September 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] idt STAC927x adat supported (sigmatel HDA codec) ?
by Eric 03 Mar '11

03 Mar '11

Is the ADAT output of the IDT (sigmatel) STAC927x HDA codec supported in alsa ? If not could someone please try to add this incredibly nice feature in the existing alsa driver ? Thank you very much

4 13

[alsa-devel] Problem with VIA VT1708S and git versions of alsa-driver.
by Mark Goldstein 31 Jan '11

31 Jan '11

Hello, Recently I'm having some strange issues with my VIA VT1708S on-board sound card. The description might be a bit long, so I'll give a brief summary first. I have ASUS P5QL/EPU motherboard with VT1708S 8-channel HDA Codec. There are 3 OSs installed on this machine: openSUSE 11.1 (my main system), openSUSE 11.3, Kubuntu 10.04. Audio worked fine until some updates at the beginning of September. Since then I have 2 different issues. 1).On 11.1 - Line In control seems inactive, instead Front Mic control controls the level from Line In jack. Front Mic is constanly disabled. alsaconf does not show hda-intel in the list of detected cards. Latest version it happened with is git20100925. When I compiled stable 1.0.23 sources, everything started working as expected again. I reported this issue on bugtrack: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5133 2) On oS 11.3 I start experiencing problems recording audio either from Mic or from Line. After some experiments I found out that the Capture1 control is somehow broken. Mute switch is reversed. When it is unmuted, there is no audio recorded. When it is muted, recording works. The same happened to at least one more user of openSUSE, see the thread "How to investigate the problem with MIC" on alsa-user mailing list. I've also added my comments to issue https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5130 but now I think it is different issue. Even when I have audio recorded normally when Capture1 is muted, I still have very distorted voice from Mic in Skype. It worked fine before the problem started. More details: On oS 11.1 I had my audio working fine after updating to 1.0.23 versions of alsa driver. I'm using opensuse/multimedia: repository that provides binary rpm of alsa-driver git snapshots. I have only one current kernel 2.6.27.48 on 11.1. On oS 11.3 audio worked right out of the box, since it uses kernels with alsa-driver 1.0.23. I have 2-3 kernels installed at the same time - normal 2.6.34 from official repository and 2.6.35 from kernel build service. I'm not sure when exactly the problems started, but definitely at the beginning of September. I've initially saw that on oS 11.1 I can't hear the audio from my TV card connected to Line In. Then I noticed that Mic does not work in Skype on both 11.1 and 11.3, while it still works on Kubuntu. When I unmuted Front Mic on 11.1 I've suddenly found that it actually enabled Line In. I think now that something went bad with card detecting. Probably on 11.1 with latest git versions my card is not properly detected and some "defaults" are used. I'm attaching here 2 outputs of alsa-info.sh - 11.1_stable captured with alsa-driver compiled from stable 1.0.23 sources. 11.1_git25 captured with driver compiled with git20100925 version from opensuse/multimedia: repository. For me the problem with 11.1 is resolved, but probably it's worth checking why latest version from git behave this way. I'll continue with 11.3 investigation later on. Regards, -- Mark Goldstein

2 21

[alsa-devel] low volume with hda base card (see attached alsa-info output
by Halim Sahin 13 Dec '10

13 Dec '10

Hi, I need to set all mixer stuff to 100% and the output isn't ok. Volume is too low. When I switch to another input on my amp. it's about 500% louder. Please have a look to attached ala-info output. I tested with opensuse 11.2 and also with a newer kernel for the distro 2.6.34. Can you please help? Thx. Halim -- Halim Sahin E-Mail: halim.sahin (at) t-online.de

2 4

[alsa-devel] alsactl adds volume controls?
by David Henningsson 24 Oct '10

24 Oct '10

So I've discovered that my sound card has a "PCM Playback Volume" control, but changing that control does not alter the volume. This is not only confusing but also a problem for PulseAudio. (PA merges this control into it's overall volume, and so when the user moves PA's volume control, in some ranges nothing happens.) Interestingly enough, this control does not come from the HDA parser, it is added by alsactl at boot time...! (Through alsa-lib's snd_ctl_elem_add_integer, which does an snd_ctl_ioctl.) So the question is, why is alsactl creating volume controls, and why is userspace allowed to add controls in the first place, if the kernel can't use them for anything anyway? That doesn't make sense to me. This a generic problem, but sound card info, if that matters, is available here: https://launchpad.net/bugs/625149 Codec: http://launchpadlibrarian.net/54478760/Card0.Codecs.codec.0.txt -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

9 73

[alsa-devel] how to mandate the use of PCM plugin?
by Stas Sergeev 22 Oct '10

22 Oct '10

Hi guys. Is it possible to somehow mandate the use of a PCM plugin? For example, the PC-Speaker.conf defines the use of softvol plugin for "default" and "front" devices. It is still possible for the software to bypass it by the use of "hw", and some software does exactly that. Is it possible to somehow make some plugin a mandatory?

6 23

[alsa-devel] [PATCH] Per Application Volume Control plugin for ALSA
by James Huk 18 Oct '10

