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Re: [alsa-devel] plug:dmix slows playback
by Jaroslav Kysela 12 Jun '09

12 Jun '09

On Fri, 12 Jun 2009, Steve Langstaff wrote: >> From: alsa-devel-bounces(a)alsa-project.org [mailto:alsa-devel- >> bounces(a)alsa-project.org] On Behalf Of Jaroslav Kysela > >> On Fri, 12 Jun 2009, Steve Langstaff wrote: >> >>> Hi All. >>> >>> I've got an installation of ALSA (1.0.19 on an embedded device) where >> using >>> plug and dmix as my output device slows down playback - voices and >> tones are >>> lower in pitch and the playback takes longer than when not using >> plug:dmix, >>> e.g. >> >> Does 48000Hz 2 channel source work OK with direct device (without dmix) >> for your hardware? > > Yes, I converted the original sample to 48000 Hz, Stereo, and it sounds OK > when I don't use dmix. > > /mnt # time aplay -v ignition48000s.wav .... > Slave: Hardware PCM card 0 'LM4549' device 0 subdevice 0 > msbits : 16 If I remember correctly, this value was 32 in your dmix setup. Could you try command: aplay -D"plug:'dmix:FORMAT=S16_LE'" <filename> Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] plug:dmix slows playback
by Jaroslav Kysela 12 Jun '09

12 Jun '09

On Fri, 12 Jun 2009, Steve Langstaff wrote: > Hi All. > > I've got an installation of ALSA (1.0.19 on an embedded device) where using > plug and dmix as my output device slows down playback - voices and tones are > lower in pitch and the playback takes longer than when not using plug:dmix, > e.g. Does 48000Hz 2 channel source work OK with direct device (without dmix) for your hardware? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] plug:dmix slows playback
by Steve Langstaff 12 Jun '09

12 Jun '09

Hi All. I've got an installation of ALSA (1.0.19 on an embedded device) where using plug and dmix as my output device slows down playback - voices and tones are lower in pitch and the playback takes longer than when not using plug:dmix, e.g. /usr/share/sounds/alsa # time aplay ignition.wav Playing WAVE 'ignition.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono real 0m 3.08s user 0m 0.04s sys 0m 0.07s /usr/share/sounds/alsa # time aplay -D plug:dmix ignition.wav Playing WAVE 'ignition.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono real 0m 6.00s user 0m 0.09s sys 0m 0.08s See how the 'real' time for playing the file goes from 3.08s to 6.00s I have also noticed that the audio is played out of both (L+R) channels when not using -D plug:dmix, but is only played out of the right channel when using -D plug:dmix. I am an ALSA beginner - any ideas where I should start looking? Using the -v option on aplay we get more output... /usr/share/sounds/alsa # aplay -v ignition.wav Playing WAVE 'ignition.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 0 'LM4549' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 /usr/share/sounds/alsa # aplay -v -D plug:dmix ignition.wav Playing WAVE 'ignition.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono Plug PCM: Rate conversion PCM (48000, sformat=U8) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 2720 period_size : 170 period_time : 21250 tstamp_mode : NONE period_step : 1 avail_min : 170 period_event : 0 start_threshold : 2720 stop_threshold : 2720 silence_threshold: 0 silence_size : 0 boundary : 356515840 Slave: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 8 buffer_size : 16320 period_size : 1020 period_time : 21250 tstamp_mode : NONE period_step : 1 avail_min : 1020 period_event : 0 start_threshold : 16320 stop_threshold : 16320 silence_threshold: 0 silence_size : 0 boundary : 2139095040 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16320 period_size : 1020 period_time : 21250 tstamp_mode : NONE period_step : 1 avail_min : 1020 period_event : 0 start_threshold : 16320 stop_threshold : 16320 silence_threshold: 0 silence_size : 0 boundary : 2139095040 Hardware PCM card 0 'LM4549' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16320 period_size : 1020 period_time : 21250 tstamp_mode : ENABLE period_step : 1 avail_min : 1020 period_event : 0 start_threshold : 1 stop_threshold : 2139095040 silence_threshold: 0 silence_size : 2139095040 boundary : 2139095040 appl_ptr : 0 hw_ptr : 0

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Re: [alsa-devel] no sound in ubuntu 8.10 on AD1986 sound card
by Daniel Chen 12 Jun '09

12 Jun '09

Your 'PCM' is muted - is that intentional? Also, unless you're explicitly using digital out, you should mute 'IEC958'. Lastly, if you're using headphones, unmute that element, too. On Jun 10, 2009 12:18 PM, "sandip gangakhedkar" <sandipfloyd(a)gmail.com> wrote: Hi all, I am facing a very perplexing issue with sound in Ubuntu 8.10 which I have never encountered before with linux. I do not hear any sound from my speakers. The sound card is AD 1986 (Analog Devices) and it worked wonderfully without any hiccups before on 8.10 and other Ubuntu versions. My alsa-info script is here: http://www.alsa-project.org/db/?f=7f34b4e28d97178d7f709740f91ff66ea08a9468 I suspect pulseaudio is the culprit: Output of : /var/log/syslog | grep pulseaudio is here: http://paste.ubuntu.com/192629/ Running : speaker-test -c2 -D hw:0,0 -t wav -l1 gives no sound. My motherboard is ASUS A8N-VM on a AMD 64 Athlon 3200 Desktop PC.The sound card is detected so no problems there. I have reinstalled ALSA, pulseaudio and all the related packages but to no avail. I am pretty sure this is not a problem of inadeuate support because sound was working before without any issues. Any ideas/suggestions on how to fix this problem will be greatly appreciated. I am also raising an email bug report at alsa-devel so anyone can canter around there and provide expert help :) Thanks, Sandip _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] snd-ctxfi oops
by The Source 12 Jun '09

