING "asoc: failed to add codec dapm sysfs entries\n"); - if (platform->probe) { - ret = platform->probe(pdev); + /* add codec sysfs entries */ + ret = device_create_file(codec->dev, &dev_attr_codec_reg); if (ret < 0) - goto platform_err; + printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + + soc_init_codec_debugfs(codec); } - /* DAPM stream work */ - INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work); -#ifdef CONFIG_PM - /* deferred resume work */ - INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); -#endif + /* probe the platform */ + if (!platform->probed) { + if (platform->driver->probe) { + ret = platform->driver->probe(platform); + if (ret < 0) + return ret; + } + /* mark platform as probed and add to card platform list */ + platform->probed = 1; + list_add(&platform->card_list, &card->platform_dev_list); + } - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].init) { - ret = card->dai_link[i].init(codec); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to init %s\n", - card->dai_link[i].stream_name); - continue; - } + /* probe the CODEC DAI */ + if (!codec_dai->probed) { + if (codec_dai->driver->probe) { + ret = codec_dai->driver->probe(codec_dai); + if (ret < 0) + return ret; } - if (card->dai_link[i].codec_dai->ac97_control) - ac97 = 1; + + /* mark cpu_dai as probed and add to card cpu_dai list */ + codec_dai->probed = 1; + list_add(&codec_dai->card_list, &card->dai_dev_list); } - snprintf(codec->card->shortname, sizeof(codec->card->shortname), - "%s", card->name); - snprintf(codec->card->longname, sizeof(codec->card->longname), - "%s (%s)", card->name, codec->name); + /* DAPM dai link stream work */ + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(codec); + /* now that all clients have probed, initialise the DAI link */ + if (dai_link->init) { + ret = dai_link->init(rtd); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to init %s\n", dai_link->stream_name); + return ret; + } + } - ret = snd_card_register(codec->card); + /* register the rtd device */ + rtd->dev.init_name = rtd->dai_link->stream_name; + ret = device_register(&rtd->dev); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for %s\n", - codec->name); - goto card_err; + printk(KERN_ERR "asoc: failed to register DAI runtime device %d\n", ret); + return ret; } + rtd->dev_registered = 1; + ret = device_create_file(&rtd->dev, &dev_attr_pmdown_time); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); + + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + printk(KERN_ERR "asoc: can't create pcm %s\n", dai_link->stream_name); + return ret; + } + + /* add platform data for AC97 devices */ + if (dai_link->codec_dai_drv->ac97_control) + snd_ac97_dev_add_pdata(codec->ac97, rtd->cpu_dai->ac97_pdata); + + return 0; +} - mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS +static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + /* Only instantiate AC97 if not already done by the adaptor * for the generic AC97 subsystem. */ - if (ac97 && strcmp(codec->name, "AC97") != 0) { - ret = soc_ac97_dev_register(codec); + if (rtd->codec_dai->driver->ac97_control && !rtd->codec->ac97_registered) { + + ret = soc_ac97_dev_register(rtd->codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); - snd_card_free(codec->card); - mutex_unlock(&codec->mutex); - goto card_err; + return ret; } + + rtd->codec->ac97_registered = 1; } + return 0; +} + +static void soc_unregister_ac97_dai_link(struct snd_soc_codec *codec) +{ + if (codec->ac97_registered) { + soc_ac97_dev_unregister(codec); + codec->ac97_registered = 0; + } +} #endif - ret = snd_soc_dapm_sys_add(card->socdev->dev); - if (ret < 0) - printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); +static void snd_soc_instantiate_card(struct snd_soc_card *card) +{ + struct platform_device *pdev = to_platform_device(card->dev); + int ret, i; - ret = device_create_file(card->socdev->dev, &dev_attr_pmdown_time); - if (ret < 0) - printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); + mutex_lock(&card->mutex); - ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg); - if (ret < 0) - printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + if (card->instantiated) { + mutex_unlock(&card->mutex); + return; + } - soc_init_codec_debugfs(codec); - mutex_unlock(&codec->mutex); + /* bind DAIs */ + for (i = 0; i < card->num_links; i++) + soc_bind_dai_link(card, i); - card->instantiated = 1; + /* bind completed ? */ + if (card->num_rtd != card->num_links) { + mutex_unlock(&card->mutex); + return; + } - return; + /* card bind complete so register a sound card */ + ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + card->owner, 0, &card->snd_card); + if (ret < 0) { + printk(KERN_ERR "asoc: can't create sound card for card %s\n", + card->name); + mutex_unlock(&card->mutex); + return; + } + card->snd_card->dev = card->dev; -card_err: - if (platform->remove) - platform->remove(pdev); + /* initialise the sound card only once */ + if (card->probe) { +#ifdef CONFIG_PM + /* deferred resume work */ + INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); +#endif + ret = card->probe(pdev); + if (ret < 0) + goto card_probe_error; + } -platform_err: - if (codec_dev->remove) - codec_dev->remove(pdev); + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_dai_link(card, i); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to instanciate card %s\n", card->name); + goto probe_dai_err; + } + } + + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), + "%s", card->name); + snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), + "%s", card->name); -cpu_dai_err: - for (i--; i >= 0; i--) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->remove) - cpu_dai->remove(pdev, cpu_dai); + ret = snd_card_register(card->snd_card); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); + goto probe_dai_err; } +#ifdef CONFIG_SND_SOC_AC97_BUS + /* register any AC97 codecs */ + for (i = 0; i < card->num_rtd; i++) { + ret = soc_register_ac97_dai_link(&card->rtd[i]); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + goto probe_dai_err; + } + } +#endif + + card->instantiated = 1; + mutex_unlock(&card->mutex); + return; + +probe_dai_err: + for (i = 0; i < card->num_links; i++) + soc_remove_dai_link(card, i); + +card_probe_error: if (card->remove) card->remove(pdev); + + snd_card_free(card->snd_card); + + mutex_unlock(&card->mutex); } /* @@ -1322,15 +1517,15 @@ static void snd_soc_instantiate_cards(void) /* probes a new socdev */ static int soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); int ret = 0; - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - - /* Bodge while we push things out of socdev */ - card->socdev = socdev; /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; + INIT_LIST_HEAD(&card->dai_dev_list); + INIT_LIST_HEAD(&card->codec_dev_list); + INIT_LIST_HEAD(&card->platform_dev_list); + ret = snd_soc_register_card(card); if (ret != 0) { dev_err(&pdev->dev, "Failed to register card\n"); @@ -1343,50 +1538,49 @@ static int soc_probe(struct platform_device *pdev) /* removes a socdev */ static int soc_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); int i; - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - if (card->instantiated) { - run_delayed_work(&card->delayed_work); - - if (platform->remove) - platform->remove(pdev); - - if (codec_dev->remove) - codec_dev->remove(pdev); + if (card->instantiated) { - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->remove) - cpu_dai->remove(pdev, cpu_dai); + /* make sure any delayed work runs */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + run_delayed_work(&rtd->delayed_work); } + /* remove and free each DAI */ + for (i = 0; i < card->num_rtd; i++) + soc_remove_dai_link(card, i); + + /* remove the card */ if (card->remove) card->remove(pdev); - } + kfree(card->rtd); + snd_card_free(card->snd_card); + } snd_soc_unregister_card(card); - return 0; } static int soc_poweroff(struct device *dev) { struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; + struct snd_soc_card *card = platform_get_drvdata(pdev); + int i; if (!card->instantiated) return 0; /* Flush out pmdown_time work - we actually do want to run it * now, we're shutting down so no imminent restart. */ - run_delayed_work(&card->delayed_work); + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + run_delayed_work(&rtd->delayed_work); + } - snd_soc_dapm_shutdown(socdev); + snd_soc_dapm_shutdown(card); return 0; } @@ -1409,53 +1603,42 @@ static struct platform_driver soc_driver = { }; /* create a new pcm */ -static int soc_new_pcm(struct snd_soc_device *socdev, - struct snd_soc_dai_link *dai_link, int num) -{ - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = card->codec; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *codec_dai = dai_link->codec_dai; - struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; - struct snd_soc_pcm_runtime *rtd; +static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; - rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); - if (rtd == NULL) - return -ENOMEM; - - rtd->dai = dai_link; - rtd->socdev = socdev; - codec_dai->codec = card->codec; - /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", - dai_link->stream_name, codec_dai->name, num); + rtd->dai_link->stream_name, codec_dai->driver->name, num); - if (codec_dai->playback.channels_min) + if (codec_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->capture.channels_min) + if (codec_dai->driver->capture.channels_min) capture = 1; - ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback, - capture, &pcm); + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); + ret = snd_pcm_new(rtd->card->snd_card, new_name, + num, playback, capture, &pcm); if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", - codec->name); - kfree(rtd); + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); return ret; } - dai_link->pcm = pcm; + rtd->pcm = pcm; pcm->private_data = rtd; - soc_pcm_ops.mmap = platform->pcm_ops->mmap; - soc_pcm_ops.ioctl = platform->pcm_ops->ioctl; - soc_pcm_ops.copy = platform->pcm_ops->copy; - soc_pcm_ops.silence = platform->pcm_ops->silence; - soc_pcm_ops.ack = platform->pcm_ops->ack; - soc_pcm_ops.page = platform->pcm_ops->page; + soc_pcm_ops.mmap = platform->driver->ops->mmap; + soc_pcm_ops.pointer = platform->driver->ops->pointer; + soc_pcm_ops.ioctl = platform->driver->ops->ioctl; + soc_pcm_ops.copy = platform->driver->ops->copy; + soc_pcm_ops.silence = platform->driver->ops->silence; + soc_pcm_ops.ack = platform->driver->ops->ack; + soc_pcm_ops.page = platform->driver->ops->page; if (playback) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); @@ -1463,16 +1646,15 @@ static int soc_new_pcm(struct snd_soc_device *socdev, if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); - ret = platform->pcm_new(codec->card, codec_dai, pcm); + ret = platform->driver->pcm_new(rtd->card->snd_card, codec_dai, pcm); if (ret < 0) { printk(KERN_ERR "asoc: platform pcm constructor failed\n"); - kfree(rtd); return ret; } - pcm->private_free = platform->pcm_free; - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); + pcm->private_free = platform->driver->pcm_free; + printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->driver->name, + cpu_dai->driver->name); return ret; } @@ -1486,8 +1668,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, */ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) { - if (codec->volatile_register) - return codec->volatile_register(reg); + if (codec->driver->volatile_register) + return codec->driver->volatile_register(reg); else return 0; } @@ -1522,7 +1704,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, codec->ac97->bus->ops = ops; codec->ac97->num = num; - codec->dev = &codec->ac97->dev; mutex_unlock(&codec->mutex); return 0; } @@ -1537,6 +1718,9 @@ EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) { mutex_lock(&codec->mutex); +#ifdef CONFIG_SND_SOC_AC97_BUS + soc_unregister_ac97_dai_link(codec); +#endif kfree(codec->ac97->bus); kfree(codec->ac97); codec->ac97 = NULL; @@ -1623,95 +1807,6 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, EXPORT_SYMBOL_GPL(snd_soc_test_bits); /** - * snd_soc_new_pcms - create new sound card and pcms - * @socdev: the SoC audio device - * @idx: ALSA card index - * @xid: card identification - * - * Create a new sound card based upon the codec and interface pcms. - * - * Returns 0 for success, else error. - */ -int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) -{ - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = card->codec; - int ret, i; - - mutex_lock(&codec->mutex); - - /* register a sound card */ - ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card); - if (ret < 0) { - printk(KERN_ERR "asoc: can't create sound card for codec %s\n", - codec->name); - mutex_unlock(&codec->mutex); - return ret; - } - - codec->socdev = socdev; - codec->card->dev = socdev->dev; - codec->card->private_data = codec; - strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); - - /* create the pcms */ - for (i = 0; i < card->num_links; i++) { - ret = soc_new_pcm(socdev, &card->dai_link[i], i); - if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm %s\n", - card->dai_link[i].stream_name); - mutex_unlock(&codec->mutex); - return ret; - } - /* Check for codec->ac97 to handle the ac97.c fun */ - if (card->dai_link[i].codec_dai->ac97_control && codec->ac97) { - snd_ac97_dev_add_pdata(codec->ac97, - card->dai_link[i].cpu_dai->ac97_pdata); - } - } - - mutex_unlock(&codec->mutex); - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_new_pcms); - -/** - * snd_soc_free_pcms - free sound card and pcms - * @socdev: the SoC audio device - * - * Frees sound card and pcms associated with the socdev. - * Also unregister the codec if it is an AC97 device. - */ -void snd_soc_free_pcms(struct snd_soc_device *socdev) -{ - struct snd_soc_codec *codec = socdev->card->codec; -#ifdef CONFIG_SND_SOC_AC97_BUS - struct snd_soc_dai *codec_dai; - int i; -#endif - - mutex_lock(&codec->mutex); - soc_cleanup_codec_debugfs(codec); -#ifdef CONFIG_SND_SOC_AC97_BUS - for (i = 0; i < codec->num_dai; i++) { - codec_dai = &codec->dai[i]; - if (codec_dai->ac97_control && codec->ac97 && - strcmp(codec->name, "AC97") != 0) { - soc_ac97_dev_unregister(codec); - goto free_card; - } - } -free_card: -#endif - - if (codec->card) - snd_card_free(codec->card); - device_remove_file(socdev->dev, &dev_attr_codec_reg); - mutex_unlock(&codec->mutex); -} -EXPORT_SYMBOL_GPL(snd_soc_free_pcms); - -/** * snd_soc_set_runtime_hwparams - set the runtime hardware parameters * @substream: the pcm substream * @hw: the hardware parameters @@ -1772,7 +1867,7 @@ EXPORT_SYMBOL_GPL(snd_soc_cnew); int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { - struct snd_card *card = codec->card; + struct snd_card *card = codec->snd_card; int err, i; for (i = 0; i < num_controls; i++) { @@ -2327,7 +2422,7 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max) { - struct snd_card *card = codec->card; + struct snd_card *card = codec->snd_card; struct snd_kcontrol *kctl; struct soc_mixer_control *mc; int found = 0; @@ -2459,8 +2554,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->ops && dai->ops->set_sysclk) - return dai->ops->set_sysclk(dai, clk_id, freq, dir); + if (dai->driver && dai->driver->ops->set_sysclk) + return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; } @@ -2479,8 +2574,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->ops && dai->ops->set_clkdiv) - return dai->ops->set_clkdiv(dai, div_id, div); + if (dai->driver && dai->driver->ops->set_clkdiv) + return dai->driver->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; } @@ -2499,8 +2594,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - if (dai->ops && dai->ops->set_pll) - return dai->ops->set_pll(dai, pll_id, source, + if (dai->driver && dai->driver->ops->set_pll) + return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); else return -EINVAL; @@ -2516,8 +2611,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->ops && dai->ops->set_fmt) - return dai->ops->set_fmt(dai, fmt); + if (dai->driver && dai->driver->ops->set_fmt) + return dai->driver->ops->set_fmt(dai, fmt); else return -EINVAL; } @@ -2537,8 +2632,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + if (dai->driver && dai->driver->ops->set_tdm_slot) + return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, slots, slot_width); else return -EINVAL; @@ -2561,8 +2656,8 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot) { - if (dai->ops && dai->ops->set_channel_map) - return dai->ops->set_channel_map(dai, tx_num, tx_slot, + if (dai->driver && dai->driver->ops->set_channel_map) + return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot, rx_num, rx_slot); else return -EINVAL; @@ -2578,8 +2673,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->ops && dai->ops->set_tristate) - return dai->ops->set_tristate(dai, tristate); + if (dai->driver && dai->driver->ops->set_tristate) + return dai->driver->ops->set_tristate(dai, tristate); else return -EINVAL; } @@ -2594,8 +2689,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->ops && dai->ops->digital_mute) - return dai->ops->digital_mute(dai, mute); + if (dai->driver && dai->driver->ops->digital_mute) + return dai->driver->ops->digital_mute(dai, mute); else return -EINVAL; } @@ -2612,11 +2707,22 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ static int snd_soc_register_card(struct snd_soc_card *card) { + int i; + if (!card->name || !card->dev) return -EINVAL; + card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * card->num_links, + GFP_KERNEL); + if (card->rtd == NULL) + return -ENOMEM; + + for (i = 0; i < card->num_links; i++) + card->rtd[i].dai_link = &card->dai_link[i]; + INIT_LIST_HEAD(&card->list); card->instantiated = 0; + mutex_init(&card->mutex); mutex_lock(&client_mutex); list_add(&card->list, &card_list); @@ -2642,7 +2748,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) mutex_lock(&client_mutex); list_del(&card->list); mutex_unlock(&client_mutex); - dev_dbg(card->dev, "Unregistered card '%s'\n", card->name); return 0; @@ -2653,19 +2758,32 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) * * @dai: DAI to register */ -int snd_soc_register_dai(struct snd_soc_dai *dai) +int snd_soc_register_dai(struct device *dev, int id, + struct snd_soc_dai_driver *dai_drv) { - if (!dai->name) - return -EINVAL; + struct snd_soc_dai *dai; + char name[NAME_SIZE]; - /* The device should become mandatory over time */ - if (!dai->dev) - printk(KERN_WARNING "No device for DAI %s\n", dai->name); + dev_dbg(dev, "dai register %s %d\n", dai_drv->name, id); - if (!dai->ops) - dai->ops = &null_dai_ops; + if (!dai_drv->name) + return -EINVAL; - INIT_LIST_HEAD(&dai->list); + dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); + if (dai == NULL) + return -ENOMEM; + snprintf(name, NAME_SIZE, "%s.%d", dai_drv->name, id); + dai->name = kstrdup(name, GFP_KERNEL); + if (dai->name == NULL) { + kfree(dai); + return -ENOMEM; + } + + dai->dev = dev; + dai->driver = dai_drv; + if (!dai->driver->ops) + dai->driver->ops = &null_dai_ops; + dai->id = id; mutex_lock(&client_mutex); list_add(&dai->list, &dai_list); @@ -2683,13 +2801,24 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dai); * * @dai: DAI to unregister */ -void snd_soc_unregister_dai(struct snd_soc_dai *dai) +void snd_soc_unregister_dai(struct device *dev, int id) { + struct snd_soc_dai *dai; + + list_for_each_entry(dai, &dai_list, list) { + if (dai->id == id && dev == dai->dev) + goto found; + } + return; + +found: mutex_lock(&client_mutex); list_del(&dai->list); mutex_unlock(&client_mutex); pr_debug("Unregistered DAI '%s'\n", dai->name); + kfree(dai->name); + kfree(dai); } EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); @@ -2699,12 +2828,13 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); * @dai: Array of DAIs to register * @count: Number of DAIs */ -int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count) +int snd_soc_register_dais(struct device *dev, + struct snd_soc_dai_driver *dai_drv, size_t count) { int i, ret; for (i = 0; i < count; i++) { - ret = snd_soc_register_dai(&dai[i]); + ret = snd_soc_register_dai(dev, i, &dai_drv[i]); if (ret != 0) goto err; } @@ -2713,7 +2843,7 @@ int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count) err: for (i--; i >= 0; i--) - snd_soc_unregister_dai(&dai[i]); + snd_soc_unregister_dai(dev, i); return ret; } @@ -2725,12 +2855,12 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dais); * @dai: Array of DAIs to unregister * @count: Number of DAIs */ -void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count) +void snd_soc_unregister_dais(struct device *dev, size_t count) { int i; for (i = 0; i < count; i++) - snd_soc_unregister_dai(&dai[i]); + snd_soc_unregister_dai(dev, i); } EXPORT_SYMBOL_GPL(snd_soc_unregister_dais); @@ -2739,19 +2869,37 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dais); * * @platform: platform to register */ -int snd_soc_register_platform(struct snd_soc_platform *platform) +int snd_soc_register_platform(struct device *dev, int id, + struct snd_soc_platform_driver *platform_drv) { - if (!platform->name) + struct snd_soc_platform *platform; + char name[NAME_SIZE]; + + dev_dbg(dev, "platform register %s %d\n", platform_drv->name, id); + + if (!platform_drv->name) return -EINVAL; - INIT_LIST_HEAD(&platform->list); + platform = kzalloc(sizeof(struct snd_soc_platform), GFP_KERNEL); + if (platform == NULL) + return -ENOMEM; + snprintf(name, NAME_SIZE, "%s.%d", platform_drv->name, id); + platform->name = kstrdup(name, GFP_KERNEL); + if (platform->name == NULL) { + kfree(platform); + return -ENOMEM; + } + + platform->dev = dev; + platform->driver = platform_drv; + platform->id = id; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); snd_soc_instantiate_cards(); mutex_unlock(&client_mutex); - pr_debug("Registered platform '%s'\n", platform->name); + pr_debug("Registered platform '%s'\n", platform_drv->name); return 0; } @@ -2762,13 +2910,24 @@ EXPORT_SYMBOL_GPL(snd_soc_register_platform); * * @platform: platform to unregister */ -void snd_soc_unregister_platform(struct snd_soc_platform *platform) +void snd_soc_unregister_platform(struct device *dev, int id) { + struct snd_soc_platform *platform; + + list_for_each_entry(platform, &platform_list, list) { + if (platform->id == id && dev == platform->dev) + goto found; + } + return; + +found: mutex_lock(&client_mutex); list_del(&platform->list); mutex_unlock(&client_mutex); pr_debug("Unregistered platform '%s'\n", platform->name); + kfree(platform->name); + kfree(platform); } EXPORT_SYMBOL_GPL(snd_soc_unregister_platform); @@ -2810,22 +2969,58 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) * * @codec: codec to register */ -int snd_soc_register_codec(struct snd_soc_codec *codec) +int snd_soc_register_codec(struct device *dev, int id, + struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai) { - int i; + struct snd_soc_codec *codec; + char name[NAME_SIZE]; + int ret, i; + + dev_dbg(dev, "codec register %s %d\n", codec_drv->name, id); - if (!codec->name) + if (!codec_drv->name) return -EINVAL; - /* The device should become mandatory over time */ - if (!codec->dev) - printk(KERN_WARNING "No device for codec %s\n", codec->name); + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + snprintf(name, NAME_SIZE, "%s.%d", codec_drv->name, id); + codec->name = kstrdup(name, GFP_KERNEL); + if (codec->name == NULL) { + kfree(codec); + return -ENOMEM; + } + + /* allocate CODEC register cache */ + if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { - INIT_LIST_HEAD(&codec->list); + if (codec_drv->reg_cache_default) + codec->reg_cache = kmemdup(codec_drv->reg_cache_default, + codec_drv->reg_cache_size * codec_drv->reg_word_size, GFP_KERNEL); + else + codec->reg_cache = kzalloc(codec_drv->reg_cache_size * + codec_drv->reg_word_size, GFP_KERNEL); - for (i = 0; i < codec->num_dai; i++) { - fixup_codec_formats(&codec->dai[i].playback); - fixup_codec_formats(&codec->dai[i].capture); + if (codec->reg_cache == NULL) { + kfree(codec->name); + kfree(codec); + return -ENOMEM; + } + } + + codec->id = id; + codec->dev = dev; + codec->driver = codec_drv; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->num_dai = num_dai; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + for (i = 0; i < num_dai; i++) { + fixup_codec_formats(&dai_drv[i].playback); + fixup_codec_formats(&dai_drv[i].capture); } mutex_lock(&client_mutex); @@ -2833,9 +3028,29 @@ int snd_soc_register_codec(struct snd_soc_codec *codec) snd_soc_instantiate_cards(); mutex_unlock(&client_mutex); - pr_debug("Registered codec '%s'\n", codec->name); + /* register DAIs */ + for (i = 0; i < num_dai; i++) { + ret = snd_soc_register_dai(dev, i, &dai_drv[i]); + if (ret != 0) + goto error; + } + pr_debug("Registered codec '%s'\n", codec->name); return 0; + +error: + for (i--; i >= 0; i--) + snd_soc_unregister_dai(dev, i); + + mutex_lock(&client_mutex); + list_del(&codec->list); + mutex_unlock(&client_mutex); + + if (codec->reg_cache) + kfree(codec->reg_cache); + kfree(codec->name); + kfree(codec); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_codec); @@ -2844,13 +3059,31 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); * * @codec: codec to unregister */ -void snd_soc_unregister_codec(struct snd_soc_codec *codec) +void snd_soc_unregister_codec(struct device *dev, int id) { + struct snd_soc_codec *codec; + int i; + + list_for_each_entry(codec, &codec_list, list) { + if (codec->id == id && dev == codec->dev) + goto found; + } + return; + +found: + for (i = 0; i < codec->num_dai; i++) + snd_soc_unregister_dai(dev, id); + mutex_lock(&client_mutex); list_del(&codec->list); mutex_unlock(&client_mutex); pr_debug("Unregistered codec '%s'\n", codec->name); + + if (codec->reg_cache) + kfree(codec->reg_cache); + kfree(codec->name); + kfree(codec); } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); -- 1.7.0.4

2 2

[alsa-devel] Introduction from Robert.
by robert 01 Jul '10

01 Jul '10

Dear All, My name is Robert. I have chosen ALSA to develop my own audio application for Linux. How are you? Cheers, Robert

1 0

[alsa-devel] [PATCH 1/2] ASoC: uda134x: add DATA011 register found in codecs family
by Vladimir Zapolskiy 30 Jun '10

30 Jun '10

In UDA1340, UDA1344 and UDA1345 codecs there is one more functional register in part of DATA0 tranfser. For UDA1341 this register coincides with EA register. Signed-off-by: Vladimir Zapolskiy <vzapolskiy(a)gmail.com> --- sound/soc/codecs/uda134x.c | 3 ++- sound/soc/codecs/uda134x.h | 5 +++-- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index b3039b1..e770029 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -57,7 +57,7 @@ static const char uda134x_reg[UDA134X_REGS_NUM] = { /* Extended address registers */ 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, /* Status, data regs */ - 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, + 0x00, 0x83, 0x00, 0x40, 0x80, 0xC0, 0x00, }; /* @@ -116,6 +116,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, case UDA134X_DATA000: case UDA134X_DATA001: case UDA134X_DATA010: + case UDA134X_DATA011: addr = UDA134X_DATA0_ADDR; break; case UDA134X_DATA1: diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h index 94f4404..205f03b 100644 --- a/sound/soc/codecs/uda134x.h +++ b/sound/soc/codecs/uda134x.h @@ -23,9 +23,10 @@ #define UDA134X_DATA000 10 #define UDA134X_DATA001 11 #define UDA134X_DATA010 12 -#define UDA134X_DATA1 13 +#define UDA134X_DATA011 13 +#define UDA134X_DATA1 14 -#define UDA134X_REGS_NUM 14 +#define UDA134X_REGS_NUM 15 #define STATUS0_DAIFMT_MASK (~(7<<1)) #define STATUS0_SYSCLK_MASK (~(3<<4)) -- 1.7.0.4

2 10

[alsa-devel] [PATCH] ASoC: DaVinci: Added support for cpu clocking I2S
by Raffaele Recalcati 30 Jun '10

30 Jun '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added audio playback support with [frame sync master - clock master] mode and with [frame sync master - clock slave]. 1. SND_SOC_DAIFMT_CBS_CFS (the cpu generates clock and frame sync) I need to clarify how the McBSP is clocked internally. Reading sprufi3a-1.pdf (TMS320DM36x DMSoC Multichannel Buffered Serial Port (McBSP) Interface) at "2.5.3 Data Clock Generation" paragraph this info not appears really clear if we are in the case of "McBSP internal input clock". I hope some Ti developers can help me. There is a FIXME message in the patch. i2s_fast_clock switch can be used to have better approximate or symmetric waveforms. clk_input_pin board info can be used to select it depending on hw connections 2. SND_SOC_DAIFMT_CBS_CFM (the cpu get clock from external pin and generates frame sync) Clock slave can be important when the external codec need system clock and bit clock synchronized (tested with uda1345). 3. We haven't changed the evmdm365 support (due also to CPLD that doesn't help to understand) We don't know in this mode if audio stereo works on evmdm365. Probably it does. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 16 ++++ sound/soc/davinci/davinci-i2s.c | 126 ++++++++++++++++++++++++++---- 2 files changed, 128 insertions(+), 14 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..f9b6da2 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,16 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not simmetrical, but sampling + * frequency is better approximated + */ + int i2s_fast_clock; + + /* To be used when cpu gets clock from extenal pin */ + int clk_input_pin; + /* McASP specific fields */ int tdm_slots; u8 op_mode; @@ -78,6 +88,12 @@ enum { MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; +enum { + MCBSP_CLKR = 0, /* DM365 */ + MCBSP_CLKS, +}; + + #define INACTIVE_MODE 0 #define TX_MODE 1 #define RX_MODE 2 diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..786ade6 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -68,16 +68,21 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE (1 << 31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE (1 << 31) #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM (1 << 29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,6 +149,11 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + int i2s_fast_clock; + unsigned int fmt; + int clk_div; + int clk_input_pin; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -254,10 +264,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = cpu_dai->private_data; unsigned int pcr; unsigned int srgr; + /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + dev->fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -268,11 +280,18 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: - /* McBSP CLKR pin is the input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + if (dev->clk_input_pin == MCBSP_CLKS) + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + else + /* + * McBSP CLKR pin is the input for the Sample Rate + * Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + pcr |= DAVINCI_MCBSP_PCR_SCLKME; break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -372,6 +391,16 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + int srgr; + + dev->clk_div = div; + return 0; +} + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,11 +409,12 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; + struct clk *clk; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -396,12 +426,56 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + if (master == SND_SOC_DAIFMT_CBS_CFS) { + clk = clk_get(NULL, "pll1_sysclk6"); + if (clk) + freq = clk_get_rate(clk); + freq = 122000000; /* FIXME ask to Texas */ + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } + clk_div &= 0xFF; + srgr |= clk_div; + } else if (master == SND_SOC_DAIFMT_CBM_CFS) { + /* Clock given on CLKS */ + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = dev->clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + } else { + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,12 +500,29 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -442,6 +533,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,6 +595,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; @@ -552,6 +648,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->i2s_fast_clock = pdata->i2s_fast_clock; + dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { -- 1.7.0.4

3 6

[alsa-devel] [PATCH v2] ASoC: ak4642: Add Digital Playback Volume control
by Kuninori Morimoto 30 Jun '10

30 Jun '10

This patch add Digital Playback Volume control. And it moves default digital playback volume settings from ak4642_dai_startup to ak4642_init Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 o "Playback Volume" => "Digital Playback Volume" o Add explain on commit log sound/soc/codecs/ak4642.c | 28 +++++++++++++++++++++++++--- 1 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7528a54..7b96311 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -36,6 +36,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -111,6 +112,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -204,7 +222,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -214,8 +231,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -463,6 +478,10 @@ static int ak4642_init(struct ak4642_priv *ak4642) goto reg_cache_err; } + /* default volume */ + ak4642_write(codec, L_DVC, 0x28); + ak4642_write(codec, R_DVC, 0x28); + return ret; reg_cache_err: @@ -548,6 +567,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -- 1.7.0.4

2 1

[alsa-devel] alsamixer-qt4 release
by Sebastian H. 30 Jun '10

30 Jun '10

Hello list Here is (yet another) mixer application for ALSA: alsamixer-qt4 It took a bit longer that expected but today I've erased the last items from the TODO list for the first release version 0.1.0. All files are hosted at sourceforge http://sourceforge.net/projects/alsamixer-qt4/files/ And here's a small web page http://xwmw.org/alsamixer-qt4 At the moment it is a volume only mixer and the dB value is only displayed but the scaling sliders are worth a look (The style is homebrewn ;). So far. Regards, Sebastian H.

2 2

[alsa-devel] snddevices issue
by Adish Kuvelker 29 Jun '10

29 Jun '10

Hello, I find that the device nodes created in the /dev/snd for Control, Timer and pcmC0D0 have different minor numbers from the class devices created by the ALSA core. I just did kernel prints to capture the minor numbers of audio Control, Timer and pcmC0D0p creatd by ALSA core and found that they are different. My ALSA core version is ALSA 1.0.13 and the snddevices file is also from the ALSA 1.0.13. How do I do a workabout for this? Regards; Aadish

2 2

[alsa-devel] alsa-utils configuration query
by Adish Kuvelker 29 Jun '10

29 Jun '10

Hello, I am cross-compiling alsa-utils for my ARM target. I have successfully configured and installed the alsa-lib (alsa-lib-1.0.16). Now to configure the alsa-utils(alsa-utils-1.0.16) i pass the following command LD=arm-linux-ld ./configure --with-alsa-prefix=/usr/lib --build=i686-linux --disable-nls The libasound.la & libasound.a are present in the /usr/lib but the above command gives an error checking for snd_ctl_open in -lasound... no configure: error: No linkable libasound was found Can anyone give me a pointer of how to tackle this? Regards; Aadish

1 0

Re: [alsa-devel] [PATCH] - remove "front" , "surround" , "rear" , "surround40" , "surround51" from output of arecord -L
by Raymond Yau 29 Jun '10

29 Jun '10

2010/6/29 Jerone Young <jerone.young(a)canonical.com> > Hey Raymod, > A bunch of things wrong with this. > > 1) You need to post the whole diff patch .. not must the lines .. > the > diff patch will contain the file this is patching. > > 2) Needs a Signed-off-by: ME <me(a)somewhere.com> line. > > Jerone > > On Tue, 2010-06-29 at 12:05 +0800, Raymond Yau wrote: > > Since arecord -L will list "front", ..... devices which are not suitable > > for recording (e.g. HDA has softvol plugin in front device , emu10k1 and > > other drivers has hook ctl and lock ) > > > > > > > > > > while (*n != NULL) { > > name = snd_device_name_get_hint(*n, "NAME"); > > descr = snd_device_name_get_hint(*n, "DESC"); > > io = snd_device_name_get_hint(*n, "IOID"); > > + if ( stream == SND_PCM_STREAM_CAPTURE ) > > + if (( strstr(descr,"speaker") > 0 ) || ( > > strstr(descr,"Output") > 0 )) > > + goto __end; > > if (io != NULL && strcmp(io, filter) != 0) > > goto __end; > > printf("%s\n", name); > > if ((descr1 = descr) != NULL) { > I am not sure the fix should be at arecord or namehint since those "Output" and "speaker" may be translated to different language and use "front" device for capturing is not recommeded since there are many drivers have hook ctl and lock (e.g. "Channel Mode" , "Line-In As Surround", Surround Down Mix) in front or surroundxx device and emu10k1 have ctl of pcm_route/pcm_volume ATIIXP.conf: lock true ATIIXP-SPDMA.conf: lock true Audigy2.conf: lock true Audigy.conf: lock true Aureon51.conf: lock true Aureon71.conf: lock true CA0106.conf: lock true CMI8338.conf: lock true CMI8738-MC6.conf: lock true CMI8738-MC8.conf: lock true CMI8788.conf: lock true CS46xx.conf: lock true EMU10K1.conf: lock true EMU10K1X.conf: lock true ENS1370.conf: lock true ENS1371.conf: lock true HDA-Intel.conf: lock true ICE1712.conf: lock true ICE1724.conf: lock true ICH4.conf: lock true ICH.conf: lock true NFORCE.conf: lock true PS3.conf: lock true RME9652.conf: lock true SB-XFi.conf: lock true SI7018.conf: lock true TRID4DWAVENX.conf: lock true VIA686A.conf: lock true VIA8233A.conf: lock true VIA8237.conf: lock true VX222.conf: lock true VXPocket.conf: lock true YMF744.conf: lock true

1 0

[alsa-devel] Card information in /proc or /sys
by Florian Faber 29 Jun '10

29 Jun '10

Hi, I want to expose more information about the sound card's ports (names, connectors etc.) to userspace in a standardized manner. Does ALSA have predefined locations for this in /sys, or would /proc (snd_card_proc_new) still be the way to go? Flo -- Machines can do the work, so people have time to think. public key DA43FEF4 x-hkp://wwwkeys.eu.pgp.net

2 1

[alsa-devel] [PATCH] - remove "front" , "surround" , "rear" , "surround40" , "surround51" from output of arecord -L
by Raymond Yau 29 Jun '10

29 Jun '10

Since arecord -L will list "front", ..... devices which are not suitable for recording (e.g. HDA has softvol plugin in front device , emu10k1 and other drivers has hook ctl and lock ) while (*n != NULL) { name = snd_device_name_get_hint(*n, "NAME"); descr = snd_device_name_get_hint(*n, "DESC"); io = snd_device_name_get_hint(*n, "IOID"); + if ( stream == SND_PCM_STREAM_CAPTURE ) + if (( strstr(descr,"speaker") > 0 ) || ( strstr(descr,"Output") > 0 )) + goto __end; if (io != NULL && strcmp(io, filter) != 0) goto __end; printf("%s\n", name); if ((descr1 = descr) != NULL) {

1 0

Re: [alsa-devel] [PATCH 1/4] dump_audiostreaming_interface(): fix ac-3 decoder
by Daniel Mack 28 Jun '10

28 Jun '10

Hi Greg, On Tue, Jun 01, 2010 at 10:06:58AM -0700, Greg KH wrote: > On Tue, Jun 01, 2010 at 06:52:12PM +0200, Daniel Mack wrote: > > Greg, > > > > I believe you're maintaining usb-utils now, right? At least it was > > you posting an announcement for a new release the other day. > > Yes I am. > > > Let me know if I should do anything else to get these patches merged :) > > Let me wake up and get a chance to get around to these patches? :) > > I'll get to them in a day or so, they look great to me. Just a reminder - could you have a look at them already? For easier access, the six patches I sent can now also be pulled from git://github.com/zonque/usbutils.git FWIW, I got positive feedback from people developing USB audio class 2 devices - my patches seem to display the right things for them :) Thanks, Daniel

2 3

[alsa-devel] newbie, arecord
by javi perez 28 Jun '10

28 Jun '10

I am trying to do a speech recognition system. In the aplay file I would like to conver the unsigned char buffer to a float buffer to be able to make comparisons the way I would like. The floats that I am getting after the conversion are these. How should I do the conversion from unsigned char to float array? thanks in advance. 819 -464604244644772930012428885483847680.000000 820 -542490915911345743841838949652758528.000000 821 -495758857691688295559156595670646784.000000 822 -480181610589352498484045935735209984.000000 823 -441239146465853748477054432634208256.000000 824 -376333296573780517983308202832297984.000000 825 -503547144523165508473277153076576256.000000 826 -485374263974618640302095603012206592.000000 827 -495758501164956981369637424722870272.000000 828 -433450265423157678580401923648651264.000000 829 -415277147190123267616207592952430592.000000 830 -425662216276168008459294146874572800.000000 831 -477585818686816398859299858518376448.000000 832 -394508395510877785555942372127277056.000000 833 -472393086073387742776912597697429504.000000 834 -482778551300245054941687119339323392.000000 835 -584029726534518193323939684778246144.000000 836 -511336897075649235277643191045586944.000000 837 -558067806486950226727430438640418816.000000 838 -513933085118997906224077235982172160.000000 839 -302397753578834695235772573852631040.000000 float* float_array; while (rest > 0 && recycle_capture_file == 0) { size_t c = (rest <= (off64_t)chunk_bytes) ? (size_t)rest : chunk_bytes; size_t f = c * 8 / bits_per_frame; if (pcm_read(audiobuf, f) != f) break; float_array=(float*)malloc(f); memcpy(float_array,audiobuf,f); int z=0; for(z=0; z<f/sizeof(float);z++){ printf("%d %f\n",z,float_array[z]); } /*if (write(fd, audiobuf, c) != c) { perror(name); prg_exit(EXIT_FAILURE); }*/ count -= c; rest -= c; fdcount += c; }

2 1

Re: [alsa-devel] [PATCH RFC 1/3] ASoC: uda1380: use callbacks instead of gpiolib
by Mark Brown 28 Jun '10

28 Jun '10

On Mon, Jun 28, 2010 at 04:20:43PM +0200, pHilipp Zabel wrote: > The reset GPIO is connected directly to the UDA1380 RESET pin. The > power source is external. OK, the power source is definitely a regulator then.