18 Oct '10

Hello everybody. This is my first "post" here, so please be "gentle". I have created PAVC plugin for ALSA, it is based on COPY plugin, and use SOFTVOL plugin for sound output. As far as I can tell from current tests - It works perfectly, and it doesn't break compatibility with any apps I use (and PulseAudio does). So I was wondering - is there any chance this plugin would be added to the next official ALSA release? Here is the patch that enables it on current alsa-lib-1.0.23 (hopefully this mailing list is a good place to "post" this): --- a/configure.in 2010-09-23 17:03:00.000000000 +0200 +++ b/configure.in 2010-09-23 17:09:11.029900400 +0200 @@ -445,7 +445,7 @@ pcm_plugins="" fi -PCM_PLUGIN_LIST="copy linear route mulaw alaw adpcm rate plug multi shm file null empty share meter hooks lfloat ladspa dmix dshare dsnoop asym iec958 softvol extplug ioplug mmap_emul" +PCM_PLUGIN_LIST="copy linear route mulaw alaw adpcm rate plug multi shm file null empty share meter hooks lfloat ladspa dmix dshare dsnoop asym iec958 softvol extplug ioplug mmap_emul pavc" build_pcm_plugin="no" for t in $PCM_PLUGIN_LIST; do @@ -516,6 +516,7 @@ AM_CONDITIONAL(BUILD_PCM_PLUGIN_EXTPLUG, test x$build_pcm_extplug = xyes) AM_CONDITIONAL(BUILD_PCM_PLUGIN_IOPLUG, test x$build_pcm_ioplug = xyes) AM_CONDITIONAL(BUILD_PCM_PLUGIN_MMAP_EMUL, test x$build_pcm_mmap_emul = xyes) +AM_CONDITIONAL(BUILD_PCM_PLUGIN_PAVC, test x$build_pcm_pavc = xyes) dnl Defines for plug plugin if test "$build_pcm_rate" = "yes"; then --- a/src/pcm/Makefile.am 2010-04-16 13:11:05.000000000 +0200 +++ b/src/pcm/Makefile.am 2010-09-23 17:20:10.601899695 +0200 @@ -102,6 +102,9 @@ if BUILD_PCM_PLUGIN_MMAP_EMUL libpcm_la_SOURCES += pcm_mmap_emul.c endif +if BUILD_PCM_PLUGIN_PAVC +libpcm_la_SOURCES += pcm_pavc.c +endif EXTRA_DIST = pcm_dmix_i386.c pcm_dmix_x86_64.c pcm_dmix_generic.c --- a/src/pcm/pcm.c 2010-04-16 13:11:05.000000000 +0200 +++ b/src/pcm/pcm.c 2010-09-23 17:10:43.813900406 +0200 @@ -2047,7 +2047,7 @@ static const char *const build_in_pcms[] = { "adpcm", "alaw", "copy", "dmix", "file", "hooks", "hw", "ladspa", "lfloat", "linear", "meter", "mulaw", "multi", "null", "empty", "plug", "rate", "route", "share", - "shm", "dsnoop", "dshare", "asym", "iec958", "softvol", "mmap_emul", + "shm", "dsnoop", "dshare", "asym", "iec958", "softvol", "mmap_emul","pavc", NULL }; --- a/include/pcm.h 2010-04-16 13:11:05.000000000 +0200 +++ b/include/pcm.h 2010-09-23 17:11:58.389899158 +0200 @@ -376,7 +376,9 @@ SND_PCM_TYPE_EXTPLUG, /** Mmap-emulation plugin */ SND_PCM_TYPE_MMAP_EMUL, - SND_PCM_TYPE_LAST = SND_PCM_TYPE_MMAP_EMUL + /** PAVC plugin */ + SND_PCM_TYPE_PAVC, + SND_PCM_TYPE_LAST = SND_PCM_TYPE_PAVC }; /** PCM type */ --- a/src/pcm/pcm_pavc.c +++ b/src/pcm/pcm_pavc.c @@ -0,0 +1,846 @@ +/* + * PCM - PAVC plugin + * Based on PCM - Copy conversion plugin by Abramo Bagnara <abramo(a)alsa-project.org> + * + * Copyright (c) 2010 by Adrian Kobyliński <huk256(a)gmail.com> + * + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <byteswap.h> +#include "pcm_local.h" +#include "pcm_plugin.h" +#include <stdio.h> +#include <string.h> +#include <unistd.h> +#include <sys/wait.h> + + +#ifndef PIC +/* entry for static linking */ +const char *_snd_module_pcm_pavc = ""; +#endif + +#ifndef DOC_HIDDEN +typedef struct { + /* This field need to be the first */ + snd_pcm_plugin_t plug; +} snd_pcm_pavc_t; +#endif + +static int snd_pcm_pavc_hw_refine_cprepare(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *params) +{ + int err; + snd_pcm_access_mask_t access_mask = { SND_PCM_ACCBIT_SHM }; + err = _snd_pcm_hw_param_set_mask(params, SND_PCM_HW_PARAM_ACCESS, + &access_mask); + if (err < 0) + return err; + params->info &= ~(SND_PCM_INFO_MMAP | SND_PCM_INFO_MMAP_VALID); + return 0; +} + +static int snd_pcm_pavc_hw_refine_sprepare(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *sparams) +{ + snd_pcm_access_mask_t saccess_mask = { SND_PCM_ACCBIT_MMAP }; + _snd_pcm_hw_params_any(sparams); + _snd_pcm_hw_param_set_mask(sparams, SND_PCM_HW_PARAM_ACCESS, + &saccess_mask); + return 0; +} + +static int snd_pcm_pavc_hw_refine_schange(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *params, + snd_pcm_hw_params_t *sparams) +{ + int err; + unsigned int links = ~SND_PCM_HW_PARBIT_ACCESS; + err = _snd_pcm_hw_params_refine(sparams, links, params); + if (err < 0) + return err; + return 0; +} + +static int snd_pcm_pavc_hw_refine_cchange(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *params, + snd_pcm_hw_params_t *sparams) +{ + int err; + unsigned int links = ~SND_PCM_HW_PARBIT_ACCESS; + err = _snd_pcm_hw_params_refine(params, links, sparams); + if (err < 0) + return err; + return 0; +} + +static int snd_pcm_pavc_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) +{ + return snd_pcm_hw_refine_slave(pcm, params, + snd_pcm_pavc_hw_refine_cprepare, + snd_pcm_pavc_hw_refine_cchange, + snd_pcm_pavc_hw_refine_sprepare, + snd_pcm_pavc_hw_refine_schange, + snd_pcm_generic_hw_refine); +} + +static int snd_pcm_pavc_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) +{ + return snd_pcm_hw_params_slave(pcm, params, + snd_pcm_pavc_hw_refine_cchange, + snd_pcm_pavc_hw_refine_sprepare, + snd_pcm_pavc_hw_refine_schange, + snd_pcm_generic_hw_params); +} + +static snd_pcm_uframes_t + snd_pcm_pavc_write_areas(snd_pcm_t *pcm, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, + snd_pcm_uframes_t size, + const snd_pcm_channel_area_t *slave_areas, + snd_pcm_uframes_t slave_offset, + snd_pcm_uframes_t *slave_sizep) +{ + if (size > *slave_sizep) + size = *slave_sizep; + snd_pcm_areas_copy(slave_areas, slave_offset, + areas, offset, + pcm->channels, size, pcm->format); + *slave_sizep = size; + return size; +} + +static snd_pcm_uframes_t + snd_pcm_pavc_read_areas(snd_pcm_t *pcm, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, + snd_pcm_uframes_t size, + const snd_pcm_channel_area_t *slave_areas, + snd_pcm_uframes_t slave_offset, + snd_pcm_uframes_t *slave_sizep) +{ + if (size > *slave_sizep) + size = *slave_sizep; + snd_pcm_areas_copy(areas, offset, + slave_areas, slave_offset, + pcm->channels, size, pcm->format); + *slave_sizep = size; + return size; +} + +static void snd_pcm_pavc_dump(snd_pcm_t *pcm, snd_output_t *out) +{ + snd_pcm_pavc_t *pavc = pcm->private_data; + snd_output_printf(out, "Copy conversion PCM\n"); + if (pcm->setup) { + snd_output_printf(out, "Its setup is:\n"); + snd_pcm_dump_setup(pcm, out); + } + snd_output_printf(out, "Slave: "); + snd_pcm_dump(pavc->plug.gen.slave, out); +} + +static const snd_pcm_ops_t snd_pcm_pavc_ops = { + .close = snd_pcm_generic_close, + .info = snd_pcm_generic_info, + .hw_refine = snd_pcm_pavc_hw_refine, + .hw_params = snd_pcm_pavc_hw_params, + .hw_free = snd_pcm_generic_hw_free, + .sw_params = snd_pcm_generic_sw_params, + .channel_info = snd_pcm_generic_channel_info, + .dump = snd_pcm_pavc_dump, + .nonblock = snd_pcm_generic_nonblock, + .async = snd_pcm_generic_async, + .mmap = snd_pcm_generic_mmap, + .munmap = snd_pcm_generic_munmap, +}; + +/** + * \brief Creates a new copy PCM + * \param pcmp Returns created PCM handle + * \param name Name of PCM + * \param slave Slave PCM handle + * \param close_slave When set, the slave PCM handle is closed with copy PCM + * \retval zero on success otherwise a negative error code + * \warning Using of this function might be dangerous in the sense + * of compatibility reasons. The prototype might be freely + * changed in future. + */ +int snd_pcm_pavc_open(snd_pcm_t **pcmp, const char *name, snd_pcm_t *slave, int close_slave) +{ + snd_pcm_t *pcm; + snd_pcm_pavc_t *pavc; + int err; + assert(pcmp && slave); + pavc = calloc(1, sizeof(snd_pcm_pavc_t)); + if (!pavc) { + return -ENOMEM; + } + snd_pcm_plugin_init(&pavc->plug); + pavc->plug.read = snd_pcm_pavc_read_areas; + pavc->plug.write = snd_pcm_pavc_write_areas; + pavc->plug.undo_read = snd_pcm_plugin_undo_read_generic; + pavc->plug.undo_write = snd_pcm_plugin_undo_write_generic; + pavc->plug.gen.slave = slave; + pavc->plug.gen.close_slave = close_slave; + + err = snd_pcm_new(&pcm, SND_PCM_TYPE_PAVC, name, slave->stream, slave->mode); + if (err < 0) { + free(pavc); + return err; + } + pcm->ops = &snd_pcm_pavc_ops; + pcm->fast_ops = &snd_pcm_plugin_fast_ops; + pcm->private_data = pavc; + pcm->poll_fd = slave->poll_fd; + pcm->poll_events = slave->poll_events; + pcm->monotonic = slave->monotonic; + snd_pcm_set_hw_ptr(pcm, &pavc->plug.hw_ptr, -1, 0); + snd_pcm_set_appl_ptr(pcm, &pavc->plug.appl_ptr, -1, 0); + *pcmp = pcm; + + return 0; +} + +int countString(char * x,char x2,int count) +{ + int i; + int z=0; + for(i=0;i<count;++i) + { + if(x[i]==x2) + { + ++z; + } + } + return z; +} + +char *deleteChar(char *x, char x2, int count) +{ + char *ret; + ret=(char*) malloc(count+1); + int i; + for(i=0;i<count;++i) + { + ret[i]=0; + } + for(i=0;i<count;++i) + { + if(x[i]!=x2) + { + ret[i]=x[i]; + } + else + { + ret[i]=' '; + } + } + return ret; +} + +char *getProceses() +{ + FILE *pipe; + if ( !(pipe = (FILE*)popen("lsof -F p /dev/snd/pcmC0D0p","r")) ) + { // If fpipe is NULL + perror("Problems with pipe"); + return NULL; + } + + int i=0,j=0; + int c; + + do { + c = fgetc (pipe); + ++i; + } while (c != EOF); + fclose (pipe); + + char *buf=(char*)malloc(sizeof(char)*i); + + for(j=0;j<i;++j) + { + buf[j]=0; + } + + i=0; + + if ( !(pipe = (FILE*)popen("lsof -F p /dev/snd/pcmC0D0p","r")) ) + { // If fpipe is NULL + perror("Problems with pipe"); + return NULL; + } + + do { + c = fgetc (pipe); + if(c!=EOF && c!='p' && c!='\n') + { + buf[i]=c; + } + else if(c=='p') + { + buf[i]=' '; + } + else if(c=='\n') + { + buf[i]=','; + } + ++i; + } while (c != EOF); + fclose (pipe); + + return buf; +} + +char *loadPid(char *name) +{ + FILE *file; + int c,i=0,j=0; + + file=fopen(name,"r"); + if(file!=NULL) + { + do{ + c=fgetc(file); + ++i; + }while(c!=EOF); + fclose(file); + + char *buf=malloc((sizeof(char))*i); + for(j=0;j<i;++j) + { + buf[j]=0; + } + + i=0; + + file=fopen(name,"r"); + if(file!=NULL) + { + do{ + c=fgetc(file); + if(c!=EOF && c!='\n') + buf[i]=c; + ++i; + }while(c!=EOF); + fclose(file); + + return buf; + } + else + { + return NULL; + } + + } + else + { + return NULL; + } +} + +void savePid(char *pid,char *file) +{ + int i; + char command3[256]; + + for(i=0;i<256;++i) + { + command3[i]=0; + } + + strcat(command3,"echo "); + strcat(command3,pid); + strcat(command3," > "); + strcat(command3,file); + + for(i=5;i<256;++i) + { + if(command3[i]!=EOF && command3[i]!='\n' && command3[i]!='*') + { + command3[i]=command3[i]; + } + else + { + command3[i]=' '; + } + } + + system(command3); +} + +int checkDir(char *name) +{ + char command[128]; + memset(command,0,128); + + strcat(command,"ls -aF "); + strcat(command,getenv("HOME")); + strcat(command," | grep \\./$ | grep -w "); + strcat(command,name); + + FILE *pipe; + if ( !(pipe = (FILE*)popen(command,"r")) ) + { + perror("Problems with pipe"); + return -1; + } + + int i=0; + int c; + + do { + c = fgetc (pipe); + if(c!=EOF && c!='\n') + ++i; + } while (c != EOF); + fclose (pipe); + + if(i>0) + { + return 1; + } + else + { + return 0; + } +} + +int createDir(char *name) +{ + pid_t PID; + int status; + + PID=fork(); + + if(PID>0) + { + char command[128]; + memset(command,0,128); + + strcat(command,getenv("HOME")); + strcat(command,"/"); + strcat(command,".softvol"); + + execlp("mkdir","mkdir",command,NULL); + } + else + { + pid_t wpid = waitpid(PID, &status, WUNTRACED); + } + + return WEXITSTATUS(status); +} + +char *getDir(char *name) +{ + static char command[128]; + memset(command,0,128); + + strcat(command,getenv("HOME")); + strcat(command,"/"); + strcat(command,name); + + return command; +} + +char *getSoftvolNumber(int channelCount,int current) +{ + size_t i; + static char ret[4]; + memset(ret,0,5); + char buf1[4],buf2[4]; + sprintf(buf1,"%i",channelCount); + sprintf(buf2,"%i",current); + + if(strlen(buf2)<strlen(buf1)) + { + for(i=0;i<(strlen(buf1)-1);++i) + { + ret[i]='0'; + } + + strcat(ret,buf2); + + return ret; + } + else if(strlen(buf2)==strlen(buf1)) + { + strcat(ret,buf2); + return ret; + } + else + { + //this should never happen + return NULL; + } +} + +int checkSoftvolProcess(int channelcount) +{ + int i=0,c=0,ret=channelcount-1,j=0; + char nameBuf[1024]; + char pidBuf[32]; + char numBuf[4]; + char command[128]; + char fileName[64]; + + FILE *fpipe; + + for(i=0;i<channelcount;++i) + { + memset(nameBuf,0,1024); + memset(numBuf,0,4); + memset(command,0,128); + memset(fileName,0,64); + + strcat(command,"ps -A -o pid= | grep -w -f "); + strcat(command,getDir(".softvol")); + strcat(command,"/SOFTVOL"); + strcat(command,getSoftvolNumber(channelcount,i)); + strcat(command,".pid"); + + strcat(fileName,getDir(".softvol")); + strcat(fileName,"/SOFTVOL"); + strcat(fileName,getSoftvolNumber(channelcount,i)); + strcat(fileName,".pid"); + + //Here we check if current process is already conected to something + FILE *temp=fopen(fileName,"r"); + if(temp!=NULL) + { + j=0; + do{ + c=fgetc(temp); + if(c!=EOF && c!='\n') + { + nameBuf[j]=c; + } + else + { + nameBuf[j]='\0'; + } + + ++j; + }while(c!=EOF); + } + else + { + memset(nameBuf,0,1024); + } + + sprintf(pidBuf,"%d",getpid()); + + if(strcmp(nameBuf,pidBuf)==0) + { + //Current process wants to re-initialize the sound - so we connect it + //to the output it is currently using + ret=i; + break; + } + else + { + //Current process is not connected to anything - so we connect it + //to the firs unused softvol output + memset(fileName,0,64); + } + } + + if(fileName[0]!=0) + { + //Current process is already using some softvol output - so we + //reconnect it to this output + for(i=0;i<channelcount;++i) + { + memset(numBuf,0,4); + memset(command,0,128); + memset(fileName,0,64); + + strcat(command,"ps -A -o pid= | grep -w -f "); + strcat(command,getDir(".softvol")); + strcat(command,"/SOFTVOL"); + strcat(command,getSoftvolNumber(channelcount,i)); + strcat(command,".pid 2> /dev/null"); + + strcat(fileName,getDir(".softvol")); + strcat(fileName,"/SOFTVOL"); + strcat(fileName,getSoftvolNumber(channelcount,i)); + strcat(fileName,".pid"); + + //Here we check if process is alive... + if ( !(fpipe = (FILE*)popen(command,"r")) ) + { // If fpipe is NULL + perror("Problems with pipe"); + } + + j=0; + + do { + c = fgetc (fpipe); + if(c!=EOF) + ++j; + } while (c != EOF); + fclose (fpipe); + + //...and if it is the current process + if(j>0) + { + FILE *temp=fopen(fileName,"r"); + j=0; + do{ + c=fgetc(temp); + if(c!=EOF && c!='\n') + { + nameBuf[j]=c; + } + else + { + nameBuf[j]='\0'; + } + + ++j; + }while(c!=EOF); + + char pidBuf[32]; + sprintf(pidBuf,"%d",getpid()); + + if(strcmp(nameBuf,pidBuf)==0) + { + //Current process wants to re-initialize the sound - so we connect it + //to the output it is currently using + ret=i; + break; + } + else + { + //It is some other process - so we connect it + //to the first unused softvol output + ret=i+1; + } + } + else + { + ret=i; + } + } + } + else + { + //Process isn't using any softvol output - so we connect it + //to the first unused on the list + + for(i=0;i<channelcount;++i) + { + memset(numBuf,0,4); + memset(command,0,128); + memset(fileName,0,64); + + strcat(command,"ps -A -o pid= | grep -w -f "); + strcat(command,getDir(".softvol")); + strcat(command,"/SOFTVOL"); + strcat(command,getSoftvolNumber(channelcount,i)); + strcat(command,".pid 2> /dev/null"); + + strcat(fileName,getDir(".softvol")); + strcat(fileName,"/SOFTVOL"); + strcat(fileName,getSoftvolNumber(channelcount,i)); + strcat(fileName,".pid"); + + //Here we check if process is alive... + if ( !(fpipe = (FILE*)popen(command,"r")) ) + { // If fpipe is NULL + perror("Problems with pipe"); + } + + j=0; + + do { + c = fgetc (fpipe); + if(c!=EOF) + ++j; + } while (c != EOF); + fclose (fpipe); + + //...and if it is the current process + if(j>0) + { + + } + else + { + ret=i; + break; + } + } + } + return ret; +} + +/*! \page pcm_plugins + +\section pcm_plugins_copy Plugin: copy + +This plugin copies samples from master copy PCM to given slave PCM. +The channel count, format and rate must match for both of them. + +\code +pcm.name { + type copy # Copy PCM + slave STR # Slave name + # or + slave { # Slave definition + pcm STR # Slave PCM name + # or + pcm { } # Slave PCM definition + } +} +\endcode + +\subsection pcm_plugins_copy_funcref Function reference + +<UL> + <LI>snd_pcm_copy_open() + <LI>_snd_pcm_copy_open() +</UL> + +*/ + +/** + * \brief Creates a new copy PCM + * \param pcmp Returns created PCM handle + * \param name Name of PCM + * \param root Root configuration node + * \param conf Configuration node with copy PCM description + * \param stream Stream type + * \param mode Stream mode + * \retval zero on success otherwise a negative error code + * \warning Using of this function might be dangerous in the sense + * of compatibility reasons. The prototype might be freely + * changed in future. + */ +int _snd_pcm_pavc_open(snd_pcm_t **pcmp, const char *name, + snd_config_t *root, snd_config_t *conf, + snd_pcm_stream_t stream, int mode) +{ + long channelcount=-1; + int ret=0; + char softvolName[64]; + char pidBuf[32]; + char fileName[64]; + char helpBuf[32]; + + memset(softvolName,0,64); + + snd_config_iterator_t i, next; + int err; + snd_pcm_t *spcm; + snd_config_t *slave = NULL, *sconf; + + snd_config_t *tempConfig; + err=snd_config_search(conf,"channelcount",&tempConfig); + if(err<0) + { + SNDERR("channelcount field not found"); + return -ENOENT; + } + err=snd_config_get_integer(tempConfig,&channelcount); + if(err<0) + { + SNDERR("channelcount vaule must be integer"); + return -EINVAL; + } + + snd_config_t *softvol[channelcount]; + + int counter=0; + + sprintf(softvolName,"softvol%i",counter); + + snd_config_for_each(i, next, conf) + { + snd_config_t *n = snd_config_iterator_entry(i); + const char *id; + if (snd_config_get_id(n, &id) < 0) + continue; + if (snd_pcm_conf_generic_id(id)) + continue; + if (strcmp(id, "slave") == 0) + { + slave = n; + continue; + } + if (strcmp(id, "channelcount") == 0) + { + continue; + } + if((counter<channelcount) && strcmp(id,softvolName)==0) + { + softvol[counter]=n; + ++counter; + sprintf(softvolName,"softvol%i",counter); + continue; + } + SNDERR("Unknown field %s", id); + return -EINVAL; + } + + if (!slave) { + SNDERR("slave is not defined"); + return -EINVAL; + } + + if(checkDir(".softvol")) + { + //Directory exists - we do nothing + } + else + { + //Directory doesn't exists we need to create it + if((err=createDir(".softvol"))!=0) + { + SNDERR("Unable to create ~/.softvol directory - unknown error %i",err); + return err; + } + } + + ret=checkSoftvolProcess(channelcount); + + err = snd_pcm_slave_conf(root, softvol[ret], &sconf, 0); + + if (err < 0) + return err; + err = snd_pcm_open_slave(&spcm, root, sconf, stream, mode, conf); + snd_config_delete(sconf); + if (err < 0) + return err; + err = snd_pcm_pavc_open(pcmp, name, spcm, 1); + if (err < 0) + snd_pcm_close(spcm); + + sprintf(pidBuf,"%d",getpid()); + memset(fileName,0,64); + strcat(fileName,getDir(".softvol")); + strcat(fileName,"/SOFTVOL"); + + strcat(fileName,getSoftvolNumber(channelcount,ret)); + strcat(fileName,".pid"); + + savePid(pidBuf,fileName); + + return err; +} +#ifndef DOC_HIDDEN +SND_DLSYM_BUILD_VERSION(_snd_pcm_pavc_open, SND_PCM_DLSYM_VERSION); +#endif To test PAVC, we have to rebuild alsa-lib, and create .asoundrc or /etc/asound.conf that looks like this: pcm.!default { type plug slave.pcm "asymed" } pcm.asymed { type asym playback.pcm "pavcp" capture.pcm "dsnooped" } pcm.dmixer { type dmix ipc_key 1025 slave { pcm "hw:0" period_time 0 period_size 256 #buffer_size 4096 periods 128 rate 44100 } } pcm.dsnooped { type dsnoop ipc_key 1026 slave { pcm "hw:0" channels 2 period_size 256 #buffer_size 4096 rate 44100 periods 0 period_time 0 } } pcm.softvol00 { type softvol slave { pcm "dmixer" } control { name "Softvol00" card 0 } } pcm.softvol01 { type softvol slave { pcm "dmixer" } control { name "Softvol01" card 0 } } pcm.softvol02 { type softvol slave { pcm "dmixer" } control { name "Softvol02" card 0 } } pcm.softvol03 { type softvol slave { pcm "dmixer" } control { name "Softvol03" card 0 } } pcm.softvol04 { type softvol slave { pcm "dmixer" } control { name "Softvol04" card 0 } } pcm.softvol05 { type softvol slave { pcm "dmixer" } control { name "Softvol05" card 0 } } pcm.softvol06 { type softvol slave { pcm "dmixer" } control { name "Softvol06" card 0 } } pcm.softvol07 { type softvol slave { pcm "dmixer" } control { name "Softvol07" card 0 } } pcm.softvol08 { type softvol slave { pcm "dmixer" } control { name "Softvol08" card 0 } } pcm.softvol09 { type softvol slave { pcm "dmixer" } control { name "Softvol09" card 0 } } pcm.softvol10 { type softvol slave { pcm "dmixer" } control { name "Softvol10" card 0 } } pcm.softvol11 { type softvol slave { pcm "dmixer" } control { name "Softvol11" card 0 } } ctl.dmixer { type hw card 0 } ctl.dsnooped { type hw card 0 } pcm.pavcp { type pavc slave.pcm "dmixer" channelcount 12 softvol0.pcm "softvol00" softvol1.pcm "softvol01" softvol2.pcm "softvol02" softvol3.pcm "softvol03" softvol4.pcm "softvol04" softvol5.pcm "softvol05" softvol6.pcm "softvol06" softvol7.pcm "softvol07" softvol8.pcm "softvol08" softvol9.pcm "softvol09" softvol10.pcm "softvol10" softvol11.pcm "softvol11" } With this, anything connected to "default" will be automatically redirected to the first unused softvol channel. PIDs of apps that currently use softvol channels are saved to ~/.softvol/SOFTVOLXX.pid, so this can be checked at any time. If some application will try to restart only its sound subsystem, PAVC plugin will check those files to determine if it should be connected to the new softvol channel, or the one it currently uses. I tried to mimic the way this works with OSS4. Current limitations: -Since there is no "mute and slider at the same time" support in SOFTVOL plugin - there is no mute switch for software channels (or maybe there is a way I don't know about?) -Software channels must be named SOFTVOL00,01,02 and so on -SOFTVOL channels have both PLAYBACK and CAPTURE capabilities by default - this makes some mixers go haywire, and display double sliders - maybe there is a way around this I don't know about (like setting PLAYBACK only capability)? -Code of PAVC, could probably look nicer Hopefully someone will take a look at this, of course I am open to suggestions and critics. Best regards. P.S Sorry for my English ;]

4 7

[alsa-devel] [PATCH] ASoC: Add max98088 CODEC driver
by Peter Hsiang 15 Oct '10

15 Oct '10

This patch adds the MAX98088 CODEC driver. Signed-off-by: Peter Hsiang <peter.hsiang(a)maxim-ic.com> --- include/sound/max98088.h | 54 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98088.c | 2355 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98088.h | 190 ++++ 5 files changed, 2605 insertions(+), 0 deletions(-) create mode 100644 include/sound/max98088.h create mode 100644 sound/soc/codecs/max98088.c create mode 100644 sound/soc/codecs/max98088.h diff --git a/include/sound/max98088.h b/include/sound/max98088.h new file mode 100644 index 0000000..30652be --- /dev/null +++ b/include/sound/max98088.h @@ -0,0 +1,54 @@ +/* + * Platform data for MAX98088 + * + * Copyright 2010 Maxim Integrated Products + * + * 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 __SOUND_MAX98088_PDATA_H__ +#define __SOUND_MAX98088_PDATA_H__ + +#define EQ_CFG_MAX 32 + +/* Equalizer filter response configuration */ +struct max98088_eq_cfg { + const char *name; + unsigned int rate; + u16 band1[5]; + u16 band2[5]; + u16 band3[5]; + u16 band4[5]; + u16 band5[5]; +}; + +/* codec platform data */ +struct max98088_pdata { + + /* Equalizers for DAI1 and DAI2 */ + struct max98088_eq_cfg *eq1_cfg; + struct max98088_eq_cfg *eq2_cfg; + unsigned int eq1_cfgcnt; + unsigned int eq2_cfgcnt; + + /* Receiver output can be configured as power amplifier or LINE out */ + /* Set receiver_mode to: + * 0 = amplifier output, or + * 1 = LINE level output + */ + unsigned int receiver_mode:1; + + /* Analog/digital microphone configuration: + * 0 = analog microphone input (normal setting) + * 1 = digital microphone input + */ + unsigned int digmic_left_mode:1; + unsigned int digmic_right_mode:1; + +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4ccc2b7..4e6713c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -27,6 +27,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740 if SOC_JZ4740 + select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF @@ -157,6 +158,9 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate +config SND_SOC_MAX98088 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 23e7e2c..7184611 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-l3-objs := l3.o +snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o @@ -88,6 +89,7 @@ obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c new file mode 100644 index 0000000..52f57d5 --- /dev/null +++ b/sound/soc/codecs/max98088.c @@ -0,0 +1,2355 @@ +/* + * max98088.c -- MAX98088 ALSA SoC Audio driver + * + * Copyright 2010 Maxim Integrated Products + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/slab.h> +#include <sound/max98088.h> +#include "max98088.h" + +/* Configurations associated with each channel of DAI stream */ +struct max98088_cdata { + unsigned int rate; + unsigned int fmt; + int eq_textcnt; + const char *eq_texts[EQ_CFG_MAX]; + int eq_sel; + struct soc_enum eq_enum; +}; + +/* Codec private data */ +struct max98088_priv { + u8 reg_cache[M98088_REG_CNT]; + void *control_data; + struct max98088_pdata *pdata; + + unsigned int sysclk; + struct max98088_cdata dai[2]; + u8 power_state; + unsigned int ex_mode; + unsigned int digmic; + unsigned int mic1pre; + unsigned int mic2pre; + unsigned int extmic_mode; +}; + +static const u8 max98088_reg[M98088_REG_CNT] = { + 0x00, /* 00 IRQ status */ + 0x00, /* 01 MIC status */ + 0x00, /* 02 jack status */ + 0x00, /* 03 battery voltage */ + 0x00, /* 04 */ + 0x00, /* 05 */ + 0x00, /* 06 */ + 0x00, /* 07 */ + 0x00, /* 08 */ + 0x00, /* 09 */ + 0x00, /* 0A */ + 0x00, /* 0B */ + 0x00, /* 0C */ + 0x00, /* 0D */ + 0x00, /* 0E */ + 0x00, /* 0F interrupt enable */ + + 0x00, /* 10 master clock */ + 0x00, /* 11 DAI1 clock mode */ + 0x00, /* 12 DAI1 clock control */ + 0x00, /* 13 DAI1 clock control */ + 0x00, /* 14 DAI1 format */ + 0x00, /* 15 DAI1 clock */ + 0x00, /* 16 DAI1 config */ + 0x00, /* 17 DAI1 TDM */ + 0x00, /* 18 DAI1 filters */ + 0x00, /* 19 DAI2 clock mode */ + 0x00, /* 1A DAI2 clock control */ + 0x00, /* 1B DAI2 clock control */ + 0x00, /* 1C DAI2 format */ + 0x00, /* 1D DAI2 clock */ + 0x00, /* 1E DAI2 config */ + 0x00, /* 1F DAI2 TDM */ + + 0x00, /* 20 DAI2 filters */ + 0x00, /* 21 data config */ + 0x00, /* 22 DAC mixer */ + 0x00, /* 23 left ADC mixer */ + 0x00, /* 24 right ADC mixer */ + 0x00, /* 25 left HP mixer */ + 0x00, /* 26 right HP mixer */ + 0x00, /* 27 HP control */ + 0x00, /* 28 left REC mixer */ + 0x00, /* 29 right REC mixer */ + 0x00, /* 2A REC control */ + 0x00, /* 2B left SPK mixer */ + 0x00, /* 2C right SPK mixer */ + 0x00, /* 2D SPK control */ + 0x00, /* 2E sidetone */ + 0x00, /* 2F DAI1 playback level */ + + 0x00, /* 30 DAI1 playback level */ + 0x00, /* 31 DAI2 playback level */ + 0x00, /* 32 DAI2 playbakc level */ + 0x00, /* 33 left ADC level */ + 0x00, /* 34 right ADC level */ + 0x00, /* 35 MIC1 level */ + 0x00, /* 36 MIC2 level */ + 0x00, /* 37 INA level */ + 0x00, /* 38 INB level */ + 0x00, /* 39 left HP volume */ + 0x00, /* 3A right HP volume */ + 0x00, /* 3B left REC volume */ + 0x00, /* 3C right REC volume */ + 0x00, /* 3D left SPK volume */ + 0x00, /* 3E right SPK volume */ + 0x00, /* 3F MIC config */ + + 0x00, /* 40 MIC threshold */ + 0x00, /* 41 excursion limiter filter */ + 0x00, /* 42 excursion limiter threshold */ + 0x00, /* 43 ALC */ + 0x00, /* 44 power limiter threshold */ + 0x00, /* 45 power limiter config */ + 0x00, /* 46 distortion limiter config */ + 0x00, /* 47 audio input */ + 0x00, /* 48 microphone */ + 0x00, /* 49 level control */ + 0x00, /* 4A bypass switches */ + 0x00, /* 4B jack detect */ + 0x00, /* 4C input enable */ + 0x00, /* 4D output enable */ + 0xF0, /* 4E bias control */ + 0x00, /* 4F DAC power */ + + 0x0F, /* 50 DAC power */ + 0x00, /* 51 system */ + 0x00, /* 52 DAI1 EQ1 */ + 0x00, /* 53 DAI1 EQ1 */ + 0x00, /* 54 DAI1 EQ1 */ + 0x00, /* 55 DAI1 EQ1 */ + 0x00, /* 56 DAI1 EQ1 */ + 0x00, /* 57 DAI1 EQ1 */ + 0x00, /* 58 DAI1 EQ1 */ + 0x00, /* 59 DAI1 EQ1 */ + 0x00, /* 5A DAI1 EQ1 */ + 0x00, /* 5B DAI1 EQ1 */ + 0x00, /* 5C DAI1 EQ2 */ + 0x00, /* 5D DAI1 EQ2 */ + 0x00, /* 5E DAI1 EQ2 */ + 0x00, /* 5F DAI1 EQ2 */ + + 0x00, /* 60 DAI1 EQ2 */ + 0x00, /* 61 DAI1 EQ2 */ + 0x00, /* 62 DAI1 EQ2 */ + 0x00, /* 63 DAI1 EQ2 */ + 0x00, /* 64 DAI1 EQ2 */ + 0x00, /* 65 DAI1 EQ2 */ + 0x00, /* 66 DAI1 EQ3 */ + 0x00, /* 67 DAI1 EQ3 */ + 0x00, /* 68 DAI1 EQ3 */ + 0x00, /* 69 DAI1 EQ3 */ + 0x00, /* 6A DAI1 EQ3 */ + 0x00, /* 6B DAI1 EQ3 */ + 0x00, /* 6C DAI1 EQ3 */ + 0x00, /* 6D DAI1 EQ3 */ + 0x00, /* 6E DAI1 EQ3 */ + 0x00, /* 6F DAI1 EQ3 */ + + 0x00, /* 70 DAI1 EQ4 */ + 0x00, /* 71 DAI1 EQ4 */ + 0x00, /* 72 DAI1 EQ4 */ + 0x00, /* 73 DAI1 EQ4 */ + 0x00, /* 74 DAI1 EQ4 */ + 0x00, /* 75 DAI1 EQ4 */ + 0x00, /* 76 DAI1 EQ4 */ + 0x00, /* 77 DAI1 EQ4 */ + 0x00, /* 78 DAI1 EQ4 */ + 0x00, /* 79 DAI1 EQ4 */ + 0x00, /* 7A DAI1 EQ5 */ + 0x00, /* 7B DAI1 EQ5 */ + 0x00, /* 7C DAI1 EQ5 */ + 0x00, /* 7D DAI1 EQ5 */ + 0x00, /* 7E DAI1 EQ5 */ + 0x00, /* 7F DAI1 EQ5 */ + + 0x00, /* 80 DAI1 EQ5 */ + 0x00, /* 81 DAI1 EQ5 */ + 0x00, /* 82 DAI1 EQ5 */ + 0x00, /* 83 DAI1 EQ5 */ + 0x00, /* 84 DAI2 EQ1 */ + 0x00, /* 85 DAI2 EQ1 */ + 0x00, /* 86 DAI2 EQ1 */ + 0x00, /* 87 DAI2 EQ1 */ + 0x00, /* 88 DAI2 EQ1 */ + 0x00, /* 89 DAI2 EQ1 */ + 0x00, /* 8A DAI2 EQ1 */ + 0x00, /* 8B DAI2 EQ1 */ + 0x00, /* 8C DAI2 EQ1 */ + 0x00, /* 8D DAI2 EQ1 */ + 0x00, /* 8E DAI2 EQ2 */ + 0x00, /* 8F DAI2 EQ2 */ + + 0x00, /* 90 DAI2 EQ2 */ + 0x00, /* 91 DAI2 EQ2 */ + 0x00, /* 92 DAI2 EQ2 */ + 0x00, /* 93 DAI2 EQ2 */ + 0x00, /* 94 DAI2 EQ2 */ + 0x00, /* 95 DAI2 EQ2 */ + 0x00, /* 96 DAI2 EQ2 */ + 0x00, /* 97 DAI2 EQ2 */ + 0x00, /* 98 DAI2 EQ3 */ + 0x00, /* 99 DAI2 EQ3 */ + 0x00, /* 9A DAI2 EQ3 */ + 0x00, /* 9B DAI2 EQ3 */ + 0x00, /* 9C DAI2 EQ3 */ + 0x00, /* 9D DAI2 EQ3 */ + 0x00, /* 9E DAI2 EQ3 */ + 0x00, /* 9F DAI2 EQ3 */ + + 0x00, /* A0 DAI2 EQ3 */ + 0x00, /* A1 DAI2 EQ3 */ + 0x00, /* A2 DAI2 EQ4 */ + 0x00, /* A3 DAI2 EQ4 */ + 0x00, /* A4 DAI2 EQ4 */ + 0x00, /* A5 DAI2 EQ4 */ + 0x00, /* A6 DAI2 EQ4 */ + 0x00, /* A7 DAI2 EQ4 */ + 0x00, /* A8 DAI2 EQ4 */ + 0x00, /* A9 DAI2 EQ4 */ + 0x00, /* AA DAI2 EQ4 */ + 0x00, /* AB DAI2 EQ4 */ + 0x00, /* AC DAI2 EQ5 */ + 0x00, /* AD DAI2 EQ5 */ + 0x00, /* AE DAI2 EQ5 */ + 0x00, /* AF DAI2 EQ5 */ + + 0x00, /* B0 DAI2 EQ5 */ + 0x00, /* B1 DAI2 EQ5 */ + 0x00, /* B2 DAI2 EQ5 */ + 0x00, /* B3 DAI2 EQ5 */ + 0x00, /* B4 DAI2 EQ5 */ + 0x00, /* B5 DAI2 EQ5 */ + 0x00, /* B6 DAI1 biquad */ + 0x00, /* B7 DAI1 biquad */ + 0x00, /* B8 DAI1 biquad */ + 0x00, /* B9 DAI1 biquad */ + 0x00, /* BA DAI1 biquad */ + 0x00, /* BB DAI1 biquad */ + 0x00, /* BC DAI1 biquad */ + 0x00, /* BD DAI1 biquad */ + 0x00, /* BE DAI1 biquad */ + 0x00, /* BF DAI1 biquad */ + + 0x00, /* C0 DAI2 biquad */ + 0x00, /* C1 DAI2 biquad */ + 0x00, /* C2 DAI2 biquad */ + 0x00, /* C3 DAI2 biquad */ + 0x00, /* C4 DAI2 biquad */ + 0x00, /* C5 DAI2 biquad */ + 0x00, /* C6 DAI2 biquad */ + 0x00, /* C7 DAI2 biquad */ + 0x00, /* C8 DAI2 biquad */ + 0x00, /* C9 DAI2 biquad */ + 0x00, /* CA */ + 0x00, /* CB */ + 0x00, /* CC */ + 0x00, /* CD */ + 0x00, /* CE */ + 0x00, /* CF */ + + 0x00, /* D0 */ + 0x00, /* D1 */ + 0x00, /* D2 */ + 0x00, /* D3 */ + 0x00, /* D4 */ + 0x00, /* D5 */ + 0x00, /* D6 */ + 0x00, /* D7 */ + 0x00, /* D8 */ + 0x00, /* D9 */ + 0x00, /* DA */ + 0x70, /* DB */ + 0x00, /* DC */ + 0x00, /* DD */ + 0x00, /* DE */ + 0x00, /* DF */ + + 0x00, /* E0 */ + 0x00, /* E1 */ + 0x00, /* E2 */ + 0x00, /* E3 */ + 0x00, /* E4 */ + 0x00, /* E5 */ + 0x00, /* E6 */ + 0x00, /* E7 */ + 0x00, /* E8 */ + 0x00, /* E9 */ + 0x00, /* EA */ + 0x00, /* EB */ + 0x00, /* EC */ + 0x00, /* ED */ + 0x00, /* EE */ + 0x00, /* EF */ + + 0x00, /* F0 */ + 0x00, /* F1 */ + 0x00, /* F2 */ + 0x00, /* F3 */ + 0x00, /* F4 */ + 0x00, /* F5 */ + 0x00, /* F6 */ + 0x00, /* F7 */ + 0x00, /* F8 */ + 0x00, /* F9 */ + 0x00, /* FA */ + 0x00, /* FB */ + 0x00, /* FC */ + 0x00, /* FD */ + 0x00, /* FE */ + 0x00, /* FF */ +}; + +static struct { + int readable; + int writable; + int vol; +} max98088_access[M98088_REG_CNT] = { + { 0xFF, 0xFF, 1 }, /* 00 IRQ status */ + { 0xFF, 0x00, 1 }, /* 01 MIC status */ + { 0xFF, 0x00, 1 }, /* 02 jack status */ + { 0x1F, 0x1F, 1 }, /* 03 battery voltage */ + { 0xFF, 0xFF, 0 }, /* 04 */ + { 0xFF, 0xFF, 0 }, /* 05 */ + { 0xFF, 0xFF, 0 }, /* 06 */ + { 0xFF, 0xFF, 0 }, /* 07 */ + { 0xFF, 0xFF, 0 }, /* 08 */ + { 0xFF, 0xFF, 0 }, /* 09 */ + { 0xFF, 0xFF, 0 }, /* 0A */ + { 0xFF, 0xFF, 0 }, /* 0B */ + { 0xFF, 0xFF, 0 }, /* 0C */ + { 0xFF, 0xFF, 0 }, /* 0D */ + { 0xFF, 0xFF, 0 }, /* 0E */ + { 0xFF, 0xFF, 0 }, /* 0F interrupt enable */ + + { 0xFF, 0xFF, 0 }, /* 10 master clock */ + { 0xFF, 0xFF, 0 }, /* 11 DAI1 clock mode */ + { 0xFF, 0xFF, 0 }, /* 12 DAI1 clock control */ + { 0xFF, 0xFF, 0 }, /* 13 DAI1 clock control */ + { 0xFF, 0xFF, 0 }, /* 14 DAI1 format */ + { 0xFF, 0xFF, 0 }, /* 15 DAI1 clock */ + { 0xFF, 0xFF, 0 }, /* 16 DAI1 config */ + { 0xFF, 0xFF, 0 }, /* 17 DAI1 TDM */ + { 0xFF, 0xFF, 0 }, /* 18 DAI1 filters */ + { 0xFF, 0xFF, 0 }, /* 19 DAI2 clock mode */ + { 0xFF, 0xFF, 0 }, /* 1A DAI2 clock control */ + { 0xFF, 0xFF, 0 }, /* 1B DAI2 clock control */ + { 0xFF, 0xFF, 0 }, /* 1C DAI2 format */ + { 0xFF, 0xFF, 0 }, /* 1D DAI2 clock */ + { 0xFF, 0xFF, 0 }, /* 1E DAI2 config */ + { 0xFF, 0xFF, 0 }, /* 1F DAI2 TDM */ + + { 0xFF, 0xFF, 0 }, /* 20 DAI2 filters */ + { 0xFF, 0xFF, 0 }, /* 21 data config */ + { 0xFF, 0xFF, 0 }, /* 22 DAC mixer */ + { 0xFF, 0xFF, 0 }, /* 23 left ADC mixer */ + { 0xFF, 0xFF, 0 }, /* 24 right ADC mixer */ + { 0xFF, 0xFF, 0 }, /* 25 left HP mixer */ + { 0xFF, 0xFF, 0 }, /* 26 right HP mixer */ + { 0xFF, 0xFF, 0 }, /* 27 HP control */ + { 0xFF, 0xFF, 0 }, /* 28 left REC mixer */ + { 0xFF, 0xFF, 0 }, /* 29 right REC mixer */ + { 0xFF, 0xFF, 0 }, /* 2A REC control */ + { 0xFF, 0xFF, 0 }, /* 2B left SPK mixer */ + { 0xFF, 0xFF, 0 }, /* 2C right SPK mixer */ + { 0xFF, 0xFF, 0 }, /* 2D SPK control */ + { 0xFF, 0xFF, 0 }, /* 2E sidetone */ + { 0xFF, 0xFF, 0 }, /* 2F DAI1 playback level */ + + { 0xFF, 0xFF, 0 }, /* 30 DAI1 playback level */ + { 0xFF, 0xFF, 0 }, /* 31 DAI2 playback level */ + { 0xFF, 0xFF, 0 }, /* 32 DAI2 playbakc level */ + { 0xFF, 0xFF, 0 }, /* 33 left ADC level */ + { 0xFF, 0xFF, 0 }, /* 34 right ADC level */ + { 0xFF, 0xFF, 0 }, /* 35 MIC1 level */ + { 0xFF, 0xFF, 0 }, /* 36 MIC2 level */ + { 0xFF, 0xFF, 0 }, /* 37 INA level */ + { 0xFF, 0xFF, 0 }, /* 38 INB level */ + { 0xFF, 0xFF, 0 }, /* 39 left HP volume */ + { 0xFF, 0xFF, 0 }, /* 3A right HP volume */ + { 0xFF, 0xFF, 0 }, /* 3B left REC volume */ + { 0xFF, 0xFF, 0 }, /* 3C right REC volume */ + { 0xFF, 0xFF, 0 }, /* 3D left SPK volume */ + { 0xFF, 0xFF, 0 }, /* 3E right SPK volume */ + { 0xFF, 0xFF, 0 }, /* 3F MIC config */ + + { 0xFF, 0xFF, 0 }, /* 40 MIC threshold */ + { 0xFF, 0xFF, 0 }, /* 41 excursion limiter filter */ + { 0xFF, 0xFF, 0 }, /* 42 excursion limiter threshold */ + { 0xFF, 0xFF, 0 }, /* 43 ALC */ + { 0xFF, 0xFF, 0 }, /* 44 power limiter threshold */ + { 0xFF, 0xFF, 0 }, /* 45 power limiter config */ + { 0xFF, 0xFF, 0 }, /* 46 distortion limiter config */ + { 0xFF, 0xFF, 0 }, /* 47 audio input */ + { 0xFF, 0xFF, 0 }, /* 48 microphone */ + { 0xFF, 0xFF, 0 }, /* 49 level control */ + { 0xFF, 0xFF, 0 }, /* 4A bypass switches */ + { 0xFF, 0xFF, 0 }, /* 4B jack detect */ + { 0xFF, 0xFF, 0 }, /* 4C input enable */ + { 0xFF, 0xFF, 0 }, /* 4D output enable */ + { 0xFF, 0xFF, 0 }, /* 4E bias control */ + { 0xFF, 0xFF, 0 }, /* 4F DAC power */ + + { 0xFF, 0xFF, 0 }, /* 50 DAC power */ + { 0xFF, 0xFF, 0 }, /* 51 system */ + { 0xFF, 0xFF, 0 }, /* 52 DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 53 DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 54 DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 55 DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 56 DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 57 DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 58 DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 59 DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 5A DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 5B DAI1 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 5C DAI1 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 5D DAI1 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 5E DAI1 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 5F DAI1 EQ2 */ + + { 0xFF, 0xFF, 0 }, /* 60 DAI1 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 61 DAI1 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 62 DAI1 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 63 DAI1 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 64 DAI1 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 65 DAI1 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 66 DAI1 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 67 DAI1 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 68 DAI1 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 69 DAI1 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 6A DAI1 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 6B DAI1 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 6C DAI1 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 6D DAI1 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 6E DAI1 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 6F DAI1 EQ3 */ + + { 0xFF, 0xFF, 0 }, /* 70 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 71 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 72 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 73 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 74 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 75 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 76 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 77 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 78 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 79 DAI1 EQ4 */ + { 0xFF, 0xFF, 0 }, /* 7A DAI1 EQ5 */ + { 0xFF, 0xFF, 0 }, /* 7B DAI1 EQ5 */ + { 0xFF, 0xFF, 0 }, /* 7C DAI1 EQ5 */ + { 0xFF, 0xFF, 0 }, /* 7D DAI1 EQ5 */ + { 0xFF, 0xFF, 0 }, /* 7E DAI1 EQ5 */ + { 0xFF, 0xFF, 0 }, /* 7F DAI1 EQ5 */ + + { 0xFF, 0xFF, 0 }, /* 80 DAI1 EQ5 */ + { 0xFF, 0xFF, 0 }, /* 81 DAI1 EQ5 */ + { 0xFF, 0xFF, 0 }, /* 82 DAI1 EQ5 */ + { 0xFF, 0xFF, 0 }, /* 83 DAI1 EQ5 */ + { 0xFF, 0xFF, 0 }, /* 84 DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 85 DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 86 DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 87 DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 88 DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 89 DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 8A DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 8B DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 8C DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 8D DAI2 EQ1 */ + { 0xFF, 0xFF, 0 }, /* 8E DAI2 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 8F DAI2 EQ2 */ + + { 0xFF, 0xFF, 0 }, /* 90 DAI2 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 91 DAI2 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 92 DAI2 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 93 DAI2 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 94 DAI2 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 95 DAI2 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 96 DAI2 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 97 DAI2 EQ2 */ + { 0xFF, 0xFF, 0 }, /* 98 DAI2 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 99 DAI2 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 9A DAI2 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 9B DAI2 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 9C DAI2 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 9D DAI2 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 9E DAI2 EQ3 */ + { 0xFF, 0xFF, 0 }, /* 9F DAI2 EQ3 */ + + { 0xFF, 0xFF, 0 }, /* A0 DAI2 EQ3 */ + { 0xFF, 0xFF, 0 }, /* A1 DAI2 EQ3 */ + { 0xFF, 0xFF, 0 }, /* A2 DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* A3 DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* A4 DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* A5 DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* A6 DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* A7 DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* A8 DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* A9 DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* AA DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* AB DAI2 EQ4 */ + { 0xFF, 0xFF, 0 }, /* AC DAI2 EQ5 */ + { 0xFF, 0xFF, 0 }, /* AD DAI2 EQ5 */ + { 0xFF, 0xFF, 0 }, /* AE DAI2 EQ5 */ + { 0xFF, 0xFF, 0 }, /* AF DAI2 EQ5 */ + + { 0xFF, 0xFF, 0 }, /* B0 DAI2 EQ5 */ + { 0xFF, 0xFF, 0 }, /* B1 DAI2 EQ5 */ + { 0xFF, 0xFF, 0 }, /* B2 DAI2 EQ5 */ + { 0xFF, 0xFF, 0 }, /* B3 DAI2 EQ5 */ + { 0xFF, 0xFF, 0 }, /* B4 DAI2 EQ5 */ + { 0xFF, 0xFF, 0 }, /* B5 DAI2 EQ5 */ + { 0xFF, 0xFF, 0 }, /* B6 DAI1 biquad */ + { 0xFF, 0xFF, 0 }, /* B7 DAI1 biquad */ + { 0xFF, 0xFF, 0 }, /* B8 DAI1 biquad */ + { 0xFF, 0xFF, 0 }, /* B9 DAI1 biquad */ + { 0xFF, 0xFF, 0 }, /* BA DAI1 biquad */ + { 0xFF, 0xFF, 0 }, /* BB DAI1 biquad */ + { 0xFF, 0xFF, 0 }, /* BC DAI1 biquad */ + { 0xFF, 0xFF, 0 }, /* BD DAI1 biquad */ + { 0xFF, 0xFF, 0 }, /* BE DAI1 biquad */ + { 0xFF, 0xFF, 0 }, /* BF DAI1 biquad */ + + { 0xFF, 0xFF, 0 }, /* C0 DAI2 biquad */ + { 0xFF, 0xFF, 0 }, /* C1 DAI2 biquad */ + { 0xFF, 0xFF, 0 }, /* C2 DAI2 biquad */ + { 0xFF, 0xFF, 0 }, /* C3 DAI2 biquad */ + { 0xFF, 0xFF, 0 }, /* C4 DAI2 biquad */ + { 0xFF, 0xFF, 0 }, /* C5 DAI2 biquad */ + { 0xFF, 0xFF, 0 }, /* C6 DAI2 biquad */ + { 0xFF, 0xFF, 0 }, /* C7 DAI2 biquad */ + { 0xFF, 0xFF, 0 }, /* C8 DAI2 biquad */ + { 0xFF, 0xFF, 0 }, /* C9 DAI2 biquad */ + { 0x00, 0x00, 0 }, /* CA */ + { 0x00, 0x00, 0 }, /* CB */ + { 0x00, 0x00, 0 }, /* CC */ + { 0x00, 0x00, 0 }, /* CD */ + { 0x00, 0x00, 0 }, /* CE */ + { 0x00, 0x00, 0 }, /* CF */ + + { 0x00, 0x00, 0 }, /* D0 */ + { 0x00, 0x00, 0 }, /* D1 */ + { 0x00, 0x00, 0 }, /* D2 */ + { 0x00, 0x00, 0 }, /* D3 */ + { 0x00, 0x00, 0 }, /* D4 */ + { 0x00, 0x00, 0 }, /* D5 */ + { 0x00, 0x00, 0 }, /* D6 */ + { 0x00, 0x00, 0 }, /* D7 */ + { 0x00, 0x00, 0 }, /* D8 */ + { 0x00, 0x00, 0 }, /* D9 */ + { 0x00, 0x00, 0 }, /* DA */ + { 0x00, 0x00, 0 }, /* DB */ + { 0x00, 0x00, 0 }, /* DC */ + { 0x00, 0x00, 0 }, /* DD */ + { 0x00, 0x00, 0 }, /* DE */ + { 0x00, 0x00, 0 }, /* DF */ + + { 0x00, 0x00, 0 }, /* E0 */ + { 0x00, 0x00, 0 }, /* E1 */ + { 0x00, 0x00, 0 }, /* E2 */ + { 0x00, 0x00, 0 }, /* E3 */ + { 0x00, 0x00, 0 }, /* E4 */ + { 0x00, 0x00, 0 }, /* E5 */ + { 0x00, 0x00, 0 }, /* E6 */ + { 0x00, 0x00, 0 }, /* E7 */ + { 0x00, 0x00, 0 }, /* E8 */ + { 0x00, 0x00, 0 }, /* E9 */ + { 0x00, 0x00, 0 }, /* EA */ + { 0x00, 0x00, 0 }, /* EB */ + { 0x00, 0x00, 0 }, /* EC */ + { 0x00, 0x00, 0 }, /* ED */ + { 0x00, 0x00, 0 }, /* EE */ + { 0x00, 0x00, 0 }, /* EF */ + + { 0x00, 0x00, 0 }, /* F0 */ + { 0x00, 0x00, 0 }, /* F1 */ + { 0x00, 0x00, 0 }, /* F2 */ + { 0x00, 0x00, 0 }, /* F3 */ + { 0x00, 0x00, 0 }, /* F4 */ + { 0x00, 0x00, 0 }, /* F5 */ + { 0x00, 0x00, 0 }, /* F6 */ + { 0x00, 0x00, 0 }, /* F7 */ + { 0x00, 0x00, 0 }, /* F8 */ + { 0x00, 0x00, 0 }, /* F9 */ + { 0x00, 0x00, 0 }, /* FA */ + { 0x00, 0x00, 0 }, /* FB */ + { 0x00, 0x00, 0 }, /* FC */ + { 0x00, 0x00, 0 }, /* FD */ + { 0x00, 0x00, 0 }, /* FE */ + { 0xFF, 0x00, 1 }, /* FF */ +}; + +static int max98088_volatile_register(unsigned int reg) +{ + return max98088_access[reg].vol; +} + +static int max98088_hw_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + data[0] = reg; + data[1] = value; + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +/* + * For kernels compiled without unsigned long long int division + */ +unsigned long long int ulldiv(unsigned long long int dividend, + unsigned long long int divisor) +{ + unsigned long long int quotient = 0; + int shift = 1; + + BUG_ON(divisor == 0); + + /* Result is 1.0 if divisor and dividend are equal */ + if (divisor == dividend) + return 1; + + /* Normalize divisor */ + while (!(divisor & 0x8000000000000000ULL)) { + divisor <<= 1; + ++shift; + } + + /* Shift and subtract */ + while (shift--) { + quotient <<= 1; + + if (divisor <= dividend) { + dividend -= divisor; + ++quotient; + } + divisor >>= 1; + } + + /* Round up */ + if (dividend > divisor) + ++quotient; + + return quotient; +} + +#define INA1_PGA_BIT 0x01 +#define INA2_PGA_BIT 0x02 +#define INB1_PGA_BIT 0x04 +#define INB2_PGA_BIT 0x08 +/* + * The INx1 and INx2 PGAs share a power control signal. + * This function OR's the two power events to keep an unpowered INx + * from turning off it's counterpart. + * The control names are used to identify the PGA. + */ +static int max98088_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + u8 *state = &max98088->power_state; + unsigned int val; + unsigned int pga; + unsigned int mask; + + BUG_ON(w->reg != M98088_REG_4C_PWR_EN_IN); + + if (strncmp(w->name, "INA1", 4) == 0) { + pga = INA1_PGA_BIT; + mask = INA1_PGA_BIT | INA2_PGA_BIT; + } else if (strncmp(w->name, "INA2", 4) == 0) { + pga = INA2_PGA_BIT; + mask = INA1_PGA_BIT | INA2_PGA_BIT; + } else if (strncmp(w->name, "INB1", 4) == 0) { + pga = INB1_PGA_BIT; + mask = INB1_PGA_BIT | INB2_PGA_BIT; + } else if (strncmp(w->name, "INB2", 4) == 0) { + pga = INB2_PGA_BIT; + mask = INB1_PGA_BIT | INB2_PGA_BIT; + } else { + return -EINVAL; + } + + if (event == SND_SOC_DAPM_POST_PMU) { + /* ON */ + *state |= pga; + + /* Turn on, avoiding unnecessary writes */ + val = snd_soc_read(codec, w->reg); + if (!(val & (1 << w->shift))) { + val |= (1 << w->shift); + snd_soc_write(codec, w->reg, val); + } + } else if (event == SND_SOC_DAPM_POST_PMD) { + /* OFF */ + *state &= ~pga; + + /* Turn off if both are off, avoiding unnecessary writes */ + if (!(*state & mask)) { + val = snd_soc_read(codec, w->reg); + if (val & (1 << w->shift)) { + val &= ~(1 << w->shift); + snd_soc_write(codec, w->reg, val); + } + } + } else { + return -EINVAL; + } + + return 0; +} + +/* + * Load equalizer DSP coefficient configurations registers + */ +void m98088_eq_band(struct snd_soc_codec *codec, unsigned int dai, + unsigned int band, u16 *coefs) +{ + unsigned int eq_reg; + unsigned int i; + + BUG_ON(band > 4); + BUG_ON(dai > 1); + + /* Load the base register address */ + eq_reg = dai ? M98088_REG_84_DAI2_EQ_BASE : M98088_REG_52_DAI1_EQ_BASE; + + /* Add the band address offset, note adjustment for word address */ + eq_reg += band * (M98088_COEFS_PER_BAND << 1); + + /* Step through the registers and coefs */ + for (i = 0; i < M98088_COEFS_PER_BAND; i++) { + snd_soc_write(codec, eq_reg++, M98088_BYTE1(coefs[i])); + snd_soc_write(codec, eq_reg++, M98088_BYTE0(coefs[i])); + } + + return; +} + +/* + * Excursion limiter modes + */ +static const char *max98088_ex_mode[] = { + "Off", + "100Hz", + "400Hz", + "600Hz", + "800Hz", + "1000Hz", + "200-400Hz", + "400-600Hz", + "400-800Hz", +}; + +static const unsigned int ex_mode_table[] = { + 0x00, /* disabled */ + (0<<4)|3, /* 100Hz */ + (1<<4)|0, /* 400Hz */ + (2<<4)|0, /* 600Hz */ + (3<<4)|0, /* 800Hz */ + (4<<4)|0, /* 1000Hz */ + (1<<4)|1, /* 200-400Hz */ + (2<<4)|2, /* 400-600Hz */ + (3<<4)|2, /* 400-800Hz */ +}; + +static const struct soc_enum max98088_ex_mode_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max98088_ex_mode), max98088_ex_mode), +}; + +static int max98088_ex_mode_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + unsigned int *mode = &max98088->ex_mode; + int sel = ucontrol->value.integer.value[0]; + + if (sel >= ARRAY_SIZE(ex_mode_table)) + return -EINVAL; + + *mode = ucontrol->value.integer.value[0]; + snd_soc_write(codec, M98088_REG_41_SPKDHP, + ex_mode_table[*mode]); + + return 0; +} + +static int max98088_ex_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + unsigned int *mode = &max98088->ex_mode; + + ucontrol->value.integer.value[0] = *mode; + return 0; +} + +static const char *max98088_ex_thresh[] = { /* volts PP */ + "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"}; +static const struct soc_enum max98088_ex_thresh_enum[] = { + SOC_ENUM_SINGLE(M98088_REG_42_SPKDHP_THRESH, 0, 8, + max98088_ex_thresh), +}; + +static const char *max98088_fltr_mode[] = {"Voice", "Music" }; +static const struct soc_enum max98088_filter_mode_enum[] = { + SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 7, 2, max98088_fltr_mode), +}; + +static const char *max98088_dai1_fltr[] = { + "Off", "fc=258/fs=16k", "fc=500/fs=16k", + "fc=258/fs=8k", "fc=500/fs=8k", "fc=200"}; +static const struct soc_enum max98088_dai1_dac_filter_enum[] = { + SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 0, 6, max98088_dai1_fltr), +}; +static const struct soc_enum max98088_dai1_adc_filter_enum[] = { + SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 4, 6, max98088_dai1_fltr), +}; + +static const char *max98088_micpre[] = { + "0dB", + "20dB", + "30dB", +}; + +static const struct soc_enum max98088_micpre_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max98088_micpre), max98088_micpre), +}; + +static const char *max98088_extmic[] = { + "Off", + "MIC1", + "MIC2", +}; + +static const struct soc_enum max98088_extmic_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max98088_extmic), max98088_extmic), +}; + +static int max98088_mic1pre_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + unsigned int *mode = &max98088->mic1pre; + int sel = ucontrol->value.integer.value[0]; + + if (sel >= ARRAY_SIZE(max98088_micpre)) + return -EINVAL; + + *mode = ucontrol->value.integer.value[0]; + return 0; +} + +static int max98088_mic1pre_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + unsigned int *mode = &max98088->mic1pre; + + ucontrol->value.integer.value[0] = *mode; + return 0; +} + +static int max98088_mic2pre_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + unsigned int *mode = &max98088->mic2pre; + int sel = ucontrol->value.integer.value[0]; + + if (sel >= ARRAY_SIZE(max98088_micpre)) + return -EINVAL; + + *mode = ucontrol->value.integer.value[0]; + return 0; +} + +static int max98088_mic2pre_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + unsigned int *mode = &max98088->mic2pre; + + ucontrol->value.integer.value[0] = *mode; + return 0; +} + +static int max98088_extmic_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + unsigned int *mode = &max98088->extmic_mode; + int sel = ucontrol->value.integer.value[0]; + + if (sel >= ARRAY_SIZE(max98088_extmic)) + return -EINVAL; + + *mode = sel; + snd_soc_update_bits(codec, M98088_REG_48_CFG_MIC, + M98088_EXTMIC_MASK, sel); + + return 0; +} + +static int max98088_extmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + unsigned int *mode = &max98088->extmic_mode; + + ucontrol->value.integer.value[0] = *mode; + return 0; +} + +static const struct snd_kcontrol_new max98088_snd_controls[] = { + + /* Analog outputs */ + + SOC_DOUBLE_R("Headphone Volume", M98088_REG_39_LVL_HP_L, + M98088_REG_3A_LVL_HP_R, 0, 31, 0), + SOC_DOUBLE_R("Speaker Volume", M98088_REG_3D_LVL_SPK_L, + M98088_REG_3E_LVL_SPK_R, 0, 31, 0), + SOC_DOUBLE_R("Receiver Volume", M98088_REG_3B_LVL_REC_L, + M98088_REG_3C_LVL_REC_R, 0, 31, 0), + + SOC_DOUBLE_R("Headphone Switch", M98088_REG_39_LVL_HP_L, + M98088_REG_3A_LVL_HP_R, 7, 1, 1), + SOC_DOUBLE_R("Speaker Switch", M98088_REG_3D_LVL_SPK_L, + M98088_REG_3E_LVL_SPK_R, 7, 1, 1), + SOC_DOUBLE_R("Receiver Switch", M98088_REG_3B_LVL_REC_L, + M98088_REG_3C_LVL_REC_R, 7, 1, 1), + + /* Analog inputs */ + + SOC_SINGLE("MIC1 Volume", M98088_REG_35_LVL_MIC1, 0, 31, 1), + SOC_SINGLE("MIC2 Volume", M98088_REG_36_LVL_MIC2, 0, 31, 1), + + SOC_ENUM_EXT("MIC1 Boost Volume", max98088_micpre_enum, + max98088_mic1pre_get, max98088_mic1pre_set), + + SOC_ENUM_EXT("MIC2 Boost Volume", max98088_micpre_enum, + max98088_mic2pre_get, max98088_mic2pre_set), + + SOC_ENUM_EXT("Ext MIC Switch", max98088_extmic_enum, + max98088_extmic_get, max98088_extmic_set), + + SOC_SINGLE("INA Volume", M98088_REG_37_LVL_INA, 0, 7, 1), + SOC_SINGLE("INB Volume", M98088_REG_38_LVL_INB, 0, 7, 1), + + /* ADC input digital gains and volume controls */ + + SOC_SINGLE("ADCL Volume", M98088_REG_33_LVL_ADC_L, 0, 15, 0), + SOC_SINGLE("ADCR Volume", M98088_REG_34_LVL_ADC_R, 0, 15, 0), + + SOC_SINGLE("ADCL Boost Volume", M98088_REG_33_LVL_ADC_L, 4, 3, 0), + SOC_SINGLE("ADCR Boost Volume", M98088_REG_34_LVL_ADC_R, 4, 3, 0), + + /* Equalizer */ + + SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0), + SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0), + + /* Excursion limiter */ + + SOC_ENUM_EXT("EX Limiter Mode", max98088_ex_mode_enum, + max98088_ex_mode_get, max98088_ex_mode_set), + SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum), + + /* Voice/music filters */ + + SOC_ENUM("DAI1 Filter Mode", max98088_filter_mode_enum), + SOC_ENUM("DAI1 DAC Filter", max98088_dai1_dac_filter_enum), + SOC_ENUM("DAI1 ADC Filter", max98088_dai1_adc_filter_enum), + SOC_SINGLE("DAI2 DC Block Switch", M98088_REG_20_DAI2_FILTERS, + 0, 1, 0), + + /* Automatic level control (for both DAI1/DAI2) */ + + SOC_SINGLE("ALC Switch", M98088_REG_43_SPKALC_COMP, 7, 1, 0), + SOC_SINGLE("ALC Threshold", M98088_REG_43_SPKALC_COMP, 0, 7, 0), + SOC_SINGLE("ALC Multiband", M98088_REG_43_SPKALC_COMP, 3, 1, 0), + SOC_SINGLE("ALC Release Time", M98088_REG_43_SPKALC_COMP, 4, 7, 0), + + /* Power limiter */ + + SOC_SINGLE("PWR Limiter Threshold", M98088_REG_44_PWRLMT_CFG, + 4, 15, 0), + SOC_SINGLE("PWR Limiter Weight", M98088_REG_44_PWRLMT_CFG, 0, 7, 0), + SOC_SINGLE("PWR Limiter Time1", M98088_REG_45_PWRLMT_TIME, 0, 15, 0), + SOC_SINGLE("PWR Limiter Time2", M98088_REG_45_PWRLMT_TIME, 4, 15, 0), + + /* THD distortion limiter */ + + SOC_SINGLE("THD Limiter Threshold", M98088_REG_46_THDLMT_CFG, 4, 15, 0), + SOC_SINGLE("THD Limiter Time", M98088_REG_46_THDLMT_CFG, 0, 7, 0), +}; + +/* Left speaker mixer switch */ +static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1, 0), + SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1, 0), + SOC_DAPM_SINGLE("INA2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 2, 1, 0), + SOC_DAPM_SINGLE("INB1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 3, 1, 0), + SOC_DAPM_SINGLE("INB2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 4, 1, 0), +}; + +/* Right speaker mixer switch */ +static const struct snd_kcontrol_new max98088_right_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 5, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 6, 1, 0), + SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 1, 1, 0), + SOC_DAPM_SINGLE("INA2 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 2, 1, 0), + SOC_DAPM_SINGLE("INB1 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 3, 1, 0), + SOC_DAPM_SINGLE("INB2 Switch", M98088_REG_2C_MIX_SPK_RIGHT, 4, 1, 0), +}; + +/* Left headphone mixer switch */ +static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1, 0), + SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1, 0), + SOC_DAPM_SINGLE("INA2 Switch", M98088_REG_25_MIX_HP_LEFT, 2, 1, 0), + SOC_DAPM_SINGLE("INB1 Switch", M98088_REG_25_MIX_HP_LEFT, 3, 1, 0), + SOC_DAPM_SINGLE("INB2 Switch", M98088_REG_25_MIX_HP_LEFT, 4, 1, 0), +}; + +/* Right headphone mixer switch */ +static const struct snd_kcontrol_new max98088_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_26_MIX_HP_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_26_MIX_HP_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_26_MIX_HP_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_26_MIX_HP_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_26_MIX_HP_RIGHT, 5, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_26_MIX_HP_RIGHT, 6, 1, 0), + SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_26_MIX_HP_RIGHT, 1, 1, 0), + SOC_DAPM_SINGLE("INA2 Switch", M98088_REG_26_MIX_HP_RIGHT, 2, 1, 0), + SOC_DAPM_SINGLE("INB1 Switch", M98088_REG_26_MIX_HP_RIGHT, 3, 1, 0), + SOC_DAPM_SINGLE("INB2 Switch", M98088_REG_26_MIX_HP_RIGHT, 4, 1, 0), +}; + +/* Left earpiece/receiver mixer switch */ +static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1, 0), + SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1, 0), + SOC_DAPM_SINGLE("INA2 Switch", M98088_REG_28_MIX_REC_LEFT, 2, 1, 0), + SOC_DAPM_SINGLE("INB1 Switch", M98088_REG_28_MIX_REC_LEFT, 3, 1, 0), + SOC_DAPM_SINGLE("INB2 Switch", M98088_REG_28_MIX_REC_LEFT, 4, 1, 0), +}; + +/* Right earpiece/receiver mixer switch */ +static const struct snd_kcontrol_new max98088_right_rec_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_29_MIX_REC_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_29_MIX_REC_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_29_MIX_REC_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_29_MIX_REC_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_29_MIX_REC_RIGHT, 5, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_29_MIX_REC_RIGHT, 6, 1, 0), + SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_29_MIX_REC_RIGHT, 1, 1, 0), + SOC_DAPM_SINGLE("INA2 Switch", M98088_REG_29_MIX_REC_RIGHT, 2, 1, 0), + SOC_DAPM_SINGLE("INB1 Switch", M98088_REG_29_MIX_REC_RIGHT, 3, 1, 0), + SOC_DAPM_SINGLE("INB2 Switch", M98088_REG_29_MIX_REC_RIGHT, 4, 1, 0), +}; + +/* Left ADC mixer switch */ +static const struct snd_kcontrol_new max98088_left_ADC_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_23_MIX_ADC_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_23_MIX_ADC_LEFT, 6, 1, 0), + SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_23_MIX_ADC_LEFT, 3, 1, 0), + SOC_DAPM_SINGLE("INA2 Switch", M98088_REG_23_MIX_ADC_LEFT, 2, 1, 0), + SOC_DAPM_SINGLE("INB1 Switch", M98088_REG_23_MIX_ADC_LEFT, 1, 1, 0), + SOC_DAPM_SINGLE("INB2 Switch", M98088_REG_23_MIX_ADC_LEFT, 0, 1, 0), +}; + +/* Right ADC mixer switch */ +static const struct snd_kcontrol_new max98088_right_ADC_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_24_MIX_ADC_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_24_MIX_ADC_RIGHT, 6, 1, 0), + SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_24_MIX_ADC_RIGHT, 3, 1, 0), + SOC_DAPM_SINGLE("INA2 Switch", M98088_REG_24_MIX_ADC_RIGHT, 2, 1, 0), + SOC_DAPM_SINGLE("INB1 Switch", M98088_REG_24_MIX_ADC_RIGHT, 1, 1, 0), + SOC_DAPM_SINGLE("INB2 Switch", M98088_REG_24_MIX_ADC_RIGHT, 0, 1, 0), +}; + +static int