1 0

[alsa-devel] [PATCH RFC 0/3] asoc: uda1380 cleanup
by Vasily Khoruzhick 28 Jun '10

28 Jun '10

This patch series cleanups uda1380 codec driver a bit: - it introduces callbacks instead of direct gpiolib usage, as some machines requires more actions than just plain gpio toggle to enable/disable/reset codec; - it makes driver more powersave-friendly and fixes suspend/resume

2 24

[alsa-devel] [PATCH] ASoC: ak4642: Add Playback Volume control
by Kuninori Morimoto 28 Jun '10

28 Jun '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch works for me. But I'm not sure it is correct settings. Please check it carefully sound/soc/codecs/ak4642.c | 28 +++++++++++++++++++++++++--- 1 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 9a6157f..ce4e6ec 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -28,6 +28,7 @@ #include <linux/slab.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -103,6 +104,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -190,7 +208,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -200,8 +217,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -446,6 +461,10 @@ static int ak4642_init(struct ak4642_priv *ak4642) return ret; } + /* default volume */ + ak4642_write(codec, L_DVC, 0x28); + ak4642_write(codec, R_DVC, 0x28); + return ret; } @@ -524,6 +543,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -- 1.7.0.4

3 5

[alsa-devel] [PATCH] sis7019: increase reset delays
by David Dillow 28 Jun '10

28 Jun '10

A few boards using this controller are reported to need a little extra time during their reset cycle. Reported-by: Michael Goeke <michael.goeke(a)icachip.de> Signed-off-by: Dave Dillow <dave(a)thedillows.org> --- Wow, a horrible maintainer am I. Working on the virtual timing period reminded me of of another timing issue from two years back that I forgot to push upstream. This applies on top of the vperiod fixes, but should not conflict if the ordering is changed. diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 614ff6e..1b8f674 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1049,7 +1049,7 @@ static int sis_chip_free(struct sis7019 *sis) /* Reset the chip, and disable all interrputs. */ outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); - udelay(10); + udelay(25); outl(0, sis->ioport + SIS_GCR); outl(0, sis->ioport + SIS_GIER); @@ -1085,7 +1085,7 @@ static int sis_chip_init(struct sis7019 *sis) /* Reset the audio controller */ outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); - udelay(10); + udelay(25); outl(0, io + SIS_GCR); /* Get the AC-link semaphore, and reset the codecs @@ -1098,7 +1098,7 @@ static int sis_chip_init(struct sis7019 *sis) return -EIO; outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); - udelay(10); + udelay(250); count = 0xffff; while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)

2 1

[alsa-devel] [PATCH] ASoC: ak4642: Add ak4642_fill_cache function
by Kuninori Morimoto 28 Jun '10

28 Jun '10

ak4642 driver is used for ak4642/ak4643. So, reading real register value is better than using constant value. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 50 +++++++++++++++----------------------------- 1 files changed, 17 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 8d56811..9a6157f 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -106,25 +106,19 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; + u8 reg_cache[AK4642_CACHEREGNUM]; }; static struct snd_soc_codec *ak4642_codec; -/* - * ak4642 register cache - */ -static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0001, 0x0000, - 0x0002, 0x0000, 0x0000, 0x0000, - 0x00e1, 0x00e1, 0x0018, 0x0000, - 0x00e1, 0x0018, 0x0011, 0x0008, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, -}; +static void ak4642_fill_cache(struct snd_soc_codec *codec) +{ + int i; + u8 *cache = codec->reg_cache; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + i2c_master_recv(codec->control_data, cache + i, 1); +} /* * read ak4642 register cache @@ -132,7 +126,7 @@ static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; if (reg >= AK4642_CACHEREGNUM) return -1; return cache[reg]; @@ -144,7 +138,7 @@ static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; if (reg >= AK4642_CACHEREGNUM) return; @@ -175,7 +169,7 @@ static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, static int ak4642_sync(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; int i, r = 0; for (i = 0; i < AK4642_CACHEREGNUM; i++) @@ -432,12 +426,9 @@ static int ak4642_init(struct ak4642_priv *ak4642) codec->dai = &ak4642_dai; codec->num_dai = 1; codec->hw_write = (hw_write_t)i2c_master_send; - codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); - codec->reg_cache = kmemdup(ak4642_reg, - sizeof(ak4642_reg), GFP_KERNEL); - - if (!codec->reg_cache) - return -ENOMEM; + codec->reg_cache_size = AK4642_CACHEREGNUM; + codec->reg_cache = ak4642->reg_cache; + ak4642_fill_cache(codec); ak4642_dai.dev = codec->dev; ak4642_codec = codec; @@ -445,23 +436,17 @@ static int ak4642_init(struct ak4642_priv *ak4642) ret = snd_soc_register_codec(codec); if (ret) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - goto reg_cache_err; + return ret; } ret = snd_soc_register_dai(&ak4642_dai); if (ret) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); snd_soc_unregister_codec(codec); - goto reg_cache_err; + return ret; } return ret; - -reg_cache_err: - kfree(codec->reg_cache); - codec->reg_cache = NULL; - - return ret; } #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -495,7 +480,6 @@ static int ak4642_i2c_remove(struct i2c_client *client) snd_soc_unregister_dai(&ak4642_dai); snd_soc_unregister_codec(&ak4642->codec); - kfree(ak4642->codec.reg_cache); kfree(ak4642); ak4642_codec = NULL; -- 1.7.0.4

3 3

[alsa-devel] ALSA SoC device tree bindings
by Jeremy Kerr 27 Jun '10

27 Jun '10

Hi all, From some discussions at the previous Ubuntu Developer Summit, I've put up a page on the devicetree.org wiki with the ultimate goal of describing device tree bindings for ALSA System on Chip devices. At this stage, we're just looking at how to structure device in the device tree. The bindings for specific devices will come as required (ie, as we add device tree support for them). So - if you're interested in the ALSA and/or device tree areas, please take a look at the notes from UDS: http://devicetree.org/SoC_Audio - if you have any comments or suggestions, please let me know, or add to the page (or the discussion for the page). I intend for this document to become a more formal specification, but we still have a way to go before that happens. If you'd like an introduction to device trees, Grant has a good walkthrough- style document at: http://devicetree.org/Device_Tree_Usage Cheers, Jeremy

4 5

[alsa-devel] Modifying the AD193X driver and integrating with the Blackfin platform
by Adam Rosenberg 27 Jun '10

27 Jun '10

I see that Liam just modified the code (bf5xx-ad193x.c) that I was looking at modifying to add support for the CS42448 CODEC. I would like to understand the architecture of the ASoC and Blackfin TDM PCM driver that it uses because I need to enable the secondary data lines to support multiple chips. Where can I find the documentation for this code? If I want to integrate this code into the uClinux-dist-2009R1.1-RC4 distribution, is it simply a matter of copying the files in "[linux-2.6-asoc.git]/sound/" to the linux-2.6/sound directory for uClinux? Or is it better if I git clone linux-2.6-asoc and move my custom board development files there? Is there a driver installation guide somewhere that I can read about this? I am concerned that the official Blackfin distribution (http://blackfin.uclinux.org/gf/download/frsrelease/473/7285/uClinux-dist-20…) will not be compatible with the ASoC distribution (git://opensource.wolfsonmicro.com/linux-2.6-asoc). Can you tell me if there are any differences between these? To sum up: 1. Where can I find the ASoC documentation? 2. Is there an installation guide to use ASoC with an existing Kernel? 3. What are the differences between the wolfsonmicro.com linux-2.6-asoc and blackfin.uclinux.org uClinux-dist-2009R1.1-RC4? Thank you, Adam Adam Rosenberg Software Engineer Alcorn McBride Inc. 3300 South Hiawassee Building 105 Orlando, FL 32835 (407) 296 - 5800 ext. 5490

2 1

[alsa-devel] Detecting if a pcm device is a digital or analog one
by Jean-Yves Avenard 25 Jun '10

25 Jun '10

Hi there... I was wondering if there was any way to accurately distinguish between audio device, if it has an analog or a digital output. I would prefer not parsing the name (like analog71 is obviously an analog device, while "hdmi" is digital) I see that on some devices the IEC958 bit is set, but this doesn't seem reliable (on another PC, the IEC958 bit isn't set). Any suggestions would be greatly appreciated... Thank you Jean-Yves

3 2

[alsa-devel] [PATCH] ASoC: SAMSUNG: Change s3c24xx directory name to samsung
by 최찬우 25 Jun '10

25 Jun '10

This patch changes s3c24xx directory name to samsung. (sound/soc/s3c24xx -> sound/soc/samsung) Currently, the source files related to samsung SoCs(s3c24xx & s3c64xx) are included in sound/soc/s3c24xx. It would be better to change s3c24xx directory name to samsung, because s3c24xx directory will support many board based on samsung SoCs(s5p, etc). Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/s3c24xx/Kconfig | 127 ---- sound/soc/s3c24xx/Makefile | 43 -- sound/soc/s3c24xx/jive_wm8750.c | 196 ------ sound/soc/s3c24xx/lm4857.h | 32 - sound/soc/s3c24xx/ln2440sbc_alc650.c | 85 --- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 506 --------------- sound/soc/s3c24xx/neo1973_wm8753.c | 707 --------------------- sound/soc/s3c24xx/regs-i2s-v2.h | 115 ---- sound/soc/s3c24xx/s3c-ac97.c | 521 ---------------- sound/soc/s3c24xx/s3c-ac97.h | 23 - sound/soc/s3c24xx/s3c-dma.c | 483 --------------- sound/soc/s3c24xx/s3c-dma.h | 31 - sound/soc/s3c24xx/s3c-i2s-v2.c | 776 ------------------------ sound/soc/s3c24xx/s3c-i2s-v2.h | 103 ---- sound/soc/s3c24xx/s3c-pcm.c | 554 ----------------- sound/soc/s3c24xx/s3c-pcm.h | 123 ---- sound/soc/s3c24xx/s3c2412-i2s.c | 190 ------ sound/soc/s3c24xx/s3c2412-i2s.h | 29 - sound/soc/s3c24xx/s3c24xx-i2s.c | 503 --------------- sound/soc/s3c24xx/s3c24xx-i2s.h | 37 -- sound/soc/s3c24xx/s3c24xx_simtec.c | 394 ------------ sound/soc/s3c24xx/s3c24xx_simtec.h | 22 - sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 153 ----- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 137 ----- sound/soc/s3c24xx/s3c24xx_uda134x.c | 373 ------------ sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 209 ------- sound/soc/s3c24xx/s3c64xx-i2s.c | 190 ------ sound/soc/s3c24xx/s3c64xx-i2s.h | 42 -- sound/soc/s3c24xx/smdk2443_wm9710.c | 81 --- sound/soc/s3c24xx/smdk64xx_wm8580.c | 266 -------- sound/soc/s3c24xx/smdk_wm9713.c | 94 --- sound/soc/samsung/Kconfig | 127 ++++ sound/soc/samsung/Makefile | 43 ++ sound/soc/samsung/jive_wm8750.c | 196 ++++++ sound/soc/samsung/lm4857.h | 32 + sound/soc/samsung/ln2440sbc_alc650.c | 85 +++ sound/soc/samsung/neo1973_gta02_wm8753.c | 506 +++++++++++++++ sound/soc/samsung/neo1973_wm8753.c | 707 +++++++++++++++++++++ sound/soc/samsung/regs-i2s-v2.h | 115 ++++ sound/soc/samsung/s3c-ac97.c | 521 ++++++++++++++++ sound/soc/samsung/s3c-ac97.h | 23 + sound/soc/samsung/s3c-dma.c | 483 +++++++++++++++ sound/soc/samsung/s3c-dma.h | 31 + sound/soc/samsung/s3c-i2s-v2.c | 776 ++++++++++++++++++++++++ sound/soc/samsung/s3c-i2s-v2.h | 103 ++++ sound/soc/samsung/s3c-pcm.c | 554 +++++++++++++++++ sound/soc/samsung/s3c-pcm.h | 123 ++++ sound/soc/samsung/s3c2412-i2s.c | 190 ++++++ sound/soc/samsung/s3c2412-i2s.h | 29 + sound/soc/samsung/s3c24xx-i2s.c | 503 +++++++++++++++ sound/soc/samsung/s3c24xx-i2s.h | 37 ++ sound/soc/samsung/s3c24xx_simtec.c | 394 ++++++++++++ sound/soc/samsung/s3c24xx_simtec.h | 22 + sound/soc/samsung/s3c24xx_simtec_hermes.c | 153 +++++ sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 137 +++++ sound/soc/samsung/s3c24xx_uda134x.c | 373 ++++++++++++ sound/soc/samsung/s3c64xx-i2s-v4.c | 209 +++++++ sound/soc/samsung/s3c64xx-i2s.c | 190 ++++++ sound/soc/samsung/s3c64xx-i2s.h | 42 ++ sound/soc/samsung/smdk2443_wm9710.c | 81 +++ sound/soc/samsung/smdk64xx_wm8580.c | 266 ++++++++ sound/soc/samsung/smdk_wm9713.c | 94 +++ 64 files changed, 7147 insertions(+), 7147 deletions(-) delete mode 100644 sound/soc/s3c24xx/Kconfig delete mode 100644 sound/soc/s3c24xx/Makefile delete mode 100644 sound/soc/s3c24xx/jive_wm8750.c delete mode 100644 sound/soc/s3c24xx/lm4857.h delete mode 100644 sound/soc/s3c24xx/ln2440sbc_alc650.c delete mode 100644 sound/soc/s3c24xx/neo1973_gta02_wm8753.c delete mode 100644 sound/soc/s3c24xx/neo1973_wm8753.c delete mode 100644 sound/soc/s3c24xx/regs-i2s-v2.h delete mode 100644 sound/soc/s3c24xx/s3c-ac97.c delete mode 100644 sound/soc/s3c24xx/s3c-ac97.h delete mode 100644 sound/soc/s3c24xx/s3c-dma.c delete mode 100644 sound/soc/s3c24xx/s3c-dma.h delete mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.c delete mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.h delete mode 100644 sound/soc/s3c24xx/s3c-pcm.c delete mode 100644 sound/soc/s3c24xx/s3c-pcm.h delete mode 100644 sound/soc/s3c24xx/s3c2412-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c2412-i2s.h delete mode 100644 sound/soc/s3c24xx/s3c24xx-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c24xx-i2s.h delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.c delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.h delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_hermes.c delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c delete mode 100644 sound/soc/s3c24xx/s3c24xx_uda134x.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s-v4.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.h delete mode 100644 sound/soc/s3c24xx/smdk2443_wm9710.c delete mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c delete mode 100644 sound/soc/s3c24xx/smdk_wm9713.c create mode 100644 sound/soc/samsung/Kconfig create mode 100644 sound/soc/samsung/Makefile create mode 100644 sound/soc/samsung/jive_wm8750.c create mode 100644 sound/soc/samsung/lm4857.h create mode 100644 sound/soc/samsung/ln2440sbc_alc650.c create mode 100644 sound/soc/samsung/neo1973_gta02_wm8753.c create mode 100644 sound/soc/samsung/neo1973_wm8753.c create mode 100644 sound/soc/samsung/regs-i2s-v2.h create mode 100644 sound/soc/samsung/s3c-ac97.c create mode 100644 sound/soc/samsung/s3c-ac97.h create mode 100644 sound/soc/samsung/s3c-dma.c create mode 100644 sound/soc/samsung/s3c-dma.h create mode 100644 sound/soc/samsung/s3c-i2s-v2.c create mode 100644 sound/soc/samsung/s3c-i2s-v2.h create mode 100644 sound/soc/samsung/s3c-pcm.c create mode 100644 sound/soc/samsung/s3c-pcm.h create mode 100644 sound/soc/samsung/s3c2412-i2s.c create mode 100644 sound/soc/samsung/s3c2412-i2s.h create mode 100644 sound/soc/samsung/s3c24xx-i2s.c create mode 100644 sound/soc/samsung/s3c24xx-i2s.h create mode 100644 sound/soc/samsung/s3c24xx_simtec.c create mode 100644 sound/soc/samsung/s3c24xx_simtec.h create mode 100644 sound/soc/samsung/s3c24xx_simtec_hermes.c create mode 100644 sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c create mode 100644 sound/soc/samsung/s3c24xx_uda134x.c create mode 100644 sound/soc/samsung/s3c64xx-i2s-v4.c create mode 100644 sound/soc/samsung/s3c64xx-i2s.c create mode 100644 sound/soc/samsung/s3c64xx-i2s.h create mode 100644 sound/soc/samsung/smdk2443_wm9710.c create mode 100644 sound/soc/samsung/smdk64xx_wm8580.c create mode 100644 sound/soc/samsung/smdk_wm9713.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index b1749bc..5d467ef 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -32,7 +32,7 @@ source "sound/soc/fsl/Kconfig" source "sound/soc/imx/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" -source "sound/soc/s3c24xx/Kconfig" +source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/txx9/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1470141..9572925 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,7 +10,7 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ -obj-$(CONFIG_SND_SOC) += s3c24xx/ +obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig deleted file mode 100644 index 2a7cc22..0000000 --- a/sound/soc/s3c24xx/Kconfig +++ /dev/null @@ -1,127 +0,0 @@ -config SND_S3C24XX_SOC - tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX - select S3C64XX_DMA if ARCH_S3C64XX - help - Say Y or M if you want to add support for codecs attached to - the S3C24XX AC97 or I2S interfaces. You will also need to - select the audio interfaces to support below. - -config SND_S3C24XX_SOC_I2S - tristate - select S3C2410_DMA - -config SND_S3C_I2SV2_SOC - tristate - -config SND_S3C2412_SOC_I2S - tristate - select SND_S3C_I2SV2_SOC - select S3C2410_DMA - -config SND_S3C64XX_SOC_I2S - tristate - select SND_S3C_I2SV2_SOC - select S3C64XX_DMA - -config SND_S3C64XX_SOC_I2S_V4 - tristate - select SND_S3C_I2SV2_SOC - select S3C64XX_DMA - -config SND_S3C_SOC_PCM - tristate - -config SND_S3C_SOC_AC97 - tristate - select SND_SOC_AC97_BUS - -config SND_S3C24XX_SOC_NEO1973_WM8753 - tristate "SoC I2S Audio support for NEO1973 - WM8753" - depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 - select SND_S3C24XX_SOC_I2S - select SND_SOC_WM8753 - help - Say Y if you want to add support for SoC audio on smdk2440 - with the WM8753. - -config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753 - tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" - depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02 - select SND_S3C24XX_SOC_I2S - select SND_SOC_WM8753 - help - This driver provides audio support for the Openmoko Neo FreeRunner - smartphone. - -config SND_S3C24XX_SOC_JIVE_WM8750 - tristate "SoC I2S Audio support for Jive" - depends on SND_S3C24XX_SOC && MACH_JIVE - select SND_SOC_WM8750 - select SND_S3C2412_SOC_I2S - help - Sat Y if you want to add support for SoC audio on the Jive. - -config SND_S3C64XX_SOC_WM8580 - tristate "SoC I2S Audio support for WM8580 on SMDK64XX" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 - select SND_SOC_WM8580 - select SND_S3C64XX_SOC_I2S_V4 - help - Say Y if you want to add support for SoC audio on the SMDK6410. - -config SND_S3C24XX_SOC_SMDK2443_WM9710 - tristate "SoC AC97 Audio support for SMDK2443 - WM9710" - depends on SND_S3C24XX_SOC && MACH_SMDK2443 - select S3C2410_DMA - select AC97_BUS - select SND_SOC_AC97_CODEC - select SND_S3C_SOC_AC97 - help - Say Y if you want to add support for SoC audio on smdk2443 - with the WM9710. - -config SND_S3C24XX_SOC_LN2440SBC_ALC650 - tristate "SoC AC97 Audio support for LN2440SBC - ALC650" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select S3C2410_DMA - select AC97_BUS - select SND_SOC_AC97_CODEC - select SND_S3C_SOC_AC97 - help - Say Y if you want to add support for SoC audio on ln2440sbc - with the ALC650. - -config SND_S3C24XX_SOC_S3C24XX_UDA134X - tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C24XX_SOC_I2S - select SND_SOC_L3 - select SND_SOC_UDA134X - -config SND_S3C24XX_SOC_SIMTEC - tristate - help - Internal node for common S3C24XX/Simtec suppor - -config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 - tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C24XX_SOC_I2S - select SND_SOC_TLV320AIC23 - select SND_S3C24XX_SOC_SIMTEC - -config SND_S3C24XX_SOC_SIMTEC_HERMES - tristate "SoC I2S Audio support for Simtec Hermes board" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C24XX_SOC_I2S - select SND_SOC_TLV320AIC3X - select SND_S3C24XX_SOC_SIMTEC - -config SND_SOC_SMDK_WM9713 - tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 - select SND_SOC_WM9713 - select SND_S3C_SOC_AC97 - help - Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile deleted file mode 100644 index 81d8dc5..0000000 --- a/sound/soc/s3c24xx/Makefile +++ /dev/null @@ -1,43 +0,0 @@ -# S3c24XX Platform Support -snd-soc-s3c24xx-objs := s3c-dma.o -snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o -snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o -snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o -snd-soc-s3c-ac97-objs := s3c-ac97.o -snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o -snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o -snd-soc-s3c-pcm-objs := s3c-pcm.o - -obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o -obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o -obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o -obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o -obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o -obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o -obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o -obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o - -# S3C24XX Machine Support -snd-soc-jive-wm8750-objs := jive_wm8750.o -snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o -snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o -snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o -snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o -snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o -snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o -snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o -snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o -snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o -snd-soc-smdk-wm9713-objs := smdk_wm9713.o - -obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o -obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o -obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o -obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o -obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o -obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o -obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o -obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o -obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o -obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o -obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c deleted file mode 100644 index 8c108b1..0000000 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ /dev/null @@ -1,196 +0,0 @@ -/* sound/soc/s3c24xx/jive_wm8750.c - * - * Copyright 2007,2008 Simtec Electronics - * - * Based on sound/soc/pxa/spitz.c - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/clk.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <asm/mach-types.h> - -#include "s3c-dma.h" -#include "s3c2412-i2s.h" - -#include "../codecs/wm8750.h" - -static const struct snd_soc_dapm_route audio_map[] = { - { "Headphone Jack", NULL, "LOUT1" }, - { "Headphone Jack", NULL, "ROUT1" }, - { "Internal Speaker", NULL, "LOUT2" }, - { "Internal Speaker", NULL, "ROUT2" }, - { "LINPUT1", NULL, "Line Input" }, - { "RINPUT1", NULL, "Line Input" }, -}; - -static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Internal Speaker", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), -}; - -static int jive_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->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct s3c_i2sv2_rate_calc div; - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - clk = 12288000; - break; - case 11025: - case 22050: - case 44100: - clk = 11289600; - break; - } - - s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), - s3c_i2sv2_get_clock(cpu_dai)); - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, - div.clk_div - 1); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops jive_ops = { - .hw_params = jive_hw_params, -}; - -static int jive_wm8750_init(struct snd_soc_codec *codec) -{ - int err; - - /* These endpoints are not being used. */ - snd_soc_dapm_nc_pin(codec, "LINPUT2"); - snd_soc_dapm_nc_pin(codec, "RINPUT2"); - snd_soc_dapm_nc_pin(codec, "LINPUT3"); - snd_soc_dapm_nc_pin(codec, "RINPUT3"); - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "MONO"); - - /* Add jive specific widgets */ - err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, - ARRAY_SIZE(wm8750_dapm_widgets)); - if (err) { - printk(KERN_ERR "%s: failed to add widgets (%d)\n", - __func__, err); - return err; - } - - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link jive_dai = { - .name = "wm8750", - .stream_name = "WM8750", - .cpu_dai = &s3c2412_i2s_dai, - .codec_dai = &wm8750_dai, - .init = jive_wm8750_init, - .ops = &jive_ops, -}; - -/* jive audio machine driver */ -static struct snd_soc_card snd_soc_machine_jive = { - .name = "Jive", - .platform = &s3c24xx_soc_platform, - .dai_link = &jive_dai, - .num_links = 1, -}; - -/* jive audio subsystem */ -static struct snd_soc_device jive_snd_devdata = { - .card = &snd_soc_machine_jive, - .codec_dev = &soc_codec_dev_wm8750, -}; - -static struct platform_device *jive_snd_device; - -static int __init jive_init(void) -{ - int ret; - - if (!machine_is_jive()) - return 0; - - printk("JIVE WM8750 Audio support\n"); - - jive_snd_device = platform_device_alloc("soc-audio", -1); - if (!jive_snd_device) - return -ENOMEM; - - platform_set_drvdata(jive_snd_device, &jive_snd_devdata); - jive_snd_devdata.dev = &jive_snd_device->dev; - ret = platform_device_add(jive_snd_device); - - if (ret) - platform_device_put(jive_snd_device); - - return ret; -} - -static void __exit jive_exit(void) -{ - platform_device_unregister(jive_snd_device); -} - -module_init(jive_init); -module_exit(jive_exit); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/lm4857.h b/sound/soc/s3c24xx/lm4857.h deleted file mode 100644 index 0cf5b70..0000000 --- a/sound/soc/s3c24xx/lm4857.h +++ /dev/null @@ -1,32 +0,0 @@ -/* - * lm4857.h -- ALSA Soc Audio Layer - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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. - * - * Revision history - * 18th Jun 2007 Initial version. - */ - -#ifndef LM4857_H_ -#define LM4857_H_ - -/* The register offsets in the cache array */ -#define LM4857_MVOL 0 -#define LM4857_LVOL 1 -#define LM4857_RVOL 2 -#define LM4857_CTRL 3 - -/* the shifts required to set these bits */ -#define LM4857_3D 5 -#define LM4857_WAKEUP 5 -#define LM4857_EPGAIN 4 - -#endif /*LM4857_H_*/ - diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c deleted file mode 100644 index ffa954f..0000000 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ /dev/null @@ -1,85 +0,0 @@ -/* - * SoC audio for ln2440sbc - * - * Copyright 2007 KonekTel, a.s. - * Author: Ivan Kuten - * ivan.kuten(a)promwad.com - * - * Heavily based on smdk2443_wm9710.c - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include "../codecs/ac97.h" -#include "s3c-dma.h" -#include "s3c-ac97.h" - -static struct snd_soc_card ln2440sbc; - -static struct snd_soc_dai_link ln2440sbc_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], - .codec_dai = &ac97_dai, -}, -}; - -static struct snd_soc_card ln2440sbc = { - .name = "LN2440SBC", - .platform = &s3c24xx_soc_platform, - .dai_link = ln2440sbc_dai, - .num_links = ARRAY_SIZE(ln2440sbc_dai), -}; - -static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { - .card = &ln2440sbc, - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct platform_device *ln2440sbc_snd_ac97_device; - -static int __init ln2440sbc_init(void) -{ - int ret; - - ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!ln2440sbc_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(ln2440sbc_snd_ac97_device, - &ln2440sbc_snd_ac97_devdata); - ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; - ret = platform_device_add(ln2440sbc_snd_ac97_device); - - if (ret) - platform_device_put(ln2440sbc_snd_ac97_device); - - return ret; -} - -static void __exit ln2440sbc_exit(void) -{ - platform_device_unregister(ln2440sbc_snd_ac97_device); -} - -module_init(ln2440sbc_init); -module_exit(ln2440sbc_exit); - -/* Module information */ -MODULE_AUTHOR("Ivan Kuten"); -MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c deleted file mode 100644 index 209c259..0000000 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ /dev/null @@ -1,506 +0,0 @@ -/* - * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02) - * - * Copyright 2007 Openmoko Inc - * Author: Graeme Gregory <graeme(a)openmoko.org> - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory <linux(a)wolfsonmicro.com> - * Copyright 2009 Wolfson Microelectronics - * - * 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/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <asm/mach-types.h> - -#include <plat/regs-iis.h> - -#include <mach/regs-clock.h> -#include <asm/io.h> -#include <mach/gta02.h> -#include "../codecs/wm8753.h" -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" - -static struct snd_soc_card neo1973_gta02; - -static int neo1973_gta02_hifi_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->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int pll_out = 0, bclk = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - switch (params_rate(params)) { - case 8000: - case 16000: - pll_out = 12288000; - break; - case 48000: - bclk = WM8753_BCLK_DIV_4; - pll_out = 12288000; - break; - case 96000: - bclk = WM8753_BCLK_DIV_2; - pll_out = 12288000; - break; - case 11025: - bclk = WM8753_BCLK_DIV_16; - pll_out = 11289600; - break; - case 22050: - bclk = WM8753_BCLK_DIV_8; - pll_out = 11289600; - break; - case 44100: - bclk = WM8753_BCLK_DIV_4; - pll_out = 11289600; - break; - case 88200: - bclk = WM8753_BCLK_DIV_2; - pll_out = 11289600; - break; - } - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set codec BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, - WM8753_BCLKDIV, bclk); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4, 4)); - if (ret < 0) - return ret; - - /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, - iis_clkrate / 4, pll_out); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); -} - -/* - * Neo1973 WM8753 HiFi DAI opserations. - */ -static struct snd_soc_ops neo1973_gta02_hifi_ops = { - .hw_params = neo1973_gta02_hifi_hw_params, - .hw_free = neo1973_gta02_hifi_hw_free, -}; - -static int neo1973_gta02_voice_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->dai->codec_dai; - unsigned int pcmdiv = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - if (params_rate(params) != 8000) - return -EINVAL; - if (params_channels(params) != 1) - return -EINVAL; - - pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, - 12288000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec PCM division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, - pcmdiv); - if (ret < 0) - return ret; - - /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, - iis_clkrate / 4, 12288000); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); -} - -static struct snd_soc_ops neo1973_gta02_voice_ops = { - .hw_params = neo1973_gta02_voice_hw_params, - .hw_free = neo1973_gta02_voice_hw_free, -}; - -#define LM4853_AMP 1 -#define LM4853_SPK 2 - -static u8 lm4853_state; - -/* This has no effect, it exists only to maintain compatibility with - * existing ALSA state files. - */ -static int lm4853_set_state(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int val = ucontrol->value.integer.value[0]; - - if (val) - lm4853_state |= LM4853_AMP; - else - lm4853_state &= ~LM4853_AMP; - - return 0; -} - -static int lm4853_get_state(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP; - - return 0; -} - -static int lm4853_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int val = ucontrol->value.integer.value[0]; - - if (val) { - lm4853_state |= LM4853_SPK; - gpio_set_value(GTA02_GPIO_HP_IN, 0); - } else { - lm4853_state &= ~LM4853_SPK; - gpio_set_value(GTA02_GPIO_HP_IN, 1); - } - - return 0; -} - -static int lm4853_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1; - - return 0; -} - -static int lm4853_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, - int event) -{ - gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value)); - - return 0; -} - -static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Handset Mic", NULL), - SND_SOC_DAPM_SPK("Handset Spk", NULL), -}; - - -/* example machine audio_mapnections */ -static const struct snd_soc_dapm_route audio_map[] = { - - /* Connections to the lm4853 amp */ - {"Stereo Out", NULL, "LOUT1"}, - {"Stereo Out", NULL, "ROUT1"}, - - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Handset Mic"}, - - /* Call Speaker */ - {"Handset Spk", NULL, "LOUT2"}, - {"Handset Spk", NULL, "ROUT2"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { - SOC_DAPM_PIN_SWITCH("Stereo Out"), - SOC_DAPM_PIN_SWITCH("GSM Line Out"), - SOC_DAPM_PIN_SWITCH("GSM Line In"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Spk"), - - /* This has no effect, it exists only to maintain compatibility with - * existing ALSA state files. - */ - SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0, - lm4853_get_state, - lm4853_set_state), - SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0, - lm4853_get_spk, - lm4853_set_spk), -}; - -/* - * This is an example machine initialisation for a wm8753 connected to a - * neo1973 GTA02. - */ -static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) -{ - int err; - - /* set up NC codec pins */ - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "OUT4"); - snd_soc_dapm_nc_pin(codec, "LINE1"); - snd_soc_dapm_nc_pin(codec, "LINE2"); - - /* Add neo1973 gta02 specific widgets */ - snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - - /* add neo1973 gta02 specific controls */ - err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls, - ARRAY_SIZE(wm8753_neo1973_gta02_controls)); - - if (err < 0) - return err; - - /* set up neo1973 gta02 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - - /* set endpoints to default off mode */ - snd_soc_dapm_disable_pin(codec, "Stereo Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Handset Mic"); - snd_soc_dapm_disable_pin(codec, "Handset Spk"); - - /* allow audio paths from the GSM modem to run during suspend */ - snd_soc_dapm_ignore_suspend(codec, "Stereo Out"); - snd_soc_dapm_ignore_suspend(codec, "GSM Line Out"); - snd_soc_dapm_ignore_suspend(codec, "GSM Line In"); - snd_soc_dapm_ignore_suspend(codec, "Headset Mic"); - snd_soc_dapm_ignore_suspend(codec, "Handset Mic"); - snd_soc_dapm_ignore_suspend(codec, "Handset Spk"); - - snd_soc_dapm_sync(codec); - - return 0; -} - -/* - * BT Codec DAI - */ -static struct snd_soc_dai bt_dai = { - .name = "Bluetooth", - .id = 0, - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static struct snd_soc_dai_link neo1973_gta02_dai[] = { -{ /* Hifi Playback - for similatious use with voice below */ - .name = "WM8753", - .stream_name = "WM8753 HiFi", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], - .init = neo1973_gta02_wm8753_init, - .ops = &neo1973_gta02_hifi_ops, -}, -{ /* Voice via BT */ - .name = "Bluetooth", - .stream_name = "Voice", - .cpu_dai = &bt_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], - .ops = &neo1973_gta02_voice_ops, -}, -}; - -static struct snd_soc_card neo1973_gta02 = { - .name = "neo1973-gta02", - .platform = &s3c24xx_soc_platform, - .dai_link = neo1973_gta02_dai, - .num_links = ARRAY_SIZE(neo1973_gta02_dai), -}; - -static struct snd_soc_device neo1973_gta02_snd_devdata = { - .card = &neo1973_gta02, - .codec_dev = &soc_codec_dev_wm8753, -}; - -static struct platform_device *neo1973_gta02_snd_device; - -static int __init neo1973_gta02_init(void) -{ - int ret; - - if (!machine_is_neo1973_gta02()) { - printk(KERN_INFO - "Only GTA02 is supported by this ASoC driver\n"); - return -ENODEV; - } - - /* register bluetooth DAI here */ - ret = snd_soc_register_dai(&bt_dai); - if (ret) - return ret; - - neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_gta02_snd_device) - return -ENOMEM; - - platform_set_drvdata(neo1973_gta02_snd_device, - &neo1973_gta02_snd_devdata); - neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev; - ret = platform_device_add(neo1973_gta02_snd_device); - - if (ret) { - platform_device_put(neo1973_gta02_snd_device); - return ret; - } - - /* Initialise GPIOs used by amp */ - ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN"); - if (ret) { - pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN); - goto err_unregister_device; - } - - ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1); - if (ret) { - pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN); - goto err_free_gpio_hp_in; - } - - ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT"); - if (ret) { - pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT); - goto err_free_gpio_hp_in; - } - - ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1); - if (ret) { - pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT); - goto err_free_gpio_amp_shut; - } - - return 0; - -err_free_gpio_amp_shut: - gpio_free(GTA02_GPIO_AMP_SHUT); -err_free_gpio_hp_in: - gpio_free(GTA02_GPIO_HP_IN); -err_unregister_device: - platform_device_unregister(neo1973_gta02_snd_device); - return ret; -} -module_init(neo1973_gta02_init); - -static void __exit neo1973_gta02_exit(void) -{ - snd_soc_unregister_dai(&bt_dai); - platform_device_unregister(neo1973_gta02_snd_device); - gpio_free(GTA02_GPIO_HP_IN); - gpio_free(GTA02_GPIO_AMP_SHUT); -} -module_exit(neo1973_gta02_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c deleted file mode 100644 index 0cb4f86..0000000 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ /dev/null @@ -1,707 +0,0 @@ -/* - * neo1973_wm8753.c -- SoC audio for Neo1973 - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/tlv.h> - -#include <asm/mach-types.h> -#include <asm/hardware/scoop.h> -#include <mach/regs-clock.h> -#include <mach/regs-gpio.h> -#include <mach/hardware.h> -#include <linux/io.h> -#include <mach/spi-gpio.h> - -#include <plat/regs-iis.h> - -#include "../codecs/wm8753.h" -#include "lm4857.h" -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" - -/* define the scenarios */ -#define NEO_AUDIO_OFF 0 -#define NEO_GSM_CALL_AUDIO_HANDSET 1 -#define NEO_GSM_CALL_AUDIO_HEADSET 2 -#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 -#define NEO_STEREO_TO_SPEAKERS 4 -#define NEO_STEREO_TO_HEADPHONES 5 -#define NEO_CAPTURE_HANDSET 6 -#define NEO_CAPTURE_HEADSET 7 -#define NEO_CAPTURE_BLUETOOTH 8 - -static struct snd_soc_card neo1973; -static struct i2c_client *i2c; - -static int neo1973_hifi_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->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int pll_out = 0, bclk = 0; - int ret = 0; - unsigned long iis_clkrate; - - pr_debug("Entered %s\n", __func__); - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - switch (params_rate(params)) { - case 8000: - case 16000: - pll_out = 12288000; - break; - case 48000: - bclk = WM8753_BCLK_DIV_4; - pll_out = 12288000; - break; - case 96000: - bclk = WM8753_BCLK_DIV_2; - pll_out = 12288000; - break; - case 11025: - bclk = WM8753_BCLK_DIV_16; - pll_out = 11289600; - break; - case 22050: - bclk = WM8753_BCLK_DIV_8; - pll_out = 11289600; - break; - case 44100: - bclk = WM8753_BCLK_DIV_4; - pll_out = 11289600; - break; - case 88200: - bclk = WM8753_BCLK_DIV_2; - pll_out = 11289600; - break; - } - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set codec BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4, 4)); - if (ret < 0) - return ret; - - /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, - iis_clkrate / 4, pll_out); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - pr_debug("Entered %s\n", __func__); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); -} - -/* - * Neo1973 WM8753 HiFi DAI opserations. - */ -static struct snd_soc_ops neo1973_hifi_ops = { - .hw_params = neo1973_hifi_hw_params, - .hw_free = neo1973_hifi_hw_free, -}; - -static int neo1973_voice_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->dai->codec_dai; - unsigned int pcmdiv = 0; - int ret = 0; - unsigned long iis_clkrate; - - pr_debug("Entered %s\n", __func__); - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - if (params_rate(params) != 8000) - return -EINVAL; - if (params_channels(params) != 1) - return -EINVAL; - - pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec PCM division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); - if (ret < 0) - return ret; - - /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, - iis_clkrate / 4, 12288000); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - pr_debug("Entered %s\n", __func__); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); -} - -static struct snd_soc_ops neo1973_voice_ops = { - .hw_params = neo1973_voice_hw_params, - .hw_free = neo1973_voice_hw_free, -}; - -static int neo1973_scenario; - -static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = neo1973_scenario; - return 0; -} - -static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) -{ - pr_debug("Entered %s\n", __func__); - - switch (neo1973_scenario) { - case NEO_AUDIO_OFF: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_CAPTURE_HANDSET: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Call Mic"); - break; - case NEO_CAPTURE_HEADSET: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - default: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - } - - snd_soc_dapm_sync(codec); - - return 0; -} - -static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - pr_debug("Entered %s\n", __func__); - - if (neo1973_scenario == ucontrol->value.integer.value[0]) - return 0; - - neo1973_scenario = ucontrol->value.integer.value[0]; - set_scenario_endpoints(codec, neo1973_scenario); - return 1; -} - -static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; - -static void lm4857_write_regs(void) -{ - pr_debug("Entered %s\n", __func__); - - if (i2c_master_send(i2c, lm4857_regs, 4) != 4) - printk(KERN_ERR "lm4857: i2c write failed\n"); -} - -static int lm4857_get_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - pr_debug("Entered %s\n", __func__); - - ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; - return 0; -} - -static int lm4857_set_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - if (((lm4857_regs[reg] >> shift) & mask) == - ucontrol->value.integer.value[0]) - return 0; - - lm4857_regs[reg] &= ~(mask << shift); - lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; - lm4857_write_regs(); - return 1; -} - -static int lm4857_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; - - pr_debug("Entered %s\n", __func__); - - if (value) - value -= 5; - - ucontrol->value.integer.value[0] = value; - return 0; -} - -static int lm4857_set_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = ucontrol->value.integer.value[0]; - - pr_debug("Entered %s\n", __func__); - - if (value) - value += 5; - - if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) - return 0; - - lm4857_regs[LM4857_CTRL] &= 0xF0; - lm4857_regs[LM4857_CTRL] |= value; - lm4857_write_regs(); - return 1; -} - -static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Audio Out", NULL), - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Call Mic", NULL), -}; - - -static const struct snd_soc_dapm_route dapm_routes[] = { - - /* Connections to the lm4857 amp */ - {"Audio Out", NULL, "LOUT1"}, - {"Audio Out", NULL, "ROUT1"}, - - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Call Mic"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const char *lm4857_mode[] = { - "Off", - "Call Speaker", - "Stereo Speakers", - "Stereo Speakers + Headphones", - "Headphones" -}; - -static const struct soc_enum lm4857_mode_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), -}; - -static const char *neo_scenarios[] = { - "Off", - "GSM Handset", - "GSM Headset", - "GSM Bluetooth", - "Speakers", - "Headphones", - "Capture Handset", - "Capture Headset", - "Capture Bluetooth" -}; - -static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), -}; - -static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); -static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); - -static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, mono_tlv), - SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], - lm4857_get_mode, lm4857_set_mode), - SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], - neo1973_get_scenario, neo1973_set_scenario), - SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, - lm4857_get_reg, lm4857_set_reg), -}; - -/* - * This is an example machine initialisation for a wm8753 connected to a - * neo1973 II. It is missing logic to detect hp/mic insertions and logic - * to re-route the audio in such an event. - */ -static int neo1973_wm8753_init(struct snd_soc_codec *codec) -{ - int err; - - pr_debug("Entered %s\n", __func__); - - /* set up NC codec pins */ - snd_soc_dapm_nc_pin(codec, "LOUT2"); - snd_soc_dapm_nc_pin(codec, "ROUT2"); - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "OUT4"); - snd_soc_dapm_nc_pin(codec, "LINE1"); - snd_soc_dapm_nc_pin(codec, "LINE2"); - - /* Add neo1973 specific widgets */ - snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - - /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); - - /* add neo1973 specific controls */ - err = snd_soc_add_controls(codec, wm8753_neo1973_controls, - ARRAY_SIZE(8753_neo1973_controls)); - if (err < 0) - return err; - - /* set up neo1973 specific audio routes */ - err = snd_soc_dapm_add_routes(codec, dapm_routes, - ARRAY_SIZE(dapm_routes)); - - snd_soc_dapm_sync(codec); - return 0; -} - -/* - * BT Codec DAI - */ -static struct snd_soc_dai bt_dai = { - .name = "Bluetooth", - .id = 0, - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static struct snd_soc_dai_link neo1973_dai[] = { -{ /* Hifi Playback - for similatious use with voice below */ - .name = "WM8753", - .stream_name = "WM8753 HiFi", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], - .init = neo1973_wm8753_init, - .ops = &neo1973_hifi_ops, -}, -{ /* Voice via BT */ - .name = "Bluetooth", - .stream_name = "Voice", - .cpu_dai = &bt_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], - .ops = &neo1973_voice_ops, -}, -}; - -static struct snd_soc_card neo1973 = { - .name = "neo1973", - .platform = &s3c24xx_soc_platform, - .dai_link = neo1973_dai, - .num_links = ARRAY_SIZE(neo1973_dai), -}; - -static struct snd_soc_device neo1973_snd_devdata = { - .card = &neo1973, - .codec_dev = &soc_codec_dev_wm8753, -}; - -static int lm4857_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - pr_debug("Entered %s\n", __func__); - - i2c = client; - - lm4857_write_regs(); - return 0; -} - -static int lm4857_i2c_remove(struct i2c_client *client) -{ - pr_debug("Entered %s\n", __func__); - - i2c = NULL; - - return 0; -} - -static u8 lm4857_state; - -static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_suspend\n"); - lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); - } - return 0; -} - -static int lm4857_resume(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); - lm4857_write_regs(); - } - return 0; -} - -static void lm4857_shutdown(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_shutdown\n"); - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); -} - -static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 }, - { } -}; - -static struct i2c_driver lm4857_i2c_driver = { - .driver = { - .name = "LM4857 I2C Amp", - .owner = THIS_MODULE, - }, - .suspend = lm4857_suspend, - .resume = lm4857_resume, - .shutdown = lm4857_shutdown, - .probe = lm4857_i2c_probe, - .remove = lm4857_i2c_remove, - .id_table = lm4857_i2c_id, -}; - -static struct platform_device *neo1973_snd_device; - -static int __init neo1973_init(void) -{ - int ret; - - pr_debug("Entered %s\n", __func__); - - if (!machine_is_neo1973_gta01()) { - printk(KERN_INFO - "Only GTA01 hardware supported by ASoC driver\n"); - return -ENODEV; - } - - neo1973_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_snd_device) - return -ENOMEM; - - platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata); - neo1973_snd_devdata.dev = &neo1973_snd_device->dev; - ret = platform_device_add(neo1973_snd_device); - - if (ret) { - platform_device_put(neo1973_snd_device); - return ret; - } - - ret = i2c_add_driver(&lm4857_i2c_driver); - - if (ret != 0) - platform_device_unregister(neo1973_snd_device); - - return ret; -} - -static void __exit neo1973_exit(void) -{ - pr_debug("Entered %s\n", __func__); - - i2c_del_driver(&lm4857_i2c_driver); - platform_device_unregister(neo1973_snd_device); -} - -module_init(neo1973_init); -module_exit(neo1973_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h deleted file mode 100644 index 5e5e568..0000000 --- a/sound/soc/s3c24xx/regs-i2s-v2.h +++ /dev/null @@ -1,115 +0,0 @@ -/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h - * - * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> - * http://armlinux.simtec.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. - * - * S3C2412 IIS register definition -*/ - -#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H -#define __ASM_ARCH_REGS_S3C2412_IIS_H - -#define S3C2412_IISCON (0x00) -#define S3C2412_IISMOD (0x04) -#define S3C2412_IISFIC (0x08) -#define S3C2412_IISPSR (0x0C) -#define S3C2412_IISTXD (0x10) -#define S3C2412_IISRXD (0x14) - -#define S5PC1XX_IISFICS 0x18 -#define S5PC1XX_IISTXDS 0x1C - -#define S5PC1XX_IISCON_SW_RST (1 << 31) -#define S5PC1XX_IISCON_FRXOFSTATUS (1 << 26) -#define S5PC1XX_IISCON_FRXORINTEN (1 << 25) -#define S5PC1XX_IISCON_FTXSURSTAT (1 << 24) -#define S5PC1XX_IISCON_FTXSURINTEN (1 << 23) -#define S5PC1XX_IISCON_TXSDMAPAUSE (1 << 20) -#define S5PC1XX_IISCON_TXSDMACTIVE (1 << 18) - -#define S3C64XX_IISCON_FTXURSTATUS (1 << 17) -#define S3C64XX_IISCON_FTXURINTEN (1 << 16) -#define S3C64XX_IISCON_TXFIFO2_EMPTY (1 << 15) -#define S3C64XX_IISCON_TXFIFO1_EMPTY (1 << 14) -#define S3C64XX_IISCON_TXFIFO2_FULL (1 << 13) -#define S3C64XX_IISCON_TXFIFO1_FULL (1 << 12) - -#define S3C2412_IISCON_LRINDEX (1 << 11) -#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) -#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) -#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) -#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) -#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) -#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) -#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) -#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) -#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) -#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) -#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) - -#define S5PC1XX_IISMOD_OPCLK_CDCLK_OUT (0 << 30) -#define S5PC1XX_IISMOD_OPCLK_CDCLK_IN (1 << 30) -#define S5PC1XX_IISMOD_OPCLK_BCLK_OUT (2 << 30) -#define S5PC1XX_IISMOD_OPCLK_PCLK (3 << 30) -#define S5PC1XX_IISMOD_OPCLK_MASK (3 << 30) -#define S5PC1XX_IISMOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ -#define S5PC1XX_IISMOD_BLCS_MASK 0x3 -#define S5PC1XX_IISMOD_BLCS_SHIFT 26 -#define S5PC1XX_IISMOD_BLCP_MASK 0x3 -#define S5PC1XX_IISMOD_BLCP_SHIFT 24 - -#define S3C64XX_IISMOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ -#define S3C64XX_IISMOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ -#define S3C64XX_IISMOD_C1DD_HHALF (1 << 19) -#define S3C64XX_IISMOD_C1DD_LHALF (1 << 18) -#define S3C64XX_IISMOD_DC2_EN (1 << 17) -#define S3C64XX_IISMOD_DC1_EN (1 << 16) -#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) -#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) -#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) -#define S3C64XX_IISMOD_BLC_MASK (3 << 13) - -#define S3C2412_IISMOD_IMS_SYSMUX (1 << 10) -#define S3C2412_IISMOD_SLAVE (1 << 11) -#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) -#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) -#define S3C2412_IISMOD_MODE_TXRX (2 << 8) -#define S3C2412_IISMOD_MODE_MASK (3 << 8) -#define S3C2412_IISMOD_LR_LLOW (0 << 7) -#define S3C2412_IISMOD_LR_RLOW (1 << 7) -#define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (1 << 5) -#define S3C2412_IISMOD_SDF_LSB (2 << 5) -#define S3C2412_IISMOD_SDF_MASK (3 << 5) -#define S3C2412_IISMOD_RCLK_256FS (0 << 3) -#define S3C2412_IISMOD_RCLK_512FS (1 << 3) -#define S3C2412_IISMOD_RCLK_384FS (2 << 3) -#define S3C2412_IISMOD_RCLK_768FS (3 << 3) -#define S3C2412_IISMOD_RCLK_MASK (3 << 3) -#define S3C2412_IISMOD_BCLK_32FS (0 << 1) -#define S3C2412_IISMOD_BCLK_48FS (1 << 1) -#define S3C2412_IISMOD_BCLK_16FS (2 << 1) -#define S3C2412_IISMOD_BCLK_24FS (3 << 1) -#define S3C2412_IISMOD_BCLK_MASK (3 << 1) -#define S3C2412_IISMOD_8BIT (1 << 0) - -#define S3C64XX_IISMOD_CDCLKCON (1 << 12) - -#define S3C2412_IISPSR_PSREN (1 << 15) - -#define S3C64XX_IISFIC_TX2COUNT(x) (((x) >> 24) & 0xf) -#define S3C64XX_IISFIC_TX1COUNT(x) (((x) >> 16) & 0xf) - -#define S3C2412_IISFIC_TXFLUSH (1 << 15) -#define S3C2412_IISFIC_RXFLUSH (1 << 7) -#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) -#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) - -#define S5PC1XX_IISFICS_TXFLUSH (1 << 15) -#define S5PC1XX_IISFICS_TXCOUNT(x) (((x) >> 8) & 0x7f) - -#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c deleted file mode 100644 index ecf4fd0..0000000 --- a/sound/soc/s3c24xx/s3c-ac97.c +++ /dev/null @@ -1,521 +0,0 @@ -/* sound/soc/s3c24xx/s3c-ac97.c - * - * ALSA SoC Audio Layer - S3C AC97 Controller driver - * Evolved from s3c2443-ac97.c - * - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * Credits: Graeme Gregory, Sean Choi - * - * 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/init.h> -#include <linux/module.h> -#include <linux/io.h> -#include <linux/delay.h> -#include <linux/clk.h> - -#include <sound/soc.h> - -#include <plat/regs-ac97.h> -#include <mach/dma.h> -#include <plat/audio.h> - -#include "s3c-dma.h" -#include "s3c-ac97.h" - -#define AC_CMD_ADDR(x) (x << 16) -#define AC_CMD_DATA(x) (x & 0xffff) - -struct s3c_ac97_info { - unsigned state; - struct clk *ac97_clk; - void __iomem *regs; - struct mutex lock; - struct completion done; -}; -static struct s3c_ac97_info s3c_ac97; - -static struct s3c2410_dma_client s3c_dma_client_out = { - .name = "AC97 PCMOut" -}; - -static struct s3c2410_dma_client s3c_dma_client_in = { - .name = "AC97 PCMIn" -}; - -static struct s3c2410_dma_client s3c_dma_client_micin = { - .name = "AC97 MicIn" -}; - -static struct s3c_dma_params s3c_ac97_pcm_out = { - .client = &s3c_dma_client_out, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c_ac97_pcm_in = { - .client = &s3c_dma_client_in, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c_ac97_mic_in = { - .client = &s3c_dma_client_micin, - .dma_size = 4, -}; - -static void s3c_ac97_activate(struct snd_ac97 *ac97) -{ - u32 ac_glbctrl, stat; - - stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; - if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) - return; /* Return if already active */ - - INIT_COMPLETION(s3c_ac97.done); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - printk(KERN_ERR "AC97: Unable to activate!"); -} - -static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - u32 ac_glbctrl, ac_codec_cmd; - u32 stat, addr, data; - - mutex_lock(&s3c_ac97.lock); - - s3c_ac97_activate(ac97); - - INIT_COMPLETION(s3c_ac97.done); - - ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); - writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - printk(KERN_ERR "AC97: Unable to read!"); - - stat = readl(s3c_ac97.regs + S3C_AC97_STAT); - addr = (stat >> 16) & 0x7f; - data = (stat & 0xffff); - - if (addr != reg) - printk(KERN_ERR "s3c-ac97: req addr = %02x, rep addr = %02x\n", reg, addr); - - mutex_unlock(&s3c_ac97.lock); - - return (unsigned short)data; -} - -static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - u32 ac_glbctrl, ac_codec_cmd; - - mutex_lock(&s3c_ac97.lock); - - s3c_ac97_activate(ac97); - - INIT_COMPLETION(s3c_ac97.done); - - ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); - writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - printk(KERN_ERR "AC97: Unable to write!"); - - ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; - writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); - - mutex_unlock(&s3c_ac97.lock); -} - -static void s3c_ac97_cold_reset(struct snd_ac97 *ac97) -{ - writel(S3C_AC97_GLBCTRL_COLDRESET, - s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); -} - -static void s3c_ac97_warm_reset(struct snd_ac97 *ac97) -{ - u32 stat; - - stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; - if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) - return; /* Return if already active */ - - writel(S3C_AC97_GLBCTRL_WARMRESET, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - s3c_ac97_activate(ac97); -} - -static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) -{ - u32 ac_glbctrl, ac_glbstat; - - ac_glbstat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT); - - if (ac_glbstat & S3C_AC97_GLBSTAT_CODECREADY) { - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - complete(&s3c_ac97.done); - } - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= (1<<30); /* Clear interrupt */ - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - return IRQ_HANDLED; -} - -struct snd_ac97_bus_ops soc_ac97_ops = { - .read = s3c_ac97_read, - .write = s3c_ac97_write, - .warm_reset = s3c_ac97_warm_reset, - .reset = s3c_ac97_cold_reset, -}; -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - -static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; - else - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; - else - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - break; - } - - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - -static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ac_glbctrl |= S3C_AC97_GLBCTRL_MICINTM_DMA; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - break; - } - - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -static struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, - .trigger = s3c_ac97_trigger, -}; - -static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, - .trigger = s3c_ac97_mic_trigger, -}; - -struct snd_soc_dai s3c_ac97_dai[] = { - [S3C_AC97_DAI_PCM] = { - .name = "s3c-ac97", - .id = S3C_AC97_DAI_PCM, - .ac97_control = 1, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c_ac97_dai_ops, - }, - [S3C_AC97_DAI_MIC] = { - .name = "s3c-ac97-mic", - .id = S3C_AC97_DAI_MIC, - .ac97_control = 1, - .capture = { - .stream_name = "AC97 Mic Capture", - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c_ac97_mic_dai_ops, - }, -}; -EXPORT_SYMBOL_GPL(s3c_ac97_dai); - -static __devinit int s3c_ac97_probe(struct platform_device *pdev) -{ - struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; - struct s3c_audio_pdata *ac97_pdata; - int ret; - - ac97_pdata = pdev->dev.platform_data; - if (!ac97_pdata || !ac97_pdata->cfg_gpio) { - dev_err(&pdev->dev, "cfg_gpio callback not provided!\n"); - return -EINVAL; - } - - /* Check for availability of necessary resource */ - dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmatx_res) { - dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n"); - return -ENXIO; - } - - dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!dmarx_res) { - dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n"); - return -ENXIO; - } - - dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2); - if (!dmamic_res) { - dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n"); - return -ENXIO; - } - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "Unable to get register resource\n"); - return -ENXIO; - } - - irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!irq_res) { - dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); - return -ENXIO; - } - - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "s3c-ac97")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - return -EBUSY; - } - - s3c_ac97_pcm_out.channel = dmatx_res->start; - s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; - s3c_ac97_pcm_in.channel = dmarx_res->start; - s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; - s3c_ac97_mic_in.channel = dmamic_res->start; - s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA; - - init_completion(&s3c_ac97.done); - mutex_init(&s3c_ac97.lock); - - s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); - if (s3c_ac97.regs == NULL) { - dev_err(&pdev->dev, "Unable to ioremap register region\n"); - ret = -ENXIO; - goto err1; - } - - s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); - if (IS_ERR(s3c_ac97.ac97_clk)) { - dev_err(&pdev->dev, "s3c-ac97 failed to get ac97_clock\n"); - ret = -ENODEV; - goto err2; - } - clk_enable(s3c_ac97.ac97_clk); - - if (ac97_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err3; - } - - ret = request_irq(irq_res->start, s3c_ac97_irq, - IRQF_DISABLED, "AC97", NULL); - if (ret < 0) { - printk(KERN_ERR "s3c-ac97: interrupt request failed.\n"); - goto err4; - } - - s3c_ac97_dai[S3C_AC97_DAI_PCM].dev = &pdev->dev; - s3c_ac97_dai[S3C_AC97_DAI_MIC].dev = &pdev->dev; - - ret = snd_soc_register_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); - if (ret) - goto err5; - - return 0; - -err5: - free_irq(irq_res->start, NULL); -err4: -err3: - clk_disable(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); -err2: - iounmap(s3c_ac97.regs); -err1: - release_mem_region(mem_res->start, resource_size(mem_res)); - - return ret; -} - -static __devexit int s3c_ac97_remove(struct platform_device *pdev) -{ - struct resource *mem_res, *irq_res; - - snd_soc_unregister_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); - - irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (irq_res) - free_irq(irq_res->start, NULL); - - clk_disable(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); - - iounmap(s3c_ac97.regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (mem_res) - release_mem_region(mem_res->start, resource_size(mem_res)); - - return 0; -} - -static struct platform_driver s3c_ac97_driver = { - .probe = s3c_ac97_probe, - .remove = s3c_ac97_remove, - .driver = { - .name = "s3c-ac97", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c_ac97_init(void) -{ - return platform_driver_register(&s3c_ac97_driver); -} -module_init(s3c_ac97_init); - -static void __exit s3c_ac97_exit(void) -{ - platform_driver_unregister(&s3c_ac97_driver); -} -module_exit(s3c_ac97_exit); - -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); -MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-ac97.h b/sound/soc/s3c24xx/s3c-ac97.h deleted file mode 100644 index 2781983..0000000 --- a/sound/soc/s3c24xx/s3c-ac97.h +++ /dev/null @@ -1,23 +0,0 @@ -/* sound/soc/s3c24xx/s3c-ac97.h - * - * ALSA SoC Audio Layer - S3C AC97 Controller driver - * Evolved from s3c2443-ac97.h - * - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * Credits: Graeme Gregory, Sean Choi - * - * 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 __S3C_AC97_H_ -#define __S3C_AC97_H_ - -#define S3C_AC97_DAI_PCM 0 -#define S3C_AC97_DAI_MIC 1 - -extern struct snd_soc_dai s3c_ac97_dai[]; - -#endif /* __S3C_AC97_H_ */ diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c deleted file mode 100644 index 1b61c23..0000000 --- a/sound/soc/s3c24xx/s3c-dma.c +++ /dev/null @@ -1,483 +0,0 @@ -/* - * s3c-dma.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * 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/init.h> -#include <linux/io.