max98088_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 status; + + BUG_ON(event != SND_SOC_DAPM_PRE_PMD); + + /* powering down headphone gracefully */ + status = snd_soc_read(codec, M98088_REG_4D_PWR_EN_OUT); + if ((status & M98088_HPEN) == M98088_HPEN) { + max98088_hw_write(codec, M98088_REG_4D_PWR_EN_OUT, + (status & ~M98088_HPEN)); + } + schedule_timeout(msecs_to_jiffies(20)); + + return 0; +} + +static int max98088_mic_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (w->reg == M98088_REG_35_LVL_MIC1) { + snd_soc_update_bits(codec, w->reg, M98088_MICPRE_MASK, + (1+max98088->mic1pre)<<M98088_MICPRE_SHIFT); + } else { + snd_soc_update_bits(codec, w->reg, M98088_MICPRE_MASK, + (1+max98088->mic2pre)<<M98088_MICPRE_SHIFT); + } + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, M98088_MICPRE_MASK, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* DAPM widgets top level */ +static const struct snd_soc_dapm_widget max98088_dapm_widgets[] = { + + SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", M98088_REG_4C_PWR_EN_IN, 1, 0), + SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", M98088_REG_4C_PWR_EN_IN, 0, 0), + + SND_SOC_DAPM_DAC("DACL1", "HiFi Playback", + M98088_REG_4D_PWR_EN_OUT, 1, 0), + SND_SOC_DAPM_DAC("DACR1", "HiFi Playback", + M98088_REG_4D_PWR_EN_OUT, 0, 0), + SND_SOC_DAPM_DAC("DACL2", "Aux Playback", + M98088_REG_4D_PWR_EN_OUT, 1, 0), + SND_SOC_DAPM_DAC("DACR2", "Aux Playback", + M98088_REG_4D_PWR_EN_OUT, 0, 0), + + SND_SOC_DAPM_PGA_E("HP Left Out", M98088_REG_4D_PWR_EN_OUT, + 7, 0, NULL, 0, max98088_hp_event, SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("HP Right Out", M98088_REG_4D_PWR_EN_OUT, + 6, 0, NULL, 0, max98088_hp_event, SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_PGA("SPK Left Out", M98088_REG_4D_PWR_EN_OUT, + 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK Right Out", M98088_REG_4D_PWR_EN_OUT, + 4, 0, NULL, 0), + + SND_SOC_DAPM_PGA("REC Left Out", M98088_REG_4D_PWR_EN_OUT, + 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("REC Right Out", M98088_REG_4D_PWR_EN_OUT, + 2, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, + &max98088_left_hp_mixer_controls[0], + ARRAY_SIZE(max98088_left_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right HP Mixer", SND_SOC_NOPM, 0, 0, + &max98088_right_hp_mixer_controls[0], + ARRAY_SIZE(max98088_right_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left SPK Mixer", SND_SOC_NOPM, 0, 0, + &max98088_left_speaker_mixer_controls[0], + ARRAY_SIZE(max98088_left_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right SPK Mixer", SND_SOC_NOPM, 0, 0, + &max98088_right_speaker_mixer_controls[0], + ARRAY_SIZE(max98088_right_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left REC Mixer", SND_SOC_NOPM, 0, 0, + &max98088_left_rec_mixer_controls[0], + ARRAY_SIZE(max98088_left_rec_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right REC Mixer", SND_SOC_NOPM, 0, 0, + &max98088_right_rec_mixer_controls[0], + ARRAY_SIZE(max98088_right_rec_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98088_left_ADC_mixer_controls[0], + ARRAY_SIZE(max98088_left_ADC_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98088_right_ADC_mixer_controls[0], + ARRAY_SIZE(max98088_right_ADC_mixer_controls)), + + SND_SOC_DAPM_PGA_E("MIC1 Input", M98088_REG_35_LVL_MIC1, + 5, 0, NULL, 0, max98088_mic_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC2 Input", M98088_REG_36_LVL_MIC2, + 5, 0, NULL, 0, max98088_mic_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("INA1 Input", M98088_REG_4C_PWR_EN_IN, + 7, 0, NULL, 0, max98088_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("INA2 Input", M98088_REG_4C_PWR_EN_IN, + 7, 0, NULL, 0, max98088_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("INB1 Input", M98088_REG_4C_PWR_EN_IN, + 6, 0, NULL, 0, max98088_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("INB2 Input", M98088_REG_4C_PWR_EN_IN, + 6, 0, NULL, 0, max98088_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MICBIAS("MICBIAS", M98088_REG_4C_PWR_EN_IN, 3, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("RECL"), + SND_SOC_DAPM_OUTPUT("RECR"), + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("INA1"), + SND_SOC_DAPM_INPUT("INA2"), + SND_SOC_DAPM_INPUT("INB1"), + SND_SOC_DAPM_INPUT("INB2"), +}; + +/* DAPM AUDIO_MAP: */ +static const struct snd_soc_dapm_route audio_map[] = { + /* Left headphone output mixer */ + {"Left HP Mixer", "Left DAC1 Switch", "DACL1"}, + {"Left HP Mixer", "Left DAC2 Switch", "DACL2"}, + {"Left HP Mixer", "Right DAC1 Switch", "DACR1"}, + {"Left HP Mixer", "Right DAC2 Switch", "DACR2"}, + {"Left HP Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left HP Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left HP Mixer", "INA1 Switch", "INA1 Input"}, + {"Left HP Mixer", "INA2 Switch", "INA2 Input"}, + {"Left HP Mixer", "INB1 Switch", "INB1 Input"}, + {"Left HP Mixer", "INB2 Switch", "INB2 Input"}, + + /* Right headphone output mixer */ + {"Right HP Mixer", "Left DAC1 Switch", "DACL1"}, + {"Right HP Mixer", "Left DAC2 Switch", "DACL2" }, + {"Right HP Mixer", "Right DAC1 Switch", "DACR1"}, + {"Right HP Mixer", "Right DAC2 Switch", "DACR2"}, + {"Right HP Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right HP Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right HP Mixer", "INA1 Switch", "INA1 Input"}, + {"Right HP Mixer", "INA2 Switch", "INA2 Input"}, + {"Right HP Mixer", "INB1 Switch", "INB1 Input"}, + {"Right HP Mixer", "INB2 Switch", "INB2 Input"}, + + /* Left speaker output mixer */ + {"Left SPK Mixer", "Left DAC1 Switch", "DACL1"}, + {"Left SPK Mixer", "Left DAC2 Switch", "DACL2"}, + {"Left SPK Mixer", "Right DAC1 Switch", "DACR1"}, + {"Left SPK Mixer", "Right DAC2 Switch", "DACR2"}, + {"Left SPK Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left SPK Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left SPK Mixer", "INA1 Switch", "INA1 Input"}, + {"Left SPK Mixer", "INA2 Switch", "INA2 Input"}, + {"Left SPK Mixer", "INB1 Switch", "INB1 Input"}, + {"Left SPK Mixer", "INB2 Switch", "INB2 Input"}, + + /* Right speaker output mixer */ + {"Right SPK Mixer", "Left DAC1 Switch", "DACL1"}, + {"Right SPK Mixer", "Left DAC2 Switch", "DACL2"}, + {"Right SPK Mixer", "Right DAC1 Switch", "DACR1"}, + {"Right SPK Mixer", "Right DAC2 Switch", "DACR2"}, + {"Right SPK Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right SPK Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right SPK Mixer", "INA1 Switch", "INA1 Input"}, + {"Right SPK Mixer", "INA2 Switch", "INA2 Input"}, + {"Right SPK Mixer", "INB1 Switch", "INB1 Input"}, + {"Right SPK Mixer", "INB2 Switch", "INB2 Input"}, + + /* Earpiece/Receiver output mixer */ + {"Left REC Mixer", "Left DAC1 Switch", "DACL1"}, + {"Left REC Mixer", "Left DAC2 Switch", "DACL2"}, + {"Left REC Mixer", "Right DAC1 Switch", "DACR1"}, + {"Left REC Mixer", "Right DAC2 Switch", "DACR2"}, + {"Left REC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left REC Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left REC Mixer", "INA1 Switch", "INA1 Input"}, + {"Left REC Mixer", "INA2 Switch", "INA2 Input"}, + {"Left REC Mixer", "INB1 Switch", "INB1 Input"}, + {"Left REC Mixer", "INB2 Switch", "INB2 Input"}, + + /* Earpiece/Receiver output mixer */ + {"Right REC Mixer", "Left DAC1 Switch", "DACL1"}, + {"Right REC Mixer", "Left DAC2 Switch", "DACL2"}, + {"Right REC Mixer", "Right DAC1 Switch", "DACR1"}, + {"Right REC Mixer", "Right DAC2 Switch", "DACR2"}, + {"Right REC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right REC Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right REC Mixer", "INA1 Switch", "INA1 Input"}, + {"Right REC Mixer", "INA2 Switch", "INA2 Input"}, + {"Right REC Mixer", "INB1 Switch", "INB1 Input"}, + {"Right REC Mixer", "INB2 Switch", "INB2 Input"}, + + {"HP Left Out", NULL, "Left HP Mixer"}, + {"HP Right Out", NULL, "Right HP Mixer"}, + {"SPK Left Out", NULL, "Left SPK Mixer"}, + {"SPK Right Out", NULL, "Right SPK Mixer"}, + {"REC Left Out", NULL, "Left REC Mixer"}, + {"REC Right Out", NULL, "Right REC Mixer"}, + + {"HPL", NULL, "HP Left Out"}, + {"HPR", NULL, "HP Right Out"}, + {"SPKL", NULL, "SPK Left Out"}, + {"SPKR", NULL, "SPK Right Out"}, + {"RECL", NULL, "REC Left Out"}, + {"RECR", NULL, "REC Right Out"}, + + /* Left ADC input mixer */ + {"Left ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left ADC Mixer", "INA1 Switch", "INA1 Input"}, + {"Left ADC Mixer", "INA2 Switch", "INA2 Input"}, + {"Left ADC Mixer", "INB1 Switch", "INB1 Input"}, + {"Left ADC Mixer", "INB2 Switch", "INB2 Input"}, + + /* Right ADC input mixer */ + {"Right ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right ADC Mixer", "INA1 Switch", "INA1 Input"}, + {"Right ADC Mixer", "INA2 Switch", "INA2 Input"}, + {"Right ADC Mixer", "INB1 Switch", "INB1 Input"}, + {"Right ADC Mixer", "INB2 Switch", "INB2 Input"}, + + /* inputs */ + {"ADCL", NULL, "Left ADC Mixer"}, + {"ADCR", NULL, "Right ADC Mixer"}, + {"INA1 Input", NULL, "INA1"}, + {"INA2 Input", NULL, "INA2"}, + {"INB1 Input", NULL, "INB1"}, + {"INB2 Input", NULL, "INB2"}, + {"MIC1 Input", NULL, "MIC1"}, + {"MIC2 Input", NULL, "MIC2"}, +}; + +static int max98088_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, max98088_dapm_widgets, + ARRAY_SIZE(max98088_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_add_controls(codec, max98088_snd_controls, + ARRAY_SIZE(max98088_snd_controls)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +/* codec mclk clock divider coefficients */ +static const struct { + u32 rate; + u8 sr; +} rate_table[] = { + {8000, 0x10}, + {11025, 0x20}, + {16000, 0x30}, + {22050, 0x40}, + {24000, 0x50}, + {32000, 0x60}, + {44100, 0x70}, + {48000, 0x80}, + {88200, 0x90}, + {96000, 0xA0}, +}; + +static inline int rate_value(int rate, u8 *value) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i].rate >= rate) { + *value = rate_table[i].sr; + return 0; + } + } + *value = rate_table[0].sr; + return -EINVAL; +} + +static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_cdata *cdata; + unsigned int rate; + u8 regval; + u16 ni; + + cdata = &max98088->dai[0]; + + rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, + M98088_DAI_WS, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, + M98088_DAI_WS, M98088_DAI_WS); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, M98088_SHDNRUN, 0); + + if (rate != cdata->rate) { + /* set DAI1 SR1 value for the DSP; FREQ1:0=anyclock */ + if (rate_value(rate, &regval)) + return -EINVAL; + + snd_soc_write(codec, M98088_REG_11_DAI1_CLKMODE, regval); + cdata->rate = rate; + } + + /* Configure NI when operating as master */ + if (snd_soc_read(codec, M98088_REG_14_DAI1_FORMAT) + & M98088_DAI_MAS) { + if (max98088->sysclk == 0) + return -EINVAL; + ni = (u16)ulldiv(65536ULL * (rate < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)rate, + (unsigned long long int)max98088->sysclk); + snd_soc_write(codec, M98088_REG_12_DAI1_CLKCFG_HI, + (ni >> 8) & 0x7f); + snd_soc_write(codec, M98088_REG_13_DAI1_CLKCFG_LO, + ni & 0xff); + } + + /* Update sample rate mode */ + if (rate < 50000) + snd_soc_update_bits(codec, M98088_REG_18_DAI1_FILTERS, + M98088_DAI_DHF, 0); + else + snd_soc_update_bits(codec, M98088_REG_18_DAI1_FILTERS, + M98088_DAI_DHF, M98088_DAI_DHF); + + snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, 0, M98088_SHDNRUN); + + return 0; +} + +static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_cdata *cdata; + unsigned int rate; + u8 regval; + u16 ni; + + cdata = &max98088->dai[1]; + + rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, + M98088_DAI_WS, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, + M98088_DAI_WS, M98088_DAI_WS); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, M98088_SHDNRUN, 0); + + if (rate != cdata->rate) { + /* set DAI2 SR2 value for the DSP */ + if (rate_value(rate, &regval)) + return -EINVAL; + + snd_soc_write(codec, M98088_REG_19_DAI2_CLKMODE, regval); + cdata->rate = rate; + } + + /* Configure NI when operating as master */ + if (snd_soc_read(codec, M98088_REG_1C_DAI2_FORMAT) + & M98088_DAI_MAS) { + if (max98088->sysclk == 0) + return -EINVAL; + ni = (u16)ulldiv(65536ULL * (rate < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)rate, + (unsigned long long int)max98088->sysclk); + snd_soc_write(codec, M98088_REG_1A_DAI2_CLKCFG_HI, + (ni >> 8) & 0x7f); + snd_soc_write(codec, M98088_REG_1B_DAI2_CLKCFG_LO, + ni & 0xff); + } + + /* Update sample rate mode */ + if (rate < 50000) + snd_soc_update_bits(codec, M98088_REG_20_DAI2_FILTERS, + M98088_DAI_DHF, 0); + else + snd_soc_update_bits(codec, M98088_REG_20_DAI2_FILTERS, + M98088_DAI_DHF, M98088_DAI_DHF); + + snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, 0, M98088_SHDNRUN); + + return 0; +} + +static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + + /* requested clock frequency is already setup */ + if (freq == max98088->sysclk) + return 0; + + max98088->sysclk = freq; /* remember current sysclk */ + + /* setup clocks for slave mode, and using the PLL + * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) + * 0x02 (when master clk is 20MHz to 30MHz).. + */ + if ((freq >= 10000000) && (freq < 20000000)) { + snd_soc_write(codec, M98088_REG_10_SYS_CLK, 0x10); + } else if ((freq >= 20000000) && (freq < 30000000)) { + snd_soc_write(codec, M98088_REG_10_SYS_CLK, 0x20); + } else { + dev_err(codec->dev, "Invalid master clock frequency\n"); + return -EINVAL; + } + + if (snd_soc_read(codec, M98088_REG_51_PWR_SYS) & M98088_SHDNRUN) { + snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, + M98088_SHDNRUN, 0); + snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, + 0, M98088_SHDNRUN); + } + + dev_dbg(dai->dev, "Clock source is %d at %uHz\n", clk_id, freq); + + max98088->sysclk = freq; + return 0; +} + +static int max98088_dai1_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_cdata *cdata; + u8 reg15val; + u8 reg14msk = 0; + u8 reg14val = 0; + + cdata = &max98088->dai[0]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* mask MAS to select slave mode */ + reg14msk |= M98088_DAI_MAS; + /* slave mode PLL */ + snd_soc_write(codec, M98088_REG_12_DAI1_CLKCFG_HI, + 0x80); + snd_soc_write(codec, M98088_REG_13_DAI1_CLKCFG_LO, + 0x00); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* set to master mode */ + reg14val |= M98088_DAI_MAS; + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "Clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg14val |= M98088_DAI_DLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg14msk |= M98088_DAI_DLY; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + reg14msk |= M98088_DAI_BCI|M98088_DAI_WCI; + break; + case SND_SOC_DAIFMT_NB_IF: + reg14msk |= M98088_DAI_BCI; + reg14val |= M98088_DAI_WCI; + break; + case SND_SOC_DAIFMT_IB_NF: + reg14msk |= M98088_DAI_WCI; + reg14val |= M98088_DAI_BCI; + break; + case SND_SOC_DAIFMT_IB_IF: + reg14val |= M98088_DAI_BCI|M98088_DAI_WCI; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, + reg14msk, reg14val); + + reg15val = M98088_DAI_BSEL64; + if (max98088->digmic) + reg15val |= M98088_DAI_OSR64; + snd_soc_write(codec, M98088_REG_15_DAI1_CLOCK, reg15val); + } + + return 0; +} + +static int max98088_dai2_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_cdata *cdata; + u8 reg1Cmsk = 0; + u8 reg1Cval = 0; + + cdata = &max98088->dai[1]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* mask MAS to select slave mode */ + reg1Cmsk |= M98088_DAI_MAS; + /* slave mode PLL */ + snd_soc_write(codec, M98088_REG_1A_DAI2_CLKCFG_HI, + 0x80); + snd_soc_write(codec, M98088_REG_1B_DAI2_CLKCFG_LO, + 0x00); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* set to master mode */ + reg1Cval |= M98088_DAI_MAS; + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "Clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg1Cval |= M98088_DAI_DLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg1Cmsk |= M98088_DAI_DLY; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + reg1Cmsk |= M98088_DAI_BCI|M98088_DAI_WCI; + break; + case SND_SOC_DAIFMT_NB_IF: + reg1Cmsk |= M98088_DAI_BCI; + reg1Cval |= M98088_DAI_WCI; + break; + case SND_SOC_DAIFMT_IB_NF: + reg1Cmsk |= M98088_DAI_WCI; + reg1Cval |= M98088_DAI_BCI; + break; + case SND_SOC_DAIFMT_IB_IF: + reg1Cval |= M98088_DAI_BCI|M98088_DAI_WCI; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, + reg1Cmsk, reg1Cval); + + snd_soc_write(codec, M98088_REG_1D_DAI2_CLOCK, + M98088_DAI_BSEL64); + } + + return 0; +} + +static void max98088_sync_cache(struct snd_soc_codec *codec) +{ + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + int i; + + if (!codec->cache_sync) + return; + + codec->cache_only = 0; + + /* write back cached values if they're writeable and + * different from the hardware default. + */ + for (i = 1; i < ARRAY_SIZE(max98088->reg_cache); i++) { + if (!max98088_access[i].writable) + continue; + + if (max98088->reg_cache[i] == max98088_reg[i]) + continue; + + snd_soc_write(codec, i, max98088->reg_cache[i]); + } + + codec->cache_sync = 0; +} + +static int max98088_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + max98088_sync_cache(codec); + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, + M98088_MBEN, M98088_MBEN); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, + M98088_MBEN, 0); +#ifdef CONFIG_REGULATOR + codec->cache_sync = 1; +#endif + break; + } + codec->bias_level = level; + return 0; +} + +#define MAX98088_RATES SNDRV_PCM_RATE_8000_96000 +#define MAX98088_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max98088_dai1_ops = { + .set_sysclk = max98088_dai_set_sysclk, + .set_fmt = max98088_dai1_set_fmt, + .hw_params = max98088_dai1_hw_params, +}; + +static struct snd_soc_dai_ops max98088_dai2_ops = { + .set_sysclk = max98088_dai_set_sysclk, + .set_fmt = max98088_dai2_set_fmt, + .hw_params = max98088_dai2_hw_params, +}; + +static struct snd_soc_dai_driver max98088_dai[] = { +{ + .name = "HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98088_RATES, + .formats = MAX98088_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98088_RATES, + .formats = MAX98088_FORMATS, + }, + .ops = &max98088_dai1_ops, +}, +{ + .name = "Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98088_RATES, + .formats = MAX98088_FORMATS, + }, + .ops = &max98088_dai2_ops, +} +}; + +static void max98088_setup_eq1(struct snd_soc_codec *codec) +{ + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_pdata *pdata = max98088->pdata; + struct max98088_eq_cfg *coef_set; + int best, best_val, save, i, sel, fs; + struct max98088_cdata *cdata; + + cdata = &max98088->dai[0]; + + if (!pdata || !cdata->eq_textcnt) + return; + + /* Find the selected configuration with nearest sample rate */ + fs = cdata->rate; + sel = cdata->eq_sel; + + best = 0; + best_val = INT_MAX; + for (i = 0; i < pdata->eq1_cfgcnt; i++) { + if (strcmp(pdata->eq1_cfg[i].name, cdata->eq_texts[sel]) == 0 && + abs(pdata->eq1_cfg[i].rate - fs) < best_val) { + best = i; + best_val = abs(pdata->eq1_cfg[i].rate - fs); + } + } + + dev_dbg(codec->dev, "Selected %s/%dHz for %dHz sample rate\n", + pdata->eq1_cfg[best].name, + pdata->eq1_cfg[best].rate, fs); + + /* Disable EQ while configuring, and save current on/off state */ + save = snd_soc_read(codec, M98088_REG_49_CFG_LEVEL); + snd_soc_update_bits(codec, M98088_REG_49_CFG_LEVEL, M98088_EQ1EN, 0); + + coef_set = &pdata->eq1_cfg[sel]; + + m98088_eq_band(codec, 0, 0, coef_set->band1); + m98088_eq_band(codec, 0, 1, coef_set->band2); + m98088_eq_band(codec, 0, 2, coef_set->band3); + m98088_eq_band(codec, 0, 3, coef_set->band4); + m98088_eq_band(codec, 0, 4, coef_set->band5); + + /* restore original on/off state */ + snd_soc_update_bits(codec, M98088_REG_49_CFG_LEVEL, M98088_EQ1EN, save); +} + +static void max98088_setup_eq2(struct snd_soc_codec *codec) +{ + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_pdata *pdata = max98088->pdata; + struct max98088_eq_cfg *coef_set; + int best, best_val, save, i, sel, fs; + struct max98088_cdata *cdata; + + cdata = &max98088->dai[1]; + + if (!pdata || !cdata->eq_textcnt) + return; + + /* Find the selected configuration with nearest sample rate */ + fs = cdata->rate; + + sel = cdata->eq_sel; + best = 0; + best_val = INT_MAX; + for (i = 0; i < pdata->eq2_cfgcnt; i++) { + if (strcmp(pdata->eq2_cfg[i].name, cdata->eq_texts[sel]) == 0 && + abs(pdata->eq2_cfg[i].rate - fs) < best_val) { + best = i; + best_val = abs(pdata->eq2_cfg[i].rate - fs); + } + } + + dev_dbg(codec->dev, "Selected %s/%dHz for %dHz sample rate\n", + pdata->eq2_cfg[best].name, + pdata->eq2_cfg[best].rate, fs); + + /* Disable EQ while configuring, and save current on/off state */ + save = snd_soc_read(codec, M98088_REG_49_CFG_LEVEL); + snd_soc_update_bits(codec, M98088_REG_49_CFG_LEVEL, M98088_EQ2EN, 0); + + coef_set = &pdata->eq2_cfg[sel]; + + m98088_eq_band(codec, 1, 0, coef_set->band1); + m98088_eq_band(codec, 1, 1, coef_set->band2); + m98088_eq_band(codec, 1, 2, coef_set->band3); + m98088_eq_band(codec, 1, 3, coef_set->band4); + m98088_eq_band(codec, 1, 4, coef_set->band5); + + /* restore original on/off state */ + snd_soc_update_bits(codec, M98088_REG_49_CFG_LEVEL, M98088_EQ2EN, + save); +} + + +static int max98088_put_eq1_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_pdata *pdata = max98088->pdata; + struct max98088_cdata *cdata; + int sel = ucontrol->value.integer.value[0]; + + cdata = &max98088->dai[0]; + + if (sel >= pdata->eq1_cfgcnt) + return -EINVAL; + + cdata->eq_sel = sel; + max98088_setup_eq1(codec); + return 0; +} + +static int max98088_put_eq2_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_pdata *pdata = max98088->pdata; + struct max98088_cdata *cdata; + int sel = ucontrol->value.integer.value[0]; + + cdata = &max98088->dai[1]; + + if (sel >= pdata->eq2_cfgcnt) + return -EINVAL; + + cdata->eq_sel = sel; + max98088_setup_eq2(codec); + return 0; +} + +static int max98088_get_eq1_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_cdata *cdata; + + cdata = &max98088->dai[0]; + + ucontrol->value.enumerated.item[0] = cdata->eq_sel; + return 0; +} + +static int max98088_get_eq2_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_cdata *cdata; + + cdata = &max98088->dai[1]; + + ucontrol->value.enumerated.item[0] = cdata->eq_sel; + return 0; +} + +static void max98088_handle_eq1_pdata(struct snd_soc_codec *codec) +{ + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_pdata *pdata = max98088->pdata; + struct max98088_cdata *cdata; + int ret, i, j; + + struct snd_kcontrol_new eq1control = + SOC_ENUM_EXT("EQ1 Mode", + max98088->dai[0].eq_enum, + max98088_get_eq1_enum, + max98088_put_eq1_enum); + + cdata = &max98088->dai[0]; + + /* Build an array of texts for the enum API. The number + * of texts is likely fewer than the number of configurations + * due to multiple sample rates for the same text name. */ + cdata->eq_textcnt = 0; + for (i = 0; i < pdata->eq1_cfgcnt; i++) { + for (j = 0; j < cdata->eq_textcnt; j++) { + if (strcmp(pdata->eq1_cfg[i].name, + cdata->eq_texts[j]) == 0) { + break; + } + } + + if (j != cdata->eq_textcnt) + continue; + + /* ...and remember the new version. */ + cdata->eq_texts[i] = pdata->eq1_cfg[i].name; + cdata->eq_textcnt++; + + if (cdata->eq_textcnt >= EQ_CFG_MAX) { + dev_err(codec->dev, "Too many EQ config entries\n"); + cdata->eq_textcnt--; + break; + } + } + + dev_dbg(codec->dev, "Installed %d EQ1 configurations\n", + cdata->eq_textcnt); + + /* now point the soc_enum to .texts array items */ + cdata->eq_enum.texts = cdata->eq_texts; + cdata->eq_enum.max = cdata->eq_textcnt; + + ret = snd_soc_add_controls(codec, &eq1control, 1); + if (ret != 0) + dev_err(codec->dev, "Failed to add EQ control: %d\n", ret); +} + +static void max98088_handle_eq2_pdata(struct snd_soc_codec *codec) +{ + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_pdata *pdata = max98088->pdata; + struct max98088_cdata *cdata; + int ret, i, j; + + struct snd_kcontrol_new eq2control = + SOC_ENUM_EXT("EQ2 Mode", + max98088->dai[1].eq_enum, + max98088_get_eq2_enum, + max98088_put_eq2_enum); + + cdata = &max98088->dai[1]; + + /* Build an array of texts for the enum API. The number + * of texts is likely fewer than the number of configurations + * due to multiple sample rates for the same text name. */ + cdata->eq_textcnt = 0; + for (i = 0; i < pdata->eq2_cfgcnt; i++) { + for (j = 0; j < cdata->eq_textcnt; j++) { + if (strcmp(pdata->eq2_cfg[i].name, + cdata->eq_texts[j]) == 0) { + break; + } + } + + if (j != cdata->eq_textcnt) + continue; + + cdata->eq_texts[i] = pdata->eq2_cfg[i].name; + cdata->eq_textcnt++; + + if (cdata->eq_textcnt >= EQ_CFG_MAX) { + dev_err(codec->dev, "Too many EQ config entries\n"); + cdata->eq_textcnt--; + break; + } + } + + dev_dbg(codec->dev, "Installed %d EQ2 configurations\n", + cdata->eq_textcnt); + + /* now point the soc_enum to .texts array items */ + cdata->eq_enum.texts = cdata->eq_texts; + cdata->eq_enum.max = cdata->eq_textcnt; + + ret = snd_soc_add_controls(codec, &eq2control, 1); + if (ret != 0) + printk(KERN_ERR "Failed to add EQ control: %d\n", ret); +} + +static void max98088_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_pdata *pdata = max98088->pdata; + u8 regval = 0; + + if (!pdata) { + dev_dbg(codec->dev, "No platform data\n"); + return; + } + + /* configure mic for analog/digital mic mode */ + if (pdata->digmic_left_mode) + regval |= M98088_DIGMIC_L; + + if (pdata->digmic_right_mode) + regval |= M98088_DIGMIC_R; + + max98088->digmic = (regval ? 1 : 0); + + snd_soc_write(codec, M98088_REG_48_CFG_MIC, regval); + + /* configure receiver output */ + regval = ((pdata->receiver_mode) ? M98088_REC_LINEMODE : 0); + snd_soc_update_bits(codec, M98088_REG_2A_MIC_REC_CNTL, + M98088_REC_LINEMODE_MASK, regval); + + /* configure equalizers */ + if (pdata->eq1_cfgcnt) + max98088_handle_eq1_pdata(codec); + + if (pdata->eq2_cfgcnt) + max98088_handle_eq2_pdata(codec); +} + +#ifdef CONFIG_PM +static int max98088_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + max98088_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int max98088_resume(struct snd_soc_codec *codec) +{ + int i; + u8 *cache = codec->reg_cache; + + max98088_set_bias_level(codec, SND_SOC_BIAS_OFF); + + /* Sync reg_cache with the hardware */ + for (i = 0; i < M98088_REG_CNT; i++) { + if (i == M98088_REG_51_PWR_SYS) + continue; + + if (!max98088_access[i].writable) + continue; + + max98088_hw_write(codec, i, cache[i]); + } + + /* now enter into the resume mode bias level */ + max98088_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define max98088_suspend NULL +#define max98088_resume NULL +#endif + +static int max98088_probe(struct snd_soc_codec *codec) +{ + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + struct max98088_cdata *cdata; + int ret = 0; + + codec->cache_sync = 1; + memcpy(codec->reg_cache, max98088_reg, sizeof(max98088_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* initalize private data */ + + max98088->sysclk = (unsigned)-1; + + cdata = &max98088->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + cdata->eq_textcnt = 0; + cdata->eq_sel = 0; + + cdata = &max98088->dai[1]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + cdata->eq_textcnt = 0; + cdata->eq_sel = 0; + + max98088->power_state = 0; /* INA INB power enable state */ + max98088->ex_mode = 0; /* excursion limiter mode */ + max98088->digmic = 0; /* 0=analog, 1=digital */ + max98088->mic1pre = 0; + max98088->mic2pre = 0; + + ret = snd_soc_read(codec, M98088_REG_FF_REV_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_access; + } + dev_info(codec->dev, "revision %c\n", ret + 'A'); + + snd_soc_write(codec, M98088_REG_51_PWR_SYS, M98088_PWRSV); + + /* initialize registers cache to hardware default */ + max98088_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_write(codec, M98088_REG_0F_IRQ_ENABLE, 0x00); + + snd_soc_write(codec, M98088_REG_22_MIX_DAC, + M98088_DAI1L_TO_DACL|M98088_DAI2L_TO_DACL| + M98088_DAI1R_TO_DACR|M98088_DAI2R_TO_DACR); + + snd_soc_write(codec, M98088_REG_4E_BIAS_CNTL, 0xF0); + snd_soc_write(codec, M98088_REG_50_DAC_BIAS2, 0x0F); + + snd_soc_write(codec, M98088_REG_16_DAI1_IOCFG, + M98088_S1NORMAL|M98088_SDATA); + + snd_soc_write(codec, M98088_REG_1E_DAI2_IOCFG, + M98088_S2NORMAL|M98088_SDATA); + + max98088_handle_pdata(codec); + + max98088_add_widgets(codec); + +err_access: + return ret; +} + +static int max98088_remove(struct snd_soc_codec *codec) +{ + if (codec->control_data) + max98088_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98088 = { + .probe = max98088_probe, + .remove = max98088_remove, + .suspend = max98088_suspend, + .resume = max98088_resume, + .set_bias_level = max98088_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max98088_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = max98088_reg, + .volatile_register = max98088_volatile_register, +}; + +static int max98088_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98088_priv *max98088; + int ret; + + max98088 = kzalloc(sizeof(struct max98088_priv), GFP_KERNEL); + if (max98088 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98088); + max98088->control_data = i2c; + max98088->pdata = i2c->dev.platform_data; + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max98088, &max98088_dai[0], 2); + if (ret < 0) + kfree(max98088); + return ret; +} + +static int max98088_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id max98088_i2c_id[] = { + { "max98088", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); + +static struct i2c_driver max98088_i2c_driver = { + .driver = { + .name = "max98088-codec", + .owner = THIS_MODULE, + }, + .probe = max98088_i2c_probe, + .remove = __devexit_p(max98088_i2c_remove), + .id_table = max98088_i2c_id, +}; + +static int __init max98088_init(void) +{ + int ret; + + ret = i2c_add_driver(&max98088_i2c_driver); + if (ret) + pr_err("Failed to register max98088 I2C driver: %d\n", ret); + + return ret; +} + +static void __exit max98088_exit(void) +{ + i2c_del_driver(&max98088_i2c_driver); +} + +module_init(max98088_init); +module_exit(max98088_exit); + +MODULE_DESCRIPTION("ALSA SoC MAX98088 driver"); +MODULE_AUTHOR("Peter Hsiang, Jesse Marroquin"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h new file mode 100644 index 0000000..9e9d8da --- /dev/null +++ b/sound/soc/codecs/max98088.h @@ -0,0 +1,190 @@ +/* + * max98088.h -- MAX98088 ALSA SoC Audio driver + * + * Copyright 2010 Maxim Integrated Products + * + * 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 _MAX98088_H +#define _MAX98088_H + +/* + * MAX98088 Registers Definition + */ +#define M98088_REG_00_IRQ_STATUS 0x00 +#define M98088_REG_01_MIC_STATUS 0x01 +#define M98088_REG_02_JACK_STAUS 0x02 +#define M98088_REG_03_BATTERY_VOLTAGE 0x03 +#define M98088_REG_0F_IRQ_ENABLE 0x0F +#define M98088_REG_10_SYS_CLK 0x10 +#define M98088_REG_11_DAI1_CLKMODE 0x11 +#define M98088_REG_12_DAI1_CLKCFG_HI 0x12 +#define M98088_REG_13_DAI1_CLKCFG_LO 0x13 +#define M98088_REG_14_DAI1_FORMAT 0x14 +#define M98088_REG_15_DAI1_CLOCK 0x15 +#define M98088_REG_16_DAI1_IOCFG 0x16 +#define M98088_REG_17_DAI1_TDM 0x17 +#define M98088_REG_18_DAI1_FILTERS 0x18 +#define M98088_REG_19_DAI2_CLKMODE 0x19 +#define M98088_REG_1A_DAI2_CLKCFG_HI 0x1A +#define M98088_REG_1B_DAI2_CLKCFG_LO 0x1B +#define M98088_REG_1C_DAI2_FORMAT 0x1C +#define M98088_REG_1D_DAI2_CLOCK 0x1D +#define M98088_REG_1E_DAI2_IOCFG 0x1E +#define M98088_REG_1F_DAI2_TDM 0x1F +#define M98088_REG_20_DAI2_FILTERS 0x20 +#define M98088_REG_21_SRC 0x21 +#define M98088_REG_22_MIX_DAC 0x22 +#define M98088_REG_23_MIX_ADC_LEFT 0x23 +#define M98088_REG_24_MIX_ADC_RIGHT 0x24 +#define M98088_REG_25_MIX_HP_LEFT 0x25 +#define M98088_REG_26_MIX_HP_RIGHT 0x26 +#define M98088_REG_27_MIX_HP_CNTL 0x27 +#define M98088_REG_28_MIX_REC_LEFT 0x28 +#define M98088_REG_29_MIX_REC_RIGHT 0x29 +#define M98088_REG_2A_MIC_REC_CNTL 0x2A +#define M98088_REG_2B_MIX_SPK_LEFT 0x2B +#define M98088_REG_2C_MIX_SPK_RIGHT 0x2C +#define M98088_REG_2D_MIX_SPK_CNTL 0x2D +#define M98088_REG_2E_LVL_SIDETONE 0x2E +#define M98088_REG_2F_LVL_DAI1_PLAY 0x2F +#define M98088_REG_30_LVL_DAI1_PLAY_EQ 0x30 +#define M98088_REG_31_LVL_DAI2_PLAY 0x31 +#define M98088_REG_32_LVL_DAI2_PLAY_EQ 0x32 +#define M98088_REG_33_LVL_ADC_L 0x33 +#define M98088_REG_34_LVL_ADC_R 0x34 +#define M98088_REG_35_LVL_MIC1 0x35 +#define M98088_REG_36_LVL_MIC2 0x36 +#define M98088_REG_37_LVL_INA 0x37 +#define M98088_REG_38_LVL_INB 0x38 +#define M98088_REG_39_LVL_HP_L 0x39 +#define M98088_REG_3A_LVL_HP_R 0x3A +#define M98088_REG_3B_LVL_REC_L 0x3B +#define M98088_REG_3C_LVL_REC_R 0x3C +#define M98088_REG_3D_LVL_SPK_L 0x3D +#define M98088_REG_3E_LVL_SPK_R 0x3E +#define M98088_REG_3F_MICAGC_CFG 0x3F +#define M98088_REG_40_MICAGC_THRESH 0x40 +#define M98088_REG_41_SPKDHP 0x41 +#define M98088_REG_42_SPKDHP_THRESH 0x42 +#define M98088_REG_43_SPKALC_COMP 0x43 +#define M98088_REG_44_PWRLMT_CFG 0x44 +#define M98088_REG_45_PWRLMT_TIME 0x45 +#define M98088_REG_46_THDLMT_CFG 0x46 +#define M98088_REG_47_CFG_AUDIO_IN 0x47 +#define M98088_REG_48_CFG_MIC 0x48 +#define M98088_REG_49_CFG_LEVEL 0x49 +#define M98088_REG_4A_CFG_BYPASS 0x4A +#define M98088_REG_4B_CFG_JACKDET 0x4B +#define M98088_REG_4C_PWR_EN_IN 0x4C +#define M98088_REG_4D_PWR_EN_OUT 0x4D +#define M98088_REG_4E_BIAS_CNTL 0x4E +#define M98088_REG_4F_DAC_BIAS1 0x4F +#define M98088_REG_50_DAC_BIAS2 0x50 +#define M98088_REG_51_PWR_SYS 0x51 +#define M98088_REG_52_DAI1_EQ_BASE 0x52 +#define M98088_REG_84_DAI2_EQ_BASE 0x84 +#define M98088_REG_B6_DAI1_BIQUAD_BASE 0xB6 +#define M98088_REG_C0_DAI2_BIQUAD_BASE 0xC0 +#define M98088_REG_FF_REV_ID 0xFF + +#define M98088_REG_CNT (0xFF+1) + +/* MAX98088 Registers Bit Fields */ + +/* M98088_REG_14_DAI1_FORMAT, M98088_REG_1C_DAI2_FORMAT */ + #define M98088_DAI_MAS (1<<7) + #define M98088_DAI_WCI (1<<6) + #define M98088_DAI_BCI (1<<5) + #define M98088_DAI_DLY (1<<4) + #define M98088_DAI_TDM (1<<2) + #define M98088_DAI_FSW (1<<1) + #define M98088_DAI_WS (1<<0) + +/* M98088_REG_15_DAI1_CLOCK, M98088_REG_1D_DAI2_CLOCK */ + #define M98088_DAI_BSEL64 (1<<0) + #define M98088_DAI_OSR64 (1<<6) + +/* M98088_REG_16_DAI1_IOCFG, M98088_REG_1E_DAI2_IOCFG */ + #define M98088_S1NORMAL (1<<6) + #define M98088_S2NORMAL (2<<6) + #define M98088_SDATA (3<<0) + +/* M98088_REG_18_DAI1_FILTERS, M98088_REG_20_DAI2_FILTERS */ + #define M98088_DAI_DHF (1<<3) + +/* M98088_REG_22_MIX_DAC */ + #define M98088_DAI1L_TO_DACL (1<<7) + #define M98088_DAI1R_TO_DACL (1<<6) + #define M98088_DAI2L_TO_DACL (1<<5) + #define M98088_DAI2R_TO_DACL (1<<4) + #define M98088_DAI1L_TO_DACR (1<<3) + #define M98088_DAI1R_TO_DACR (1<<2) + #define M98088_DAI2L_TO_DACR (1<<1) + #define M98088_DAI2R_TO_DACR (1<<0) + +/* M98088_REG_2A_MIC_REC_CNTL */ + #define M98088_REC_LINEMODE (1<<7) + #define M98088_REC_LINEMODE_MASK (1<<7) + +/* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */ + #define M98088_MICPRE_MASK (3<<5) + #define M98088_MICPRE_SHIFT 5 + +/* M98088_REG_3A_LVL_HP_R */ + #define M98088_HP_MUTE (1<<7) + +/* M98088_REG_3C_LVL_REC_R */ + #define M98088_REC_MUTE (1<<7) + +/* M98088_REG_3E_LVL_SPK_R */ + #define M98088_SP_MUTE (1<<7) + +/* M98088_REG_48_CFG_MIC */ + #define M98088_EXTMIC_MASK (3<<0) + #define M98088_DIGMIC_L (1<<5) + #define M98088_DIGMIC_R (1<<4) + +/* M98088_REG_49_CFG_LEVEL */ + #define M98088_VSEN (1<<6) + #define M98088_ZDEN (1<<5) + #define M98088_EQ2EN (1<<1) + #define M98088_EQ1EN (1<<0) + +/* M98088_REG_4C_PWR_EN_IN */ + #define M98088_INAEN (1<<7) + #define M98088_INBEN (1<<6) + #define M98088_MBEN (1<<3) + #define M98088_ADLEN (1<<1) + #define M98088_ADREN (1<<0) + +/* M98088_REG_4D_PWR_EN_OUT */ + #define M98088_HPLEN (1<<7) + #define M98088_HPREN (1<<6) + #define M98088_HPEN ((1<<7)|(1<<6)) + #define M98088_SPLEN (1<<5) + #define M98088_SPREN (1<<4) + #define M98088_RECEN (1<<3) + #define M98088_DALEN (1<<1) + #define M98088_DAREN (1<<0) + +/* M98088_REG_51_PWR_SYS */ + #define M98088_SHDNRUN (1<<7) + #define M98088_PERFMODE (1<<3) + #define M98088_HPPLYBACK (1<<2) + #define M98088_PWRSV8K (1<<1) + #define M98088_PWRSV (1<<0) + +#define M98088_COEFS_PER_BAND 5 + +#define M98088_BYTE1(w) ((w >> 8) & 0xff) +#define M98088_BYTE0(w) (w & 0xff) + +struct max98088_setup_data { + unsigned short i2c_address; +}; + +#endif -- 1.6.3.3