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> -#include <mach/hardware.h> -#include <mach/dma.h> - -#include "s3c-dma.h" - -static const struct snd_pcm_hardware s3c_dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct s3c24xx_runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_limit; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -/* s3c_dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. -*/ -static void s3c_dma_enqueue(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - dma_addr_t pos = prtd->dma_pos; - unsigned int limit; - int ret; - - pr_debug("Entered %s\n", __func__); - - if (s3c_dma_has_circular()) - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - else - limit = prtd->dma_limit; - - pr_debug("%s: loaded %d, limit %d\n", - __func__, prtd->dma_loaded, limit); - - while (prtd->dma_loaded < limit) { - unsigned long len = prtd->dma_period; - - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + len) > prtd->dma_end) { - len = prtd->dma_end - pos; - pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", - __func__, len); - } - - ret = s3c2410_dma_enqueue(prtd->params->channel, - substream, pos, len); - - if (ret == 0) { - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } else - break; - } - - prtd->dma_pos = pos; -} - -static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, - void *dev_id, int size, - enum s3c2410_dma_buffresult result) -{ - struct snd_pcm_substream *substream = dev_id; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) - return; - - prtd = substream->runtime->private_data; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { - prtd->dma_loaded--; - s3c_dma_enqueue(substream); - } - - spin_unlock(&prtd->lock); -} - -static int s3c_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - unsigned long totbytes = params_buffer_bytes(params); - struct s3c_dma_params *dma = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - int ret = 0; - - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - ret = s3c2410_dma_request(prtd->params->channel, - prtd->params->client, NULL); - - if (ret < 0) { - printk(KERN_ERR "failed to get dma channel\n"); - return ret; - } - - /* use the circular buffering if we have it available. */ - if (s3c_dma_has_circular()) - s3c2410_dma_setflags(prtd->params->channel, - S3C2410_DMAF_CIRCULAR); - } - - s3c2410_dma_set_buffdone_fn(prtd->params->channel, - s3c24xx_audio_buffdone); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_limit = runtime->hw.periods_min; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int s3c_dma_hw_free(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - /* TODO - do we need to ensure DMA flushed */ - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - s3c2410_dma_free(prtd->params->channel, prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int s3c_dma_prepare(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* channel needs configuring for mem=>device, increment memory addr, - * sync to pclk, half-word transfers to the IIS-FIFO. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_MEM, - prtd->params->dma_addr); - } else { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_HW, - prtd->params->dma_addr); - } - - s3c2410_dma_config(prtd->params->channel, - prtd->params->dma_size); - - /* flush the DMA channel */ - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - s3c_dma_enqueue(substream); - - return ret; -} - -static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - prtd->state |= ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - prtd->state &= ~ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -s3c_dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - unsigned long res; - dma_addr_t src, dst; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - s3c2410_dma_getposition(prtd->params->channel, &src, &dst); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - res = dst - prtd->dma_start; - else - res = src - prtd->dma_start; - - spin_unlock(&prtd->lock); - - pr_debug("Pointer %x %x\n", src, dst); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * callled... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int s3c_dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); - - prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -static int s3c_dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("s3c_dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int s3c_dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops s3c_dma_ops = { - .open = s3c_dma_open, - .close = s3c_dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c_dma_hw_params, - .hw_free = s3c_dma_hw_free, - .prepare = s3c_dma_prepare, - .trigger = s3c_dma_trigger, - .pointer = s3c_dma_pointer, - .mmap = s3c_dma_mmap, -}; - -static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = s3c_dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static u64 s3c_dma_mask = DMA_BIT_MASK(32); - -static int s3c_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c_dma_mask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -struct snd_soc_platform s3c24xx_soc_platform = { - .name = "s3c24xx-audio", - .pcm_ops = &s3c_dma_ops, - .pcm_new = s3c_dma_new, - .pcm_free = s3c_dma_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); - -static int __init s3c24xx_soc_platform_init(void) -{ - return snd_soc_register_platform(&s3c24xx_soc_platform); -} -module_init(s3c24xx_soc_platform_init); - -static void __exit s3c24xx_soc_platform_exit(void) -{ - snd_soc_unregister_platform(&s3c24xx_soc_platform); -} -module_exit(s3c24xx_soc_platform_exit); - -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-dma.h b/sound/soc/s3c24xx/s3c-dma.h deleted file mode 100644 index 69bb6bf..0000000 --- a/sound/soc/s3c24xx/s3c-dma.h +++ /dev/null @@ -1,31 +0,0 @@ -/* - * s3c-dma.h -- - * - * 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. - * - * ALSA PCM interface for the Samsung S3C24xx CPU - */ - -#ifndef _S3C_AUDIO_H -#define _S3C_AUDIO_H - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ - int channel; /* Channel ID */ - dma_addr_t dma_addr; - int dma_size; /* Size of the DMA transfer */ -}; - -#define S3C24XX_DAI_I2S 0 - -/* platform data */ -extern struct snd_soc_platform s3c24xx_soc_platform; -extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; - -#endif diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c deleted file mode 100644 index 13311c8..0000000 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ /dev/null @@ -1,776 +0,0 @@ -/* sound/soc/s3c24xx/s3c-i2c-v2.c - * - * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. - * - * Copyright (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com - * linux(a)wolfsonmicro.com - * - * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * 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/delay.h> -#include <linux/clk.h> -#include <linux/io.h> - -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/dma.h> - -#include "regs-i2s-v2.h" -#include "s3c-i2s-v2.h" -#include "s3c-dma.h" - -#undef S3C_IIS_V2_SUPPORTED - -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) -#define S3C_IIS_V2_SUPPORTED -#endif - -#ifdef CONFIG_PLAT_S3C64XX -#define S3C_IIS_V2_SUPPORTED -#endif - -#ifndef S3C_IIS_V2_SUPPORTED -#error Unsupported CPU model -#endif - -#define S3C2412_I2S_DEBUG_CON 0 - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -#define bit_set(v, b) (((v) & (b)) ? 1 : 0) - -#if S3C2412_I2S_DEBUG_CON -static void dbg_showcon(const char *fn, u32 con) -{ - printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, - bit_set(con, S3C2412_IISCON_LRINDEX), - bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_TXFIFO_FULL), - bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); - - printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", - fn, - bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_TXCH_PAUSE), - bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); - printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, - bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); -} -#else -static inline void dbg_showcon(const char *fn, u32 con) -{ -} -#endif - - -/* Turn on or off the transmission path. */ -static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) -{ - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - pr_debug("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_TXDMA_PAUSE; - con &= ~S3C2412_IISCON_TXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXONLY: - case S3C2412_IISMOD_MODE_TXRX: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_RXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } else { - /* Note, we do not have any indication that the FIFO problems - * tha the S3C2410/2440 had apply here, so we should be able - * to disable the DMA and TX without resetting the FIFOS. - */ - - con |= S3C2412_IISCON_TXDMA_PAUSE; - con |= S3C2412_IISCON_TXCH_PAUSE; - con &= ~S3C2412_IISCON_TXDMA_ACTIVE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_RXONLY; - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - con &= ~S3C2412_IISCON_IIS_ACTIVE; - break; - - default: - dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } - - fic = readl(regs + S3C2412_IISFIC); - dbg_showcon(__func__, con); - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) -{ - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - pr_debug("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_RXDMA_PAUSE; - con &= ~S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - case S3C2412_IISMOD_MODE_RXONLY: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } else { - /* See txctrl notes on FIFOs. */ - - con &= ~S3C2412_IISCON_RXDMA_ACTIVE; - con |= S3C2412_IISCON_RXDMA_PAUSE; - con |= S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_RXONLY: - con &= ~S3C2412_IISCON_IIS_ACTIVE; - mod &= ~S3C2412_IISMOD_MODE_MASK; - break; - - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXONLY; - break; - - default: - dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } - - fic = readl(regs + S3C2412_IISFIC); - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) -{ - u32 iiscon; - unsigned long loops = msecs_to_loops(5); - - pr_debug("Entered %s\n", __func__); - - while (--loops) { - iiscon = readl(i2s->regs + S3C2412_IISCON); - if (iiscon & S3C2412_IISCON_LRINDEX) - break; - - cpu_relax(); - } - - if (!loops) { - printk(KERN_ERR "%s: timeout\n", __func__); - return -ETIMEDOUT; - } - - return 0; -} - -/* - * Set S3C2412 I2S DAI format - */ -static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - i2s->master = 0; - iismod |= S3C2412_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_CBS_CFS: - i2s->master = 1; - iismod &= ~S3C2412_IISMOD_SLAVE; - break; - default: - pr_err("unknwon master/slave format\n"); - return -EINVAL; - } - - iismod &= ~S3C2412_IISMOD_SDF_MASK; - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_RIGHT_J: - iismod |= S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_MSB; - break; - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_LSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod &= ~S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_IIS; - break; - default: - pr_err("Unknown data format\n"); - return -EINVAL; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("hw_params w: IISMOD: %x \n", iismod); - return 0; -} - -static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *socdai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai = rtd->dai; - struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); - struct s3c_dma_params *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); - - /* Working copies of register */ - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - - iismod &= ~S3C64XX_IISMOD_BLC_MASK; - /* Sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C64XX_IISMOD_BLC_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - break; - case SNDRV_PCM_FORMAT_S24_LE: - iismod |= S3C64XX_IISMOD_BLC_24BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static int s3c_i2sv2_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - pr_debug("Entered %s\n", __func__); - pr_debug("%s r: IISMOD: %x\n", __func__, iismod); - - switch (clk_id) { - case S3C_I2SV2_CLKSRC_PCLK: - iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C_I2SV2_CLKSRC_AUDIOBUS: - iismod |= S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C_I2SV2_CLKSRC_CDCLK: - /* Error if controller doesn't have the CDCLKCON bit */ - if (!(i2s->feature & S3C_FEATURE_CDCLKCON)) - return -EINVAL; - - switch (dir) { - case SND_SOC_CLOCK_IN: - iismod |= S3C64XX_IISMOD_CDCLKCON; - break; - case SND_SOC_CLOCK_OUT: - iismod &= ~S3C64XX_IISMOD_CDCLKCON; - break; - default: - return -EINVAL; - } - break; - - default: - return -EINVAL; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai); - int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - unsigned long irqs; - int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - pr_debug("Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* On start, ensure that the FIFOs are cleared and reset. */ - - writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - /* clear again, just in case */ - writel(0x0, i2s->regs + S3C2412_IISFIC); - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!i2s->master) { - ret = s3c2412_snd_lrsync(i2s); - if (ret) - goto exit_err; - } - - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(i2s, 1); - else - s3c2412_snd_txctrl(i2s, 1); - - local_irq_restore(irqs); - - /* - * Load the next buffer to DMA to meet the reqirement - * of the auto reload mechanism of S3C24XX. - * This call won't bother S3C64XX. - */ - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(i2s, 0); - else - s3c2412_snd_txctrl(i2s, 0); - - local_irq_restore(irqs); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* - * Set S3C2412 Clock dividers - */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 reg; - - pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); - - switch (div_id) { - case S3C_I2SV2_DIV_BCLK: - switch (div) { - case 16: - div = S3C2412_IISMOD_BCLK_16FS; - break; - - case 32: - div = S3C2412_IISMOD_BCLK_32FS; - break; - - case 24: - div = S3C2412_IISMOD_BCLK_24FS; - break; - - case 48: - div = S3C2412_IISMOD_BCLK_48FS; - break; - - default: - return -EINVAL; - } - - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_BCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - - pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C_I2SV2_DIV_RCLK: - switch (div) { - case 256: - div = S3C2412_IISMOD_RCLK_256FS; - break; - - case 384: - div = S3C2412_IISMOD_RCLK_384FS; - break; - - case 512: - div = S3C2412_IISMOD_RCLK_512FS; - break; - - case 768: - div = S3C2412_IISMOD_RCLK_768FS; - break; - - default: - return -EINVAL; - } - - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_RCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C_I2SV2_DIV_PRESCALER: - if (div >= 0) { - writel((div << 8) | S3C2412_IISPSR_PSREN, - i2s->regs + S3C2412_IISPSR); - } else { - writel(0x0, i2s->regs + S3C2412_IISPSR); - } - pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); - break; - - default: - return -EINVAL; - } - - return 0; -} - -static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 reg = readl(i2s->regs + S3C2412_IISFIC); - snd_pcm_sframes_t delay; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - delay = S3C2412_IISFIC_TXCOUNT(reg); - else - delay = S3C2412_IISFIC_RXCOUNT(reg); - - return delay; -} - -struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISMOD_IMS_SYSMUX) - return i2s->iis_cclk; - else - return i2s->iis_pclk; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_get_clock); - -/* default table of all avaialable root fs divisors */ -static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; - -int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk) -{ - unsigned long clkrate = clk_get_rate(clk); - unsigned int div; - unsigned int fsclk; - unsigned int actual; - unsigned int fs; - unsigned int fsdiv; - signed int deviation = 0; - unsigned int best_fs = 0; - unsigned int best_div = 0; - unsigned int best_rate = 0; - unsigned int best_deviation = INT_MAX; - - pr_debug("Input clock rate %ldHz\n", clkrate); - - if (fstab == NULL) - fstab = iis_fs_tab; - - for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) { - fsdiv = iis_fs_tab[fs]; - - fsclk = clkrate / fsdiv; - div = fsclk / rate; - - if ((fsclk % rate) > (rate / 2)) - div++; - - if (div <= 1) - continue; - - actual = clkrate / (fsdiv * div); - deviation = actual - rate; - - printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", - fsdiv, div, actual, deviation); - - deviation = abs(deviation); - - if (deviation < best_deviation) { - best_fs = fsdiv; - best_div = div; - best_rate = actual; - best_deviation = deviation; - } - - if (deviation == 0) - break; - } - - printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", - best_fs, best_div, best_rate); - - info->fs_div = best_fs; - info->clk_div = best_div; - - return 0; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); - -int s3c_i2sv2_probe(struct platform_device *pdev, - struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base) -{ - struct device *dev = &pdev->dev; - unsigned int iismod; - - i2s->dev = dev; - - /* record our i2s structure for later use in the callbacks */ - dai->private_data = i2s; - - if (!base) { - struct resource *res = platform_get_resource(pdev, - IORESOURCE_MEM, - 0); - if (!res) { - dev_err(dev, "Unable to get register resource\n"); - return -ENXIO; - } - - if (!request_mem_region(res->start, resource_size(res), - "s3c64xx-i2s-v4")) { - dev_err(dev, "Unable to request register region\n"); - return -EBUSY; - } - - base = res->start; - } - - i2s->regs = ioremap(base, 0x100); - if (i2s->regs == NULL) { - dev_err(dev, "cannot ioremap registers\n"); - return -ENXIO; - } - - i2s->iis_pclk = clk_get(dev, "iis"); - if (IS_ERR(i2s->iis_pclk)) { - dev_err(dev, "failed to get iis_clock\n"); - iounmap(i2s->regs); - return -ENOENT; - } - - clk_enable(i2s->iis_pclk); - - /* Mark ourselves as in TXRX mode so we can run through our cleanup - * process without warnings. */ - iismod = readl(i2s->regs + S3C2412_IISMOD); - iismod |= S3C2412_IISMOD_MODE_TXRX; - writel(iismod, i2s->regs + S3C2412_IISMOD); - s3c2412_snd_txctrl(i2s, 0); - s3c2412_snd_rxctrl(i2s, 0); - - return 0; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); - -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warning("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warning("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warning("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif - -int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) -{ - struct snd_soc_dai_ops *ops = dai->ops; - - ops->trigger = s3c2412_i2s_trigger; - if (!ops->hw_params) - ops->hw_params = s3c_i2sv2_hw_params; - ops->set_fmt = s3c2412_i2s_set_fmt; - ops->set_clkdiv = s3c2412_i2s_set_clkdiv; - ops->set_sysclk = s3c_i2sv2_set_sysclk; - - /* Allow overriding by (for example) IISv4 */ - if (!ops->delay) - ops->delay = s3c2412_i2s_delay; - - dai->suspend = s3c2412_i2s_suspend; - dai->resume = s3c2412_i2s_resume; - - return snd_soc_register_dai(dai); -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); - -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h deleted file mode 100644 index 766f43a..0000000 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ /dev/null @@ -1,103 +0,0 @@ -/* sound/soc/s3c24xx/s3c-i2s-v2.h - * - * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver - * - * Copyright (c) 2007 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * 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 code is the core support for the I2S block found in a number of - * Samsung SoC devices which is unofficially named I2S-V2. Currently the - * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S - * channels via configurable GPIO. - */ - -#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H -#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__ - -#define S3C_I2SV2_DIV_BCLK (1) -#define S3C_I2SV2_DIV_RCLK (2) -#define S3C_I2SV2_DIV_PRESCALER (3) - -#define S3C_I2SV2_CLKSRC_PCLK 0 -#define S3C_I2SV2_CLKSRC_AUDIOBUS 1 -#define S3C_I2SV2_CLKSRC_CDCLK 2 - -/* Set this flag for I2S controllers that have the bit IISMOD[12] - * bridge/break RCLK signal and external Xi2sCDCLK pin. - */ -#define S3C_FEATURE_CDCLKCON (1 << 0) - -/** - * struct s3c_i2sv2_info - S3C I2S-V2 information - * @dev: The parent device passed to use from the probe. - * @regs: The pointer to the device registe block. - * @feature: Set of bit-flags indicating features of the controller. - * @master: True if the I2S core is the I2S bit clock master. - * @dma_playback: DMA information for playback channel. - * @dma_capture: DMA information for capture channel. - * @suspend_iismod: PM save for the IISMOD register. - * @suspend_iiscon: PM save for the IISCON register. - * @suspend_iispsr: PM save for the IISPSR register. - * - * This is the private codec state for the hardware associated with an - * I2S channel such as the register mappings and clock sources. - */ -struct s3c_i2sv2_info { - struct device *dev; - void __iomem *regs; - - u32 feature; - - struct clk *iis_pclk; - struct clk *iis_cclk; - - unsigned char master; - - struct s3c_dma_params *dma_playback; - struct s3c_dma_params *dma_capture; - - u32 suspend_iismod; - u32 suspend_iiscon; - u32 suspend_iispsr; -}; - -extern struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai); - -struct s3c_i2sv2_rate_calc { - unsigned int clk_div; /* for prescaler */ - unsigned int fs_div; /* for root frame clock */ -}; - -extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk); - -/** - * s3c_i2sv2_probe - probe for i2s device helper - * @pdev: The platform device supplied to the original probe. - * @dai: The ASoC DAI structure supplied to the original probe. - * @i2s: Our local i2s structure to fill in. - * @base: The base address for the registers. - */ -extern int s3c_i2sv2_probe(struct platform_device *pdev, - struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base); - -/** - * s3c_i2sv2_register_dai - register dai with soc core - * @dai: The snd_soc_dai structure to register - * - * Fill in any missing fields and then register the given dai with the - * soc core. - */ -extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai); - -#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c deleted file mode 100644 index 326f0a9..0000000 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ /dev/null @@ -1,554 +0,0 @@ -/* sound/soc/s3c24xx/s3c-pcm.c - * - * ALSA SoC Audio Layer - S3C PCM-Controller driver - * - * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * based upon I2S drivers by Ben Dooks. - * - * 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/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/kernel.h> -#include <linux/gpio.h> -#include <linux/io.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <plat/audio.h> -#include <plat/dma.h> - -#include "s3c-dma.h" -#include "s3c-pcm.h" - -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { - .name = "PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { - .name = "PCM Stereo in" -}; - -static struct s3c_dma_params s3c_pcm_stereo_out[] = { - [0] = { - .client = &s3c_pcm_dma_client_out, - .dma_size = 4, - }, - [1] = { - .client = &s3c_pcm_dma_client_out, - .dma_size = 4, - }, -}; - -static struct s3c_dma_params s3c_pcm_stereo_in[] = { - [0] = { - .client = &s3c_pcm_dma_client_in, - .dma_size = 4, - }, - [1] = { - .client = &s3c_pcm_dma_client_in, - .dma_size = 4, - }, -}; - -static struct s3c_pcm_info s3c_pcm[2]; - -static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on) -{ - void __iomem *regs = pcm->regs; - u32 ctl, clkctl; - - clkctl = readl(regs + S3C_PCM_CLKCTL); - ctl = readl(regs + S3C_PCM_CTL); - ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK - << S3C_PCM_CTL_TXDIPSTICK_SHIFT); - - if (on) { - ctl |= S3C_PCM_CTL_TXDMA_EN; - ctl |= S3C_PCM_CTL_TXFIFO_EN; - ctl |= S3C_PCM_CTL_ENABLE; - ctl |= (0x20<<S3C_PCM_CTL_TXDIPSTICK_SHIFT); - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } else { - ctl &= ~S3C_PCM_CTL_TXDMA_EN; - ctl &= ~S3C_PCM_CTL_TXFIFO_EN; - - if (!(ctl & S3C_PCM_CTL_RXFIFO_EN)) { - ctl &= ~S3C_PCM_CTL_ENABLE; - if (!pcm->idleclk) - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } - } - - writel(clkctl, regs + S3C_PCM_CLKCTL); - writel(ctl, regs + S3C_PCM_CTL); -} - -static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on) -{ - void __iomem *regs = pcm->regs; - u32 ctl, clkctl; - - ctl = readl(regs + S3C_PCM_CTL); - clkctl = readl(regs + S3C_PCM_CLKCTL); - - if (on) { - ctl |= S3C_PCM_CTL_RXDMA_EN; - ctl |= S3C_PCM_CTL_RXFIFO_EN; - ctl |= S3C_PCM_CTL_ENABLE; - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } else { - ctl &= ~S3C_PCM_CTL_RXDMA_EN; - ctl &= ~S3C_PCM_CTL_RXFIFO_EN; - - if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) { - ctl &= ~S3C_PCM_CTL_ENABLE; - if (!pcm->idleclk) - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } - } - - writel(clkctl, regs + S3C_PCM_CLKCTL); - writel(ctl, regs + S3C_PCM_CTL); -} - -static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai); - unsigned long flags; - - dev_dbg(pcm->dev, "Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - spin_lock_irqsave(&pcm->lock, flags); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c_pcm_snd_rxctrl(pcm, 1); - else - s3c_pcm_snd_txctrl(pcm, 1); - - spin_unlock_irqrestore(&pcm->lock, flags); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - spin_lock_irqsave(&pcm->lock, flags); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c_pcm_snd_rxctrl(pcm, 0); - else - s3c_pcm_snd_txctrl(pcm, 0); - - spin_unlock_irqrestore(&pcm->lock, flags); - break; - - default: - return -EINVAL; - } - - return 0; -} - -static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *socdai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai = rtd->dai; - struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); - struct s3c_dma_params *dma_data; - void __iomem *regs = pcm->regs; - struct clk *clk; - int sclk_div, sync_div; - unsigned long flags; - u32 clkctl; - - dev_dbg(pcm->dev, "Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); - - /* Strictly check for sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - break; - default: - return -EINVAL; - } - - spin_lock_irqsave(&pcm->lock, flags); - - /* Get hold of the PCMSOURCE_CLK */ - clkctl = readl(regs + S3C_PCM_CLKCTL); - if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK) - clk = pcm->pclk; - else - clk = pcm->cclk; - - /* Set the SCLK divider */ - sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs / - params_rate(params) / 2 - 1; - - clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK - << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); - clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK) - << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); - - /* Set the SYNC divider */ - sync_div = pcm->sclk_per_fs - 1; - - clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK - << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); - clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK) - << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); - - writel(clkctl, regs + S3C_PCM_CLKCTL); - - spin_unlock_irqrestore(&pcm->lock, flags); - - dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs SCLK_DIV=%d SYNC_DIV=%d\n", - clk_get_rate(clk), pcm->sclk_per_fs, - sclk_div, sync_div); - - return 0; -} - -static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct s3c_pcm_info *pcm = to_info(cpu_dai); - void __iomem *regs = pcm->regs; - unsigned long flags; - int ret = 0; - u32 ctl; - - dev_dbg(pcm->dev, "Entered %s\n", __func__); - - spin_lock_irqsave(&pcm->lock, flags); - - ctl = readl(regs + S3C_PCM_CTL); - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - /* Nothing to do, NB_NF by default */ - break; - default: - dev_err(pcm->dev, "Unsupported clock inversion!\n"); - ret = -EINVAL; - goto exit; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* Nothing to do, Master by default */ - break; - default: - dev_err(pcm->dev, "Unsupported master/slave format!\n"); - ret = -EINVAL; - goto exit; - } - - switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { - case SND_SOC_DAIFMT_CONT: - pcm->idleclk = 1; - break; - case SND_SOC_DAIFMT_GATED: - pcm->idleclk = 0; - break; - default: - dev_err(pcm->dev, "Invalid Clock gating request!\n"); - ret = -EINVAL; - goto exit; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC; - ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC; - break; - case SND_SOC_DAIFMT_DSP_B: - ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC; - ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC; - break; - default: - dev_err(pcm->dev, "Unsupported data format!\n"); - ret = -EINVAL; - goto exit; - } - - writel(ctl, regs + S3C_PCM_CTL); - -exit: - spin_unlock_irqrestore(&pcm->lock, flags); - - return ret; -} - -static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct s3c_pcm_info *pcm = to_info(cpu_dai); - - switch (div_id) { - case S3C_PCM_SCLK_PER_FS: - pcm->sclk_per_fs = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - -static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct s3c_pcm_info *pcm = to_info(cpu_dai); - void __iomem *regs = pcm->regs; - u32 clkctl = readl(regs + S3C_PCM_CLKCTL); - - switch (clk_id) { - case S3C_PCM_CLKSRC_PCLK: - clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK; - break; - - case S3C_PCM_CLKSRC_MUX: - clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK; - - if (clk_get_rate(pcm->cclk) != freq) - clk_set_rate(pcm->cclk, freq); - - break; - - default: - return -EINVAL; - } - - writel(clkctl, regs + S3C_PCM_CLKCTL); - - return 0; -} - -static struct snd_soc_dai_ops s3c_pcm_dai_ops = { - .set_sysclk = s3c_pcm_set_sysclk, - .set_clkdiv = s3c_pcm_set_clkdiv, - .trigger = s3c_pcm_trigger, - .hw_params = s3c_pcm_hw_params, - .set_fmt = s3c_pcm_set_fmt, -}; - -#define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 - -#define S3C_PCM_DECLARE(n) \ -{ \ - .name = "samsung-pcm", \ - .id = (n), \ - .symmetric_rates = 1, \ - .ops = &s3c_pcm_dai_ops, \ - .playback = { \ - .channels_min = 2, \ - .channels_max = 2, \ - .rates = S3C_PCM_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ - }, \ - .capture = { \ - .channels_min = 2, \ - .channels_max = 2, \ - .rates = S3C_PCM_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ - }, \ -} - -struct snd_soc_dai s3c_pcm_dai[] = { - S3C_PCM_DECLARE(0), - S3C_PCM_DECLARE(1), -}; -EXPORT_SYMBOL_GPL(s3c_pcm_dai); - -static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) -{ - struct s3c_pcm_info *pcm; - struct snd_soc_dai *dai; - struct resource *mem_res, *dmatx_res, *dmarx_res; - struct s3c_audio_pdata *pcm_pdata; - int ret; - - /* Check for valid device index */ - if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) { - dev_err(&pdev->dev, "id %d out of range\n", pdev->id); - return -EINVAL; - } - - pcm_pdata = pdev->dev.platform_data; - - /* Check for availability of necessary resource */ - dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmatx_res) { - dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n"); - return -ENXIO; - } - - dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!dmarx_res) { - dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n"); - return -ENXIO; - } - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "Unable to get register resource\n"); - return -ENXIO; - } - - if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - return -EINVAL; - } - - pcm = &s3c_pcm[pdev->id]; - pcm->dev = &pdev->dev; - - spin_lock_init(&pcm->lock); - - dai = &s3c_pcm_dai[pdev->id]; - dai->dev = &pdev->dev; - - /* Default is 128fs */ - pcm->sclk_per_fs = 128; - - pcm->cclk = clk_get(&pdev->dev, "audio-bus"); - if (IS_ERR(pcm->cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus\n"); - ret = PTR_ERR(pcm->cclk); - goto err1; - } - clk_enable(pcm->cclk); - - /* record our pcm structure for later use in the callbacks */ - dai->private_data = pcm; - - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "samsung-pcm")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - ret = -EBUSY; - goto err2; - } - - pcm->regs = ioremap(mem_res->start, 0x100); - if (pcm->regs == NULL) { - dev_err(&pdev->dev, "cannot ioremap registers\n"); - ret = -ENXIO; - goto err3; - } - - pcm->pclk = clk_get(&pdev->dev, "pcm"); - if (IS_ERR(pcm->pclk)) { - dev_err(&pdev->dev, "failed to get pcm_clock\n"); - ret = -ENOENT; - goto err4; - } - clk_enable(pcm->pclk); - - ret = snd_soc_register_dai(dai); - if (ret != 0) { - dev_err(&pdev->dev, "failed to get pcm_clock\n"); - goto err5; - } - - s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start - + S3C_PCM_RXFIFO; - s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start - + S3C_PCM_TXFIFO; - - s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start; - s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start; - - pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; - pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; - - return 0; - -err5: - clk_disable(pcm->pclk); - clk_put(pcm->pclk); -err4: - iounmap(pcm->regs); -err3: - release_mem_region(mem_res->start, resource_size(mem_res)); -err2: - clk_disable(pcm->cclk); - clk_put(pcm->cclk); -err1: - return ret; -} - -static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) -{ - struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; - struct resource *mem_res; - - iounmap(pcm->regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem_res->start, resource_size(mem_res)); - - clk_disable(pcm->cclk); - clk_disable(pcm->pclk); - clk_put(pcm->pclk); - clk_put(pcm->cclk); - - return 0; -} - -static struct platform_driver s3c_pcm_driver = { - .probe = s3c_pcm_dev_probe, - .remove = s3c_pcm_dev_remove, - .driver = { - .name = "samsung-pcm", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c_pcm_init(void) -{ - return platform_driver_register(&s3c_pcm_driver); -} -module_init(s3c_pcm_init); - -static void __exit s3c_pcm_exit(void) -{ - platform_driver_unregister(&s3c_pcm_driver); -} -module_exit(s3c_pcm_exit); - -/* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); -MODULE_DESCRIPTION("S3C PCM Controller Driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-pcm.h b/sound/soc/s3c24xx/s3c-pcm.h deleted file mode 100644 index 69ff997..0000000 --- a/sound/soc/s3c24xx/s3c-pcm.h +++ /dev/null @@ -1,123 +0,0 @@ -/* sound/soc/s3c24xx/s3c-pcm.h - * - * 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 __S3C_PCM_H -#define __S3C_PCM_H __FILE__ - -/*Register Offsets */ -#define S3C_PCM_CTL (0x00) -#define S3C_PCM_CLKCTL (0x04) -#define S3C_PCM_TXFIFO (0x08) -#define S3C_PCM_RXFIFO (0x0C) -#define S3C_PCM_IRQCTL (0x10) -#define S3C_PCM_IRQSTAT (0x14) -#define S3C_PCM_FIFOSTAT (0x18) -#define S3C_PCM_CLRINT (0x20) - -/* PCM_CTL Bit-Fields */ -#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) -#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) -#define S3C_PCM_CTL_RXDIPSTICK_MSK (0x3f<<7) -#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) -#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) -#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) -#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) -#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) -#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) -#define S3C_PCM_CTL_ENABLE (0x1<<0) - -/* PCM_CLKCTL Bit-Fields */ -#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) -#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) -#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) -#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) -#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) -#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) - -/* PCM_TXFIFO Bit-Fields */ -#define S3C_PCM_TXFIFO_DVALID (0x1<<16) -#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) - -/* PCM_RXFIFO Bit-Fields */ -#define S3C_PCM_RXFIFO_DVALID (0x1<<16) -#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) - -/* PCM_IRQCTL Bit-Fields */ -#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) -#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) -#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) -#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) -#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) -#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) -#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) -#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) -#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) -#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) -#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) -#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) -#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) -#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) - -/* PCM_IRQSTAT Bit-Fields */ -#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) -#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) -#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) -#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) -#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) -#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) -#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) -#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) -#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) -#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) -#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) -#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) -#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) -#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) - -/* PCM_FIFOSTAT Bit-Fields */ -#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) -#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) -#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) -#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) -#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) -#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) -#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) -#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) -#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) -#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) - -#define S3C_PCM_CLKSRC_PCLK 0 -#define S3C_PCM_CLKSRC_MUX 1 - -#define S3C_PCM_SCLK_PER_FS 0 - -/** - * struct s3c_pcm_info - S3C PCM Controller information - * @dev: The parent device passed to use from the probe. - * @regs: The pointer to the device register block. - * @dma_playback: DMA information for playback channel. - * @dma_capture: DMA information for capture channel. - */ -struct s3c_pcm_info { - spinlock_t lock; - struct device *dev; - void __iomem *regs; - - unsigned int sclk_per_fs; - - /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ - unsigned int idleclk; - - struct clk *pclk; - struct clk *cclk; - - struct s3c_dma_params *dma_playback; - struct s3c_dma_params *dma_capture; -}; - -#endif /* __S3C_PCM_H */ diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c deleted file mode 100644 index 709adef..0000000 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ /dev/null @@ -1,190 +0,0 @@ -/* sound/soc/s3c24xx/s3c2412-i2s.c - * - * ALSA Soc Audio Layer - S3C2412 I2S driver - * - * Copyright (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com - * linux(a)wolfsonmicro.com - * - * Copyright (c) 2007, 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * 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/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/gpio.h> -#include <linux/clk.h> -#include <linux/kernel.h> -#include <linux/io.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> -#include <mach/hardware.h> - -#include <mach/regs-gpio.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "regs-i2s-v2.h" -#include "s3c2412-i2s.h" - -#define S3C2412_I2S_DEBUG 0 - -static struct s3c2410_dma_client s3c2412_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2412_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - -static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = &s3c2412_dma_client_out, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, - .dma_size = 4, -}; - -static struct s3c_i2sv2_info s3c2412_i2s; - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static int s3c2412_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - int ret; - - pr_debug("Entered %s\n", __func__); - - ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS); - if (ret) - return ret; - - s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; - s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - - s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); - if (s3c2412_i2s.iis_cclk == NULL) { - pr_err("failed to get i2sclk clock\n"); - iounmap(s3c2412_i2s.regs); - return -ENODEV; - } - - /* Set MPLL as the source for IIS CLK */ - - clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - clk_enable(s3c2412_i2s.iis_cclk); - - s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; - - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); - - return 0; -} - -static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - struct s3c_dma_params *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C2412_IISMOD_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - iismod &= ~S3C2412_IISMOD_8BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -#define S3C2412_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { - .hw_params = s3c2412_i2s_hw_params, -}; - -struct snd_soc_dai s3c2412_i2s_dai = { - .name = "s3c2412-i2s", - .id = 0, - .probe = s3c2412_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C2412_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C2412_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &s3c2412_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); - -static int __init s3c2412_i2s_init(void) -{ - return s3c_i2sv2_register_dai(&s3c2412_i2s_dai); -} -module_init(s3c2412_i2s_init); - -static void __exit s3c2412_i2s_exit(void) -{ - snd_soc_unregister_dai(&s3c2412_i2s_dai); -} -module_exit(s3c2412_i2s_exit); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h deleted file mode 100644 index 0b5686b..0000000 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ /dev/null @@ -1,29 +0,0 @@ -/* sound/soc/s3c24xx/s3c2412-i2s.c - * - * ALSA Soc Audio Layer - S3C2412 I2S driver - * - * Copyright (c) 2007 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * 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 __SND_SOC_S3C24XX_S3C2412_I2S_H -#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ - -#include "s3c-i2s-v2.h" - -#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK -#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK -#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER - -#define S3C2412_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK -#define S3C2412_CLKSRC_I2SCLK S3C_I2SV2_CLKSRC_AUDIOBUS - -extern struct snd_soc_dai s3c2412_i2s_dai; - -#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c deleted file mode 100644 index c3ac890..0000000 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ /dev/null @@ -1,503 +0,0 @@ -/* - * s3c24xx-i2s.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * 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/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/jiffies.h> -#include <linux/io.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/regs-gpio.h> -#include <mach/regs-clock.h> - -#include <asm/dma.h> -#include <mach/dma.h> - -#include <plat/regs-iis.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" - -static struct s3c2410_dma_client s3c24xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c24xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - -static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = &s3c24xx_dma_client_out, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, - .dma_size = 2, -}; - -static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, - .dma_size = 2, -}; - -struct s3c24xx_i2s_info { - void __iomem *regs; - struct clk *iis_clk; - u32 iiscon; - u32 iismod; - u32 iisfcon; - u32 iispsr; -}; -static struct s3c24xx_i2s_info s3c24xx_i2s; - -static void s3c24xx_snd_txctrl(int on) -{ - u32 iisfcon; - u32 iiscon; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); - - if (on) { - iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; - iiscon |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN; - iiscon &= ~S3C2410_IISCON_TXIDLE; - iismod |= S3C2410_IISMOD_TXMODE; - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - } else { - /* note, we have to disable the FIFOs otherwise bad things - * seem to happen when the DMA stops. According to the - * Samsung supplied kernel, this should allow the DMA - * engine and FIFOs to reset. If this isn't allowed, the - * DMA engine will simply freeze randomly. - */ - - iisfcon &= ~S3C2410_IISFCON_TXENABLE; - iisfcon &= ~S3C2410_IISFCON_TXDMA; - iiscon |= S3C2410_IISCON_TXIDLE; - iiscon &= ~S3C2410_IISCON_TXDMAEN; - iismod &= ~S3C2410_IISMOD_TXMODE; - - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - } - - pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); -} - -static void s3c24xx_snd_rxctrl(int on) -{ - u32 iisfcon; - u32 iiscon; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); - - if (on) { - iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; - iiscon |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN; - iiscon &= ~S3C2410_IISCON_RXIDLE; - iismod |= S3C2410_IISMOD_RXMODE; - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - } else { - /* note, we have to disable the FIFOs otherwise bad things - * seem to happen when the DMA stops. According to the - * Samsung supplied kernel, this should allow the DMA - * engine and FIFOs to reset. If this isn't allowed, the - * DMA engine will simply freeze randomly. - */ - - iisfcon &= ~S3C2410_IISFCON_RXENABLE; - iisfcon &= ~S3C2410_IISFCON_RXDMA; - iiscon |= S3C2410_IISCON_RXIDLE; - iiscon &= ~S3C2410_IISCON_RXDMAEN; - iismod &= ~S3C2410_IISMOD_RXMODE; - - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - } - - pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); -} - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c24xx_snd_lrsync(void) -{ - u32 iiscon; - int timeout = 50; /* 5ms */ - - pr_debug("Entered %s\n", __func__); - - while (1) { - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - if (iiscon & S3C2410_IISCON_LRINDEX) - break; - - if (!timeout--) - return -ETIMEDOUT; - udelay(100); - } - - return 0; -} - -/* - * Check whether CPU is the master or slave - */ -static inline int s3c24xx_snd_is_clkmaster(void) -{ - pr_debug("Entered %s\n", __func__); - - return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; -} - -/* - * Set S3C24xx I2S DAI format - */ -static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - iismod |= S3C2410_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_CBS_CFS: - iismod &= ~S3C2410_IISMOD_SLAVE; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2410_IISMOD_MSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod &= ~S3C2410_IISMOD_MSB; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %x \n", iismod); - return 0; -} - -static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c24xx_i2s_pcm_stereo_out; - else - dma_data = &s3c24xx_i2s_pcm_stereo_in; - - snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); - - /* Working copies of register */ - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %x\n", iismod); - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->dma_size = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - iismod |= S3C2410_IISMOD_16BIT; - dma_data->dma_size = 2; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %x\n", iismod); - return 0; -} - -static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - int ret = 0; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - pr_debug("Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!s3c24xx_snd_is_clkmaster()) { - ret = s3c24xx_snd_lrsync(); - if (ret) - goto exit_err; - } - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c24xx_snd_rxctrl(1); - else - s3c24xx_snd_txctrl(1); - - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c24xx_snd_rxctrl(0); - else - s3c24xx_snd_txctrl(0); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* - * Set S3C24xx Clock source - */ -static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("Entered %s\n", __func__); - - iismod &= ~S3C2440_IISMOD_MPLL; - - switch (clk_id) { - case S3C24XX_CLKSRC_PCLK: - break; - case S3C24XX_CLKSRC_MPLL: - iismod |= S3C2440_IISMOD_MPLL; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - return 0; -} - -/* - * Set S3C24xx Clock dividers - */ -static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - u32 reg; - - pr_debug("Entered %s\n", __func__); - - switch (div_id) { - case S3C24XX_DIV_BCLK: - reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; - writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); - break; - case S3C24XX_DIV_MCLK: - reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); - writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); - break; - case S3C24XX_DIV_PRESCALER: - writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR); - reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON); - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * To avoid duplicating clock code, allow machine driver to - * get the clockrate from here. - */ -u32 s3c24xx_i2s_get_clockrate(void) -{ - return clk_get_rate(s3c24xx_i2s.iis_clk); -} -EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); - -static int s3c24xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - pr_debug("Entered %s\n", __func__); - - s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; - - s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); - if (s3c24xx_i2s.iis_clk == NULL) { - pr_err("failed to get iis_clock\n"); - iounmap(s3c24xx_i2s.regs); - return -ENODEV; - } - clk_enable(s3c24xx_i2s.iis_clk); - - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); - - writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); - - s3c24xx_snd_txctrl(0); - s3c24xx_snd_rxctrl(0); - - return 0; -} - -#ifdef CONFIG_PM -static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) -{ - pr_debug("Entered %s\n", __func__); - - s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); - - clk_disable(s3c24xx_i2s.iis_clk); - - return 0; -} - -static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) -{ - pr_debug("Entered %s\n", __func__); - clk_enable(s3c24xx_i2s.iis_clk); - - writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR); - - return 0; -} -#else -#define s3c24xx_i2s_suspend NULL -#define s3c24xx_i2s_resume NULL -#endif - - -#define S3C24XX_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { - .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params, - .set_fmt = s3c24xx_i2s_set_fmt, - .set_clkdiv = s3c24xx_i2s_set_clkdiv, - .set_sysclk = s3c24xx_i2s_set_sysclk, -}; - -struct snd_soc_dai s3c24xx_i2s_dai = { - .name = "s3c24xx-i2s", - .id = 0, - .probe = s3c24xx_i2s_probe, - .suspend = s3c24xx_i2s_suspend, - .resume = s3c24xx_i2s_resume, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C24XX_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C24XX_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c24xx_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); - -static int __init s3c24xx_i2s_init(void) -{ - return snd_soc_register_dai(&s3c24xx_i2s_dai); -} -module_init(s3c24xx_i2s_init); - -static void __exit s3c24xx_i2s_exit(void) -{ - snd_soc_unregister_dai(&s3c24xx_i2s_dai); -} -module_exit(s3c24xx_i2s_exit); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h deleted file mode 100644 index 726d91c..0000000 --- a/sound/soc/s3c24xx/s3c24xx-i2s.h +++ /dev/null @@ -1,37 +0,0 @@ -/* - * s3c24xx-i2s.c -- ALSA Soc Audio Layer - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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. - * - * Revision history - * 10th Nov 2006 Initial version. - */ - -#ifndef S3C24XXI2S_H_ -#define S3C24XXI2S_H_ - -/* clock sources */ -#define S3C24XX_CLKSRC_PCLK 0 -#define S3C24XX_CLKSRC_MPLL 1 - -/* Clock dividers */ -#define S3C24XX_DIV_MCLK 0 -#define S3C24XX_DIV_BCLK 1 -#define S3C24XX_DIV_PRESCALER 2 - -/* prescaler */ -#define S3C24XX_PRESCALE(a,b) \ - (((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT)) - -u32 s3c24xx_i2s_get_clockrate(void); - -extern struct snd_soc_dai s3c24xx_i2s_dai; - -#endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c deleted file mode 100644 index 4984754..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ /dev/null @@ -1,394 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> -#include <linux/clk.h> -#include <linux/i2c.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <plat/audio-simtec.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -static struct s3c24xx_audio_simtec_pdata *pdata; -static struct clk *xtal_clk; - -static int spk_gain; -static int spk_unmute; - -/** - * speaker_gain_get - read the speaker gain setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be updated. - * - * Read the value for the AMP gain control. - */ -static int speaker_gain_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = spk_gain; - return 0; -} - -/** - * speaker_gain_set - set the value of the speaker amp gain - * @value: The value to write. - */ -static void speaker_gain_set(int value) -{ - gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); - gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); -} - -/** - * speaker_gain_put - set the speaker gain setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be set. - * - * Set the value of the speaker gain from the specified - * @ucontrol setting. - * - * Note, if the speaker amp is muted, then we do not set a gain value - * as at-least one of the ICs that is fitted will try and power up even - * if the main control is set to off. - */ -static int speaker_gain_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int value = ucontrol->value.integer.value[0]; - - spk_gain = value; - - if (!spk_unmute) - speaker_gain_set(value); - - return 0; -} - -static const struct snd_kcontrol_new amp_gain_controls[] = { - SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, - speaker_gain_get, speaker_gain_put), -}; - -/** - * spk_unmute_state - set the unmute state of the speaker - * @to: zero to unmute, non-zero to ununmute. - */ -static void spk_unmute_state(int to) -{ - pr_debug("%s: to=%d\n", __func__, to); - - spk_unmute = to; - gpio_set_value(pdata->amp_gpio, to); - - /* if we're umuting, also re-set the gain */ - if (to && pdata->amp_gain[0] > 0) - speaker_gain_set(spk_gain); -} - -/** - * speaker_unmute_get - read the speaker unmute setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be updated. - * - * Read the value for the AMP gain control. - */ -static int speaker_unmute_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = spk_unmute; - return 0; -} - -/** - * speaker_unmute_put - set the speaker unmute setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be set. - * - * Set the value of the speaker gain from the specified - * @ucontrol setting. - */ -static int speaker_unmute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - spk_unmute_state(ucontrol->value.integer.value[0]); - return 0; -} - -/* This is added as a manual control as the speaker amps create clicks - * when their power state is changed, which are far more noticeable than - * anything produced by the CODEC itself. - */ -static const struct snd_kcontrol_new amp_unmute_controls[] = { - SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0, - speaker_unmute_get, speaker_unmute_put), -}; - -void simtec_audio_init(struct snd_soc_codec *codec) -{ - if (pdata->amp_gpio > 0) { - pr_debug("%s: adding amp routes\n", __func__); - - snd_soc_add_controls(codec, amp_unmute_controls, - ARRAY_SIZE(amp_unmute_controls)); - } - - if (pdata->amp_gain[0] > 0) { - pr_debug("%s: adding amp controls\n", __func__); - snd_soc_add_controls(codec, amp_gain_controls, - ARRAY_SIZE(amp_gain_controls)); - } -} -EXPORT_SYMBOL_GPL(simtec_audio_init); - -#define CODEC_CLOCK 12000000 - -/** - * simtec_hw_params - update hardware parameters - * @substream: The audio substream instance. - * @params: The parameters requested. - * - * Update the codec data routing and configuration settings - * from the supplied data. - */ -static int simtec_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->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - /* Set the CODEC as the bus clock master, I2S */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; - } - - /* Set the CODEC as the bus clock master */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - CODEC_CLOCK, SND_SOC_CLOCK_IN); - if (ret) { - pr_err( "%s: failed setting codec sysclk\n", __func__); - return ret; - } - - if (pdata->use_mpllin) { - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, - 0, SND_SOC_CLOCK_OUT); - - if (ret) { - pr_err("%s: failed to set MPLLin as clksrc\n", - __func__); - return ret; - } - } - - if (pdata->output_cdclk) { - int cdclk_scale; - - cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; - cdclk_scale--; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - cdclk_scale); - } - - return 0; -} - -static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) -{ - /* call any board supplied startup code, this currently only - * covers the bast/vr1000 which have a CPLD in the way of the - * LRCLK */ - if (pd->startup) - pd->startup(); - - return 0; -} - -static struct snd_soc_ops simtec_snd_ops = { - .hw_params = simtec_hw_params, -}; - -/** - * attach_gpio_amp - get and configure the necessary gpios - * @dev: The device we're probing. - * @pd: The platform data supplied by the board. - * - * If there is a GPIO based amplifier attached to the board, claim - * the necessary GPIO lines for it, and set default values. - */ -static int attach_gpio_amp(struct device *dev, - struct s3c24xx_audio_simtec_pdata *pd) -{ - int ret; - - /* attach gpio amp gain (if any) */ - if (pdata->amp_gain[0] > 0) { - ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); - if (ret) { - dev_err(dev, "cannot get amp gpio gain0\n"); - return ret; - } - - ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); - if (ret) { - dev_err(dev, "cannot get amp gpio gain1\n"); - gpio_free(pdata->amp_gain[0]); - return ret; - } - - gpio_direction_output(pd->amp_gain[0], 0); - gpio_direction_output(pd->amp_gain[1], 0); - } - - /* note, currently we assume GPA0 isn't valid amp */ - if (pdata->amp_gpio > 0) { - ret = gpio_request(pd->amp_gpio, "gpio-amp"); - if (ret) { - dev_err(dev, "cannot get amp gpio %d (%d)\n", - pd->amp_gpio, ret); - goto err_amp; - } - - /* set the amp off at startup */ - spk_unmute_state(0); - } - - return 0; - -err_amp: - if (pd->amp_gain[0] > 0) { - gpio_free(pd->amp_gain[0]); - gpio_free(pd->amp_gain[1]); - } - - return ret; -} - -static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) -{ - if (pd->amp_gain[0] > 0) { - gpio_free(pd->amp_gain[0]); - gpio_free(pd->amp_gain[1]); - } - - if (pd->amp_gpio > 0) - gpio_free(pd->amp_gpio); -} - -#ifdef CONFIG_PM -int simtec_audio_resume(struct device *dev) -{ - simtec_call_startup(pdata); - return 0; -} - -const struct dev_pm_ops simtec_audio_pmops = { - .resume = simtec_audio_resume, -}; -EXPORT_SYMBOL_GPL(simtec_audio_pmops); -#endif - -int __devinit simtec_audio_core_probe(struct platform_device *pdev, - struct snd_soc_device *socdev) -{ - struct platform_device *snd_dev; - int ret; - - socdev->card->dai_link->ops = &simtec_snd_ops; - - pdata = pdev->dev.platform_data; - if (!pdata) { - dev_err(&pdev->dev, "no platform data supplied\n"); - return -EINVAL; - } - - simtec_call_startup(pdata); - - xtal_clk = clk_get(&pdev->dev, "xtal"); - if (IS_ERR(xtal_clk)) { - dev_err(&pdev->dev, "could not get clkout0\n"); - return -EINVAL; - } - - dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); - - ret = attach_gpio_amp(&pdev->dev, pdata); - if (ret) - goto err_clk; - - snd_dev = platform_device_alloc("soc-audio", -1); - if (!snd_dev) { - dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); - ret = -ENOMEM; - goto err_gpio; - } - - platform_set_drvdata(snd_dev, socdev); - socdev->dev = &snd_dev->dev; - - ret = platform_device_add(snd_dev); - if (ret) { - dev_err(&pdev->dev, "failed to add soc-audio dev\n"); - goto err_pdev; - } - - platform_set_drvdata(pdev, snd_dev); - return 0; - -err_pdev: - platform_device_put(snd_dev); - -err_gpio: - detach_gpio_amp(pdata); - -err_clk: - clk_put(xtal_clk); - return ret; -} -EXPORT_SYMBOL_GPL(simtec_audio_core_probe); - -int __devexit simtec_audio_remove(struct platform_device *pdev) -{ - struct platform_device *snd_dev = platform_get_drvdata(pdev); - - platform_device_unregister(snd_dev); - - detach_gpio_amp(pdata); - clk_put(xtal_clk); - return 0; -} -EXPORT_SYMBOL_GPL(simtec_audio_remove); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h deleted file mode 100644 index e18faee..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec.h +++ /dev/null @@ -1,22 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec.h - * - * Copyright 2009 Simtec Electronics - * - * 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. -*/ - -extern void simtec_audio_init(struct snd_soc_codec *codec); - -extern int simtec_audio_core_probe(struct platform_device *pdev, - struct snd_soc_device *socdev); - -extern int simtec_audio_remove(struct platform_device *pdev); - -#ifdef CONFIG_PM -extern const struct dev_pm_ops simtec_audio_pmops; -#define simtec_audio_pm &simtec_audio_pmops -#else -#define simtec_audio_pm NULL -#endif diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c deleted file mode 100644 index bdf8951..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ /dev/null @@ -1,153 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c - * - * Copyright 2009 Simtec Electronics - * - * 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/clk.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <plat/audio-simtec.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -#include "../codecs/tlv320aic3x.h" - -static const struct snd_soc_dapm_widget dapm_widgets[] = { - SND_SOC_DAPM_LINE("GSM Out", NULL), - SND_SOC_DAPM_LINE("GSM In", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_LINE("ZV", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_HP("Headphone Jack", NULL), -}; - -static const struct snd_soc_dapm_route base_map[] = { - /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ - - { "Headphone Jack", NULL, "HPLOUT" }, - { "Headphone Jack", NULL, "HPLCOM" }, - { "Headphone Jack", NULL, "HPROUT" }, - { "Headphone Jack", NULL, "HPRCOM" }, - - /* ZV connected to Line1 */ - - { "LINE1L", NULL, "ZV" }, - { "LINE1R", NULL, "ZV" }, - - /* Line In connected to Line2 */ - - { "LINE2L", NULL, "Line In" }, - { "LINE2R", NULL, "Line In" }, - - /* Microphone connected to MIC3R and MIC_BIAS */ - - { "MIC3L", NULL, "Mic Jack" }, - - /* GSM connected to MONO_LOUT and MIC3L (in) */ - - { "GSM Out", NULL, "MONO_LOUT" }, - { "MIC3L", NULL, "GSM In" }, - - /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are - * not using the DAPM to power it up and down as there it makes - * a click when powering up. */ -}; - -/** - * simtec_hermes_init - initialise and add controls - * @codec; The codec instance to attach to. - * - * Attach our controls and configure the necessary codec - * mappings for our sound card instance. -*/ -static int simtec_hermes_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_new_controls(codec, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); - - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - - simtec_audio_init(codec); - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct aic3x_setup_data codec_setup = { -}; - -static struct snd_soc_dai_link simtec_dai_aic33 = { - .name = "tlv320aic33", - .stream_name = "TLV320AIC33", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &aic3x_dai, - .init = simtec_hermes_init, -}; - -/* simtec audio machine driver */ -static struct snd_soc_card snd_soc_machine_simtec_aic33 = { - .name = "Simtec-Hermes", - .platform = &s3c24xx_soc_platform, - .dai_link = &simtec_dai_aic33, - .num_links = 1, -}; - -/* simtec audio subsystem */ -static struct snd_soc_device simtec_snd_devdata_aic33 = { - .card = &snd_soc_machine_simtec_aic33, - .codec_dev = &soc_codec_dev_aic3x, - .codec_data = &codec_setup, -}; - -static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) -{ - dev_info(&pd->dev, "probing....\n"); - return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); -} - -static struct platform_driver simtec_audio_hermes_platdrv = { - .driver = { - .owner = THIS_MODULE, - .name = "s3c24xx-simtec-hermes-snd", - .pm = simtec_audio_pm, - }, - .probe = simtec_audio_hermes_probe, - .remove = __devexit_p(simtec_audio_remove), -}; - -MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); - -static int __init simtec_hermes_modinit(void) -{ - return platform_driver_register(&simtec_audio_hermes_platdrv); -} - -static void __exit simtec_hermes_modexit(void) -{ - platform_driver_unregister(&simtec_audio_hermes_platdrv); -} - -module_init(simtec_hermes_modinit); -module_exit(simtec_hermes_modexit); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c deleted file mode 100644 index 185c0ac..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ /dev/null @@ -1,137 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c - * - * Copyright 2009 Simtec Electronics - * - * 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/clk.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <plat/audio-simtec.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -#include "../codecs/tlv320aic23.h" - -/* supported machines: - * - * Machine Connections AMP - * ------- ----------- --- - * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) - * VR1000 HPOUT, LIN None - * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) - * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) - * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) - */ - -static const struct snd_soc_dapm_widget dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route base_map[] = { - { "Headphone Jack", NULL, "LHPOUT"}, - { "Headphone Jack", NULL, "RHPOUT"}, - - { "Line Out", NULL, "LOUT" }, - { "Line Out", NULL, "ROUT" }, - - { "LLINEIN", NULL, "Line In"}, - { "RLINEIN", NULL, "Line In"}, - - { "MICIN", NULL, "Mic Jack"}, -}; - -/** - * simtec_tlv320aic23_init - initialise and add controls - * @codec; The codec instance to attach to. - * - * Attach our controls and configure the necessary codec - * mappings for our sound card instance. -*/ -static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_new_controls(codec, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); - - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - - simtec_audio_init(codec); - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link simtec_dai_aic23 = { - .name = "tlv320aic23", - .stream_name = "TLV320AIC23", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &tlv320aic23_dai, - .init = simtec_tlv320aic23_init, -}; - -/* simtec audio machine driver */ -static struct snd_soc_card snd_soc_machine_simtec_aic23 = { - .name = "Simtec", - .platform = &s3c24xx_soc_platform, - .dai_link = &simtec_dai_aic23, - .num_links = 1, -}; - -/* simtec audio subsystem */ -static struct snd_soc_device simtec_snd_devdata_aic23 = { - .card = &snd_soc_machine_simtec_aic23, - .codec_dev = &soc_codec_dev_tlv320aic23, -}; - -static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) -{ - return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); -} - -static struct platform_driver simtec_audio_tlv320aic23_platdrv = { - .driver = { - .owner = THIS_MODULE, - .name = "s3c24xx-simtec-tlv320aic23", - .pm = simtec_audio_pm, - }, - .probe = simtec_audio_tlv320aic23_probe, - .remove = __devexit_p(simtec_audio_remove), -}; - -MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); - -static int __init simtec_tlv320aic23_modinit(void) -{ - return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); -} - -static void __exit simtec_tlv320aic23_modexit(void) -{ - platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); -} - -module_init(simtec_tlv320aic23_modinit); -module_exit(simtec_tlv320aic23_modexit); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c deleted file mode 100644 index 052d596..0000000 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ /dev/null @@ -1,373 +0,0 @@ -/* - * Modifications by Christian Pellegrin <chripell(a)evolware.org> - * - * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver - * - * Copyright 2007 Dension Audio Systems Ltd. - * Author: Zoltan Devai - * - * 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/clk.h> -#include <linux/mutex.h> -#include <linux/gpio.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/s3c24xx_uda134x.h> -#include <sound/uda134x.h> - -#include <plat/regs-iis.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "../codecs/uda134x.h" - - -/* #define ENFORCE_RATES 1 */ -/* - Unfortunately the S3C24XX in master mode has a limited capacity of - generating the clock for the codec. If you define this only rates - that are really available will be enforced. But be careful, most - user level application just want the usual sampling frequencies (8, - 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly - operation for embedded systems. So if you aren't very lucky or your - hardware engineer wasn't very forward-looking it's better to leave - this undefined. If you do so an approximate value for the requested - sampling rate in the range -/+ 5% will be chosen. If this in not - possible an error will be returned. -*/ - -static struct clk *xtal; -static struct clk *pclk; -/* this is need because we don't have a place where to keep the - * pointers to the clocks in each substream. We get the clocks only - * when we are actually using them so we don't block stuff like - * frequency change or oscillator power-off */ -static int clk_users; -static DEFINE_MUTEX(clk_lock); - -static unsigned int rates[33 * 2]; -#ifdef ENFORCE_RATES -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; -#endif - -static struct platform_device *s3c24xx_uda134x_snd_device; - -static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) -{ - int ret = 0; -#ifdef ENFORCE_RATES - struct snd_pcm_runtime *runtime = substream->runtime; -#endif - - mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); - if (clk_users == 0) { - xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); - if (!xtal) { - printk(KERN_ERR "%s cannot get xtal\n", __func__); - ret = -EBUSY; - } else { - pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, - "pclk"); - if (!pclk) { - printk(KERN_ERR "%s cannot get pclk\n", - __func__); - clk_put(xtal); - ret = -EBUSY; - } - } - if (!ret) { - int i, j; - - for (i = 0; i < 2; i++) { - int fs = i ? 256 : 384; - - rates[i*33] = clk_get_rate(xtal) / fs; - for (j = 1; j < 33; j++) - rates[i*33 + j] = clk_get_rate(pclk) / - (j * fs); - } - } - } - clk_users += 1; - mutex_unlock(&clk_lock); - if (!ret) { -#ifdef ENFORCE_RATES - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates); - if (ret < 0) - printk(KERN_ERR "%s cannot set constraints\n", - __func__); -#endif - } - return ret; -} - -static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) -{ - mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); - clk_users -= 1; - if (clk_users == 0) { - clk_put(xtal); - xtal = NULL; - clk_put(pclk); - pclk = NULL; - } - mutex_unlock(&clk_lock); -} - -static int s3c24xx_uda134x_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->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int clk = 0; - int ret = 0; - int clk_source, fs_mode; - unsigned long rate = params_rate(params); - long err, cerr; - unsigned int div; - int i, bi; - - err = 999999; - bi = 0; - for (i = 0; i < 2*33; i++) { - cerr = rates[i] - rate; - if (cerr < 0) - cerr = -cerr; - if (cerr < err) { - err = cerr; - bi = i; - } - } - if (bi / 33 == 1) - fs_mode = S3C2410_IISMOD_256FS; - else - fs_mode = S3C2410_IISMOD_384FS; - if (bi % 33 == 0) { - clk_source = S3C24XX_CLKSRC_MPLL; - div = 1; - } else { - clk_source = S3C24XX_CLKSRC_PCLK; - div = bi % 33; - } - pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); - - clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; - pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, - fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", - clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", - div, clk, err); - - if ((err * 100 / rate) > 5) { - printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " - "too different from desired (%ld%%)\n", - err * 100 / rate); - return -EINVAL; - } - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(div, div)); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops s3c24xx_uda134x_ops = { - .startup = s3c24xx_uda134x_startup, - .shutdown = s3c24xx_uda134x_shutdown, - .hw_params = s3c24xx_uda134x_hw_params, -}; - -static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { - .name = "UDA134X", - .stream_name = "UDA134X", - .codec_dai = &uda134x_dai, - .cpu_dai = &s3c24xx_i2s_dai, - .ops = &s3c24xx_uda134x_ops, -}; - -static struct snd_soc_card snd_soc_s3c24xx_uda134x = { - .name = "S3C24XX_UDA134X", - .platform = &s3c24xx_soc_platform, - .dai_link = &s3c24xx_uda134x_dai_link, - .num_links = 1, -}; - -static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; - -static void setdat(int v) -{ - gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); -} - -static void setclk(int v) -{ - gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); -} - -static void setmode(int v) -{ - gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); -} - -static struct uda134x_platform_data s3c24xx_uda134x = { - .l3 = { - .setdat = setdat, - .setclk = setclk, - .setmode = setmode, - .data_hold = 1, - .data_setup = 1, - .clock_high = 1, - .mode_hold = 1, - .mode = 1, - .mode_setup = 1, - }, -}; - -static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { - .card = &snd_soc_s3c24xx_uda134x, - .codec_dev = &soc_codec_dev_uda134x, - .codec_data = &s3c24xx_uda134x, -}; - -static int s3c24xx_uda134x_setup_pin(int pin, char *fun) -{ - if (gpio_request(pin, "s3c24xx_uda134x") < 0) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " - "l3 %s pin already in use", fun); - return -EBUSY; - } - gpio_direction_output(pin, 0); - return 0; -} - -static int s3c24xx_uda134x_probe(struct platform_device *pdev) -{ - int ret; - - printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); - - s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; - if (s3c24xx_uda134x_l3_pins == NULL) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " - "unable to find platform data\n"); - return -ENODEV; - } - s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; - s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; - - if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data, - "data") < 0) - return -EBUSY; - if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk, - "clk") < 0) { - gpio_free(s3c24xx_uda134x_l3_pins->l3_data); - return -EBUSY; - } - if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode, - "mode") < 0) { - gpio_free(s3c24xx_uda134x_l3_pins->l3_data); - gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); - return -EBUSY; - } - - s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_uda134x_snd_device) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " - "Unable to register\n"); - return -ENOMEM; - } - - platform_set_drvdata(s3c24xx_uda134x_snd_device, - &s3c24xx_uda134x_snd_devdata); - s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; - ret = platform_device_add(s3c24xx_uda134x_snd_device); - if (ret) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); - platform_device_put(s3c24xx_uda134x_snd_device); - } - - return ret; -} - -static int s3c24xx_uda134x_remove(struct platform_device *pdev) -{ - platform_device_unregister(s3c24xx_uda134x_snd_device); - gpio_free(s3c24xx_uda134x_l3_pins->l3_data); - gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); - gpio_free(s3c24xx_uda134x_l3_pins->l3_mode); - return 0; -} - -static struct platform_driver s3c24xx_uda134x_driver = { - .probe = s3c24xx_uda134x_probe, - .remove = s3c24xx_uda134x_remove, - .driver = { - .name = "s3c24xx_uda134x", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c24xx_uda134x_init(void) -{ - return platform_driver_register(&s3c24xx_uda134x_driver); -} - -static void __exit s3c24xx_uda134x_exit(void) -{ - platform_driver_unregister(&s3c24xx_uda134x_driver); -} - - -module_init(s3c24xx_uda134x_init); -module_exit(s3c24xx_uda134x_exit); - -MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); -MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c deleted file mode 100644 index 06db130..0000000 --- a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c +++ /dev/null @@ -1,209 +0,0 @@ -/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c - * - * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/clk.h> -#include <linux/gpio.h> -#include <linux/io.h> - -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <mach/gpio-bank-c.h> -#include <mach/gpio-bank-h.h> -#include <plat/gpio-cfg.h> - -#include <mach/map.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "regs-i2s-v2.h" -#include "s3c64xx-i2s.h" - -static struct s3c2410_dma_client s3c64xx_dma_client_out = { - .name = "I2Sv4 PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c64xx_dma_client_in = { - .name = "I2Sv4 PCM Stereo in" -}; - -static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out; -static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in; -static struct s3c_i2sv2_info s3c64xx_i2sv4; - -struct snd_soc_dai s3c64xx_i2s_v4_dai; -EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai); - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static int s3c64xx_i2sv4_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - /* configure GPIO for i2s port */ - s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0); - s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1); - s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2); - s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK); - s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI); - - return 0; -} - -static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - struct s3c_dma_params *dma_data; - u32 iismod; - - dev_dbg(cpu_dai->dev, "Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod); - - iismod &= ~S3C64XX_IISMOD_BLC_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C64XX_IISMOD_BLC_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - break; - case SNDRV_PCM_FORMAT_S24_LE: - iismod |= S3C64XX_IISMOD_BLC_24BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = { - .hw_params = s3c_i2sv4_hw_params, -}; - -static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev) -{ - struct s3c_i2sv2_info *i2s; - struct snd_soc_dai *dai; - int ret; - - i2s = &s3c64xx_i2sv4; - dai = &s3c64xx_i2s_v4_dai; - - if (dai->dev) { - dev_dbg(dai->dev, "%s: \ - I2Sv4 instance already registered!\n", __func__); - return -EBUSY; - } - - dai->dev = &pdev->dev; - dai->name = "s3c64xx-i2s-v4"; - dai->id = 0; - dai->symmetric_rates = 1; - dai->playback.channels_min = 2; - dai->playback.channels_max = 2; - dai->playback.rates = S3C64XX_I2S_RATES; - dai->playback.formats = S3C64XX_I2S_FMTS; - dai->capture.channels_min = 2; - dai->capture.channels_max = 2; - dai->capture.rates = S3C64XX_I2S_RATES; - dai->capture.formats = S3C64XX_I2S_FMTS; - dai->probe = s3c64xx_i2sv4_probe; - dai->ops = &s3c64xx_i2sv4_dai_ops; - - i2s->feature |= S3C_FEATURE_CDCLKCON; - - i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in; - i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out; - - i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX; - i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX; - i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD; - - i2s->dma_capture->client = &s3c64xx_dma_client_in; - i2s->dma_capture->dma_size = 4; - i2s->dma_playback->client = &s3c64xx_dma_client_out; - i2s->dma_playback->dma_size = 4; - - i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); - if (IS_ERR(i2s->iis_cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus\n"); - ret = PTR_ERR(i2s->iis_cclk); - goto err; - } - - clk_enable(i2s->iis_cclk); - - ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); - if (ret) - goto err_clk; - - ret = s3c_i2sv2_register_dai(dai); - if (ret != 0) - goto err_i2sv2; - - return 0; - -err_i2sv2: - /* Not implemented for I2Sv2 core yet */ -err_clk: - clk_put(i2s->iis_cclk); -err: - return ret; -} - -static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev) -{ - dev_err(&pdev->dev, "Device removal not yet supported\n"); - return 0; -} - -static struct platform_driver s3c64xx_i2sv4_driver = { - .probe = s3c64xx_i2sv4_dev_probe, - .remove = s3c64xx_i2sv4_dev_remove, - .driver = { - .name = "s3c64xx-iis-v4", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c64xx_i2sv4_init(void) -{ - return platform_driver_register(&s3c64xx_i2sv4_driver); -} -module_init(s3c64xx_i2sv4_init); - -static void __exit s3c64xx_i2sv4_exit(void) -{ - platform_driver_unregister(&s3c64xx_i2sv4_driver); -} -module_exit(s3c64xx_i2sv4_exit); - -/* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); -MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c deleted file mode 100644 index 1d85cb8..0000000 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ /dev/null @@ -1,190 +0,0 @@ -/* sound/soc/s3c24xx/s3c64xx-i2s.c - * - * ALSA SoC Audio Layer - S3C64XX I2S driver - * - * Copyright 2008 Openmoko, Inc. - * Copyright 2008 Simtec Electronics - * Ben Dooks <ben(a)simtec.co.uk> - * http://armlinux.simtec.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/clk.h> -#include <linux/gpio.h> -#include <linux/io.h> - -#include <sound/soc.h> - -#include <mach/gpio-bank-d.h> -#include <mach/gpio-bank-e.h> -#include <plat/gpio-cfg.h> - -#include <mach/map.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "regs-i2s-v2.h" -#include "s3c64xx-i2s.h" - -/* The value should be set to maximum of the total number - * of I2Sv3 controllers that any supported SoC has. - */ -#define MAX_I2SV3 2 - -static struct s3c2410_dma_client s3c64xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c64xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[MAX_I2SV3]; -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[MAX_I2SV3]; -static struct s3c_i2sv2_info s3c64xx_i2s[MAX_I2SV3]; - -struct snd_soc_dai s3c64xx_i2s_dai[MAX_I2SV3]; -EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static int s3c64xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - /* configure GPIO for i2s port */ - switch (dai->id) { - case 0: - s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); - s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); - s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); - break; - case 1: - s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); - s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); - s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); - } - - return 0; -} - - -static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops; - -static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) -{ - struct s3c_i2sv2_info *i2s; - struct snd_soc_dai *dai; - int ret; - - if (pdev->id >= MAX_I2SV3) { - dev_err(&pdev->dev, "id %d out of range\n", pdev->id); - return -EINVAL; - } - - i2s = &s3c64xx_i2s[pdev->id]; - dai = &s3c64xx_i2s_dai[pdev->id]; - dai->dev = &pdev->dev; - dai->name = "s3c64xx-i2s"; - dai->id = pdev->id; - dai->symmetric_rates = 1; - dai->playback.channels_min = 2; - dai->playback.channels_max = 2; - dai->playback.rates = S3C64XX_I2S_RATES; - dai->playback.formats = S3C64XX_I2S_FMTS; - dai->capture.channels_min = 2; - dai->capture.channels_max = 2; - dai->capture.rates = S3C64XX_I2S_RATES; - dai->capture.formats = S3C64XX_I2S_FMTS; - dai->probe = s3c64xx_i2s_probe; - dai->ops = &s3c64xx_i2s_dai_ops; - - i2s->feature |= S3C_FEATURE_CDCLKCON; - - i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; - i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; - - if (pdev->id == 0) { - i2s->dma_capture->channel = DMACH_I2S0_IN; - i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_I2S0_OUT; - i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; - } else { - i2s->dma_capture->channel = DMACH_I2S1_IN; - i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_I2S1_OUT; - i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD; - } - - i2s->dma_capture->client = &s3c64xx_dma_client_in; - i2s->dma_capture->dma_size = 4; - i2s->dma_playback->client = &s3c64xx_dma_client_out; - i2s->dma_playback->dma_size = 4; - - i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); - if (IS_ERR(i2s->iis_cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus\n"); - ret = PTR_ERR(i2s->iis_cclk); - goto err; - } - - clk_enable(i2s->iis_cclk); - - ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); - if (ret) - goto err_clk; - - ret = s3c_i2sv2_register_dai(dai); - if (ret != 0) - goto err_i2sv2; - - return 0; - -err_i2sv2: - /* Not implemented for I2Sv2 core yet */ -err_clk: - clk_put(i2s->iis_cclk); -err: - return ret; -} - -static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev) -{ - dev_err(&pdev->dev, "Device removal not yet supported\n"); - return 0; -} - -static struct platform_driver s3c64xx_iis_driver = { - .probe = s3c64xx_iis_dev_probe, - .remove = s3c64xx_iis_dev_remove, - .driver = { - .name = "s3c64xx-iis", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c64xx_i2s_init(void) -{ - return platform_driver_register(&s3c64xx_iis_driver); -} -module_init(s3c64xx_i2s_init); - -static void __exit s3c64xx_i2s_exit(void) -{ - platform_driver_unregister(&s3c64xx_iis_driver); -} -module_exit(s3c64xx_i2s_exit); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h deleted file mode 100644 index 7a40f43..0000000 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ /dev/null @@ -1,42 +0,0 @@ -/* sound/soc/s3c24xx/s3c64xx-i2s.h - * - * ALSA SoC Audio Layer - S3C64XX I2S driver - * - * Copyright 2008 Openmoko, Inc. - * Copyright 2008 Simtec Electronics - * Ben Dooks <ben(a)simtec.co.uk> - * http://armlinux.simtec.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 __SND_SOC_S3C24XX_S3C64XX_I2S_H -#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ - -struct clk; - -#include "s3c-i2s-v2.h" - -#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK -#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK -#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER - -#define S3C64XX_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK -#define S3C64XX_CLKSRC_MUX S3C_I2SV2_CLKSRC_AUDIOBUS -#define S3C64XX_CLKSRC_CDCLK S3C_I2SV2_CLKSRC_CDCLK - -#define S3C64XX_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -#define S3C64XX_I2S_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) - -extern struct snd_soc_dai s3c64xx_i2s_dai[]; -extern struct snd_soc_dai s3c64xx_i2s_v4_dai; - -#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c deleted file mode 100644 index 3622588..0000000 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ /dev/null @@ -1,81 +0,0 @@ -/* - * smdk2443_wm9710.c -- SoC audio for smdk2443 - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include "../codecs/ac97.h" -#include "s3c-dma.h" -#include "s3c-ac97.h" - -static struct snd_soc_card smdk2443; - -static struct snd_soc_dai_link smdk2443_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], - .codec_dai = &ac97_dai, -}, -}; - -static struct snd_soc_card smdk2443 = { - .name = "SMDK2443", - .platform = &s3c24xx_soc_platform, - .dai_link = smdk2443_dai, - .num_links = ARRAY_SIZE(smdk2443_dai), -}; - -static struct snd_soc_device smdk2443_snd_ac97_devdata = { - .card = &smdk2443, - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct platform_device *smdk2443_snd_ac97_device; - -static int __init smdk2443_init(void) -{ - int ret; - - smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!smdk2443_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(smdk2443_snd_ac97_device, - &smdk2443_snd_ac97_devdata); - smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev; - ret = platform_device_add(smdk2443_snd_ac97_device); - - if (ret) - platform_device_put(smdk2443_snd_ac97_device); - - return ret; -} - -static void __exit smdk2443_exit(void) -{ - platform_device_unregister(smdk2443_snd_ac97_device); -} - -module_init(smdk2443_init); -module_exit(smdk2443_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme.gregory(a)wolfsonmicro.com, www.wolfsonmicro.com"); -MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c deleted file mode 100644 index 07e8e51..0000000 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ /dev/null @@ -1,266 +0,0 @@ -/* - * smdk64xx_wm8580.c - * - * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.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/platform_device.h> -#include <linux/clk.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include "../codecs/wm8580.h" -#include "s3c-dma.h" -#include "s3c64xx-i2s.h" - -/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ -#define SMDK64XX_WM8580_FREQ 12000000 - -static int smdk64xx_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - unsigned int pll_out; - int bfs, rfs, ret; - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_U8: - case SNDRV_PCM_FORMAT_S8: - bfs = 16; - break; - case SNDRV_PCM_FORMAT_U16_LE: - case SNDRV_PCM_FORMAT_S16_LE: - bfs = 32; - break; - default: - return -EINVAL; - } - - /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. - * This criterion can't be met if we request PLL output - * as {8000x256, 64000x256, 11025x256}Hz. - * As a wayout, we rather change rfs to a minimum value that - * results in (params_rate(params) * rfs), and itself, acceptable - * to both - the CODEC and the CPU. - */ - switch (params_rate(params)) { - case 16000: - case 22050: - case 32000: - case 44100: - case 48000: - case 88200: - case 96000: - rfs = 256; - break; - case 64000: - rfs = 384; - break; - case 8000: - case 11025: - rfs = 512; - break; - default: - return -EINVAL; - } - pll_out = params_rate(params) * rfs; - - /* Set the Codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* Set the AP DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* We use PCLK for basic ops in SoC-Slave mode */ - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Set WM8580 to drive MCLK from its PLLA */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_PLLA); - if (ret < 0) - return ret; - - /* Explicitly set WM8580-DAC to source from MCLK */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, - WM8580_CLKSRC_MCLK); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, - SMDK64XX_WM8580_FREQ, pll_out); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); - if (ret < 0) - return ret; - - return 0; -} - -/* - * SMDK64XX WM8580 DAI operations. - */ -static struct snd_soc_ops smdk64xx_ops = { - .hw_params = smdk64xx_hw_params, -}; - -/* SMDK64xx Playback widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { - SND_SOC_DAPM_HP("Front-L/R", NULL), - SND_SOC_DAPM_HP("Center/Sub", NULL), - SND_SOC_DAPM_HP("Rear-L/R", NULL), -}; - -/* SMDK64xx Capture widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { - SND_SOC_DAPM_MIC("MicIn", NULL), - SND_SOC_DAPM_LINE("LineIn", NULL), -}; - -/* SMDK-PAIFTX connections */ -static const struct snd_soc_dapm_route audio_map_tx[] = { - /* MicIn feeds AINL */ - {"AINL", NULL, "MicIn"}, - - /* LineIn feeds AINL/R */ - {"AINL", NULL, "LineIn"}, - {"AINR", NULL, "LineIn"}, -}; - -/* SMDK-PAIFRX connections */ -static const struct snd_soc_dapm_route audio_map_rx[] = { - /* Front Left/Right are fed VOUT1L/R */ - {"Front-L/R", NULL, "VOUT1L"}, - {"Front-L/R", NULL, "VOUT1R"}, - - /* Center/Sub are fed VOUT2L/R */ - {"Center/Sub", NULL, "VOUT2L"}, - {"Center/Sub", NULL, "VOUT2R"}, - - /* Rear Left/Right are fed VOUT3L/R */ - {"Rear-L/R", NULL, "VOUT3L"}, - {"Rear-L/R", NULL, "VOUT3R"}, -}; - -static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) -{ - /* Add smdk64xx specific Capture widgets */ - snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, - ARRAY_SIZE(wm8580_dapm_widgets_cpt)); - - /* Set up PAIFTX audio path */ - snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); - - /* Enabling the microphone requires the fitting of a 0R - * resistor to connect the line from the microphone jack. - */ - snd_soc_dapm_disable_pin(codec, "MicIn"); - - /* signal a DAPM event */ - snd_soc_dapm_sync(codec); - - return 0; -} - -static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) -{ - /* Add smdk64xx specific Playback widgets */ - snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, - ARRAY_SIZE(wm8580_dapm_widgets_pbk)); - - /* Set up PAIFRX audio path */ - snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); - - /* signal a DAPM event */ - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link smdk64xx_dai[] = { -{ /* Primary Playback i/f */ - .name = "WM8580 PAIF RX", - .stream_name = "Playback", - .cpu_dai = &s3c64xx_i2s_v4_dai, - .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], - .init = smdk64xx_wm8580_init_paifrx, - .ops = &smdk64xx_ops, -}, -{ /* Primary Capture i/f */ - .name = "WM8580 PAIF TX", - .stream_name = "Capture", - .cpu_dai = &s3c64xx_i2s_v4_dai, - .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], - .init = smdk64xx_wm8580_init_paiftx, - .ops = &smdk64xx_ops, -}, -}; - -static struct snd_soc_card smdk64xx = { - .name = "smdk64xx", - .platform = &s3c24xx_soc_platform, - .dai_link = smdk64xx_dai, - .num_links = ARRAY_SIZE(smdk64xx_dai), -}; - -static struct snd_soc_device smdk64xx_snd_devdata = { - .card = &smdk64xx, - .codec_dev = &soc_codec_dev_wm8580, -}; - -static struct platform_device *smdk64xx_snd_device; - -static int __init smdk64xx_audio_init(void) -{ - int ret; - - smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!smdk64xx_snd_device) - return -ENOMEM; - - platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); - smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; - ret = platform_device_add(smdk64xx_snd_device); - - if (ret) - platform_device_put(smdk64xx_snd_device); - - return ret; -} -module_init(smdk64xx_audio_init); - -MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); -MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c deleted file mode 100644 index 24fd39f..0000000 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ /dev/null @@ -1,94 +0,0 @@ -/* - * smdk_wm9713.c -- SoC audio for SMDK - * - * Copyright 2010 Samsung Electronics Co. Ltd. - * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.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/device.h> -#include <sound/soc.h> - -#include "../codecs/wm9713.h" -#include "s3c-dma.h" -#include "s3c-ac97.h" - -static struct snd_soc_card smdk; - -/* - * Default CFG switch settings to use this driver: - * - * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off - */ - -/* - Playback (HeadPhone):- - $ amixer sset 'Headphone' unmute - $ amixer sset 'Right Headphone Out Mux' 'Headphone' - $ amixer sset 'Left Headphone Out Mux' 'Headphone' - $ amixer sset 'Right HP Mixer PCM' unmute - $ amixer sset 'Left HP Mixer PCM' unmute - - Capture (LineIn):- - $ amixer sset 'Right Capture Source' 'Line' - $ amixer sset 'Left Capture Source' 'Line' -*/ - -static struct snd_soc_dai_link smdk_dai = { - .name = "AC97", - .stream_name = "AC97 PCM", - .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], - .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], -}; - -static struct snd_soc_card smdk = { - .name = "SMDK", - .platform = &s3c24xx_soc_platform, - .dai_link = &smdk_dai, - .num_links = 1, -}; - -static struct snd_soc_device smdk_snd_ac97_devdata = { - .card = &smdk, - .codec_dev = &soc_codec_dev_wm9713, -}; - -static struct platform_device *smdk_snd_ac97_device; - -static int __init smdk_init(void) -{ - int ret; - - smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!smdk_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(smdk_snd_ac97_device, - &smdk_snd_ac97_devdata); - smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; - - ret = platform_device_add(smdk_snd_ac97_device); - if (ret) - platform_device_put(smdk_snd_ac97_device); - - return ret; -} - -static void __exit smdk_exit(void) -{ - platform_device_unregister(smdk_snd_ac97_device); -} - -module_init(smdk_init); -module_exit(smdk_exit); - -/* Module information */ -MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); -MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig new file mode 100644 index 0000000..2a7cc22 --- /dev/null +++ b/sound/soc/samsung/Kconfig @@ -0,0 +1,127 @@ +config SND_S3C24XX_SOC + tristate "SoC Audio for the Samsung S3CXXXX chips" + depends on ARCH_S3C2410 || ARCH_S3C64XX + select S3C64XX_DMA if ARCH_S3C64XX + help + Say Y or M if you want to add support for codecs attached to + the S3C24XX AC97 or I2S interfaces. You will also need to + select the audio interfaces to support below. + +config SND_S3C24XX_SOC_I2S + tristate + select S3C2410_DMA + +config SND_S3C_I2SV2_SOC + tristate + +config SND_S3C2412_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + select S3C2410_DMA + +config SND_S3C64XX_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + +config SND_S3C64XX_SOC_I2S_V4 + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + +config SND_S3C_SOC_PCM + tristate + +config SND_S3C_SOC_AC97 + tristate + select SND_SOC_AC97_BUS + +config SND_S3C24XX_SOC_NEO1973_WM8753 + tristate "SoC I2S Audio support for NEO1973 - WM8753" + depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 + select SND_S3C24XX_SOC_I2S + select SND_SOC_WM8753 + help + Say Y if you want to add support for SoC audio on smdk2440 + with the WM8753. + +config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753 + tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" + depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02 + select SND_S3C24XX_SOC_I2S + select SND_SOC_WM8753 + help + This driver provides audio support for the Openmoko Neo FreeRunner + smartphone. + +config SND_S3C24XX_SOC_JIVE_WM8750 + tristate "SoC I2S Audio support for Jive" + depends on SND_S3C24XX_SOC && MACH_JIVE + select SND_SOC_WM8750 + select SND_S3C2412_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the Jive. + +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S_V4 + help + Say Y if you want to add support for SoC audio on the SMDK6410. + +config SND_S3C24XX_SOC_SMDK2443_WM9710 + tristate "SoC AC97 Audio support for SMDK2443 - WM9710" + depends on SND_S3C24XX_SOC && MACH_SMDK2443 + select S3C2410_DMA + select AC97_BUS + select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 + help + Say Y if you want to add support for SoC audio on smdk2443 + with the WM9710. + +config SND_S3C24XX_SOC_LN2440SBC_ALC650 + tristate "SoC AC97 Audio support for LN2440SBC - ALC650" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select S3C2410_DMA + select AC97_BUS + select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 + help + Say Y if you want to add support for SoC audio on ln2440sbc + with the ALC650. + +config SND_S3C24XX_SOC_S3C24XX_UDA134X + tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_L3 + select SND_SOC_UDA134X + +config SND_S3C24XX_SOC_SIMTEC + tristate + help + Internal node for common S3C24XX/Simtec suppor + +config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 + tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC23 + select SND_S3C24XX_SOC_SIMTEC + +config SND_S3C24XX_SOC_SIMTEC_HERMES + tristate "SoC I2S Audio support for Simtec Hermes board" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC3X + select SND_S3C24XX_SOC_SIMTEC + +config SND_SOC_SMDK_WM9713 + tristate "SoC AC97 Audio support for SMDK with WM9713" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM9713 + select SND_S3C_SOC_AC97 + help + Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile new file mode 100644 index 0000000..81d8dc5 --- /dev/null +++ b/sound/soc/samsung/Makefile @@ -0,0 +1,43 @@ +# S3c24XX Platform Support +snd-soc-s3c24xx-objs := s3c-dma.o +snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o +snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o +snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o +snd-soc-s3c-ac97-objs := s3c-ac97.o +snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o +snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o +snd-soc-s3c-pcm-objs := s3c-pcm.o + +obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o +obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o +obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o +obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o +obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o +obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o + +# S3C24XX Machine Support +snd-soc-jive-wm8750-objs := jive_wm8750.o +snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o +snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o +snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o +snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o +snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o +snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o +snd-soc-smdk-wm9713-objs := smdk_wm9713.o + +obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o +obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o +obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o +obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o +obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o +obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c new file mode 100644 index 0000000..8c108b1 --- /dev/null +++ b/sound/soc/samsung/jive_wm8750.c @@ -0,0 +1,196 @@ +/* sound/soc/s3c24xx/jive_wm8750.c + * + * Copyright 2007,2008 Simtec Electronics + * + * Based on sound/soc/pxa/spitz.c + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c2412-i2s.h" + +#include "../codecs/wm8750.h" + +static const struct snd_soc_dapm_route audio_map[] = { + { "Headphone Jack", NULL, "LOUT1" }, + { "Headphone Jack", NULL, "ROUT1" }, + { "Internal Speaker", NULL, "LOUT2" }, + { "Internal Speaker", NULL, "ROUT2" }, + { "LINPUT1", NULL, "Line Input" }, + { "RINPUT1", NULL, "Line Input" }, +}; + +static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Internal Speaker", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static int jive_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c_i2sv2_get_clock(cpu_dai)); + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops jive_ops = { + .hw_params = jive_hw_params, +}; + +static int jive_wm8750_init(struct snd_soc_codec *codec) +{ + int err; + + /* These endpoints are not being used. */ + snd_soc_dapm_nc_pin(codec, "LINPUT2"); + snd_soc_dapm_nc_pin(codec, "RINPUT2"); + snd_soc_dapm_nc_pin(codec, "LINPUT3"); + snd_soc_dapm_nc_pin(codec, "RINPUT3"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONO"); + + /* Add jive specific widgets */ + err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); + if (err) { + printk(KERN_ERR "%s: failed to add widgets (%d)\n", + __func__, err); + return err; + } + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link jive_dai = { + .name = "wm8750", + .stream_name = "WM8750", + .cpu_dai = &s3c2412_i2s_dai, + .codec_dai = &wm8750_dai, + .init = jive_wm8750_init, + .ops = &jive_ops, +}; + +/* jive audio machine driver */ +static struct snd_soc_card snd_soc_machine_jive = { + .name = "Jive", + .platform = &s3c24xx_soc_platform, + .dai_link = &jive_dai, + .num_links = 1, +}; + +/* jive audio subsystem */ +static struct snd_soc_device jive_snd_devdata = { + .card = &snd_soc_machine_jive, + .codec_dev = &soc_codec_dev_wm8750, +}; + +static struct platform_device *jive_snd_device; + +static int __init jive_init(void) +{ + int ret; + + if (!machine_is_jive()) + return 0; + + printk("JIVE WM8750 Audio support\n"); + + jive_snd_device = platform_device_alloc("soc-audio", -1); + if (!jive_snd_device) + return -ENOMEM; + + platform_set_drvdata(jive_snd_device, &jive_snd_devdata); + jive_snd_devdata.dev = &jive_snd_device->dev; + ret = platform_device_add(jive_snd_device); + + if (ret) + platform_device_put(jive_snd_device); + + return ret; +} + +static void __exit jive_exit(void) +{ + platform_device_unregister(jive_snd_device); +} + +module_init(jive_init); +module_exit(jive_exit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/lm4857.h b/sound/soc/samsung/lm4857.h new file mode 100644 index 0000000..0cf5b70 --- /dev/null +++ b/sound/soc/samsung/lm4857.h @@ -0,0 +1,32 @@ +/* + * lm4857.h -- ALSA Soc Audio Layer + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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. + * + * Revision history + * 18th Jun 2007 Initial version. + */ + +#ifndef LM4857_H_ +#define LM4857_H_ + +/* The register offsets in the cache array */ +#define LM4857_MVOL 0 +#define LM4857_LVOL 1 +#define LM4857_RVOL 2 +#define LM4857_CTRL 3 + +/* the shifts required to set these bits */ +#define LM4857_3D 5 +#define LM4857_WAKEUP 5 +#define LM4857_EPGAIN 4 + +#endif /*LM4857_H_*/ + diff --git a/sound/soc/samsung/ln2440sbc_alc650.c b/sound/soc/samsung/ln2440sbc_alc650.c new file mode 100644 index 0000000..ffa954f --- /dev/null +++ b/sound/soc/samsung/ln2440sbc_alc650.c @@ -0,0 +1,85 @@ +/* + * SoC audio for ln2440sbc + * + * Copyright 2007 KonekTel, a.s. + * Author: Ivan Kuten + * ivan.kuten(a)promwad.com + * + * Heavily based on smdk2443_wm9710.c + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card ln2440sbc; + +static struct snd_soc_dai_link ln2440sbc_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_card ln2440sbc = { + .name = "LN2440SBC", + .platform = &s3c24xx_soc_platform, + .dai_link = ln2440sbc_dai, + .num_links = ARRAY_SIZE(ln2440sbc_dai), +}; + +static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { + .card = &ln2440sbc, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *ln2440sbc_snd_ac97_device; + +static int __init ln2440sbc_init(void) +{ + int ret; + + ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!ln2440sbc_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(ln2440sbc_snd_ac97_device, + &ln2440sbc_snd_ac97_devdata); + ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; + ret = platform_device_add(ln2440sbc_snd_ac97_device); + + if (ret) + platform_device_put(ln2440sbc_snd_ac97_device); + + return ret; +} + +static void __exit ln2440sbc_exit(void) +{ + platform_device_unregister(ln2440sbc_snd_ac97_device); +} + +module_init(ln2440sbc_init); +module_exit(ln2440sbc_exit); + +/* Module information */ +MODULE_AUTHOR("Ivan Kuten"); +MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c new file mode 100644 index 0000000..209c259 --- /dev/null +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -0,0 +1,506 @@ +/* + * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02) + * + * Copyright 2007 Openmoko Inc + * Author: Graeme Gregory <graeme(a)openmoko.org> + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory <linux(a)wolfsonmicro.com> + * Copyright 2009 Wolfson Microelectronics + * + * 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/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include <plat/regs-iis.h> + +#include <mach/regs-clock.h> +#include <asm/io.h> +#include <mach/gta02.h> +#include "../codecs/wm8753.