6 43

[alsa-devel] Need expert's advice - Fast Track Ultra (8R) dropping samples
by Felix Homann 15 Oct '10

15 Oct '10

Hi, in that past months I've been trying get the Fast Track Ultra devices working properly in Alsa. We've had lots of progress, most of the code has moved to Alsa git and today I've even posted a patch for getting mixer support for these devices. Now, I need to get some expert's advice: The devices seem to drop samples or frames. Here's a report I've got today on the M-Audio forum: "I've got a subtle problem to report: I think audio playback is dropping sample frames. To hear the problem, open Audacity at 48 kHz and play a 10-kHz. sine wave. When I do that I hear a regular clicking sound, about four clicks a second. I've tried recording the output and if I'm seeing correctly, exactly one sample frame in every 13312 (13x1024) is being dropped on output. I don't see anything similar on input. When either jack or Pd has both the input and the output open, the delay from input to output gradually decreases until it forces occasional sync errors. (I haven't tried this with audacity though.)" (see http://forums.m-audio.com/showthread.php?714-Not-a-problem.-FastTrack-on-li…) I could reproduce it on my machines, even at 44.1 kHz. The clicking sound is very subtle, it goes unnoticed when not listening to pure sines without attention to clicks. How can this be sorted out. Any ideas? Kind regards, Felix

4 23

[alsa-devel] [PATCH] ASoC: Add HP iPAQ H1940 support
by Vasily Khoruzhick 15 Oct '10

15 Oct '10

This patch depends on gpiolib support for h1940 latch access, here's link to patch in ml archive: http://www.spinics.net/lists/arm-kernel/msg98042.html I hope Ben will find some time to merge it into his tree before 2.6.37 merge window :) Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Tested-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/h1940_uda1380.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 306 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/h1940_uda1380.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 7d8235d..6b50509 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -118,6 +118,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC +config SND_S3C24XX_SOC_H1940_UDA1380 + tristate "Audio support for the HP iPAQ H1940" + depends on SND_S3C24XX_SOC && ARCH_H1940 + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA1380 + help + This driver provides audio support for HP iPAQ h1940 PDA. + config SND_S3C24XX_SOC_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" depends on SND_S3C24XX_SOC && MACH_RX1950 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index dd412a9..33a7c68 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ 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-rx1950-uda1380-objs := rx1950_uda1380.o +snd-soc-h1940-uda1380-objs := h1940_uda1380.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 @@ -44,6 +45,7 @@ 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_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o +obj-$(CONFIG_SND_S3C24XX_SOC_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(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/h1940_uda1380.c b/sound/soc/s3c24xx/h1940_uda1380.c new file mode 100644 index 0000000..77550fa --- /dev/null +++ b/sound/soc/s3c24xx/h1940_uda1380.c @@ -0,0 +1,296 @@ +/* + * h1940-uda1380.c -- ALSA Soc Audio Layer + * + * Copyright (c) 2010 Arnaud Patard <arnaud.patard(a)rtp-net.org> + * Copyright (c) 2010 Vasily Khoruzhick <anarsoul(a)gmail.com> + * + * Based on version from Arnaud Patard <arnaud.patard(a)rtp-net.org> + * + * 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/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/uda1380.h> +#include <sound/jack.h> + +#include <plat/regs-iis.h> + +#include <mach/h1940-latch.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.h" + +static unsigned int rates[] = { + 11025, + 22050, + 44100, +}; + +static struct snd_pcm_hw_constraint_list hw_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static struct snd_soc_jack hp_jack; + +static struct snd_soc_jack_pin hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + +static struct snd_soc_jack_gpio hp_jack_gpios[] = { + { + .gpio = S3C2410_GPG(4), + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .invert = 1, + .debounce_time = 200, + }, +}; + +static int h1940_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.rate_min = hw_rates.list[0]; + runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; + runtime->hw.rates = SNDRV_PCM_RATE_KNOT; + + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_rates); +} + +static int h1940_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int div; + int ret; + unsigned int rate = params_rate(params); + + switch (rate) { + case 11025: + case 22050: + case 44100: + div = s3c24xx_i2s_get_clockrate() / (384 * rate); + if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate)) + div++; + break; + default: + printk(KERN_ERR "%s: rate %d is not supported\n", + __func__, rate); + return -EINVAL; + } + + /* 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; + + /* select clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + /* set BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops h1940_ops = { + .startup = h1940_startup, + .hw_params = h1940_hw_params, +}; + +static int h1940_spk_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(H1940_LATCH_AUDIO_POWER, 1); + else + gpio_set_value(H1940_LATCH_AUDIO_POWER, 0); + + return 0; +} + +/* h1940 machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", h1940_spk_power), +}; + +/* h1940 machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* headphone connected to VOUTLHP, VOUTRHP */ + {"Headphone Jack", NULL, "VOUTLHP"}, + {"Headphone Jack", NULL, "VOUTRHP"}, + + /* ext speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* mic is connected to VINM */ + {"VINM", NULL, "Mic Jack"}, +}; + +static struct platform_device *s3c24xx_snd_device; + +static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int err; + + /* Add h1940 specific widgets */ + err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + if (err) + return err; + + /* Set up h1940 specific audio path audio_mapnects */ + err = snd_soc_dapm_add_routes(codec, audio_map, + ARRAY_SIZE(audio_map)); + if (err) + return err; + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack); + + snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), + hp_jack_pins); + + snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + +/* s3c24xx digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link h1940_uda1380_dai[] = { + { + .name = "uda1380", + .stream_name = "UDA1380 Duplex", + .cpu_dai_name = "s3c24xx-iis", + .codec_dai_name = "uda1380-hifi", + .init = h1940_uda1380_init, + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "uda1380-codec.0-001a", + .ops = &h1940_ops, + }, +}; + +static struct snd_soc_card h1940_asoc = { + .name = "h1940", + .dai_link = h1940_uda1380_dai, + .num_links = ARRAY_SIZE(h1940_uda1380_dai), +}; + +static int __init h1940_init(void) +{ + int ret; + + if (!machine_is_h1940()) + return -ENODEV; + + /* configure some gpios */ + ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power"); + if (ret) + goto err_out; + + ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0); + if (ret) + goto err_gpio; + + s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_snd_device) { + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc); + ret = platform_device_add(s3c24xx_snd_device); + + if (ret) + goto err_plat; + + return 0; + +err_plat: + platform_device_put(s3c24xx_snd_device); +err_gpio: + gpio_free(H1940_LATCH_AUDIO_POWER); + +err_out: + return ret; +} + +static void __exit h1940_exit(void) +{ + platform_device_unregister(s3c24xx_snd_device); + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + gpio_free(H1940_LATCH_AUDIO_POWER); +} + +module_init(h1940_init); +module_exit(h1940_exit); + +/* Module information */ +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); +MODULE_DESCRIPTION("ALSA SoC H1940"); +MODULE_LICENSE("GPL"); -- 1.7.2.2