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +static struct snd_soc_card neo1973_gta02; + +static int neo1973_gta02_hifi_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + switch (params_rate(params)) { + case 8000: + case 16000: + pll_out = 12288000; + break; + case 48000: + bclk = WM8753_BCLK_DIV_4; + pll_out = 12288000; + break; + case 96000: + bclk = WM8753_BCLK_DIV_2; + pll_out = 12288000; + break; + case 11025: + bclk = WM8753_BCLK_DIV_16; + pll_out = 11289600; + break; + case 22050: + bclk = WM8753_BCLK_DIV_8; + pll_out = 11289600; + break; + case 44100: + bclk = WM8753_BCLK_DIV_4; + pll_out = 11289600; + break; + case 88200: + bclk = WM8753_BCLK_DIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set codec BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, + WM8753_BCLKDIV, bclk); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(4, 4)); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, + iis_clkrate / 4, pll_out); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); +} + +/* + * Neo1973 WM8753 HiFi DAI opserations. + */ +static struct snd_soc_ops neo1973_gta02_hifi_ops = { + .hw_params = neo1973_gta02_hifi_hw_params, + .hw_free = neo1973_gta02_hifi_hw_free, +}; + +static int neo1973_gta02_voice_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->dai->codec_dai; + unsigned int pcmdiv = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + if (params_rate(params) != 8000) + return -EINVAL; + if (params_channels(params) != 1) + return -EINVAL; + + pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ + + /* todo: gg check mode (DSP_B) against CSR datasheet */ + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, + 12288000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec PCM division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, + pcmdiv); + if (ret < 0) + return ret; + + /* configue and enable PLL for 12.288MHz output */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, + iis_clkrate / 4, 12288000); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); +} + +static struct snd_soc_ops neo1973_gta02_voice_ops = { + .hw_params = neo1973_gta02_voice_hw_params, + .hw_free = neo1973_gta02_voice_hw_free, +}; + +#define LM4853_AMP 1 +#define LM4853_SPK 2 + +static u8 lm4853_state; + +/* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ +static int lm4853_set_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = ucontrol->value.integer.value[0]; + + if (val) + lm4853_state |= LM4853_AMP; + else + lm4853_state &= ~LM4853_AMP; + + return 0; +} + +static int lm4853_get_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP; + + return 0; +} + +static int lm4853_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = ucontrol->value.integer.value[0]; + + if (val) { + lm4853_state |= LM4853_SPK; + gpio_set_value(GTA02_GPIO_HP_IN, 0); + } else { + lm4853_state &= ~LM4853_SPK; + gpio_set_value(GTA02_GPIO_HP_IN, 1); + } + + return 0; +} + +static int lm4853_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1; + + return 0; +} + +static int lm4853_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Handset Mic", NULL), + SND_SOC_DAPM_SPK("Handset Spk", NULL), +}; + + +/* example machine audio_mapnections */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Connections to the lm4853 amp */ + {"Stereo Out", NULL, "LOUT1"}, + {"Stereo Out", NULL, "ROUT1"}, + + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Handset Mic"}, + + /* Call Speaker */ + {"Handset Spk", NULL, "LOUT2"}, + {"Handset Spk", NULL, "ROUT2"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { + SOC_DAPM_PIN_SWITCH("Stereo Out"), + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Spk"), + + /* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ + SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0, + lm4853_get_state, + lm4853_set_state), + SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0, + lm4853_get_spk, + lm4853_set_spk), +}; + +/* + * This is an example machine initialisation for a wm8753 connected to a + * neo1973 GTA02. + */ +static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) +{ + int err; + + /* set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); + + /* Add neo1973 gta02 specific widgets */ + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); + + /* add neo1973 gta02 specific controls */ + err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls, + ARRAY_SIZE(wm8753_neo1973_gta02_controls)); + + if (err < 0) + return err; + + /* set up neo1973 gta02 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to default off mode */ + snd_soc_dapm_disable_pin(codec, "Stereo Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Spk"); + + /* allow audio paths from the GSM modem to run during suspend */ + snd_soc_dapm_ignore_suspend(codec, "Stereo Out"); + snd_soc_dapm_ignore_suspend(codec, "GSM Line Out"); + snd_soc_dapm_ignore_suspend(codec, "GSM Line In"); + snd_soc_dapm_ignore_suspend(codec, "Headset Mic"); + snd_soc_dapm_ignore_suspend(codec, "Handset Mic"); + snd_soc_dapm_ignore_suspend(codec, "Handset Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* + * BT Codec DAI + */ +static struct snd_soc_dai bt_dai = { + .name = "Bluetooth", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_dai_link neo1973_gta02_dai[] = { +{ /* Hifi Playback - for similatious use with voice below */ + .name = "WM8753", + .stream_name = "WM8753 HiFi", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], + .init = neo1973_gta02_wm8753_init, + .ops = &neo1973_gta02_hifi_ops, +}, +{ /* Voice via BT */ + .name = "Bluetooth", + .stream_name = "Voice", + .cpu_dai = &bt_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], + .ops = &neo1973_gta02_voice_ops, +}, +}; + +static struct snd_soc_card neo1973_gta02 = { + .name = "neo1973-gta02", + .platform = &s3c24xx_soc_platform, + .dai_link = neo1973_gta02_dai, + .num_links = ARRAY_SIZE(neo1973_gta02_dai), +}; + +static struct snd_soc_device neo1973_gta02_snd_devdata = { + .card = &neo1973_gta02, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static struct platform_device *neo1973_gta02_snd_device; + +static int __init neo1973_gta02_init(void) +{ + int ret; + + if (!machine_is_neo1973_gta02()) { + printk(KERN_INFO + "Only GTA02 is supported by this ASoC driver\n"); + return -ENODEV; + } + + /* register bluetooth DAI here */ + ret = snd_soc_register_dai(&bt_dai); + if (ret) + return ret; + + neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1); + if (!neo1973_gta02_snd_device) + return -ENOMEM; + + platform_set_drvdata(neo1973_gta02_snd_device, + &neo1973_gta02_snd_devdata); + neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev; + ret = platform_device_add(neo1973_gta02_snd_device); + + if (ret) { + platform_device_put(neo1973_gta02_snd_device); + return ret; + } + + /* Initialise GPIOs used by amp */ + ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_unregister_device; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_free_gpio_hp_in; + } + + ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_hp_in; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_amp_shut; + } + + return 0; + +err_free_gpio_amp_shut: + gpio_free(GTA02_GPIO_AMP_SHUT); +err_free_gpio_hp_in: + gpio_free(GTA02_GPIO_HP_IN); +err_unregister_device: + platform_device_unregister(neo1973_gta02_snd_device); + return ret; +} +module_init(neo1973_gta02_init); + +static void __exit neo1973_gta02_exit(void) +{ + snd_soc_unregister_dai(&bt_dai); + platform_device_unregister(neo1973_gta02_snd_device); + gpio_free(GTA02_GPIO_HP_IN); + gpio_free(GTA02_GPIO_AMP_SHUT); +} +module_exit(neo1973_gta02_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org"); +MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c new file mode 100644 index 0000000..0cb4f86 --- /dev/null +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -0,0 +1,707 @@ +/* + * neo1973_wm8753.c -- SoC audio for Neo1973 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include <asm/mach-types.h> +#include <asm/hardware/scoop.h> +#include <mach/regs-clock.h> +#include <mach/regs-gpio.h> +#include <mach/hardware.h> +#include <linux/io.h> +#include <mach/spi-gpio.h> + +#include <plat/regs-iis.h> + +#include "../codecs/wm8753.h" +#include "lm4857.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +/* define the scenarios */ +#define NEO_AUDIO_OFF 0 +#define NEO_GSM_CALL_AUDIO_HANDSET 1 +#define NEO_GSM_CALL_AUDIO_HEADSET 2 +#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 +#define NEO_STEREO_TO_SPEAKERS 4 +#define NEO_STEREO_TO_HEADPHONES 5 +#define NEO_CAPTURE_HANDSET 6 +#define NEO_CAPTURE_HEADSET 7 +#define NEO_CAPTURE_BLUETOOTH 8 + +static struct snd_soc_card neo1973; +static struct i2c_client *i2c; + +static int neo1973_hifi_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0; + int ret = 0; + unsigned long iis_clkrate; + + pr_debug("Entered %s\n", __func__); + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + switch (params_rate(params)) { + case 8000: + case 16000: + pll_out = 12288000; + break; + case 48000: + bclk = WM8753_BCLK_DIV_4; + pll_out = 12288000; + break; + case 96000: + bclk = WM8753_BCLK_DIV_2; + pll_out = 12288000; + break; + case 11025: + bclk = WM8753_BCLK_DIV_16; + pll_out = 11289600; + break; + case 22050: + bclk = WM8753_BCLK_DIV_8; + pll_out = 11289600; + break; + case 44100: + bclk = WM8753_BCLK_DIV_4; + pll_out = 11289600; + break; + case 88200: + bclk = WM8753_BCLK_DIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set codec BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(4, 4)); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, + iis_clkrate / 4, pll_out); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + pr_debug("Entered %s\n", __func__); + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); +} + +/* + * Neo1973 WM8753 HiFi DAI opserations. + */ +static struct snd_soc_ops neo1973_hifi_ops = { + .hw_params = neo1973_hifi_hw_params, + .hw_free = neo1973_hifi_hw_free, +}; + +static int neo1973_voice_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->dai->codec_dai; + unsigned int pcmdiv = 0; + int ret = 0; + unsigned long iis_clkrate; + + pr_debug("Entered %s\n", __func__); + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + if (params_rate(params) != 8000) + return -EINVAL; + if (params_channels(params) != 1) + return -EINVAL; + + pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ + + /* todo: gg check mode (DSP_B) against CSR datasheet */ + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec PCM division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + if (ret < 0) + return ret; + + /* configue and enable PLL for 12.288MHz output */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, + iis_clkrate / 4, 12288000); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + pr_debug("Entered %s\n", __func__); + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); +} + +static struct snd_soc_ops neo1973_voice_ops = { + .hw_params = neo1973_voice_hw_params, + .hw_free = neo1973_voice_hw_free, +}; + +static int neo1973_scenario; + +static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = neo1973_scenario; + return 0; +} + +static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) +{ + pr_debug("Entered %s\n", __func__); + + switch (neo1973_scenario) { + case NEO_AUDIO_OFF: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_GSM_CALL_AUDIO_HANDSET: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case NEO_GSM_CALL_AUDIO_HEADSET: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_GSM_CALL_AUDIO_BLUETOOTH: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_STEREO_TO_SPEAKERS: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_STEREO_TO_HEADPHONES: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_CAPTURE_HANDSET: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case NEO_CAPTURE_HEADSET: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_CAPTURE_BLUETOOTH: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + default: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + } + + snd_soc_dapm_sync(codec); + + return 0; +} + +static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + pr_debug("Entered %s\n", __func__); + + if (neo1973_scenario == ucontrol->value.integer.value[0]) + return 0; + + neo1973_scenario = ucontrol->value.integer.value[0]; + set_scenario_endpoints(codec, neo1973_scenario); + return 1; +} + +static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; + +static void lm4857_write_regs(void) +{ + pr_debug("Entered %s\n", __func__); + + if (i2c_master_send(i2c, lm4857_regs, 4) != 4) + printk(KERN_ERR "lm4857: i2c write failed\n"); +} + +static int lm4857_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + pr_debug("Entered %s\n", __func__); + + ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; + return 0; +} + +static int lm4857_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + if (((lm4857_regs[reg] >> shift) & mask) == + ucontrol->value.integer.value[0]) + return 0; + + lm4857_regs[reg] &= ~(mask << shift); + lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; + lm4857_write_regs(); + return 1; +} + +static int lm4857_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; + + pr_debug("Entered %s\n", __func__); + + if (value) + value -= 5; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int lm4857_set_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + pr_debug("Entered %s\n", __func__); + + if (value) + value += 5; + + if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) + return 0; + + lm4857_regs[LM4857_CTRL] &= 0xF0; + lm4857_regs[LM4857_CTRL] |= value; + lm4857_write_regs(); + return 1; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Audio Out", NULL), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Call Mic", NULL), +}; + + +static const struct snd_soc_dapm_route dapm_routes[] = { + + /* Connections to the lm4857 amp */ + {"Audio Out", NULL, "LOUT1"}, + {"Audio Out", NULL, "ROUT1"}, + + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Call Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const char *lm4857_mode[] = { + "Off", + "Call Speaker", + "Stereo Speakers", + "Stereo Speakers + Headphones", + "Headphones" +}; + +static const struct soc_enum lm4857_mode_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), +}; + +static const char *neo_scenarios[] = { + "Off", + "GSM Handset", + "GSM Headset", + "GSM Bluetooth", + "Speakers", + "Headphones", + "Capture Handset", + "Capture Headset", + "Capture Bluetooth" +}; + +static const struct soc_enum neo_scenario_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), +}; + +static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + +static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, mono_tlv), + SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], + lm4857_get_mode, lm4857_set_mode), + SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], + neo1973_get_scenario, neo1973_set_scenario), + SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, + lm4857_get_reg, lm4857_set_reg), +}; + +/* + * This is an example machine initialisation for a wm8753 connected to a + * neo1973 II. It is missing logic to detect hp/mic insertions and logic + * to re-route the audio in such an event. + */ +static int neo1973_wm8753_init(struct snd_soc_codec *codec) +{ + int err; + + pr_debug("Entered %s\n", __func__); + + /* set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "LOUT2"); + snd_soc_dapm_nc_pin(codec, "ROUT2"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); + + /* Add neo1973 specific widgets */ + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); + + /* set endpoints to default mode */ + set_scenario_endpoints(codec, NEO_AUDIO_OFF); + + /* add neo1973 specific controls */ + err = snd_soc_add_controls(codec, wm8753_neo1973_controls, + ARRAY_SIZE(8753_neo1973_controls)); + if (err < 0) + return err; + + /* set up neo1973 specific audio routes */ + err = snd_soc_dapm_add_routes(codec, dapm_routes, + ARRAY_SIZE(dapm_routes)); + + snd_soc_dapm_sync(codec); + return 0; +} + +/* + * BT Codec DAI + */ +static struct snd_soc_dai bt_dai = { + .name = "Bluetooth", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_dai_link neo1973_dai[] = { +{ /* Hifi Playback - for similatious use with voice below */ + .name = "WM8753", + .stream_name = "WM8753 HiFi", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], + .init = neo1973_wm8753_init, + .ops = &neo1973_hifi_ops, +}, +{ /* Voice via BT */ + .name = "Bluetooth", + .stream_name = "Voice", + .cpu_dai = &bt_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], + .ops = &neo1973_voice_ops, +}, +}; + +static struct snd_soc_card neo1973 = { + .name = "neo1973", + .platform = &s3c24xx_soc_platform, + .dai_link = neo1973_dai, + .num_links = ARRAY_SIZE(neo1973_dai), +}; + +static struct snd_soc_device neo1973_snd_devdata = { + .card = &neo1973, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static int lm4857_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + pr_debug("Entered %s\n", __func__); + + i2c = client; + + lm4857_write_regs(); + return 0; +} + +static int lm4857_i2c_remove(struct i2c_client *client) +{ + pr_debug("Entered %s\n", __func__); + + i2c = NULL; + + return 0; +} + +static u8 lm4857_state; + +static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) +{ + pr_debug("Entered %s\n", __func__); + + dev_dbg(&dev->dev, "lm4857_suspend\n"); + lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; + if (lm4857_state) { + lm4857_regs[LM4857_CTRL] &= 0xf0; + lm4857_write_regs(); + } + return 0; +} + +static int lm4857_resume(struct i2c_client *dev) +{ + pr_debug("Entered %s\n", __func__); + + if (lm4857_state) { + lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); + lm4857_write_regs(); + } + return 0; +} + +static void lm4857_shutdown(struct i2c_client *dev) +{ + pr_debug("Entered %s\n", __func__); + + dev_dbg(&dev->dev, "lm4857_shutdown\n"); + lm4857_regs[LM4857_CTRL] &= 0xf0; + lm4857_write_regs(); +} + +static const struct i2c_device_id lm4857_i2c_id[] = { + { "neo1973_lm4857", 0 }, + { } +}; + +static struct i2c_driver lm4857_i2c_driver = { + .driver = { + .name = "LM4857 I2C Amp", + .owner = THIS_MODULE, + }, + .suspend = lm4857_suspend, + .resume = lm4857_resume, + .shutdown = lm4857_shutdown, + .probe = lm4857_i2c_probe, + .remove = lm4857_i2c_remove, + .id_table = lm4857_i2c_id, +}; + +static struct platform_device *neo1973_snd_device; + +static int __init neo1973_init(void) +{ + int ret; + + pr_debug("Entered %s\n", __func__); + + if (!machine_is_neo1973_gta01()) { + printk(KERN_INFO + "Only GTA01 hardware supported by ASoC driver\n"); + return -ENODEV; + } + + neo1973_snd_device = platform_device_alloc("soc-audio", -1); + if (!neo1973_snd_device) + return -ENOMEM; + + platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata); + neo1973_snd_devdata.dev = &neo1973_snd_device->dev; + ret = platform_device_add(neo1973_snd_device); + + if (ret) { + platform_device_put(neo1973_snd_device); + return ret; + } + + ret = i2c_add_driver(&lm4857_i2c_driver); + + if (ret != 0) + platform_device_unregister(neo1973_snd_device); + + return ret; +} + +static void __exit neo1973_exit(void) +{ + pr_debug("Entered %s\n", __func__); + + i2c_del_driver(&lm4857_i2c_driver); + platform_device_unregister(neo1973_snd_device); +} + +module_init(neo1973_init); +module_exit(neo1973_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); +MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/regs-i2s-v2.h b/sound/soc/samsung/regs-i2s-v2.h new file mode 100644 index 0000000..5e5e568 --- /dev/null +++ b/sound/soc/samsung/regs-i2s-v2.h @@ -0,0 +1,115 @@ +/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h + * + * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> + * http://armlinux.simtec.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. + * + * S3C2412 IIS register definition +*/ + +#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H +#define __ASM_ARCH_REGS_S3C2412_IIS_H + +#define S3C2412_IISCON (0x00) +#define S3C2412_IISMOD (0x04) +#define S3C2412_IISFIC (0x08) +#define S3C2412_IISPSR (0x0C) +#define S3C2412_IISTXD (0x10) +#define S3C2412_IISRXD (0x14) + +#define S5PC1XX_IISFICS 0x18 +#define S5PC1XX_IISTXDS 0x1C + +#define S5PC1XX_IISCON_SW_RST (1 << 31) +#define S5PC1XX_IISCON_FRXOFSTATUS (1 << 26) +#define S5PC1XX_IISCON_FRXORINTEN (1 << 25) +#define S5PC1XX_IISCON_FTXSURSTAT (1 << 24) +#define S5PC1XX_IISCON_FTXSURINTEN (1 << 23) +#define S5PC1XX_IISCON_TXSDMAPAUSE (1 << 20) +#define S5PC1XX_IISCON_TXSDMACTIVE (1 << 18) + +#define S3C64XX_IISCON_FTXURSTATUS (1 << 17) +#define S3C64XX_IISCON_FTXURINTEN (1 << 16) +#define S3C64XX_IISCON_TXFIFO2_EMPTY (1 << 15) +#define S3C64XX_IISCON_TXFIFO1_EMPTY (1 << 14) +#define S3C64XX_IISCON_TXFIFO2_FULL (1 << 13) +#define S3C64XX_IISCON_TXFIFO1_FULL (1 << 12) + +#define S3C2412_IISCON_LRINDEX (1 << 11) +#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) +#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) +#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) +#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) +#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) +#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) +#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) +#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) +#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) +#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) +#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) + +#define S5PC1XX_IISMOD_OPCLK_CDCLK_OUT (0 << 30) +#define S5PC1XX_IISMOD_OPCLK_CDCLK_IN (1 << 30) +#define S5PC1XX_IISMOD_OPCLK_BCLK_OUT (2 << 30) +#define S5PC1XX_IISMOD_OPCLK_PCLK (3 << 30) +#define S5PC1XX_IISMOD_OPCLK_MASK (3 << 30) +#define S5PC1XX_IISMOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ +#define S5PC1XX_IISMOD_BLCS_MASK 0x3 +#define S5PC1XX_IISMOD_BLCS_SHIFT 26 +#define S5PC1XX_IISMOD_BLCP_MASK 0x3 +#define S5PC1XX_IISMOD_BLCP_SHIFT 24 + +#define S3C64XX_IISMOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ +#define S3C64XX_IISMOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ +#define S3C64XX_IISMOD_C1DD_HHALF (1 << 19) +#define S3C64XX_IISMOD_C1DD_LHALF (1 << 18) +#define S3C64XX_IISMOD_DC2_EN (1 << 17) +#define S3C64XX_IISMOD_DC1_EN (1 << 16) +#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) +#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) +#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) +#define S3C64XX_IISMOD_BLC_MASK (3 << 13) + +#define S3C2412_IISMOD_IMS_SYSMUX (1 << 10) +#define S3C2412_IISMOD_SLAVE (1 << 11) +#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) +#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) +#define S3C2412_IISMOD_MODE_TXRX (2 << 8) +#define S3C2412_IISMOD_MODE_MASK (3 << 8) +#define S3C2412_IISMOD_LR_LLOW (0 << 7) +#define S3C2412_IISMOD_LR_RLOW (1 << 7) +#define S3C2412_IISMOD_SDF_IIS (0 << 5) +#define S3C2412_IISMOD_SDF_MSB (1 << 5) +#define S3C2412_IISMOD_SDF_LSB (2 << 5) +#define S3C2412_IISMOD_SDF_MASK (3 << 5) +#define S3C2412_IISMOD_RCLK_256FS (0 << 3) +#define S3C2412_IISMOD_RCLK_512FS (1 << 3) +#define S3C2412_IISMOD_RCLK_384FS (2 << 3) +#define S3C2412_IISMOD_RCLK_768FS (3 << 3) +#define S3C2412_IISMOD_RCLK_MASK (3 << 3) +#define S3C2412_IISMOD_BCLK_32FS (0 << 1) +#define S3C2412_IISMOD_BCLK_48FS (1 << 1) +#define S3C2412_IISMOD_BCLK_16FS (2 << 1) +#define S3C2412_IISMOD_BCLK_24FS (3 << 1) +#define S3C2412_IISMOD_BCLK_MASK (3 << 1) +#define S3C2412_IISMOD_8BIT (1 << 0) + +#define S3C64XX_IISMOD_CDCLKCON (1 << 12) + +#define S3C2412_IISPSR_PSREN (1 << 15) + +#define S3C64XX_IISFIC_TX2COUNT(x) (((x) >> 24) & 0xf) +#define S3C64XX_IISFIC_TX1COUNT(x) (((x) >> 16) & 0xf) + +#define S3C2412_IISFIC_TXFLUSH (1 << 15) +#define S3C2412_IISFIC_RXFLUSH (1 << 7) +#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) +#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) + +#define S5PC1XX_IISFICS_TXFLUSH (1 << 15) +#define S5PC1XX_IISFICS_TXCOUNT(x) (((x) >> 8) & 0x7f) + +#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ diff --git a/sound/soc/samsung/s3c-ac97.c b/sound/soc/samsung/s3c-ac97.c new file mode 100644 index 0000000..ecf4fd0 --- /dev/null +++ b/sound/soc/samsung/s3c-ac97.c @@ -0,0 +1,521 @@ +/* sound/soc/s3c24xx/s3c-ac97.c + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.c + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Credits: Graeme Gregory, Sean Choi + * + * 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/init.h> +#include <linux/module.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/clk.h> + +#include <sound/soc.h> + +#include <plat/regs-ac97.h> +#include <mach/dma.h> +#include <plat/audio.h> + +#include "s3c-dma.h" +#include "s3c-ac97.h" + +#define AC_CMD_ADDR(x) (x << 16) +#define AC_CMD_DATA(x) (x & 0xffff) + +struct s3c_ac97_info { + unsigned state; + struct clk *ac97_clk; + void __iomem *regs; + struct mutex lock; + struct completion done; +}; +static struct s3c_ac97_info s3c_ac97; + +static struct s3c2410_dma_client s3c_dma_client_out = { + .name = "AC97 PCMOut" +}; + +static struct s3c2410_dma_client s3c_dma_client_in = { + .name = "AC97 PCMIn" +}; + +static struct s3c2410_dma_client s3c_dma_client_micin = { + .name = "AC97 MicIn" +}; + +static struct s3c_dma_params s3c_ac97_pcm_out = { + .client = &s3c_dma_client_out, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_pcm_in = { + .client = &s3c_dma_client_in, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_mic_in = { + .client = &s3c_dma_client_micin, + .dma_size = 4, +}; + +static void s3c_ac97_activate(struct snd_ac97 *ac97) +{ + u32 ac_glbctrl, stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + INIT_COMPLETION(s3c_ac97.done); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to activate!"); +} + +static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + u32 ac_glbctrl, ac_codec_cmd; + u32 stat, addr, data; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to read!"); + + stat = readl(s3c_ac97.regs + S3C_AC97_STAT); + addr = (stat >> 16) & 0x7f; + data = (stat & 0xffff); + + if (addr != reg) + printk(KERN_ERR "s3c-ac97: req addr = %02x, rep addr = %02x\n", reg, addr); + + mutex_unlock(&s3c_ac97.lock); + + return (unsigned short)data; +} + +static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + u32 ac_glbctrl, ac_codec_cmd; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to write!"); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + mutex_unlock(&s3c_ac97.lock); +} + +static void s3c_ac97_cold_reset(struct snd_ac97 *ac97) +{ + writel(S3C_AC97_GLBCTRL_COLDRESET, + s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); +} + +static void s3c_ac97_warm_reset(struct snd_ac97 *ac97) +{ + u32 stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + writel(S3C_AC97_GLBCTRL_WARMRESET, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + s3c_ac97_activate(ac97); +} + +static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) +{ + u32 ac_glbctrl, ac_glbstat; + + ac_glbstat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT); + + if (ac_glbstat & S3C_AC97_GLBSTAT_CODECREADY) { + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + complete(&s3c_ac97.done); + } + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= (1<<30); /* Clear interrupt */ + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + return IRQ_HANDLED; +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = s3c_ac97_read, + .write = s3c_ac97_write, + .warm_reset = s3c_ac97_warm_reset, + .reset = s3c_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_dma_params *dma_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &s3c_ac97_pcm_out; + else + dma_data = &s3c_ac97_pcm_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + return 0; +} + +static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; + else + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; + else + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + else + snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); + + return 0; +} + +static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ac_glbctrl |= S3C_AC97_GLBCTRL_MICINTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static struct snd_soc_dai_ops s3c_ac97_dai_ops = { + .hw_params = s3c_ac97_hw_params, + .trigger = s3c_ac97_trigger, +}; + +static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { + .hw_params = s3c_ac97_hw_mic_params, + .trigger = s3c_ac97_mic_trigger, +}; + +struct snd_soc_dai s3c_ac97_dai[] = { + [S3C_AC97_DAI_PCM] = { + .name = "s3c-ac97", + .id = S3C_AC97_DAI_PCM, + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_dai_ops, + }, + [S3C_AC97_DAI_MIC] = { + .name = "s3c-ac97-mic", + .id = S3C_AC97_DAI_MIC, + .ac97_control = 1, + .capture = { + .stream_name = "AC97 Mic Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_mic_dai_ops, + }, +}; +EXPORT_SYMBOL_GPL(s3c_ac97_dai); + +static __devinit int s3c_ac97_probe(struct platform_device *pdev) +{ + struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; + struct s3c_audio_pdata *ac97_pdata; + int ret; + + ac97_pdata = pdev->dev.platform_data; + if (!ac97_pdata || !ac97_pdata->cfg_gpio) { + dev_err(&pdev->dev, "cfg_gpio callback not provided!\n"); + return -EINVAL; + } + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n"); + return -ENXIO; + } + + dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2); + if (!dmamic_res) { + dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!irq_res) { + dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); + return -ENXIO; + } + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "s3c-ac97")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + return -EBUSY; + } + + s3c_ac97_pcm_out.channel = dmatx_res->start; + s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_pcm_in.channel = dmarx_res->start; + s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_mic_in.channel = dmamic_res->start; + s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA; + + init_completion(&s3c_ac97.done); + mutex_init(&s3c_ac97.lock); + + s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); + if (s3c_ac97.regs == NULL) { + dev_err(&pdev->dev, "Unable to ioremap register region\n"); + ret = -ENXIO; + goto err1; + } + + s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + if (IS_ERR(s3c_ac97.ac97_clk)) { + dev_err(&pdev->dev, "s3c-ac97 failed to get ac97_clock\n"); + ret = -ENODEV; + goto err2; + } + clk_enable(s3c_ac97.ac97_clk); + + if (ac97_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err3; + } + + ret = request_irq(irq_res->start, s3c_ac97_irq, + IRQF_DISABLED, "AC97", NULL); + if (ret < 0) { + printk(KERN_ERR "s3c-ac97: interrupt request failed.\n"); + goto err4; + } + + s3c_ac97_dai[S3C_AC97_DAI_PCM].dev = &pdev->dev; + s3c_ac97_dai[S3C_AC97_DAI_MIC].dev = &pdev->dev; + + ret = snd_soc_register_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + if (ret) + goto err5; + + return 0; + +err5: + free_irq(irq_res->start, NULL); +err4: +err3: + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); +err2: + iounmap(s3c_ac97.regs); +err1: + release_mem_region(mem_res->start, resource_size(mem_res)); + + return ret; +} + +static __devexit int s3c_ac97_remove(struct platform_device *pdev) +{ + struct resource *mem_res, *irq_res; + + snd_soc_unregister_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (irq_res) + free_irq(irq_res->start, NULL); + + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); + + iounmap(s3c_ac97.regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (mem_res) + release_mem_region(mem_res->start, resource_size(mem_res)); + + return 0; +} + +static struct platform_driver s3c_ac97_driver = { + .probe = s3c_ac97_probe, + .remove = s3c_ac97_remove, + .driver = { + .name = "s3c-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_ac97_init(void) +{ + return platform_driver_register(&s3c_ac97_driver); +} +module_init(s3c_ac97_init); + +static void __exit s3c_ac97_exit(void) +{ + platform_driver_unregister(&s3c_ac97_driver); +} +module_exit(s3c_ac97_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-ac97.h b/sound/soc/samsung/s3c-ac97.h new file mode 100644 index 0000000..2781983 --- /dev/null +++ b/sound/soc/samsung/s3c-ac97.h @@ -0,0 +1,23 @@ +/* sound/soc/s3c24xx/s3c-ac97.h + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.h + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Credits: Graeme Gregory, Sean Choi + * + * 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 __S3C_AC97_H_ +#define __S3C_AC97_H_ + +#define S3C_AC97_DAI_PCM 0 +#define S3C_AC97_DAI_MIC 1 + +extern struct snd_soc_dai s3c_ac97_dai[]; + +#endif /* __S3C_AC97_H_ */ diff --git a/sound/soc/samsung/s3c-dma.c b/sound/soc/samsung/s3c-dma.c new file mode 100644 index 0000000..1b61c23 --- /dev/null +++ b/sound/soc/samsung/s3c-dma.c @@ -0,0 +1,483 @@ +/* + * s3c-dma.c -- ALSA Soc Audio Layer + * + * (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Copyright 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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/init.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <mach/hardware.h> +#include <mach/dma.h> + +#include "s3c-dma.h" + +static const struct snd_pcm_hardware s3c_dma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128*1024, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = PAGE_SIZE*2, + .periods_min = 2, + .periods_max = 128, + .fifo_size = 32, +}; + +struct s3c24xx_runtime_data { + spinlock_t lock; + int state; + unsigned int dma_loaded; + unsigned int dma_limit; + unsigned int dma_period; + dma_addr_t dma_start; + dma_addr_t dma_pos; + dma_addr_t dma_end; + struct s3c_dma_params *params; +}; + +/* s3c_dma_enqueue + * + * place a dma buffer onto the queue for the dma system + * to handle. +*/ +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + dma_addr_t pos = prtd->dma_pos; + unsigned int limit; + int ret; + + pr_debug("Entered %s\n", __func__); + + if (s3c_dma_has_circular()) + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { + unsigned long len = prtd->dma_period; + + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); + + if ((pos + len) > prtd->dma_end) { + len = prtd->dma_end - pos; + pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", + __func__, len); + } + + ret = s3c2410_dma_enqueue(prtd->params->channel, + substream, pos, len); + + if (ret == 0) { + prtd->dma_loaded++; + pos += prtd->dma_period; + if (pos >= prtd->dma_end) + pos = prtd->dma_start; + } else + break; + } + + prtd->dma_pos = pos; +} + +static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, + void *dev_id, int size, + enum s3c2410_dma_buffresult result) +{ + struct snd_pcm_substream *substream = dev_id; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) + return; + + prtd = substream->runtime->private_data; + + if (substream) + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { + prtd->dma_loaded--; + s3c_dma_enqueue(substream); + } + + spin_unlock(&prtd->lock); +} + +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + int ret = 0; + + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!dma) + return 0; + + /* this may get called several times by oss emulation + * with different params -HW */ + if (prtd->params == NULL) { + /* prepare DMA */ + prtd->params = dma; + + pr_debug("params %p, client %p, channel %d\n", prtd->params, + prtd->params->client, prtd->params->channel); + + ret = s3c2410_dma_request(prtd->params->channel, + prtd->params->client, NULL); + + if (ret < 0) { + printk(KERN_ERR "failed to get dma channel\n"); + return ret; + } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); + } + + s3c2410_dma_set_buffdone_fn(prtd->params->channel, + s3c24xx_audio_buffdone); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + runtime->dma_bytes = totbytes; + + spin_lock_irq(&prtd->lock); + prtd->dma_loaded = 0; + prtd->dma_limit = runtime->hw.periods_min; + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + totbytes; + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + /* TODO - do we need to ensure DMA flushed */ + snd_pcm_set_runtime_buffer(substream, NULL); + + if (prtd->params) { + s3c2410_dma_free(prtd->params->channel, prtd->params->client); + prtd->params = NULL; + } + + return 0; +} + +static int s3c_dma_prepare(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->params) + return 0; + + /* channel needs configuring for mem=>device, increment memory addr, + * sync to pclk, half-word transfers to the IIS-FIFO. */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_MEM, + prtd->params->dma_addr); + } else { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_HW, + prtd->params->dma_addr); + } + + s3c2410_dma_config(prtd->params->channel, + prtd->params->dma_size); + + /* flush the DMA channel */ + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); + prtd->dma_loaded = 0; + prtd->dma_pos = prtd->dma_start; + + /* enqueue dma buffers */ + s3c_dma_enqueue(substream); + + return ret; +} + +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t +s3c_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + unsigned long res; + dma_addr_t src, dst; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + s3c2410_dma_getposition(prtd->params->channel, &src, &dst); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + res = dst - prtd->dma_start; + else + res = src - prtd->dma_start; + + spin_unlock(&prtd->lock); + + pr_debug("Pointer %x %x\n", src, dst); + + /* we seem to be getting the odd error from the pcm library due + * to out-of-bounds pointers. this is maybe due to the dma engine + * not having loaded the new values for the channel before being + * callled... (todo - fix ) + */ + + if (res >= snd_pcm_lib_buffer_bytes(substream)) { + if (res == snd_pcm_lib_buffer_bytes(substream)) + res = 0; + } + + return bytes_to_frames(substream->runtime, res); +} + +static int s3c_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); + + prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + return 0; +} + +static int s3c_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + if (!prtd) + pr_debug("s3c_dma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static int s3c_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + pr_debug("Entered %s\n", __func__); + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, +}; + +static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = s3c_dma_hardware.buffer_bytes_max; + + pr_debug("Entered %s\n", __func__); + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + pr_debug("Entered %s\n", __func__); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 s3c_dma_mask = DMA_BIT_MASK(32); + +static int s3c_dma_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s3c_dma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +struct snd_soc_platform s3c24xx_soc_platform = { + .name = "s3c24xx-audio", + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); + +static int __init s3c24xx_soc_platform_init(void) +{ + return snd_soc_register_platform(&s3c24xx_soc_platform); +} +module_init(s3c24xx_soc_platform_init); + +static void __exit s3c24xx_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&s3c24xx_soc_platform); +} +module_exit(s3c24xx_soc_platform_exit); + +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-dma.h b/sound/soc/samsung/s3c-dma.h new file mode 100644 index 0000000..69bb6bf --- /dev/null +++ b/sound/soc/samsung/s3c-dma.h @@ -0,0 +1,31 @@ +/* + * s3c-dma.h -- + * + * 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. + * + * ALSA PCM interface for the Samsung S3C24xx CPU + */ + +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +struct s3c_dma_params { + struct s3c2410_dma_client *client; /* stream identifier */ + int channel; /* Channel ID */ + dma_addr_t dma_addr; + int dma_size; /* Size of the DMA transfer */ +}; + +#define S3C24XX_DAI_I2S 0 + +/* platform data */ +extern struct snd_soc_platform s3c24xx_soc_platform; +extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; + +#endif diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c new file mode 100644 index 0000000..13311c8 --- /dev/null +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -0,0 +1,776 @@ +/* sound/soc/s3c24xx/s3c-i2c-v2.c + * + * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com + * linux(a)wolfsonmicro.com + * + * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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/delay.h> +#include <linux/clk.h> +#include <linux/io.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/dma.h> + +#include "regs-i2s-v2.h" +#include "s3c-i2s-v2.h" +#include "s3c-dma.h" + +#undef S3C_IIS_V2_SUPPORTED + +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#define S3C_IIS_V2_SUPPORTED +#endif + +#ifdef CONFIG_PLAT_S3C64XX +#define S3C_IIS_V2_SUPPORTED +#endif + +#ifndef S3C_IIS_V2_SUPPORTED +#error Unsupported CPU model +#endif + +#define S3C2412_I2S_DEBUG_CON 0 + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +#define bit_set(v, b) (((v) & (b)) ? 1 : 0) + +#if S3C2412_I2S_DEBUG_CON +static void dbg_showcon(const char *fn, u32 con) +{ + printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, + bit_set(con, S3C2412_IISCON_LRINDEX), + bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_TXFIFO_FULL), + bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); + + printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", + fn, + bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_TXCH_PAUSE), + bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); + printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, + bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); +} +#else +static inline void dbg_showcon(const char *fn, u32 con) +{ +} +#endif + + +/* Turn on or off the transmission path. */ +static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + pr_debug("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_TXDMA_PAUSE; + con &= ~S3C2412_IISCON_TXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXONLY: + case S3C2412_IISMOD_MODE_TXRX: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_RXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } else { + /* Note, we do not have any indication that the FIFO problems + * tha the S3C2410/2440 had apply here, so we should be able + * to disable the DMA and TX without resetting the FIFOS. + */ + + con |= S3C2412_IISCON_TXDMA_PAUSE; + con |= S3C2412_IISCON_TXCH_PAUSE; + con &= ~S3C2412_IISCON_TXDMA_ACTIVE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_RXONLY; + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + con &= ~S3C2412_IISCON_IIS_ACTIVE; + break; + + default: + dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } + + fic = readl(regs + S3C2412_IISFIC); + dbg_showcon(__func__, con); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + +static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + pr_debug("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_RXDMA_PAUSE; + con &= ~S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + case S3C2412_IISMOD_MODE_RXONLY: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } else { + /* See txctrl notes on FIFOs. */ + + con &= ~S3C2412_IISCON_RXDMA_ACTIVE; + con |= S3C2412_IISCON_RXDMA_PAUSE; + con |= S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_RXONLY: + con &= ~S3C2412_IISCON_IIS_ACTIVE; + mod &= ~S3C2412_IISMOD_MODE_MASK; + break; + + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXONLY; + break; + + default: + dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } + + fic = readl(regs + S3C2412_IISFIC); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + +#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) +{ + u32 iiscon; + unsigned long loops = msecs_to_loops(5); + + pr_debug("Entered %s\n", __func__); + + while (--loops) { + iiscon = readl(i2s->regs + S3C2412_IISCON); + if (iiscon & S3C2412_IISCON_LRINDEX) + break; + + cpu_relax(); + } + + if (!loops) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; + } + + return 0; +} + +/* + * Set S3C2412 I2S DAI format + */ +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("hw_params r: IISMOD: %x \n", iismod); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->master = 0; + iismod |= S3C2412_IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->master = 1; + iismod &= ~S3C2412_IISMOD_SLAVE; + break; + default: + pr_err("unknwon master/slave format\n"); + return -EINVAL; + } + + iismod &= ~S3C2412_IISMOD_SDF_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; + iismod |= S3C2412_IISMOD_SDF_MSB; + break; + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; + iismod |= S3C2412_IISMOD_SDF_LSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2412_IISMOD_LR_RLOW; + iismod |= S3C2412_IISMOD_SDF_IIS; + break; + default: + pr_err("Unknown data format\n"); + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); + + /* Working copies of register */ + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); + + iismod &= ~S3C64XX_IISMOD_BLC_MASK; + /* Sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C64XX_IISMOD_BLC_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + iismod |= S3C64XX_IISMOD_BLC_24BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static int s3c_i2sv2_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + pr_debug("Entered %s\n", __func__); + pr_debug("%s r: IISMOD: %x\n", __func__, iismod); + + switch (clk_id) { + case S3C_I2SV2_CLKSRC_PCLK: + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; + break; + + case S3C_I2SV2_CLKSRC_AUDIOBUS: + iismod |= S3C2412_IISMOD_IMS_SYSMUX; + break; + + case S3C_I2SV2_CLKSRC_CDCLK: + /* Error if controller doesn't have the CDCLKCON bit */ + if (!(i2s->feature & S3C_FEATURE_CDCLKCON)) + return -EINVAL; + + switch (dir) { + case SND_SOC_CLOCK_IN: + iismod |= S3C64XX_IISMOD_CDCLKCON; + break; + case SND_SOC_CLOCK_OUT: + iismod &= ~S3C64XX_IISMOD_CDCLKCON; + break; + default: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai); + int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + unsigned long irqs; + int ret = 0; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + pr_debug("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* On start, ensure that the FIFOs are cleared and reset. */ + + writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + /* clear again, just in case */ + writel(0x0, i2s->regs + S3C2412_IISFIC); + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!i2s->master) { + ret = s3c2412_snd_lrsync(i2s); + if (ret) + goto exit_err; + } + + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 1); + else + s3c2412_snd_txctrl(i2s, 1); + + local_irq_restore(irqs); + + /* + * Load the next buffer to DMA to meet the reqirement + * of the auto reload mechanism of S3C24XX. + * This call won't bother S3C64XX. + */ + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 0); + else + s3c2412_snd_txctrl(i2s, 0); + + local_irq_restore(irqs); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* + * Set S3C2412 Clock dividers + */ +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 reg; + + pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); + + switch (div_id) { + case S3C_I2SV2_DIV_BCLK: + switch (div) { + case 16: + div = S3C2412_IISMOD_BCLK_16FS; + break; + + case 32: + div = S3C2412_IISMOD_BCLK_32FS; + break; + + case 24: + div = S3C2412_IISMOD_BCLK_24FS; + break; + + case 48: + div = S3C2412_IISMOD_BCLK_48FS; + break; + + default: + return -EINVAL; + } + + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_BCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_RCLK: + switch (div) { + case 256: + div = S3C2412_IISMOD_RCLK_256FS; + break; + + case 384: + div = S3C2412_IISMOD_RCLK_384FS; + break; + + case 512: + div = S3C2412_IISMOD_RCLK_512FS; + break; + + case 768: + div = S3C2412_IISMOD_RCLK_768FS; + break; + + default: + return -EINVAL; + } + + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_RCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_PRESCALER: + if (div >= 0) { + writel((div << 8) | S3C2412_IISPSR_PSREN, + i2s->regs + S3C2412_IISPSR); + } else { + writel(0x0, i2s->regs + S3C2412_IISPSR); + } + pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 reg = readl(i2s->regs + S3C2412_IISFIC); + snd_pcm_sframes_t delay; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = S3C2412_IISFIC_TXCOUNT(reg); + else + delay = S3C2412_IISFIC_RXCOUNT(reg); + + return delay; +} + +struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_get_clock); + +/* default table of all avaialable root fs divisors */ +static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; + +int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) +{ + unsigned long clkrate = clk_get_rate(clk); + unsigned int div; + unsigned int fsclk; + unsigned int actual; + unsigned int fs; + unsigned int fsdiv; + signed int deviation = 0; + unsigned int best_fs = 0; + unsigned int best_div = 0; + unsigned int best_rate = 0; + unsigned int best_deviation = INT_MAX; + + pr_debug("Input clock rate %ldHz\n", clkrate); + + if (fstab == NULL) + fstab = iis_fs_tab; + + for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) { + fsdiv = iis_fs_tab[fs]; + + fsclk = clkrate / fsdiv; + div = fsclk / rate; + + if ((fsclk % rate) > (rate / 2)) + div++; + + if (div <= 1) + continue; + + actual = clkrate / (fsdiv * div); + deviation = actual - rate; + + printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", + fsdiv, div, actual, deviation); + + deviation = abs(deviation); + + if (deviation < best_deviation) { + best_fs = fsdiv; + best_div = div; + best_rate = actual; + best_deviation = deviation; + } + + if (deviation == 0) + break; + } + + printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", + best_fs, best_div, best_rate); + + info->fs_div = best_fs; + info->clk_div = best_div; + + return 0; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); + +int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base) +{ + struct device *dev = &pdev->dev; + unsigned int iismod; + + i2s->dev = dev; + + /* record our i2s structure for later use in the callbacks */ + dai->private_data = i2s; + + if (!base) { + struct resource *res = platform_get_resource(pdev, + IORESOURCE_MEM, + 0); + if (!res) { + dev_err(dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (!request_mem_region(res->start, resource_size(res), + "s3c64xx-i2s-v4")) { + dev_err(dev, "Unable to request register region\n"); + return -EBUSY; + } + + base = res->start; + } + + i2s->regs = ioremap(base, 0x100); + if (i2s->regs == NULL) { + dev_err(dev, "cannot ioremap registers\n"); + return -ENXIO; + } + + i2s->iis_pclk = clk_get(dev, "iis"); + if (IS_ERR(i2s->iis_pclk)) { + dev_err(dev, "failed to get iis_clock\n"); + iounmap(i2s->regs); + return -ENOENT; + } + + clk_enable(i2s->iis_pclk); + + /* Mark ourselves as in TXRX mode so we can run through our cleanup + * process without warnings. */ + iismod = readl(i2s->regs + S3C2412_IISMOD); + iismod |= S3C2412_IISMOD_MODE_TXRX; + writel(iismod, i2s->regs + S3C2412_IISMOD); + s3c2412_snd_txctrl(i2s, 0); + s3c2412_snd_rxctrl(i2s, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); + +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod; + + if (dai->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warning("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warning("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warning("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", + dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (dai->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + +int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) +{ + struct snd_soc_dai_ops *ops = dai->ops; + + ops->trigger = s3c2412_i2s_trigger; + if (!ops->hw_params) + ops->hw_params = s3c_i2sv2_hw_params; + ops->set_fmt = s3c2412_i2s_set_fmt; + ops->set_clkdiv = s3c2412_i2s_set_clkdiv; + ops->set_sysclk = s3c_i2sv2_set_sysclk; + + /* Allow overriding by (for example) IISv4 */ + if (!ops->delay) + ops->delay = s3c2412_i2s_delay; + + dai->suspend = s3c2412_i2s_suspend; + dai->resume = s3c2412_i2s_resume; + + return snd_soc_register_dai(dai); +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h new file mode 100644 index 0000000..766f43a --- /dev/null +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -0,0 +1,103 @@ +/* sound/soc/s3c24xx/s3c-i2s-v2.h + * + * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver + * + * Copyright (c) 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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 code is the core support for the I2S block found in a number of + * Samsung SoC devices which is unofficially named I2S-V2. Currently the + * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S + * channels via configurable GPIO. + */ + +#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H +#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__ + +#define S3C_I2SV2_DIV_BCLK (1) +#define S3C_I2SV2_DIV_RCLK (2) +#define S3C_I2SV2_DIV_PRESCALER (3) + +#define S3C_I2SV2_CLKSRC_PCLK 0 +#define S3C_I2SV2_CLKSRC_AUDIOBUS 1 +#define S3C_I2SV2_CLKSRC_CDCLK 2 + +/* Set this flag for I2S controllers that have the bit IISMOD[12] + * bridge/break RCLK signal and external Xi2sCDCLK pin. + */ +#define S3C_FEATURE_CDCLKCON (1 << 0) + +/** + * struct s3c_i2sv2_info - S3C I2S-V2 information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device registe block. + * @feature: Set of bit-flags indicating features of the controller. + * @master: True if the I2S core is the I2S bit clock master. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + * @suspend_iismod: PM save for the IISMOD register. + * @suspend_iiscon: PM save for the IISCON register. + * @suspend_iispsr: PM save for the IISPSR register. + * + * This is the private codec state for the hardware associated with an + * I2S channel such as the register mappings and clock sources. + */ +struct s3c_i2sv2_info { + struct device *dev; + void __iomem *regs; + + u32 feature; + + struct clk *iis_pclk; + struct clk *iis_cclk; + + unsigned char master; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; + + u32 suspend_iismod; + u32 suspend_iiscon; + u32 suspend_iispsr; +}; + +extern struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai); + +struct s3c_i2sv2_rate_calc { + unsigned int clk_div; /* for prescaler */ + unsigned int fs_div; /* for root frame clock */ +}; + +extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk); + +/** + * s3c_i2sv2_probe - probe for i2s device helper + * @pdev: The platform device supplied to the original probe. + * @dai: The ASoC DAI structure supplied to the original probe. + * @i2s: Our local i2s structure to fill in. + * @base: The base address for the registers. + */ +extern int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base); + +/** + * s3c_i2sv2_register_dai - register dai with soc core + * @dai: The snd_soc_dai structure to register + * + * Fill in any missing fields and then register the given dai with the + * soc core. + */ +extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai); + +#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/samsung/s3c-pcm.c b/sound/soc/samsung/s3c-pcm.c new file mode 100644 index 0000000..326f0a9 --- /dev/null +++ b/sound/soc/samsung/s3c-pcm.c @@ -0,0 +1,554 @@ +/* sound/soc/s3c24xx/s3c-pcm.c + * + * ALSA SoC Audio Layer - S3C PCM-Controller driver + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * based upon I2S drivers by Ben Dooks. + * + * 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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/audio.h> +#include <plat/dma.h> + +#include "s3c-dma.h" +#include "s3c-pcm.h" + +static struct s3c2410_dma_client s3c_pcm_dma_client_out = { + .name = "PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c_pcm_dma_client_in = { + .name = "PCM Stereo in" +}; + +static struct s3c_dma_params s3c_pcm_stereo_out[] = { + [0] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, +}; + +static struct s3c_dma_params s3c_pcm_stereo_in[] = { + [0] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, +}; + +static struct s3c_pcm_info s3c_pcm[2]; + +static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + clkctl = readl(regs + S3C_PCM_CLKCTL); + ctl = readl(regs + S3C_PCM_CTL); + ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK + << S3C_PCM_CTL_TXDIPSTICK_SHIFT); + + if (on) { + ctl |= S3C_PCM_CTL_TXDMA_EN; + ctl |= S3C_PCM_CTL_TXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + ctl |= (0x20<<S3C_PCM_CTL_TXDIPSTICK_SHIFT); + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_TXDMA_EN; + ctl &= ~S3C_PCM_CTL_TXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_RXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + ctl = readl(regs + S3C_PCM_CTL); + clkctl = readl(regs + S3C_PCM_CLKCTL); + + if (on) { + ctl |= S3C_PCM_CTL_RXDMA_EN; + ctl |= S3C_PCM_CTL_RXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_RXDMA_EN; + ctl &= ~S3C_PCM_CTL_RXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai); + unsigned long flags; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 1); + else + s3c_pcm_snd_txctrl(pcm, 1); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 0); + else + s3c_pcm_snd_txctrl(pcm, 0); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; + void __iomem *regs = pcm->regs; + struct clk *clk; + int sclk_div, sync_div; + unsigned long flags; + u32 clkctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = pcm->dma_playback; + else + dma_data = pcm->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); + + /* Strictly check for sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + return -EINVAL; + } + + spin_lock_irqsave(&pcm->lock, flags); + + /* Get hold of the PCMSOURCE_CLK */ + clkctl = readl(regs + S3C_PCM_CLKCTL); + if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK) + clk = pcm->pclk; + else + clk = pcm->cclk; + + /* Set the SCLK divider */ + sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs / + params_rate(params) / 2 - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK) + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + + /* Set the SYNC divider */ + sync_div = pcm->sclk_per_fs - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK) + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + spin_unlock_irqrestore(&pcm->lock, flags); + + dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs SCLK_DIV=%d SYNC_DIV=%d\n", + clk_get_rate(clk), pcm->sclk_per_fs, + sclk_div, sync_div); + + return 0; +} + +static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + unsigned long flags; + int ret = 0; + u32 ctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + spin_lock_irqsave(&pcm->lock, flags); + + ctl = readl(regs + S3C_PCM_CTL); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do, NB_NF by default */ + break; + default: + dev_err(pcm->dev, "Unsupported clock inversion!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Nothing to do, Master by default */ + break; + default: + dev_err(pcm->dev, "Unsupported master/slave format!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + pcm->idleclk = 1; + break; + case SND_SOC_DAIFMT_GATED: + pcm->idleclk = 0; + break; + default: + dev_err(pcm->dev, "Invalid Clock gating request!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + case SND_SOC_DAIFMT_DSP_B: + ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + default: + dev_err(pcm->dev, "Unsupported data format!\n"); + ret = -EINVAL; + goto exit; + } + + writel(ctl, regs + S3C_PCM_CTL); + +exit: + spin_unlock_irqrestore(&pcm->lock, flags); + + return ret; +} + +static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + + switch (div_id) { + case S3C_PCM_SCLK_PER_FS: + pcm->sclk_per_fs = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + u32 clkctl = readl(regs + S3C_PCM_CLKCTL); + + switch (clk_id) { + case S3C_PCM_CLKSRC_PCLK: + clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + break; + + case S3C_PCM_CLKSRC_MUX: + clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + + if (clk_get_rate(pcm->cclk) != freq) + clk_set_rate(pcm->cclk, freq); + + break; + + default: + return -EINVAL; + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + return 0; +} + +static struct snd_soc_dai_ops s3c_pcm_dai_ops = { + .set_sysclk = s3c_pcm_set_sysclk, + .set_clkdiv = s3c_pcm_set_clkdiv, + .trigger = s3c_pcm_trigger, + .hw_params = s3c_pcm_hw_params, + .set_fmt = s3c_pcm_set_fmt, +}; + +#define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 + +#define S3C_PCM_DECLARE(n) \ +{ \ + .name = "samsung-pcm", \ + .id = (n), \ + .symmetric_rates = 1, \ + .ops = &s3c_pcm_dai_ops, \ + .playback = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ +} + +struct snd_soc_dai s3c_pcm_dai[] = { + S3C_PCM_DECLARE(0), + S3C_PCM_DECLARE(1), +}; +EXPORT_SYMBOL_GPL(s3c_pcm_dai); + +static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm; + struct snd_soc_dai *dai; + struct resource *mem_res, *dmatx_res, *dmarx_res; + struct s3c_audio_pdata *pcm_pdata; + int ret; + + /* Check for valid device index */ + if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + pcm_pdata = pdev->dev.platform_data; + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + return -EINVAL; + } + + pcm = &s3c_pcm[pdev->id]; + pcm->dev = &pdev->dev; + + spin_lock_init(&pcm->lock); + + dai = &s3c_pcm_dai[pdev->id]; + dai->dev = &pdev->dev; + + /* Default is 128fs */ + pcm->sclk_per_fs = 128; + + pcm->cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(pcm->cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(pcm->cclk); + goto err1; + } + clk_enable(pcm->cclk); + + /* record our pcm structure for later use in the callbacks */ + dai->private_data = pcm; + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "samsung-pcm")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + ret = -EBUSY; + goto err2; + } + + pcm->regs = ioremap(mem_res->start, 0x100); + if (pcm->regs == NULL) { + dev_err(&pdev->dev, "cannot ioremap registers\n"); + ret = -ENXIO; + goto err3; + } + + pcm->pclk = clk_get(&pdev->dev, "pcm"); + if (IS_ERR(pcm->pclk)) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + ret = -ENOENT; + goto err4; + } + clk_enable(pcm->pclk); + + ret = snd_soc_register_dai(dai); + if (ret != 0) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + goto err5; + } + + s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + + S3C_PCM_RXFIFO; + s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start + + S3C_PCM_TXFIFO; + + s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start; + s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start; + + pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; + pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + + return 0; + +err5: + clk_disable(pcm->pclk); + clk_put(pcm->pclk); +err4: + iounmap(pcm->regs); +err3: + release_mem_region(mem_res->start, resource_size(mem_res)); +err2: + clk_disable(pcm->cclk); + clk_put(pcm->cclk); +err1: + return ret; +} + +static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; + struct resource *mem_res; + + iounmap(pcm->regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem_res->start, resource_size(mem_res)); + + clk_disable(pcm->cclk); + clk_disable(pcm->pclk); + clk_put(pcm->pclk); + clk_put(pcm->cclk); + + return 0; +} + +static struct platform_driver s3c_pcm_driver = { + .probe = s3c_pcm_dev_probe, + .remove = s3c_pcm_dev_remove, + .driver = { + .name = "samsung-pcm", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_pcm_init(void) +{ + return platform_driver_register(&s3c_pcm_driver); +} +module_init(s3c_pcm_init); + +static void __exit s3c_pcm_exit(void) +{ + platform_driver_unregister(&s3c_pcm_driver); +} +module_exit(s3c_pcm_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("S3C PCM Controller Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-pcm.h b/sound/soc/samsung/s3c-pcm.h new file mode 100644 index 0000000..69ff997 --- /dev/null +++ b/sound/soc/samsung/s3c-pcm.h @@ -0,0 +1,123 @@ +/* sound/soc/s3c24xx/s3c-pcm.h + * + * 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 __S3C_PCM_H +#define __S3C_PCM_H __FILE__ + +/*Register Offsets */ +#define S3C_PCM_CTL (0x00) +#define S3C_PCM_CLKCTL (0x04) +#define S3C_PCM_TXFIFO (0x08) +#define S3C_PCM_RXFIFO (0x0C) +#define S3C_PCM_IRQCTL (0x10) +#define S3C_PCM_IRQSTAT (0x14) +#define S3C_PCM_FIFOSTAT (0x18) +#define S3C_PCM_CLRINT (0x20) + +/* PCM_CTL Bit-Fields */ +#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) +#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) +#define S3C_PCM_CTL_RXDIPSTICK_MSK (0x3f<<7) +#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) +#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) +#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) +#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) +#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) +#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) +#define S3C_PCM_CTL_ENABLE (0x1<<0) + +/* PCM_CLKCTL Bit-Fields */ +#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) +#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) +#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) +#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) + +/* PCM_TXFIFO Bit-Fields */ +#define S3C_PCM_TXFIFO_DVALID (0x1<<16) +#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_RXFIFO Bit-Fields */ +#define S3C_PCM_RXFIFO_DVALID (0x1<<16) +#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_IRQCTL Bit-Fields */ +#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) +#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) +#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) +#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) +#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) +#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) +#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) +#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) +#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) +#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) +#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) +#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) +#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) +#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) + +/* PCM_IRQSTAT Bit-Fields */ +#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) +#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) +#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) +#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) +#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) +#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) +#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) +#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) +#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) +#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) +#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) +#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) +#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) +#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) + +/* PCM_FIFOSTAT Bit-Fields */ +#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) +#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) +#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) +#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) +#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) +#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) + +#define S3C_PCM_CLKSRC_PCLK 0 +#define S3C_PCM_CLKSRC_MUX 1 + +#define S3C_PCM_SCLK_PER_FS 0 + +/** + * struct s3c_pcm_info - S3C PCM Controller information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device register block. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + */ +struct s3c_pcm_info { + spinlock_t lock; + struct device *dev; + void __iomem *regs; + + unsigned int sclk_per_fs; + + /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ + unsigned int idleclk; + + struct clk *pclk; + struct clk *cclk; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; +}; + +#endif /* __S3C_PCM_H */ diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c new file mode 100644 index 0000000..709adef --- /dev/null +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -0,0 +1,190 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com + * linux(a)wolfsonmicro.com + * + * Copyright (c) 2007, 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <mach/hardware.h> + +#include <mach/regs-gpio.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c2412-i2s.h" + +#define S3C2412_I2S_DEBUG 0 + +static struct s3c2410_dma_client s3c2412_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c2412_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { + .client = &s3c2412_dma_client_out, + .channel = DMACH_I2S_OUT, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { + .client = &s3c2412_dma_client_in, + .channel = DMACH_I2S_IN, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .dma_size = 4, +}; + +static struct s3c_i2sv2_info s3c2412_i2s; + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c2412_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int ret; + + pr_debug("Entered %s\n", __func__); + + ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS); + if (ret) + return ret; + + s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; + s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; + + s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); + if (s3c2412_i2s.iis_cclk == NULL) { + pr_err("failed to get i2sclk clock\n"); + iounmap(s3c2412_i2s.regs); + return -ENODEV; + } + + /* Set MPLL as the source for IIS CLK */ + + clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); + clk_enable(s3c2412_i2s.iis_cclk); + + s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; + + /* Configure the I2S pins in correct mode */ + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + + return 0; +} + +static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C2412_IISMOD_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod &= ~S3C2412_IISMOD_8BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +#define S3C2412_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { + .hw_params = s3c2412_i2s_hw_params, +}; + +struct snd_soc_dai s3c2412_i2s_dai = { + .name = "s3c2412-i2s", + .id = 0, + .probe = s3c2412_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &s3c2412_i2s_dai_ops, +}; +EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); + +static int __init s3c2412_i2s_init(void) +{ + return s3c_i2sv2_register_dai(&s3c2412_i2s_dai); +} +module_init(s3c2412_i2s_init); + +static void __exit s3c2412_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c2412_i2s_dai); +} +module_exit(s3c2412_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c2412-i2s.h b/sound/soc/samsung/s3c2412-i2s.h new file mode 100644 index 0000000..0b5686b --- /dev/null +++ b/sound/soc/samsung/s3c2412-i2s.h @@ -0,0 +1,29 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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 __SND_SOC_S3C24XX_S3C2412_I2S_H +#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ + +#include "s3c-i2s-v2.h" + +#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C2412_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK +#define S3C2412_CLKSRC_I2SCLK S3C_I2SV2_CLKSRC_AUDIOBUS + +extern struct snd_soc_dai s3c2412_i2s_dai; + +#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c new file mode 100644 index 0000000..c3ac890 --- /dev/null +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -0,0 +1,503 @@ +/* + * s3c24xx-i2s.c -- ALSA Soc Audio Layer + * + * (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Copyright 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/jiffies.h> +#include <linux/io.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> +#include <mach/regs-gpio.h> +#include <mach/regs-clock.h> + +#include <asm/dma.h> +#include <mach/dma.h> + +#include <plat/regs-iis.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +static struct s3c2410_dma_client s3c24xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c24xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { + .client = &s3c24xx_dma_client_out, + .channel = DMACH_I2S_OUT, + .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .dma_size = 2, +}; + +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { + .client = &s3c24xx_dma_client_in, + .channel = DMACH_I2S_IN, + .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .dma_size = 2, +}; + +struct s3c24xx_i2s_info { + void __iomem *regs; + struct clk *iis_clk; + u32 iiscon; + u32 iismod; + u32 iisfcon; + u32 iispsr; +}; +static struct s3c24xx_i2s_info s3c24xx_i2s; + +static void s3c24xx_snd_txctrl(int on) +{ + u32 iisfcon; + u32 iiscon; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); + + if (on) { + iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; + iiscon |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN; + iiscon &= ~S3C2410_IISCON_TXIDLE; + iismod |= S3C2410_IISMOD_TXMODE; + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + } else { + /* note, we have to disable the FIFOs otherwise bad things + * seem to happen when the DMA stops. According to the + * Samsung supplied kernel, this should allow the DMA + * engine and FIFOs to reset. If this isn't allowed, the + * DMA engine will simply freeze randomly. + */ + + iisfcon &= ~S3C2410_IISFCON_TXENABLE; + iisfcon &= ~S3C2410_IISFCON_TXDMA; + iiscon |= S3C2410_IISCON_TXIDLE; + iiscon &= ~S3C2410_IISCON_TXDMAEN; + iismod &= ~S3C2410_IISMOD_TXMODE; + + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + } + + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); +} + +static void s3c24xx_snd_rxctrl(int on) +{ + u32 iisfcon; + u32 iiscon; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); + + if (on) { + iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; + iiscon |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN; + iiscon &= ~S3C2410_IISCON_RXIDLE; + iismod |= S3C2410_IISMOD_RXMODE; + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + } else { + /* note, we have to disable the FIFOs otherwise bad things + * seem to happen when the DMA stops. According to the + * Samsung supplied kernel, this should allow the DMA + * engine and FIFOs to reset. If this isn't allowed, the + * DMA engine will simply freeze randomly. + */ + + iisfcon &= ~S3C2410_IISFCON_RXENABLE; + iisfcon &= ~S3C2410_IISFCON_RXDMA; + iiscon |= S3C2410_IISCON_RXIDLE; + iiscon &= ~S3C2410_IISCON_RXDMAEN; + iismod &= ~S3C2410_IISMOD_RXMODE; + + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + } + + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); +} + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c24xx_snd_lrsync(void) +{ + u32 iiscon; + int timeout = 50; /* 5ms */ + + pr_debug("Entered %s\n", __func__); + + while (1) { + iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + if (iiscon & S3C2410_IISCON_LRINDEX) + break; + + if (!timeout--) + return -ETIMEDOUT; + udelay(100); + } + + return 0; +} + +/* + * Check whether CPU is the master or slave + */ +static inline int s3c24xx_snd_is_clkmaster(void) +{ + pr_debug("Entered %s\n", __func__); + + return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; +} + +/* + * Set S3C24xx I2S DAI format + */ +static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params r: IISMOD: %x \n", iismod); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iismod |= S3C2410_IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iismod &= ~S3C2410_IISMOD_SLAVE; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2410_IISMOD_MSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2410_IISMOD_MSB; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &s3c24xx_i2s_pcm_stereo_out; + else + dma_data = &s3c24xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); + + /* Working copies of register */ + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params r: IISMOD: %x\n", iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod &= ~S3C2410_IISMOD_16BIT; + dma_data->dma_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod |= S3C2410_IISMOD_16BIT; + dma_data->dma_size = 2; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params w: IISMOD: %x\n", iismod); + return 0; +} + +static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + pr_debug("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!s3c24xx_snd_is_clkmaster()) { + ret = s3c24xx_snd_lrsync(); + if (ret) + goto exit_err; + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c24xx_snd_rxctrl(1); + else + s3c24xx_snd_txctrl(1); + + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c24xx_snd_rxctrl(0); + else + s3c24xx_snd_txctrl(0); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* + * Set S3C24xx Clock source + */ +static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + + pr_debug("Entered %s\n", __func__); + + iismod &= ~S3C2440_IISMOD_MPLL; + + switch (clk_id) { + case S3C24XX_CLKSRC_PCLK: + break; + case S3C24XX_CLKSRC_MPLL: + iismod |= S3C2440_IISMOD_MPLL; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + return 0; +} + +/* + * Set S3C24xx Clock dividers + */ +static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + u32 reg; + + pr_debug("Entered %s\n", __func__); + + switch (div_id) { + case S3C24XX_DIV_BCLK: + reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; + writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); + break; + case S3C24XX_DIV_MCLK: + reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); + writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); + break; + case S3C24XX_DIV_PRESCALER: + writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR); + reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * To avoid duplicating clock code, allow machine driver to + * get the clockrate from here. + */ +u32 s3c24xx_i2s_get_clockrate(void) +{ + return clk_get_rate(s3c24xx_i2s.iis_clk); +} +EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); + +static int s3c24xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + pr_debug("Entered %s\n", __func__); + + s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); + if (s3c24xx_i2s.regs == NULL) + return -ENXIO; + + s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); + if (s3c24xx_i2s.iis_clk == NULL) { + pr_err("failed to get iis_clock\n"); + iounmap(s3c24xx_i2s.regs); + return -ENODEV; + } + clk_enable(s3c24xx_i2s.iis_clk); + + /* Configure the I2S pins in correct mode */ + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + + writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); + + s3c24xx_snd_txctrl(0); + s3c24xx_snd_rxctrl(0); + + return 0; +} + +#ifdef CONFIG_PM +static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) +{ + pr_debug("Entered %s\n", __func__); + + s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); + + clk_disable(s3c24xx_i2s.iis_clk); + + return 0; +} + +static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) +{ + pr_debug("Entered %s\n", __func__); + clk_enable(s3c24xx_i2s.iis_clk); + + writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR); + + return 0; +} +#else +#define s3c24xx_i2s_suspend NULL +#define s3c24xx_i2s_resume NULL +#endif + + +#define S3C24XX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { + .trigger = s3c24xx_i2s_trigger, + .hw_params = s3c24xx_i2s_hw_params, + .set_fmt = s3c24xx_i2s_set_fmt, + .set_clkdiv = s3c24xx_i2s_set_clkdiv, + .set_sysclk = s3c24xx_i2s_set_sysclk, +}; + +struct snd_soc_dai s3c24xx_i2s_dai = { + .name = "s3c24xx-i2s", + .id = 0, + .probe = s3c24xx_i2s_probe, + .suspend = s3c24xx_i2s_suspend, + .resume = s3c24xx_i2s_resume, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C24XX_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C24XX_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c24xx_i2s_dai_ops, +}; +EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); + +static int __init s3c24xx_i2s_init(void) +{ + return snd_soc_register_dai(&s3c24xx_i2s_dai); +} +module_init(s3c24xx_i2s_init); + +static void __exit s3c24xx_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c24xx_i2s_dai); +} +module_exit(s3c24xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx-i2s.h b/sound/soc/samsung/s3c24xx-i2s.h new file mode 100644 index 0000000..726d91c --- /dev/null +++ b/sound/soc/samsung/s3c24xx-i2s.h @@ -0,0 +1,37 @@ +/* + * s3c24xx-i2s.c -- ALSA Soc Audio Layer + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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. + * + * Revision history + * 10th Nov 2006 Initial version. + */ + +#ifndef S3C24XXI2S_H_ +#define S3C24XXI2S_H_ + +/* clock sources */ +#define S3C24XX_CLKSRC_PCLK 0 +#define S3C24XX_CLKSRC_MPLL 1 + +/* Clock dividers */ +#define S3C24XX_DIV_MCLK 0 +#define S3C24XX_DIV_BCLK 1 +#define S3C24XX_DIV_PRESCALER 2 + +/* prescaler */ +#define S3C24XX_PRESCALE(a,b) \ + (((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT)) + +u32 s3c24xx_i2s_get_clockrate(void); + +extern struct snd_soc_dai s3c24xx_i2s_dai; + +#endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c new file mode 100644 index 0000000..4984754 --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -0,0 +1,394 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/i2c.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +static struct s3c24xx_audio_simtec_pdata *pdata; +static struct clk *xtal_clk; + +static int spk_gain; +static int spk_unmute; + +/** + * speaker_gain_get - read the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_gain; + return 0; +} + +/** + * speaker_gain_set - set the value of the speaker amp gain + * @value: The value to write. + */ +static void speaker_gain_set(int value) +{ + gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); + gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); +} + +/** + * speaker_gain_put - set the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + * + * Note, if the speaker amp is muted, then we do not set a gain value + * as at-least one of the ICs that is fitted will try and power up even + * if the main control is set to off. + */ +static int speaker_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = ucontrol->value.integer.value[0]; + + spk_gain = value; + + if (!spk_unmute) + speaker_gain_set(value); + + return 0; +} + +static const struct snd_kcontrol_new amp_gain_controls[] = { + SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, + speaker_gain_get, speaker_gain_put), +}; + +/** + * spk_unmute_state - set the unmute state of the speaker + * @to: zero to unmute, non-zero to ununmute. + */ +static void spk_unmute_state(int to) +{ + pr_debug("%s: to=%d\n", __func__, to); + + spk_unmute = to; + gpio_set_value(pdata->amp_gpio, to); + + /* if we're umuting, also re-set the gain */ + if (to && pdata->amp_gain[0] > 0) + speaker_gain_set(spk_gain); +} + +/** + * speaker_unmute_get - read the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_unmute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_unmute; + return 0; +} + +/** + * speaker_unmute_put - set the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + */ +static int speaker_unmute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + spk_unmute_state(ucontrol->value.integer.value[0]); + return 0; +} + +/* This is added as a manual control as the speaker amps create clicks + * when their power state is changed, which are far more noticeable than + * anything produced by the CODEC itself. + */ +static const struct snd_kcontrol_new amp_unmute_controls[] = { + SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0, + speaker_unmute_get, speaker_unmute_put), +}; + +void simtec_audio_init(struct snd_soc_codec *codec) +{ + if (pdata->amp_gpio > 0) { + pr_debug("%s: adding amp routes\n", __func__); + + snd_soc_add_controls(codec, amp_unmute_controls, + ARRAY_SIZE(amp_unmute_controls)); + } + + if (pdata->amp_gain[0] > 0) { + pr_debug("%s: adding amp controls\n", __func__); + snd_soc_add_controls(codec, amp_gain_controls, + ARRAY_SIZE(amp_gain_controls)); + } +} +EXPORT_SYMBOL_GPL(simtec_audio_init); + +#define CODEC_CLOCK 12000000 + +/** + * simtec_hw_params - update hardware parameters + * @substream: The audio substream instance. + * @params: The parameters requested. + * + * Update the codec data routing and configuration settings + * from the supplied data. + */ +static int simtec_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err( "%s: failed setting codec sysclk\n", __func__); + return ret; + } + + if (pdata->use_mpllin) { + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, + 0, SND_SOC_CLOCK_OUT); + + if (ret) { + pr_err("%s: failed to set MPLLin as clksrc\n", + __func__); + return ret; + } + } + + if (pdata->output_cdclk) { + int cdclk_scale; + + cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; + cdclk_scale--; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + cdclk_scale); + } + + return 0; +} + +static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) +{ + /* call any board supplied startup code, this currently only + * covers the bast/vr1000 which have a CPLD in the way of the + * LRCLK */ + if (pd->startup) + pd->startup(); + + return 0; +} + +static struct snd_soc_ops simtec_snd_ops = { + .hw_params = simtec_hw_params, +}; + +/** + * attach_gpio_amp - get and configure the necessary gpios + * @dev: The device we're probing. + * @pd: The platform data supplied by the board. + * + * If there is a GPIO based amplifier attached to the board, claim + * the necessary GPIO lines for it, and set default values. + */ +static int attach_gpio_amp(struct device *dev, + struct s3c24xx_audio_simtec_pdata *pd) +{ + int ret; + + /* attach gpio amp gain (if any) */ + if (pdata->amp_gain[0] > 0) { + ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain0\n"); + return ret; + } + + ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain1\n"); + gpio_free(pdata->amp_gain[0]); + return ret; + } + + gpio_direction_output(pd->amp_gain[0], 0); + gpio_direction_output(pd->amp_gain[1], 0); + } + + /* note, currently we assume GPA0 isn't valid amp */ + if (pdata->amp_gpio > 0) { + ret = gpio_request(pd->amp_gpio, "gpio-amp"); + if (ret) { + dev_err(dev, "cannot get amp gpio %d (%d)\n", + pd->amp_gpio, ret); + goto err_amp; + } + + /* set the amp off at startup */ + spk_unmute_state(0); + } + + return 0; + +err_amp: + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + return ret; +} + +static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) +{ + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + if (pd->amp_gpio > 0) + gpio_free(pd->amp_gpio); +} + +#ifdef CONFIG_PM +int simtec_audio_resume(struct device *dev) +{ + simtec_call_startup(pdata); + return 0; +} + +const struct dev_pm_ops simtec_audio_pmops = { + .resume = simtec_audio_resume, +}; +EXPORT_SYMBOL_GPL(simtec_audio_pmops); +#endif + +int __devinit simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev) +{ + struct platform_device *snd_dev; + int ret; + + socdev->card->dai_link->ops = &simtec_snd_ops; + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + + simtec_call_startup(pdata); + + xtal_clk = clk_get(&pdev->dev, "xtal"); + if (IS_ERR(xtal_clk)) { + dev_err(&pdev->dev, "could not get clkout0\n"); + return -EINVAL; + } + + dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); + + ret = attach_gpio_amp(&pdev->dev, pdata); + if (ret) + goto err_clk; + + snd_dev = platform_device_alloc("soc-audio", -1); + if (!snd_dev) { + dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(snd_dev, socdev); + socdev->dev = &snd_dev->dev; + + ret = platform_device_add(snd_dev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc-audio dev\n"); + goto err_pdev; + } + + platform_set_drvdata(pdev, snd_dev); + return 0; + +err_pdev: + platform_device_put(snd_dev); + +err_gpio: + detach_gpio_amp(pdata); + +err_clk: + clk_put(xtal_clk); + return ret; +} +EXPORT_SYMBOL_GPL(simtec_audio_core_probe); + +int __devexit simtec_audio_remove(struct platform_device *pdev) +{ + struct platform_device *snd_dev = platform_get_drvdata(pdev); + + platform_device_unregister(snd_dev); + + detach_gpio_amp(pdata); + clk_put(xtal_clk); + return 0; +} +EXPORT_SYMBOL_GPL(simtec_audio_remove); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec.h b/sound/soc/samsung/s3c24xx_simtec.h new file mode 100644 index 0000000..e18faee --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec.h @@ -0,0 +1,22 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.h + * + * Copyright 2009 Simtec Electronics + * + * 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. +*/ + +extern void simtec_audio_init(struct snd_soc_codec *codec); + +extern int simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev); + +extern int simtec_audio_remove(struct platform_device *pdev); + +#ifdef CONFIG_PM +extern const struct dev_pm_ops simtec_audio_pmops; +#define simtec_audio_pm &simtec_audio_pmops +#else +#define simtec_audio_pm NULL +#endif diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c new file mode 100644 index 0000000..bdf8951 --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -0,0 +1,153 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c + * + * Copyright 2009 Simtec Electronics + * + * 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/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic3x.h" + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Out", NULL), + SND_SOC_DAPM_LINE("GSM In", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_LINE("ZV", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ + + { "Headphone Jack", NULL, "HPLOUT" }, + { "Headphone Jack", NULL, "HPLCOM" }, + { "Headphone Jack", NULL, "HPROUT" }, + { "Headphone Jack", NULL, "HPRCOM" }, + + /* ZV connected to Line1 */ + + { "LINE1L", NULL, "ZV" }, + { "LINE1R", NULL, "ZV" }, + + /* Line In connected to Line2 */ + + { "LINE2L", NULL, "Line In" }, + { "LINE2R", NULL, "Line In" }, + + /* Microphone connected to MIC3R and MIC_BIAS */ + + { "MIC3L", NULL, "Mic Jack" }, + + /* GSM connected to MONO_LOUT and MIC3L (in) */ + + { "GSM Out", NULL, "MONO_LOUT" }, + { "MIC3L", NULL, "GSM In" }, + + /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are + * not using the DAPM to power it up and down as there it makes + * a click when powering up. */ +}; + +/** + * simtec_hermes_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_hermes_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct aic3x_setup_data codec_setup = { +}; + +static struct snd_soc_dai_link simtec_dai_aic33 = { + .name = "tlv320aic33", + .stream_name = "TLV320AIC33", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &aic3x_dai, + .init = simtec_hermes_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic33 = { + .name = "Simtec-Hermes", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic33, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic33 = { + .card = &snd_soc_machine_simtec_aic33, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &codec_setup, +}; + +static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) +{ + dev_info(&pd->dev, "probing....\n"); + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); +} + +static struct platform_driver simtec_audio_hermes_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-hermes-snd", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_hermes_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); + +static int __init simtec_hermes_modinit(void) +{ + return platform_driver_register(&simtec_audio_hermes_platdrv); +} + +static void __exit simtec_hermes_modexit(void) +{ + platform_driver_unregister(&simtec_audio_hermes_platdrv); +} + +module_init(simtec_hermes_modinit); +module_exit(simtec_hermes_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c new file mode 100644 index 0000000..185c0ac --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -0,0 +1,137 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * + * Copyright 2009 Simtec Electronics + * + * 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/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic23.h" + +/* supported machines: + * + * Machine Connections AMP + * ------- ----------- --- + * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) + * VR1000 HPOUT, LIN None + * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) + */ + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + { "Headphone Jack", NULL, "LHPOUT"}, + { "Headphone Jack", NULL, "RHPOUT"}, + + { "Line Out", NULL, "LOUT" }, + { "Line Out", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line In"}, + { "RLINEIN", NULL, "Line In"}, + + { "MICIN", NULL, "Mic Jack"}, +}; + +/** + * simtec_tlv320aic23_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link simtec_dai_aic23 = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &tlv320aic23_dai, + .init = simtec_tlv320aic23_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic23 = { + .name = "Simtec", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic23, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic23 = { + .card = &snd_soc_machine_simtec_aic23, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) +{ + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); +} + +static struct platform_driver simtec_audio_tlv320aic23_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-tlv320aic23", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_tlv320aic23_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); + +static int __init simtec_tlv320aic23_modinit(void) +{ + return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); +} + +static void __exit simtec_tlv320aic23_modexit(void) +{ + platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); +} + +module_init(simtec_tlv320aic23_modinit); +module_exit(simtec_tlv320aic23_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c new file mode 100644 index 0000000..052d596 --- /dev/null +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -0,0 +1,373 @@ +/* + * Modifications by Christian Pellegrin <chripell(a)evolware.org> + * + * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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/clk.h> +#include <linux/mutex.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda134x.h> +#include <sound/uda134x.h> + +#include <plat/regs-iis.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda134x.h" + + +/* #define ENFORCE_RATES 1 */ +/* + Unfortunately the S3C24XX in master mode has a limited capacity of + generating the clock for the codec. If you define this only rates + that are really available will be enforced. But be careful, most + user level application just want the usual sampling frequencies (8, + 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly + operation for embedded systems. So if you aren't very lucky or your + hardware engineer wasn't very forward-looking it's better to leave + this undefined. If you do so an approximate value for the requested + sampling rate in the range -/+ 5% will be chosen. If this in not + possible an error will be returned. +*/ + +static struct clk *xtal; +static struct clk *pclk; +/* this is need because we don't have a place where to keep the + * pointers to the clocks in each substream. We get the clocks only + * when we are actually using them so we don't block stuff like + * frequency change or oscillator power-off */ +static int clk_users; +static DEFINE_MUTEX(clk_lock); + +static unsigned int rates[33 * 2]; +#ifdef ENFORCE_RATES +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; +#endif + +static struct platform_device *s3c24xx_uda134x_snd_device; + +static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; +#ifdef ENFORCE_RATES + struct snd_pcm_runtime *runtime = substream->runtime; +#endif + + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + if (clk_users == 0) { + xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); + if (!xtal) { + printk(KERN_ERR "%s cannot get xtal\n", __func__); + ret = -EBUSY; + } else { + pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, + "pclk"); + if (!pclk) { + printk(KERN_ERR "%s cannot get pclk\n", + __func__); + clk_put(xtal); + ret = -EBUSY; + } + } + if (!ret) { + int i, j; + + for (i = 0; i < 2; i++) { + int fs = i ? 256 : 384; + + rates[i*33] = clk_get_rate(xtal) / fs; + for (j = 1; j < 33; j++) + rates[i*33 + j] = clk_get_rate(pclk) / + (j * fs); + } + } + } + clk_users += 1; + mutex_unlock(&clk_lock); + if (!ret) { +#ifdef ENFORCE_RATES + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); + if (ret < 0) + printk(KERN_ERR "%s cannot set constraints\n", + __func__); +#endif + } + return ret; +} + +static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) +{ + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + clk_users -= 1; + if (clk_users == 0) { + clk_put(xtal); + xtal = NULL; + clk_put(pclk); + pclk = NULL; + } + mutex_unlock(&clk_lock); +} + +static int s3c24xx_uda134x_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + int ret = 0; + int clk_source, fs_mode; + unsigned long rate = params_rate(params); + long err, cerr; + unsigned int div; + int i, bi; + + err = 999999; + bi = 0; + for (i = 0; i < 2*33; i++) { + cerr = rates[i] - rate; + if (cerr < 0) + cerr = -cerr; + if (cerr < err) { + err = cerr; + bi = i; + } + } + if (bi / 33 == 1) + fs_mode = S3C2410_IISMOD_256FS; + else + fs_mode = S3C2410_IISMOD_384FS; + if (bi % 33 == 0) { + clk_source = S3C24XX_CLKSRC_MPLL; + div = 1; + } else { + clk_source = S3C24XX_CLKSRC_PCLK; + div = bi % 33; + } + pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); + + clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; + pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); + + if ((err * 100 / rate) > 5) { + printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " + "too different from desired (%ld%%)\n", + err * 100 / rate); + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s3c24xx_uda134x_ops = { + .startup = s3c24xx_uda134x_startup, + .shutdown = s3c24xx_uda134x_shutdown, + .hw_params = s3c24xx_uda134x_hw_params, +}; + +static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { + .name = "UDA134X", + .stream_name = "UDA134X", + .codec_dai = &uda134x_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &s3c24xx_uda134x_ops, +}; + +static struct snd_soc_card snd_soc_s3c24xx_uda134x = { + .name = "S3C24XX_UDA134X", + .platform = &s3c24xx_soc_platform, + .dai_link = &s3c24xx_uda134x_dai_link, + .num_links = 1, +}; + +static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; + +static void setdat(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); +} + +static void setclk(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); +} + +static void setmode(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); +} + +static struct uda134x_platform_data s3c24xx_uda134x = { + .l3 = { + .setdat = setdat, + .setclk = setclk, + .setmode = setmode, + .data_hold = 1, + .data_setup = 1, + .clock_high = 1, + .mode_hold = 1, + .mode = 1, + .mode_setup = 1, + }, +}; + +static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { + .card = &snd_soc_s3c24xx_uda134x, + .codec_dev = &soc_codec_dev_uda134x, + .codec_data = &s3c24xx_uda134x, +}; + +static int s3c24xx_uda134x_setup_pin(int pin, char *fun) +{ + if (gpio_request(pin, "s3c24xx_uda134x") < 0) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "l3 %s pin already in use", fun); + return -EBUSY; + } + gpio_direction_output(pin, 0); + return 0; +} + +static int s3c24xx_uda134x_probe(struct platform_device *pdev) +{ + int ret; + + printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); + + s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; + if (s3c24xx_uda134x_l3_pins == NULL) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "unable to find platform data\n"); + return -ENODEV; + } + s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; + s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; + + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data, + "data") < 0) + return -EBUSY; + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk, + "clk") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + return -EBUSY; + } + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode, + "mode") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + return -EBUSY; + } + + s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_uda134x_snd_device) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(s3c24xx_uda134x_snd_device, + &s3c24xx_uda134x_snd_devdata); + s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; + ret = platform_device_add(s3c24xx_uda134x_snd_device); + if (ret) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); + platform_device_put(s3c24xx_uda134x_snd_device); + } + + return ret; +} + +static int s3c24xx_uda134x_remove(struct platform_device *pdev) +{ + platform_device_unregister(s3c24xx_uda134x_snd_device); + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + gpio_free(s3c24xx_uda134x_l3_pins->l3_mode); + return 0; +} + +static struct platform_driver s3c24xx_uda134x_driver = { + .probe = s3c24xx_uda134x_probe, + .remove = s3c24xx_uda134x_remove, + .driver = { + .name = "s3c24xx_uda134x", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c24xx_uda134x_init(void) +{ + return platform_driver_register(&s3c24xx_uda134x_driver); +} + +static void __exit s3c24xx_uda134x_exit(void) +{ + platform_driver_unregister(&s3c24xx_uda134x_driver); +} + + +module_init(s3c24xx_uda134x_init); +module_exit(s3c24xx_uda134x_exit); + +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c64xx-i2s-v4.c b/sound/soc/samsung/s3c64xx-i2s-v4.c new file mode 100644 index 0000000..06db130 --- /dev/null +++ b/sound/soc/samsung/s3c64xx-i2s-v4.c @@ -0,0 +1,209 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c + * + * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include <mach/gpio-bank-c.h> +#include <mach/gpio-bank-h.h> +#include <plat/gpio-cfg.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2Sv4 PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2Sv4 PCM Stereo in" +}; + +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out; +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in; +static struct s3c_i2sv2_info s3c64xx_i2sv4; + +struct snd_soc_dai s3c64xx_i2s_v4_dai; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai); + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2sv4_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* configure GPIO for i2s port */ + s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0); + s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1); + s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2); + s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI); + + return 0; +} + +static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + dev_dbg(cpu_dai->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod); + + iismod &= ~S3C64XX_IISMOD_BLC_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C64XX_IISMOD_BLC_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + iismod |= S3C64XX_IISMOD_BLC_24BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = { + .hw_params = s3c_i2sv4_hw_params, +}; + +static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + i2s = &s3c64xx_i2sv4; + dai = &s3c64xx_i2s_v4_dai; + + if (dai->dev) { + dev_dbg(dai->dev, "%s: \ + I2Sv4 instance already registered!\n", __func__); + return -EBUSY; + } + + dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s-v4"; + dai->id = 0; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2sv4_probe; + dai->ops = &s3c64xx_i2sv4_dai_ops; + + i2s->feature |= S3C_FEATURE_CDCLKCON; + + i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in; + i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out; + + i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX; + i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX; + i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD; + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + clk_enable(i2s->iis_cclk); + + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + if (ret) + goto err_clk; + + ret = s3c_i2sv2_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_i2sv4_driver = { + .probe = s3c64xx_i2sv4_dev_probe, + .remove = s3c64xx_i2sv4_dev_remove, + .driver = { + .name = "s3c64xx-iis-v4", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c64xx_i2sv4_init(void) +{ + return platform_driver_register(&s3c64xx_i2sv4_driver); +} +module_init(s3c64xx_i2sv4_init); + +static void __exit s3c64xx_i2sv4_exit(void) +{ + platform_driver_unregister(&s3c64xx_i2sv4_driver); +} +module_exit(s3c64xx_i2sv4_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c64xx-i2s.c b/sound/soc/samsung/s3c64xx-i2s.c new file mode 100644 index 0000000..1d85cb8 --- /dev/null +++ b/sound/soc/samsung/s3c64xx-i2s.c @@ -0,0 +1,190 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.c + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.simtec.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/clk.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/soc.h> + +#include <mach/gpio-bank-d.h> +#include <mach/gpio-bank-e.h> +#include <plat/gpio-cfg.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c64xx-i2s.h" + +/* The value should be set to maximum of the total number + * of I2Sv3 controllers that any supported SoC has. + */ +#define MAX_I2SV3 2 + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[MAX_I2SV3]; +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[MAX_I2SV3]; +static struct s3c_i2sv2_info s3c64xx_i2s[MAX_I2SV3]; + +struct snd_soc_dai s3c64xx_i2s_dai[MAX_I2SV3]; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* configure GPIO for i2s port */ + switch (dai->id) { + case 0: + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); + break; + case 1: + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); + } + + return 0; +} + + +static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops; + +static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + if (pdev->id >= MAX_I2SV3) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + i2s = &s3c64xx_i2s[pdev->id]; + dai = &s3c64xx_i2s_dai[pdev->id]; + dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s"; + dai->id = pdev->id; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2s_probe; + dai->ops = &s3c64xx_i2s_dai_ops; + + i2s->feature |= S3C_FEATURE_CDCLKCON; + + i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; + i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + + if (pdev->id == 0) { + i2s->dma_capture->channel = DMACH_I2S0_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S0_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; + } else { + i2s->dma_capture->channel = DMACH_I2S1_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S1_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD; + } + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + clk_enable(i2s->iis_cclk); + + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + if (ret) + goto err_clk; + + ret = s3c_i2sv2_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_iis_driver = { + .probe = s3c64xx_iis_dev_probe, + .remove = s3c64xx_iis_dev_remove, + .driver = { + .name = "s3c64xx-iis", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c64xx_i2s_init(void) +{ + return platform_driver_register(&s3c64xx_iis_driver); +} +module_init(s3c64xx_i2s_init); + +static void __exit s3c64xx_i2s_exit(void) +{ + platform_driver_unregister(&s3c64xx_iis_driver); +} +module_exit(s3c64xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c64xx-i2s.h b/sound/soc/samsung/s3c64xx-i2s.h new file mode 100644 index 0000000..7a40f43 --- /dev/null +++ b/sound/soc/samsung/s3c64xx-i2s.h @@ -0,0 +1,42 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.h + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.simtec.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 __SND_SOC_S3C24XX_S3C64XX_I2S_H +#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ + +struct clk; + +#include "s3c-i2s-v2.h" + +#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C64XX_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK +#define S3C64XX_CLKSRC_MUX S3C_I2SV2_CLKSRC_AUDIOBUS +#define S3C64XX_CLKSRC_CDCLK S3C_I2SV2_CLKSRC_CDCLK + +#define S3C64XX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define S3C64XX_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +extern struct snd_soc_dai s3c64xx_i2s_dai[]; +extern struct snd_soc_dai s3c64xx_i2s_v4_dai; + +#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c new file mode 100644 index 0000000..3622588 --- /dev/null +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -0,0 +1,81 @@ +/* + * smdk2443_wm9710.c -- SoC audio for smdk2443 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk2443; + +static struct snd_soc_dai_link smdk2443_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_card smdk2443 = { + .name = "SMDK2443", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk2443_dai, + .num_links = ARRAY_SIZE(smdk2443_dai), +}; + +static struct snd_soc_device smdk2443_snd_ac97_devdata = { + .card = &smdk2443, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *smdk2443_snd_ac97_device; + +static int __init smdk2443_init(void) +{ + int ret; + + smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk2443_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk2443_snd_ac97_device, + &smdk2443_snd_ac97_devdata); + smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev; + ret = platform_device_add(smdk2443_snd_ac97_device); + + if (ret) + platform_device_put(smdk2443_snd_ac97_device); + + return ret; +} + +static void __exit smdk2443_exit(void) +{ + platform_device_unregister(smdk2443_snd_ac97_device); +} + +module_init(smdk2443_init); +module_exit(smdk2443_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme.gregory(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk64xx_wm8580.c b/sound/soc/samsung/smdk64xx_wm8580.c new file mode 100644 index 0000000..07e8e51 --- /dev/null +++ b/sound/soc/samsung/smdk64xx_wm8580.c @@ -0,0 +1,266 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.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/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. + * This criterion can't be met if we request PLL output + * as {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a minimum value that + * results in (params_rate(params) * rfs), and itself, acceptable + * to both - the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from its PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* Enabling the microphone requires the fitting of a 0R + * resistor to connect the line from the microphone jack. + */ + snd_soc_dapm_disable_pin(codec, "MicIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_v4_dai, + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_v4_dai, + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c new file mode 100644 index 0000000..24fd39f --- /dev/null +++ b/sound/soc/samsung/smdk_wm9713.c @@ -0,0 +1,94 @@ +/* + * smdk_wm9713.c -- SoC audio for SMDK + * + * Copyright 2010 Samsung Electronics Co. Ltd. + * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.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/device.h> +#include <sound/soc.h> + +#include "../codecs/wm9713.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk; + +/* + * Default CFG switch settings to use this driver: + * + * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off + */ + +/* + Playback (HeadPhone):- + $ amixer sset 'Headphone' unmute + $ amixer sset 'Right Headphone Out Mux' 'Headphone' + $ amixer sset 'Left Headphone Out Mux' 'Headphone' + $ amixer sset 'Right HP Mixer PCM' unmute + $ amixer sset 'Left HP Mixer PCM' unmute + + Capture (LineIn):- + $ amixer sset 'Right Capture Source' 'Line' + $ amixer sset 'Left Capture Source' 'Line' +*/ + +static struct snd_soc_dai_link smdk_dai = { + .name = "AC97", + .stream_name = "AC97 PCM", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], +}; + +static struct snd_soc_card smdk = { + .name = "SMDK", + .platform = &s3c24xx_soc_platform, + .dai_link = &smdk_dai, + .num_links = 1, +}; + +static struct snd_soc_device smdk_snd_ac97_devdata = { + .card = &smdk, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *smdk_snd_ac97_device; + +static int __init smdk_init(void) +{ + int ret; + + smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk_snd_ac97_device, + &smdk_snd_ac97_devdata); + smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; + + ret = platform_device_add(smdk_snd_ac97_device); + if (ret) + platform_device_put(smdk_snd_ac97_device); + + return ret; +} + +static void __exit smdk_exit(void) +{ + platform_device_unregister(smdk_snd_ac97_device); +} + +module_init(smdk_init); +module_exit(smdk_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