4 20

[alsa-devel] [PATCH v5] ASoC/2.6.37: fix au1x platform
by Manuel Lauss 13 Oct '10

13 Oct '10

This patch fixes up the au1x audio platform after the multi-component merge: - compile fixes and updates to get DB1200 platform audio working again, - removal of global variables in AC97/I2S/DMA(PCM) modules. The AC97 part is limited to one instance only for now due to issues with getting at driver data in the soc_ac97_ops. Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- v5: smaller patch v4: fixed a bug in the previous bugfix, and added DAI drvdata accessors. v3: fixed a bug which caused cat /proc/iomem to loop endlessly. v2: prepare PCM,I2S for multiple card operation, use dev_name() for DAI name. Against Liam's asoc/for-2.6.37 branch. Tested on DB1200 and DB1300 (here both I2S and AC97 operate as independent cards), please fold this into the other Au1x multi-component patches. Issues I observed with AC97: * AC97 is limited to a single instance since I cannot get at the driver data in the AC97 callbacks at all time (or did I miss anything?): when the AC97 codec calls snd_ac97_mixer(), it calls into the soc_ac97_ops callbacks; however ac97->bus->card->private_data (suggested by Mark) is _always_ NULL, so no way to get at the dai and ultimately driver data. * generic AC97 codec use spits out this kobject warning, which is caused by the "device_register()" in soc-core.c::soc_ac97_dev_register(): asoc: ac97-hifi <-> au1xpsc-ac97.1 mapping ok kobject (8fdc59b0): tried to init an initialized object, something is seriously wrong. Call Trace: [<804959c4>] dump_stack+0x8/0x34 [<802a30e4>] kobject_init+0x50/0xcc [<802ec27c>] device_initialize+0x2c/0x70 [<802ecb64>] device_register+0x14/0x28 [<8039e058>] snd_soc_instantiate_cards+0xa00/0xb10 [<8039e27c>] soc_probe+0x114/0x154 [<802ef0fc>] driver_probe_device+0xe4/0x1a0 [<802ee404>] bus_for_each_drv+0x60/0xb0 [<802ef370>] device_attach+0x74/0xa8 [<802ee1e8>] bus_probe_device+0x30/0x54 [<802ec9a0>] device_add+0x384/0x534 [<802f09a4>] platform_device_add+0x15c/0x1c8 [<805ccb14>] db1200_audio_load+0x70/0x9c [<801004fc>] do_one_initcall+0xfc/0x1e0 [<805ba32c>] kernel_init+0xc8/0x168 [<80105904>] kernel_thread_helper+0x10/0x18 arch/mips/alchemy/devboards/db1200/platform.c | 6 ++ sound/soc/au1x/db1200.c | 16 +++--- sound/soc/au1x/dbdma2.c | 82 ++++++++----------------- sound/soc/au1x/psc-ac97.c | 59 +++++++++++------- sound/soc/au1x/psc-i2s.c | 42 ++++--------- sound/soc/au1x/psc.h | 7 +-- 6 files changed, 91 insertions(+), 121 deletions(-) diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c index 3fa34c3..fbb5593 100644 --- a/arch/mips/alchemy/devboards/db1200/platform.c +++ b/arch/mips/alchemy/devboards/db1200/platform.c @@ -429,6 +429,11 @@ static struct platform_device db1200_audio_dev = { .resource = au1200_psc1_res, }; +static struct platform_device db1200_stac_dev = { + .name = "ac97-codec", + .id = 1, /* on PSC1 */ +}; + static struct platform_device *db1200_devs[] __initdata = { NULL, /* PSC0, selected by S6.8 */ &db1200_ide_dev, @@ -436,6 +441,7 @@ static struct platform_device *db1200_devs[] __initdata = { &db1200_rtc_dev, &db1200_nand_dev, &db1200_audio_dev, + &db1200_stac_dev, }; static int __init db1200_dev_init(void) diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index d8dc822..b62fcd3 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -27,10 +27,10 @@ static struct snd_soc_dai_link db1200_ac97_dai = { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai_name = "au1xpsc-ac97", .codec_dai_name = "ac97-hifi", - .platform_name = "au1xpsc-pcm-audio", - .codec_name = "ac97-codec", + .cpu_dai_name = "au1xpsc_ac97.1", + .platform_name = "au1xpsc-pcm.1", + .codec_name = "ac97-codec.1", }; static struct snd_soc_card db1200_ac97_machine = { @@ -75,10 +75,10 @@ static struct snd_soc_ops db1200_i2s_wm8731_ops = { static struct snd_soc_dai_link db1200_i2s_dai = { .name = "WM8731", .stream_name = "WM8731 PCM", - .cpu_dai_name = "au1xpsc", - .codec_dai_name = "wm8731-hifi" - .platform_name = "au1xpsc-pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .codec_dai_name = "wm8731-hifi", + .cpu_dai_name = "au1xpsc_i2s.1", + .platform_name = "au1xpsc-pcm.1", + .codec_name = "wm8731-codec.0-001b", .ops = &db1200_i2s_wm8731_ops, }; @@ -97,7 +97,7 @@ static int __init db1200_audio_load(void) int ret; ret = -ENOMEM; - db1200_asoc_dev = platform_device_alloc("soc-audio", -1); + db1200_asoc_dev = platform_device_alloc("soc-audio", 1); /* PSC1 */ if (!db1200_asoc_dev) goto out; diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 00fdb9c..10fdd28 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -10,9 +10,6 @@ * * DMA glue for Au1x-PSC audio. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ @@ -61,9 +58,6 @@ struct au1xpsc_audio_dmadata { int msbits; }; -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; - /* * These settings are somewhat okay, at least on my machine audio plays * almost skip-free. Especially the 64kB buffer seems to help a LOT. @@ -199,6 +193,14 @@ out: return 0; } +static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + struct au1xpsc_audio_dmadata *pcd = + snd_soc_platform_get_drvdata(rtd->platform); + return &pcd[SUBSTREAM_TYPE(ss)]; +} + static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -211,7 +213,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, goto out; stype = SUBSTREAM_TYPE(substream); - pcd = au1xpsc_audio_pcmdma[stype]; + pcd = to_dmadata(substream); DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " "runtime->min_align %d\n", @@ -249,8 +251,7 @@ static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream) static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) { - struct au1xpsc_audio_dmadata *pcd = - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); au1xxx_dbdma_reset(pcd->ddma_chan); @@ -267,7 +268,7 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan; + u32 c = to_dmadata(substream)->ddma_chan; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -287,8 +288,7 @@ static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) { - return bytes_to_frames(substream->runtime, - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos); + return bytes_to_frames(substream->runtime, to_dmadata(substream)->pos); } static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) @@ -299,7 +299,7 @@ static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]); + au1x_pcm_dbdma_free(to_dmadata(substream)); return 0; } @@ -329,35 +329,21 @@ static int au1xpsc_pcm_new(struct snd_card *card, return 0; } -static int au1xpsc_pcm_probe(struct snd_soc_platform *platform) -{ - if (!au1xpsc_audio_pcmdma[PCM_TX] || !au1xpsc_audio_pcmdma[PCM_RX]) - return -ENODEV; - - return 0; -} - /* au1xpsc audio platform */ struct snd_soc_platform_driver au1xpsc_soc_platform = { - .probe = au1xpsc_pcm_probe, .ops = &au1xpsc_pcm_ops, .pcm_new = au1xpsc_pcm_new, .pcm_free = au1xpsc_pcm_free_dma_buffers, }; -EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) { + struct au1xpsc_audio_dmadata *dmadata; struct resource *r; int ret; - if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX]) - return -EBUSY; - - /* TX DMA */ - au1xpsc_audio_pcmdma[PCM_TX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_TX]) + dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!dmadata) return -ENOMEM; r = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -365,54 +351,40 @@ static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) ret = -ENODEV; goto out1; } - (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start; + dmadata[PCM_TX].ddma_id = r->start; /* RX DMA */ - au1xpsc_audio_pcmdma[PCM_RX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_RX]) - return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) { ret = -ENODEV; - goto out2; + goto out1; } - (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; + dmadata[PCM_RX].ddma_id = r->start; + + platform_set_drvdata(pdev, dmadata); ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); if (!ret) return ret; -out2: - kfree(au1xpsc_audio_pcmdma[PCM_RX]); - au1xpsc_audio_pcmdma[PCM_RX] = NULL; out1: - kfree(au1xpsc_audio_pcmdma[PCM_TX]); - au1xpsc_audio_pcmdma[PCM_TX] = NULL; + kfree(dmadata); return ret; } static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev) { - int i; + struct au1xpsc_audio_dmadata *dmadata = platform_get_drvdata(pdev); snd_soc_unregister_platform(&pdev->dev); - - for (i = 0; i < 2; i++) { - if (au1xpsc_audio_pcmdma[i]) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); - kfree(au1xpsc_audio_pcmdma[i]); - au1xpsc_audio_pcmdma[i] = NULL; - } - } + kfree(dmadata); return 0; } static struct platform_driver au1xpsc_pcm_driver = { .driver = { - .name = "au1xpsc-pcm-audio", + .name = "au1xpsc-pcm", .owner = THIS_MODULE, }, .probe = au1xpsc_pcm_drvprobe, @@ -421,8 +393,6 @@ static struct platform_driver au1xpsc_pcm_driver = { static int __init au1xpsc_audio_dbdma_load(void) { - au1xpsc_audio_pcmdma[PCM_TX] = NULL; - au1xpsc_audio_pcmdma[PCM_RX] = NULL; return platform_driver_register(&au1xpsc_pcm_driver); } @@ -460,7 +430,7 @@ struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) res[1].start = res[1].end = id[1]; res[0].flags = res[1].flags = IORESOURCE_DMA; - pd = platform_device_alloc("au1xpsc-pcm", -1); + pd = platform_device_alloc("au1xpsc-pcm", pdev->id); if (!pd) goto out; diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 6a9516c..d0db66f 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -10,9 +10,6 @@ * * Au1xxx-PSC AC97 glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #include <linux/init.h> @@ -56,12 +53,29 @@ /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; +#if 0 + +/* this could theoretically work, but ac97->bus->card->private_data can be NULL + * when snd_ac97_mixer() is called; I don't know if the rest further down the + * chain are always valid either. + */ +static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x) +{ + struct snd_soc_card *c = x->bus->card->private_data; + return snd_soc_dai_get_drvdata(c->rtd->cpu_dai); +} + +#else + +#define ac97_to_pscdata(x) au1xpsc_ac97_workdata + +#endif + /* AC97 controller reads codec register */ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); unsigned short retry, tmo; unsigned long data; @@ -102,8 +116,7 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); unsigned int tmo, retry; au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); @@ -134,8 +147,7 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, /* AC97 controller asserts a warm reset */ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata)); au_sync(); @@ -146,8 +158,7 @@ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); int i; /* disable PSC during cold reset */ @@ -202,8 +213,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); unsigned long r, ro, stat; int chans, t, stype = SUBSTREAM_TYPE(substream); @@ -283,8 +293,7 @@ out: static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); int ret, stype = SUBSTREAM_TYPE(substream); ret = 0; @@ -325,7 +334,7 @@ static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { .hw_params = au1xpsc_ac97_hw_params, }; -struct snd_soc_dai_driver au1xpsc_ac97_dai = { +static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { .ac97_control = 1, .probe = au1xpsc_ac97_probe, .playback = { @@ -342,7 +351,6 @@ struct snd_soc_dai_driver au1xpsc_ac97_dai = { }, .ops = &au1xpsc_ac97_dai_ops, }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { @@ -351,9 +359,6 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) unsigned long sel; struct au1xpsc_audio_data *wd; - if (au1xpsc_ac97_workdata) - return -EBUSY; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); if (!wd) return -ENOMEM; @@ -387,14 +392,20 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd)); au_sync(); - ret = snd_soc_register_dai(&pdev->dev, &au1xpsc_ac97_dai); + /* name the DAI like this device instance ("au1xpsc-ac97.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_ac97_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); + + platform_set_drvdata(pdev, wd); + + ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) goto out1; wd->dmapd = au1xpsc_pcm_add(pdev); if (wd->dmapd) { - platform_set_drvdata(pdev, wd); - au1xpsc_ac97_workdata = wd; /* MDEV */ + au1xpsc_ac97_workdata = wd; return 0; } @@ -477,7 +488,7 @@ static struct dev_pm_ops au1xpscac97_pmops = { static struct platform_driver au1xpsc_ac97_driver = { .driver = { - .name = "au1xpsc-ac97", + .name = "au1xpsc_ac97", .owner = THIS_MODULE, .pm = AU1XPSCAC97_PMOPS, }, diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 94e560a..fca0912 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -10,9 +10,6 @@ * * Au1xxx-PSC I2S glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. * NOTE: so far only PSC slave mode (bit- and frameclock) is supported. */ @@ -54,13 +51,10 @@ ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_data *au1xpsc_i2s_workdata; - static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(cpu_dai); unsigned long ct; int ret; @@ -120,7 +114,7 @@ static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); int cfgbits; unsigned long stat; @@ -245,7 +239,7 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); int ret, stype = SUBSTREAM_TYPE(substream); switch (cmd) { @@ -263,19 +257,13 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int au1xpsc_i2s_probe(struct snd_soc_dai *dai) -{ - return au1xpsc_i2s_workdata ? 0 : -ENODEV; -} - static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, .set_fmt = au1xpsc_i2s_set_fmt, }; -static struct snd_soc_dai_driver au1xpsc_i2s_dai = { - .probe = au1xpsc_i2s_probe, +static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { .playback = { .rates = AU1XPSC_I2S_RATES, .formats = AU1XPSC_I2S_FMTS, @@ -298,9 +286,6 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) int ret; struct au1xpsc_audio_data *wd; - if (au1xpsc_i2s_workdata) - return -EBUSY; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); if (!wd) return -ENOMEM; @@ -337,17 +322,21 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) * time out. */ - ret = snd_soc_register_dai(&pdev->dev, &au1xpsc_i2s_dai); + /* name the DAI like this device instance ("au1xpsc-i2s.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_i2s_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); + + platform_set_drvdata(pdev, wd); + + ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) goto out1; /* finally add the DMA device for this PSC */ wd->dmapd = au1xpsc_pcm_add(pdev); - if (wd->dmapd) { - platform_set_drvdata(pdev, wd); - au1xpsc_i2s_workdata = wd; + if (wd->dmapd) return 0; - } snd_soc_unregister_dai(&pdev->dev); out1: @@ -376,8 +365,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) release_mem_region(r->start, resource_size(r)); kfree(wd); - au1xpsc_i2s_workdata = NULL; /* MDEV */ - return 0; } @@ -427,7 +414,7 @@ static struct dev_pm_ops au1xpsci2s_pmops = { static struct platform_driver au1xpsc_i2s_driver = { .driver = { - .name = "au1xpsc", + .name = "au1xpsc_i2s", .owner = THIS_MODULE, .pm = AU1XPSCI2S_PMOPS, }, @@ -437,7 +424,6 @@ static struct platform_driver au1xpsc_i2s_driver = { static int __init au1xpsc_i2s_load(void) { - au1xpsc_i2s_workdata = NULL; return platform_driver_register(&au1xpsc_i2s_driver); } diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index f281443..b30eadd 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -8,16 +8,11 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #ifndef _AU1X_PCM_H #define _AU1X_PCM_H -extern struct snd_ac97_bus_ops soc_ac97_ops; - /* DBDMA helpers */ extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev); extern void au1xpsc_pcm_destroy(struct platform_device *dmapd); @@ -28,6 +23,8 @@ struct au1xpsc_audio_data { unsigned long cfg; unsigned long rate; + struct snd_soc_dai_driver dai_drv; + unsigned long pm[2]; struct mutex lock; struct platform_device *dmapd; -- 1.7.2

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