2 1

[alsa-devel] Writing a driver for the Blackfin BF537 with two CS42448 on SPORT1
by Adam Rosenberg 25 Jun '10

25 Jun '10

I am writing my first ALSA driver for uClinux on Blackfin BF537. I have a custom board with two Cirrus Logic CS42448 chips connected to the primary and secondary data lines of SPORT1. I was basing my driver off of the driver for the Analog Devices AD1836 but it does not look like that driver was written using the latest practices. I already have the code modified well enough to communicate with both chips and produce some noise, but it is obvious that I am not filling the DMA buffer properly. Please let me know where I can find a driver that would be a better example for me to reference. When I have completed the driver I will also need help on where to submit it so that it will be supported by ALSA for others. Thank you for your help, Adam Adam Rosenberg Software Engineer Alcorn McBride Inc. 3300 South Hiawassee Building 105 Orlando, FL 32835 (407) 296 - 5800 ext. 5490

2 1

[alsa-devel] [PATCH] eukrea-tlv320: add support for our i.MX25 board
by Eric Bénard 25 Jun '10

25 Jun '10

* tdm slot has to be configured to get sound working on i.MX25 Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index fd292f1..fd616ae 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -31,7 +31,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 bool "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD + depends on MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD select SND_MXC_SOC_SSI select SND_SOC_TLV320AIC23 help diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 968380a..45f5e4b 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -60,6 +60,7 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, pr_err("%s: failed setting codec sysclk\n", __func__); return ret; } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); @@ -100,7 +101,7 @@ static int __init eukrea_tlv320_init(void) { int ret; - if (!machine_is_eukrea_cpuimx27()) + if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()) /* return happy. We might run on a totally different machine */ return 0; -- 1.6.3.3

3 13

[alsa-devel] [PATCH 1/2] ASoC: uda134x: replace a macro with a value in platform struct.
by Vladimir Zapolskiy 25 Jun '10

25 Jun '10

This change wipes out a hardcoded macro, which enables codec bias level control. Now is_powered_on_standby value shall be used instead. Signed-off-by: Vladimir Zapolskiy <vzapolskiy(a)gmail.com> --- include/sound/uda134x.h | 12 ++++++++++++ sound/soc/codecs/uda134x.c | 21 +++++---------------- 2 files changed, 17 insertions(+), 16 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index 509efb0..e475659 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -18,6 +18,18 @@ struct uda134x_platform_data { struct l3_pins l3; void (*power) (int); int model; + /* + ALSA SOC usually puts the device in standby mode when it's not used + for sometime. If you unset is_powered_on_standby the driver will + turn off the ADC/DAC when this callback is invoked and turn it back + on when needed. Unfortunately this will result in a very light bump + (it can be audible only with good earphones). If this bothers you + set is_powered_on_standby, you will have slightly higher power + consumption. Please note that sending the L3 command for ADC is + enough to make the bump, so it doesn't make difference if you + completely take off power from the codec. + */ + int is_powered_on_standby; #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index e8066b6..5c7b054 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -27,19 +27,6 @@ #include "uda134x.h" -#define POWER_OFF_ON_STANDBY 1 -/* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you define POWER_OFF_ON_STANDBY the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - just comment this line, you will have slightly higher power - consumption . Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - #define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 #define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) @@ -559,9 +546,11 @@ static int uda134x_soc_probe(struct platform_device *pdev) codec->num_dai = 1; codec->read = uda134x_read_reg_cache; codec->write = uda134x_write; -#ifdef POWER_OFF_ON_STANDBY - codec->set_bias_level = uda134x_set_bias_level; -#endif + + if (!pd->is_powered_on_standby) { + codec->set_bias_level = uda134x_set_bias_level; + } + INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); -- 1.7.0.4

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Re: [alsa-devel] redirecting audio to another desktop machine
by Hans-Kristian Arntzen 25 Jun '10

25 Jun '10

On 06/21/2010 12:00 PM, alsa-devel-request(a)alsa-project.org wrote: > Date: Mon, 21 Jun 2010 14:57:00 +0530 > From: Viral Mehta<Viral.Mehta(a)lntinfotech.com> > Subject: [alsa-devel] redirecting audio to another desktop machine > To: "alsa-devel(a)alsa-project.org"<alsa-devel(a)alsa-project.org> > Message-ID: > <70376CA23424B34D86F1C7DE6B9973430254343B3E(a)VSHINMSMBX01.vshodc.lntinfotech.com> > > Content-Type: text/plain; charset="us-ascii" > > Hi list, > > I have an embedded device running linux kernel and a minimal filesystem. > > I want to access this device remotely which means I should be able to connect to the device just like we do Remote Desktop over ethernet. > I am able to do this. I am able to send key-event and framebuffer data whenever my desktop machine connects to the device. > > But, I am not able to send audio data on remote side. > I am really not sure which device file (/dev/snd/ ???) I should read and capture the data and should send across.... > Or can I read the playback device ? I think NO. > Do I need to create some kind of virtual sound device ? > > Any pointers/suggestions are appreciated. > > Thanks, > Viral > > > If you want to reroute an ALSA application over the network, you can use output plugins. Afaik, both PulseAudio and RSound provides plugins for this. You generally want to set up a virtual device in .asoundrc or similar. -- Hans-Kristian Arntzen

3 5

[alsa-devel] AndreaElectronics PureAudio USB-SA
by rmfc 25 Jun '10

25 Jun '10

Hello all, I'm trying to use any USB sound card with some DSP processing that allows a microphone array to capture audio and use in speech recognition (mostly applied in noisy environments). The best device that I found so far is AndreaElectronics PureAudio USB-SA but unfortunately it isn't properly supported in Linux/Alsa. The device is recognized as an USB Audio device (snd_usb_audio) and I can record audio in Audacity. The problem is that none of DSP features is explicitly activated in USB configuration so it works almost like any other microphone. I then tried Asus Xonar U1 but the problem is the same, no configuration is done in such devices, so they record audio without enough quality to be used by a speech recognizer (not enough energy, audio DC components when software gain is applied in Sound Preferences...). Is there any way to configure an USB Audio device interfaces? Here's the output of "lsusb" of AndreaElectronics USB-SA sound card. ------------------------------------------------------------ Bus 003 Device 002: ID 08a8:000c Andrea Electronics Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x08a8 Andrea Electronics idProduct 0x000c bcdDevice 1.00 iManufacturer 3 Andrea Electronics iProduct 1 Andrea PureAudio USB-SA iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 293 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 128 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 9 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 8 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 8 bNrInPins 3 baSource( 0) 10 baSource( 1) 11 baSource( 2) 12 iSelector 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 1 bControlSize 2 bmaControls( 0) 0x01 bmaControls( 0) 0x02 Mute Loudness bmaControls( 1) 0x02 bmaControls( 1) 0x00 Volume bmaControls( 2) 0x02 bmaControls( 2) 0x00 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 3 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 12 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 50 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Device Status: 0x0000 (Bus Powered) ------------------------------------------------------------ Renato

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Added support for stereo I2S
by Raffaele Recalcati 25 Jun '10

25 Jun '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added audio playback support with [frame sync master - clock master] mode and with [frame sync master - clock slave]. Clock slave can be important when the external codec need system clock and bit clock synchronized. In the clock master case there is a FIXME message in the source code, because we (Davide and myself) have guessed a frequency of 122000000 that seems the base to be divided. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 7 ++ sound/soc/davinci/davinci-i2s.c | 141 ++++++++++++++++++++++++++---- 2 files changed, 129 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..b1faeb9 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,13 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not simmetrical, but sampling + * frequency is better approximated + */ + int i2s_fast_clock; + /* McASP specific fields */ int tdm_slots; u8 op_mode; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..8811d25 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -68,16 +68,23 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE (1 << 31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE (1 << 31) + +#define CLKGDV(v) (v) /* Bits 0:7 */ #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM (1 << 29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,8 +151,17 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + int i2s_fast_clock; +}; + +struct davinci_mcbsp_data { + unsigned int fmt; + int clk_div; }; +static struct davinci_mcbsp_data mcbsp_data; + static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, int reg, u32 val) { @@ -255,24 +271,27 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int pcr; unsigned int srgr; srgr = DAVINCI_MCBSP_SRGR_FSGM | - DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | - DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | + DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + /* Attention srgr is updated by hw_params! */ + mcbsp_data.fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: case SND_SOC_DAIFMT_CBS_CFS: /* cpu is master */ pcr = DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM | - DAVINCI_MCBSP_PCR_CLKXM | - DAVINCI_MCBSP_PCR_CLKRM; + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: /* McBSP CLKR pin is the input for the Sample Rate Generator. * McBSP FSR and FSX are driven by the Sample Rate Generator. */ pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -372,6 +391,17 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + int srgr; + + mcbsp_data.clk_div = div; + return 0; +} + + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,11 +410,12 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; + struct clk *clk; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -396,12 +427,60 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = mcbsp_data.fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + if (master == SND_SOC_DAIFMT_CBS_CFS) { + clk = clk_get(NULL, "pll1_sysclk6"); + if (clk) + freq = clk_get_rate(clk); + freq = 122000000; /* FIXME ask to Texas */ + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* simmetric waveforms */ + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + } + clk_div &= 0xFF; + srgr |= clk_div; + } else if (master == SND_SOC_DAIFMT_CBS_CFM) { + /* Clock given on CLKS */ + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = mcbsp_data.clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + } else { + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,22 +505,45 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + } else { + /* FIXME ask to Texas */ + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } else { + /* FIXME ask to Texas */ + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); xcr |= DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,7 +602,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, - + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; struct snd_soc_dai davinci_i2s_dai = { @@ -548,6 +650,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) } if (pdata) { dev->enable_channel_combine = pdata->enable_channel_combine; + dev->i2s_fast_clock = pdata->i2s_fast_clock; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = -- 1.7.0.4

3 3

[alsa-devel] Compilation error
by lars.bjorndal＠broadpark.no 25 Jun '10

25 Jun '10

I get the following output while trying to compile the alsa driver on my Fedora 12 system. It worked well with the previous kernel version. # ./alsa-compile.sh --install File environment.base is cached. > cd /tmp/alsa-compile-script File environment.compile is cached. alsa-driver tree alsa-driver-1.0.23.31.g2379a.273.g5b97d is present. Reusing it. Use './alsa-compile.sh --clean=alsa-driver' command to refetch and rebuild. Sources unpacked to alsa-driver-1.0.23.31.g2379a.273.g5b97d > cd alsa-driver-1.0.23.31.g2379a.273.g5b97d Running make: make dep make[1]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d' make[2]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include' make -C sound prepare make[3]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make prepare2 make[4]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[4]: Nothing to be done for `prepare2'. make[4]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[3]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[2]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include' ... make[2]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/misc' make[2]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/misc' make[1]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d' make -C /lib/modules/2.6.32.14-127.fc12.i686.PAE/build SUBDIRS=/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/kernels/2.6.32.14-127.fc12.i686.PAE' CC [M] /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.o In file included from /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.c:2: /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'alloc_stream_buffers': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1071: error: implicit declaration of function 'usb_alloc_coherent' /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1072: warning: assignment makes pointer from integer without a cast /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'free_stream_buffers': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1092: error: implicit declaration of function 'usb_free_coherent' /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'ua101_card_free': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1172: warning: unused variable 'ua' make[4]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.o] Error 1 make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc] Error 2 make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb] Error 2 make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.32.14-127.fc12.i686.PAE' make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. Lars

1 0

[alsa-devel] Verifying mixer dB data/Invalid dB data from USB cards, especially Aureon 5.1 MkII
by Lennart Poettering 25 Jun '10

25 Jun '10

Heya, a while back I wrote a little tool "dbmeasure" which can be used to determine dB attenuation data for alsa mixer controls. I have now extended this little dB tools suite to include a tiny tool "dbverify" that can be used to verify the correctness of existing dB data exposed by the drivers. It should be relatively easy to use that tool, suitable even for non-guru folks who want to debug the dB information from alsa. http://git.0pointer.de/?p=dbmeasure.git git://git.0pointer.de/dbmeasure.git Just run "make" in a git checkout and then try: $ ./dbverify Aureon51MkII Master 30 200 That will verify the dB data of the "Master" control of the sound card that goes by the id "Aureon51MkII". It will compare the discrete volume steps 30 and 200: first it will play a full amplitude sine wave at mixer volume step 30, and then will change to mixer volume step 200, but attenuate the sine wave in software compensating for the volume change. So once you hear the sine wave attenuated by the hw, and once by the software. If the card's dB data is correct both sines should have the same volume. Now, playing around with this, I could verify what I already mentioned to Jarsolav elsewhere earlier: it seems that particularly USB cards seem to expose invalid dB data (at least all mine do). For example, for the above mentioned Aureon 5.1 it's really *way* off -- if you posess that card just run the line above, and listen to the difference. It's really bad. OTOH the dB data of the HDA cards I own is pretty accurate as it seems, at least I was unable to decide just by listening which sine wave was attenuated by sw, and which by hw. Getting back to the invalid dB data from the USB cards: the question is whether the USB descriptor data is badly parsed, and the dB mismatch hence systematic for USB cards, or if these cards are just crappy and include invalid data? In which case I wonder what we could do about that? The Aureon's dB range already looks suspicous, since both the max and the min dB value are < 0 (-47.69 to -1.97), so maybe we could add some heuristics to filter out data that already looks suspicous? Invalid dB data from the driver is a real problem for the "flat volume" logic in PA. We basically allow each app to control the full hw volume range individually, and then set the hw volume to the max of what all apps wanted and attenuate the other streams accordingly. On the Aureon this doesn't work at all, since the attenuation of the streams is miscalculated due to the invalid dB data. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

8 93

[alsa-devel] [RFC 5/5] ASoC multi-component support : jack
by Liam Girdwood 24 Jun '10

24 Jun '10

Cache configuration is now stored as part of the codec driver structure. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/soc-cache.c | 34 +++++++++++++++++----------------- 1 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 472af38..83cd8ed 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -19,7 +19,7 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -31,12 +31,12 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[2]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg << 4) | ((value >> 8) & 0x000f); data[1] = value & 0x00ff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -89,7 +89,7 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -101,12 +101,12 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[2]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -161,13 +161,13 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, u8 *cache = codec->reg_cache; u8 data[2]; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); reg &= 0xff; data[0] = reg; data[1] = value & 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -188,7 +188,7 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, { u8 *cache = codec->reg_cache; reg &= 0xff; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -224,7 +224,7 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size || + if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -EINVAL; @@ -343,7 +343,7 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; reg &= 0xff; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -355,14 +355,14 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[3]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg >> 8) & 0xff; data[1] = reg & 0xff; data[2] = value; reg &= 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -451,7 +451,7 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size || + if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -EINVAL; @@ -474,7 +474,7 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = (value >> 8) & 0xff; data[3] = value & 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -571,8 +571,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return -EINVAL; } - codec->write = io_types[i].write; - codec->read = io_types[i].read; + codec->driver->write = io_types[i].write; + codec->driver->read = io_types[i].read; switch (control) { case SND_SOC_CUSTOM: -- 1.7.0.4

1 0

[alsa-devel] [RFC 4/5] ASoC multi-component support : jack
by Liam Girdwood 24 Jun '10

24 Jun '10

Jack interface is per codec rather than per card in order to support sound cards with more than one codec. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/soc-jack.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 29159e1..1dfc4fd 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -32,14 +32,14 @@ * Returns zero if successful, or a negative error code on failure. * On success jack will be initialised. */ -int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, +int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack) { - jack->card = card; + jack->codec = codec; INIT_LIST_HEAD(&jack->pins); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(card->codec->card, id, type, &jack->jack); + return snd_jack_new(codec->snd_card, id, type, &jack->jack); } EXPORT_SYMBOL_GPL(snd_soc_jack_new); @@ -67,7 +67,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) if (!jack) return; - codec = jack->card->codec; + codec = jack->codec; mutex_lock(&codec->mutex); @@ -268,7 +268,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, ret = request_irq(gpio_to_irq(gpios[i].gpio), gpio_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, - jack->card->dev->driver->name, + jack->codec->dev->driver->name, &gpios[i]); if (ret) goto err; -- 1.7.0.4

1 0

[alsa-devel] [RFC 2/5] ASoC multi-component support : DAPM
by Liam Girdwood 24 Jun '10

24 Jun '10

This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Also added void snd_soc_dapm_shutdown() to shutdown all CODECs in a card. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc-dapm.h | 8 ++-- sound/soc/soc-dapm.c | 82 +++++++++++++++++++++++----------------------- 2 files changed, 45 insertions(+), 45 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c5d9987..c4a4456 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -322,14 +322,14 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec); -void snd_soc_dapm_free(struct snd_soc_device *socdev); +void snd_soc_dapm_free(struct snd_soc_codec *codec); int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, const struct snd_soc_dapm_route *route, int num); /* dapm events */ -int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, - int event); -void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); +int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, + const char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 03cb7c0..a0d143d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -112,43 +112,41 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( /** * snd_soc_dapm_set_bias_level - set the bias level for the system - * @socdev: audio device + * @card: audio device * @level: level to configure * * Configure the bias (power) levels for the SoC audio device. * * Returns 0 for success else error. */ -static int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, - enum snd_soc_bias_level level) +static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card, + struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; switch (level) { case SND_SOC_BIAS_ON: - dev_dbg(socdev->dev, "Setting full bias\n"); + dev_dbg(codec->dev, "Setting full bias\n"); break; case SND_SOC_BIAS_PREPARE: - dev_dbg(socdev->dev, "Setting bias prepare\n"); + dev_dbg(codec->dev, "Setting bias prepare\n"); break; case SND_SOC_BIAS_STANDBY: - dev_dbg(socdev->dev, "Setting standby bias\n"); + dev_dbg(codec->dev, "Setting standby bias\n"); break; case SND_SOC_BIAS_OFF: - dev_dbg(socdev->dev, "Setting bias off\n"); + dev_dbg(codec->dev, "Setting bias off\n"); break; default: - dev_err(socdev->dev, "Setting invalid bias %d\n", level); + dev_err(codec->dev, "Setting invalid bias %d\n", level); return -EINVAL; } - if (card->set_bias_level) + if (card && card->set_bias_level) ret = card->set_bias_level(card, level); if (ret == 0) { - if (codec->set_bias_level) - ret = codec->set_bias_level(codec, level); + if (codec->driver->set_bias_level) + ret = codec->driver->set_bias_level(codec, level); else codec->bias_level = level; } @@ -370,7 +368,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, path->long_name); - ret = snd_ctl_add(codec->card, path->kcontrol); + ret = snd_ctl_add(codec->snd_card, path->kcontrol); if (ret < 0) { printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n", path->long_name, @@ -398,7 +396,7 @@ static int dapm_new_mux(struct snd_soc_codec *codec, } kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); - ret = snd_ctl_add(codec->card, kcontrol); + ret = snd_ctl_add(codec->snd_card, kcontrol); if (ret < 0) goto err; @@ -893,7 +891,7 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, */ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) { - struct snd_soc_device *socdev = codec->socdev; + struct snd_soc_card *card = codec->card; struct snd_soc_dapm_widget *w; LIST_HEAD(up_list); LIST_HEAD(down_list); @@ -966,7 +964,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_dapm_set_bias_level(socdev, + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_STANDBY); if (ret != 0) pr_err("Failed to turn on bias: %d\n", ret); @@ -975,8 +973,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* If we're changing to all on or all off then prepare */ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { - ret = snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_PREPARE); + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_PREPARE); if (ret != 0) pr_err("Failed to prepare bias: %d\n", ret); } @@ -989,8 +986,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* If we just powered the last thing off drop to standby bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) { - ret = snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_STANDBY); + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_STANDBY); if (ret != 0) pr_err("Failed to apply standby bias: %d\n", ret); } @@ -998,15 +994,14 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* If we're in standby and can support bias off then do that */ if (codec->bias_level == SND_SOC_BIAS_STANDBY && codec->idle_bias_off) { - ret = snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF); if (ret != 0) pr_err("Failed to turn off bias: %d\n", ret); } /* If we just powered up then move to active bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { - ret = snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_ON); + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_ON); if (ret != 0) pr_err("Failed to apply active bias: %d\n", ret); } @@ -1188,8 +1183,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, static ssize_t dapm_widget_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); - struct snd_soc_codec *codec = devdata->card->codec; + struct snd_soc_codec *codec = dev_get_drvdata(dev); struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; @@ -1998,9 +1992,10 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); * * Returns 0 for success else error. */ -int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, - char *stream, int event) +int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, + const char *stream, int event) { + struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_widget *w; if (stream == NULL) @@ -2168,25 +2163,19 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); /** * snd_soc_dapm_free - free dapm resources - * @socdev: SoC device + * @card: SoC device * * Free all dapm widgets and resources. */ -void snd_soc_dapm_free(struct snd_soc_device *socdev) +void snd_soc_dapm_free(struct snd_soc_codec *codec) { - struct snd_soc_codec *codec = socdev->card->codec; - - snd_soc_dapm_sys_remove(socdev->dev); + snd_soc_dapm_sys_remove(codec->dev); dapm_free_widgets(codec); } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); -/* - * snd_soc_dapm_shutdown - callback for system shutdown - */ -void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) +static void soc_dapm_shutdown_codec(struct snd_soc_codec *codec) { - struct snd_soc_codec *codec = socdev->card->codec; struct snd_soc_dapm_widget *w; LIST_HEAD(down_list); int powerdown = 0; @@ -2203,12 +2192,23 @@ void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); + snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_PREPARE); dapm_seq_run(codec, &down_list, 0, dapm_down_seq); - snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); + snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_STANDBY); } +} + +/* + * snd_soc_dapm_shutdown - callback for system shutdown + */ +void snd_soc_dapm_shutdown(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, list) + soc_dapm_shutdown_codec(codec); - snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF); } /* Module information */ -- 1.7.0.4

1 0

[alsa-devel] [RFC 1/5] ASoC multi-component support : core API
by Liam Girdwood 24 Jun '10

24 Jun '10

This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc-dai.h | 99 ++++++++++++------- include/sound/soc.h | 246 +++++++++++++++++++++++++++++------------------ sound/soc/soc-core.c | 2 + 3 files changed, 218 insertions(+), 129 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 377693a..495f0aa 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -91,15 +91,17 @@ struct snd_pcm_substream; SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) -struct snd_soc_dai_ops; +struct snd_soc_dai_driver; struct snd_soc_dai; struct snd_ac97_bus_ops; /* Digital Audio Interface registration */ -int snd_soc_register_dai(struct snd_soc_dai *dai); -void snd_soc_unregister_dai(struct snd_soc_dai *dai); -int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count); -void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count); +int snd_soc_register_dai(struct device *dev, int id, + struct snd_soc_dai_driver *dai_drv); +void snd_soc_unregister_dai(struct device *dev, int id); +int snd_soc_register_dais(struct device *dev, + struct snd_soc_dai_driver *dai_drv, size_t count); +void snd_soc_unregister_dais(struct device *dev, size_t count); /* Digital Audio Interface clocking API.*/ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, @@ -126,16 +128,6 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); -/* - * Digital Audio Interface. - * - * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 - * operations and capabilities. Codec and platform drivers will register this - * structure for every DAI they have. - * - * This structure covers the clocking, formating and ALSA operations for each - * interface. - */ struct snd_soc_dai_ops { /* * DAI clocking configuration, all optional. @@ -191,24 +183,24 @@ struct snd_soc_dai_ops { }; /* - * Digital Audio Interface runtime data. + * Digital Audio Interface Driver. * - * Holds runtime data for a DAI. + * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 + * operations and capabilities. Codec and platform drivers will register this + * structure for every DAI they have. + * + * This structure covers the clocking, formating and ALSA operations for each + * interface. */ -struct snd_soc_dai { +struct snd_soc_dai_driver { /* DAI description */ - char *name; - unsigned int id; + const char *name; + struct module *owner; int ac97_control; - struct device *dev; - void *ac97_pdata; /* platform_data for the ac97 codec */ - - /* DAI callbacks */ - int (*probe)(struct platform_device *pdev, - struct snd_soc_dai *dai); - void (*remove)(struct platform_device *pdev, - struct snd_soc_dai *dai); + /* DAI driver callbacks */ + int (*probe)(struct snd_soc_dai *dai); + int (*remove)(struct snd_soc_dai *dai); int (*suspend)(struct snd_soc_dai *dai); int (*resume)(struct snd_soc_dai *dai); @@ -219,26 +211,50 @@ struct snd_soc_dai { struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; unsigned int symmetric_rates:1; +}; + +/* + * Digital Audio Interface runtime data. + * + * Holds runtime data for a DAI. + */ +struct snd_soc_dai { + const char *name; + int id; + struct device *dev; + void *ac97_pdata; /* platform_data for the ac97 codec */ + + /* driver ops */ + struct snd_soc_dai_driver *driver; /* DAI runtime info */ - struct snd_soc_codec *codec; + unsigned int capture_active:1; /* stream is in use */ + unsigned int playback_active:1; /* stream is in use */ + unsigned int symmetric_rates:1; + struct snd_pcm_runtime *runtime; unsigned int active; unsigned char pop_wait:1; + unsigned char probed:1; - /* DAI private data */ - void *private_data; + /* DAI DMA data */ + void *playback_dma_data; + void *capture_dma_data; - /* parent platform */ - struct snd_soc_platform *platform; + /* parent platform/codec */ + union { + struct snd_soc_platform *platform; + struct snd_soc_codec *codec; + }; struct list_head list; + struct list_head card_list; }; static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, const struct snd_pcm_substream *ss) { return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - dai->playback.dma_data : dai->capture.dma_data; + dai->playback_dma_data : dai->capture_dma_data; } static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, @@ -246,9 +262,20 @@ static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, void *data) { if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback.dma_data = data; + dai->playback_dma_data = data; else - dai->capture.dma_data = data; + dai->capture_dma_data = data; +} + +static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, + void *data) +{ + dev_set_drvdata(dai->dev, data); +} + +static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) +{ + return dev_get_drvdata(dai->dev); } #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 65e9d03..b3fe7a8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -228,13 +228,17 @@ struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; struct snd_soc_dai; +struct snd_soc_dai_driver; struct snd_soc_platform; struct snd_soc_dai_link; +struct snd_soc_platform_driver; struct snd_soc_codec; +struct snd_soc_codec_driver; struct soc_enum; struct snd_soc_ac97_ops; struct snd_soc_jack; struct snd_soc_jack_pin; + #ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio; #endif @@ -249,19 +253,18 @@ enum snd_soc_control_type { SND_SOC_SPI, }; -int snd_soc_register_platform(struct snd_soc_platform *platform); -void snd_soc_unregister_platform(struct snd_soc_platform *platform); -int snd_soc_register_codec(struct snd_soc_codec *codec); -void snd_soc_unregister_codec(struct snd_soc_codec *codec); +int snd_soc_register_platform(struct device *dev, int id, + struct snd_soc_platform_driver *platform_drv); +void snd_soc_unregister_platform(struct device *dev, int id); +int snd_soc_register_codec(struct device *dev, int id, + struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); +void snd_soc_unregister_codec(struct device *dev, int id); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control); -/* pcm <-> DAI connect */ -void snd_soc_free_pcms(struct snd_soc_device *socdev); -int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); - /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); @@ -273,7 +276,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, const struct snd_pcm_hardware *hw); /* Jack reporting */ -int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, +int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack); void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, @@ -390,7 +393,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack { struct snd_jack *jack; - struct snd_soc_card *card; + struct snd_soc_codec *codec; struct list_head pins; int status; struct blocking_notifier_head notifier; @@ -398,15 +401,13 @@ struct snd_soc_jack { /* SoC PCM stream information */ struct snd_soc_pcm_stream { - char *stream_name; + const char *stream_name; u64 formats; /* SNDRV_PCM_FMTBIT_* */ unsigned int rates; /* SNDRV_PCM_RATE_* */ unsigned int rate_min; /* min rate */ unsigned int rate_max; /* max rate */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ - unsigned int active; /* stream is in use */ - void *dma_data; /* used by platform code */ }; /* SoC audio ops */ @@ -419,44 +420,37 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -/* SoC Audio Codec */ +/* SoC Audio Codec device */ struct snd_soc_codec { - char *name; - struct module *owner; - struct mutex mutex; + const char *name; + int id; + struct device *dev; - struct snd_soc_device *socdev; + struct snd_soc_codec_driver *driver; + struct mutex mutex; + struct snd_card *snd_card; + struct snd_soc_card *card; struct list_head list; - - /* callbacks */ - int (*set_bias_level)(struct snd_soc_codec *, - enum snd_soc_bias_level level); + struct list_head card_list; + int num_dai; /* runtime */ - struct snd_card *card; struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; - unsigned int pcm_devs; - void *drvdata; + unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + unsigned int cache_only:1; /* Suppress writes to hardware */ + unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ + unsigned int suspended:1; /* Codec is in suspend PM state */ + unsigned int probed:1; /* Codec has been probed */ + unsigned int ac97_registered:1; /* Codec has been AC97 registered */ + unsigned int sysfs_registered:1; /* codec has been sysfs registered */ /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - int (*display_register)(struct snd_soc_codec *, char *, - size_t, unsigned int); - int (*volatile_register)(unsigned int); - int (*readable_register)(unsigned int); hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); void *reg_cache; - short reg_cache_size; - short reg_cache_step; - - unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ - unsigned int cache_only:1; /* Suppress writes to hardware */ - unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ /* dapm */ u32 pop_time; @@ -466,10 +460,6 @@ struct snd_soc_codec { enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; - /* codec DAI's */ - struct snd_soc_dai *dai; - unsigned int num_dai; - #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_codec_root; struct dentry *debugfs_reg; @@ -478,23 +468,44 @@ struct snd_soc_codec { #endif }; -/* codec device */ -struct snd_soc_codec_device { - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); - int (*suspend)(struct platform_device *pdev, pm_message_t state); - int (*resume)(struct platform_device *pdev); +/* codec driver */ +struct snd_soc_codec_driver { + const char *name; + struct module *owner; + + /* driver ops */ + int (*probe)(struct snd_soc_codec *); + int (*remove)(struct snd_soc_codec *); + int (*suspend)(struct snd_soc_codec *, + pm_message_t state); + int (*resume)(struct snd_soc_codec *); + + /* codec IO */ + unsigned int (*read)(struct snd_soc_codec *, unsigned int); + int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); + int (*display_register)(struct snd_soc_codec *, char *, + size_t, unsigned int); + int (*volatile_register)(unsigned int); + int (*readable_register)(unsigned int); + short reg_cache_size; + short reg_cache_step; + short reg_word_size; + const void *reg_cache_default; + + /* codec bias level */ + int (*set_bias_level)(struct snd_soc_codec *, + enum snd_soc_bias_level level); }; /* SoC platform interface */ -struct snd_soc_platform { - char *name; - struct list_head list; +struct snd_soc_platform_driver { + const char *name; + struct module *owner; - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); - int (*suspend)(struct snd_soc_dai_link *dai_link); - int (*resume)(struct snd_soc_dai_link *dai_link); + int (*probe)(struct snd_soc_platform *); + int (*remove)(struct snd_soc_platform *); + int (*suspend)(struct snd_soc_dai *dai); + int (*resume)(struct snd_soc_dai *dai); /* pcm creation and destruction */ int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, @@ -509,23 +520,35 @@ struct snd_soc_platform { struct snd_soc_dai *); /* platform stream ops */ - struct snd_pcm_ops *pcm_ops; + struct snd_pcm_ops *ops; }; -/* SoC machine DAI configuration, glues a codec and cpu DAI together */ -struct snd_soc_dai_link { - char *name; /* Codec name */ - char *stream_name; /* Stream name */ +struct snd_soc_platform { + const char *name; + int id; - /* DAI */ - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; + struct device *dev; + struct snd_soc_platform_driver *driver; - /* machine stream operations */ - struct snd_soc_ops *ops; + unsigned int suspended:1; /* platform is suspended */ + unsigned int probed:1; - /* codec/machine specific init - e.g. add machine controls */ - int (*init)(struct snd_soc_codec *codec); + struct list_head list; + struct list_head card_list; +}; + +struct snd_soc_dai_link { + /* config - must be set by machine driver */ + const char *name; /* Codec name */ + const char *stream_name; /* Stream name */ + int codec_id; /* for multi-codec */ + int platform_id; /* for multi-platform */ + int cpu_dai_id; + int codec_dai_id; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_dai_driver *cpu_dai_drv; + struct snd_soc_platform_driver *platform_drv; + struct snd_soc_codec_driver *codec_drv; /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; @@ -533,21 +556,24 @@ struct snd_soc_dai_link { /* Symmetry requirements */ unsigned int symmetric_rates:1; - /* Symmetry data - only valid if symmetry is being enforced */ - unsigned int rate; + /* codec/machine specific init - e.g. add machine controls */ + int (*init)(struct snd_soc_pcm_runtime *rtd); - /* DAI pcm */ - struct snd_pcm *pcm; + /* machine stream operations */ + struct snd_soc_ops *ops; }; /* SoC card */ struct snd_soc_card { - char *name; + const char *name; struct device *dev; + struct snd_card *snd_card; + struct module *owner; struct list_head list; + struct mutex mutex; - int instantiated; + int instantiated:1; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); @@ -568,28 +594,38 @@ struct snd_soc_card { /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; + struct snd_soc_pcm_runtime *rtd; + int num_rtd; - struct snd_soc_device *socdev; - - struct snd_soc_codec *codec; - - struct snd_soc_platform *platform; - struct delayed_work delayed_work; struct work_struct deferred_resume_work; + + /* lists of probed devices belonging to this card */ + struct list_head codec_dev_list; + struct list_head platform_dev_list; + struct list_head dai_dev_list; }; -/* SoC Device - the audio subsystem */ -struct snd_soc_device { - struct device *dev; +/* SoC machine DAI configuration, glues a codec and cpu DAI together */ +struct snd_soc_pcm_runtime { + struct device dev; struct snd_soc_card *card; - struct snd_soc_codec_device *codec_dev; - void *codec_data; -}; + struct snd_soc_dai_link *dai_link; + + unsigned int complete:1; + unsigned int dev_registered:1; + + /* Symmetry data - only valid if symmetry is being enforced */ + unsigned int rate; + long pmdown_time; -/* runtime channel data */ -struct snd_soc_pcm_runtime { - struct snd_soc_dai_link *dai; - struct snd_soc_device *socdev; + /* runtime devices */ + struct snd_pcm *pcm; + struct snd_soc_codec *codec; + struct snd_soc_platform *platform; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; + + struct delayed_work delayed_work; }; /* mixer control */ @@ -615,24 +651,48 @@ struct soc_enum { static inline unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg) { - return codec->read(codec, reg); + return codec->driver->read(codec, reg); } static inline unsigned int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - return codec->write(codec, reg, val); + return codec->driver->write(codec, reg, val); } +/* device driver data */ + static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec, - void *data) + void *data) { - codec->drvdata = data; + dev_set_drvdata(codec->dev, data); } static inline void *snd_soc_codec_get_drvdata(struct snd_soc_codec *codec) { - return codec->drvdata; + return dev_get_drvdata(codec->dev); +} + +static inline void snd_soc_platform_set_drvdata(struct snd_soc_platform *platform, + void *data) +{ + dev_set_drvdata(platform->dev, data); +} + +static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platform) +{ + return dev_get_drvdata(platform->dev); +} + +static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd, + void *data) +{ + dev_set_drvdata(&rtd->dev, data); +} + +static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) +{ + return dev_get_drvdata(&rtd->dev); } #include <sound/soc-dai.h> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8b79d90..3deb226 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3,6 +3,8 @@ * * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. * * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * with code, comments and ideas from :- -- 1.7.0.4

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[alsa-devel] [PATCH 1/1] Daily snapshot not building on 2.6.32
by Brad Figg 24 Jun '10

24 Jun '10

I've had to apply the following patch to successfully build the daily snapshot on 2.6.32. Without this patch I get: CC [M] /karmic/home/work/alsa/alsa-driver/usb/urb.o /karmic/home/work/alsa/alsa-driver/usb/urb.c: In function ‘release_urb_ctx’: /karmic/home/work/alsa/alsa-driver/usb/urb.c:112: error: implicit declaration of function ‘usb_free_ coherent’ /karmic/home/work/alsa/alsa-driver/usb/urb.c: In function ‘snd_usb_init_substream_urbs’: /karmic/home/work/alsa/alsa-driver/usb/urb.c:327: error: implicit declaration of function ‘usb_alloc_coherent’ /karmic/home/work/alsa/alsa-driver/usb/urb.c:328: warning: assignment makes pointer from integer without a cast /karmic/home/work/alsa/alsa-driver/usb/urb.c:341: warning: assignment makes pointer from integer without a cast make[3]: *** [/karmic/home/work/alsa/alsa-driver/usb/urb.o] Error 1 Note: I would have preferred to provide this as a git patch but I don't know where the repository for the daily snapshot files is located. --- diff -ru a/usb/misc/Makefile b/usb/misc/Makefile --- a/usb/misc/Makefile 2010-03-10 01:12:19.000000000 -0800 +++ b/usb/misc/Makefile 2010-06-23 11:25:33.176133657 -0700 @@ -7,4 +7,8 @@ include $(SND_TOPDIR)/alsa-kernel/usb/misc/Makefile +EXTRA_CFLAGS += -I$(SND_TOPDIR)/usb + include $(SND_TOPDIR)/Rules.make + +ua101.c: usbaudio.inc $(SND_TOPDIR)/alsa-kernel/usb/misc/ua101.c diff -ru a/usb/misc/ua101.c b/usb/misc/ua101.c --- a/usb/misc/ua101.c 2010-03-10 01:12:19.000000000 -0800 +++ b/usb/misc/ua101.c 2010-06-23 11:25:53.054884851 -0700 @@ -1,3 +1,5 @@ +#define __NO_VERSION__ +#include "usbaudio.inc" #include "adriver.h" #include "../../alsa-kernel/usb/misc/ua101.c" EXPORT_NO_SYMBOLS; -- Brad Figg brad.figg(a)canonical.com http://www.canonical.com

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[alsa-devel] USB midi support for KAOSS pad
by Pedro Ferreira 24 Jun '10

24 Jun '10

Hello all, I tried connecting my KAOSS PAD 3 to a linux laptop, and apparently ALSA MIDI detects the device. However, dmesg shows several messages, all similar to: {{{ [ 7088.722283] ALSA midi.c:1597: MIDIStreaming version 01.00 [ 7088.722290] ALSA midi.c:1632: EP 02: 1 jack(s) [ 7088.722295] ALSA midi.c:1647: EP 82: 1 jack(s) [ 7088.722315] ALSA midi.c:1567: created 1 output and 1 input ports [ 7088.806326] usb 2-1.1: reset full speed USB device using ehci_hcd and address 15 }}} If i open qjackctl, i can see the device being added and removed successively. Does anyone know what can be going wrong? I would be more than happy to contribute with usbmon logs, etc... or even try to research a bit if someone points me in the right direction. Thanks in advance, Pedro

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[alsa-devel] DMA Physical Address through "snd_pcm_lib_malloc_pages"
by Adish Kuvelker 24 Jun '10

24 Jun '10

Hello, I am trying to allocate memory for DMA buffer through "snd_pcm_lib_malloc_pages" in hw_params callout function, with prior use of "snd_pcm_lib_preallocate_pages_for_all". The chapter on PCM Interface in "writing an ALSA driver" states that if the standard function "snd_pcm_lib_malloc_pages" is use for allocating the DMA buffers then the DMA fields of dma_addr, dma_area, dma_bytes are set by the ALSA middle layer. However my code walk through didn't point me to the area where this physical memory setting is done. My DMA channel is not serving a PCI device. I would be very grateful if any of you guys can point me to the code where this settings get done by calling "snd_pcm_lib_malloc_pages". Going through the code for a few devices registered under "pci" I found that the call to "snd_pcm_lib_malloc_pages" is followed by call to "snd_pcm_sgbuf_get_addr" to get the physical address. Also a call to "__pa" is done in the "parisc/harmony.c". Do I have to set the dma parameters in the runtime structure even after allocating memory using "snd_pcm_lib_malloc_pages"? Regards; Aadish

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