12 Jun '09

Hello. I use snd-ctxfi from 8 June snapshot. I hadn't oopses since I installed this driver but recently I got one again: Kernel failure message 1: BUG: sleeping function called from invalid context at mm/slub.c:1599 in_atomic(): 0, irqs_disabled(): 1, pid: 2537, name: gstreamer-prope Pid: 2537, comm: gstreamer-prope Tainted: P 2.6.29.4-167.fc11.x86_64 #1 Call Trace: [<ffffffff8103ff0f>] __might_sleep+0x10b/0x110 [<ffffffff810cd734>] __kmalloc+0x73/0x130 [<ffffffffa0b4b142>] ? daio_rsc_init+0xaa/0x125 [snd_ctxfi] [<ffffffffa0b4b212>] dao_rsc_init+0x55/0x1c0 [snd_ctxfi] [<ffffffffa0b4b3d2>] dao_rsc_reinit+0x55/0x5d [snd_ctxfi] [<ffffffff813abd6c>] ? _spin_lock_irqsave+0x32/0x3b [<ffffffffa0b454fe>] atc_spdif_out_passthru+0x92/0x136 [snd_ctxfi] [<ffffffffa0b4744d>] ct_pcm_playback_open+0x8b/0x11a [snd_ctxfi] [<ffffffffa0b34498>] snd_pcm_open_substream+0x46/0x83 [snd_pcm] [<ffffffffa0b345a8>] snd_pcm_open+0xd3/0x1db [snd_pcm] [<ffffffff810406fd>] ? default_wake_function+0x0/0x14 [<ffffffffa0b3471a>] snd_pcm_playback_open+0x31/0x36 [snd_pcm] [<ffffffff813abe16>] ? lock_kernel+0x21/0x31 [<ffffffffa00ec57d>] snd_open+0x105/0x184 [snd] [<ffffffff810d7da3>] chrdev_open+0x152/0x171 [<ffffffff810d7c51>] ? chrdev_open+0x0/0x171 [<ffffffff810d3922>] __dentry_open+0x143/0x273 [<ffffffff810d3b22>] nameidata_to_filp+0x42/0x53 [<ffffffff810df1bd>] do_filp_open+0x3fd/0x7b8 [<ffffffff810e0f0b>] ? vfs_ioctl+0x22/0x87 [<ffffffff811b95b2>] ? might_fault+0x1f/0x21 [<ffffffff811b96b1>] ? __strncpy_from_user+0x1e/0x49 [<ffffffff810e859c>] ? alloc_fd+0x11b/0x12d [<ffffffff810d3712>] do_sys_open+0x59/0xda [<ffffffff810d37c6>] sys_open+0x20/0x22 [<ffffffff8101133a>] system_call_fastpath+0x16/0x1b

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Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted:
by bill＠tellytopia.com 12 Jun '09

12 Jun '09

Yes, I have loaded the module snd-ctfxi many times, and unloaded snd-hda-intl. Also blacklisted snd-hda-intel, and put snd-ctxfi into modprobe.conf. I get more warnings on compiling the driver than you do - see below. (the messages below are from today not from Tuesday. The driver source was downloaded this morning.) Fedora Core 8: [root@localhost alsa-driver-unstable]# uname -a Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:18:33 EDT 2007 x86_64 x86_64 x86_64 GNU/Linux (the messages below are from today not from Tuesday. The driver source was downloaded this morning.) --------------------------- configure: creating ./config.status config.status: creating version config.status: creating Makefile.conf A couple of days ago I got this: config.status: WARNING: 'Makefile.conf.in' seems to ignore the --datarootdir setting but now this: $as_echo "$as_me: WARNING: $ac_file_inputs seems to ignore the --datarootdir setting" >&2;} config.status: creating snddevices -------------------------------------------------------------------------- These messages are from the "make": CC [M] /root/Desktop/alsa-driver-unstable/pci/ctxfi/ctatc.o /root/Desktop/alsa-driver-unstable/pci/ctxfi/ctatc.c: In function â€˜spdif_passthru_playback_prepareâ€™: /root/Desktop/alsa-driver-unstable/pci/ctxfi/ctatc.c:872: warning: â€˜errâ€™ may be used uninitialized in this function CC [M] /root/Desktop/alsa-driver-unstable/pci/ctxfi/ctvmem.o .......... CC [M] /root/Desktop/alsa-driver-unstable/pci/hda/hda_eld.o In file included from /root/Desktop/alsa-driver-unstable/pci/hda/hda_eld.c:4: /root/Desktop/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hda/hda_eld.c: In function â€˜hdmi_update_eldâ€™: /root/Desktop/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hda/hda_eld.c:281: warning: passing argument 1 of â€˜get_unaligned_le64â€™ discards qualifiers from pointer target type /root/Desktop/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hda/hda_eld.c:284: warning: passing argument 1 of â€˜get_unaligned_le16â€™ discards qualifiers from pointer target type /root/Desktop/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hda/hda_eld.c:285: warning: passing argument 1 of â€˜get_unaligned_le16â€™ discards qualifiers from pointer target type CC [M] /root/Desktop/alsa-driver-unstable/pci/hda/patch_nvhdmi.o ......... CC [M] /root/Desktop/alsa-driver-unstable/pci/riptide/riptide.o /root/Desktop/alsa-driver-unstable/pci/riptide/riptide.c: In function â€˜alsa_card_riptide_initâ€™: /root/Desktop/alsa-driver-unstable/pci/riptide/riptide.c:2206: warning: ignoring return value of â€˜pci_register_driverâ€™, declared with attribute warn_unused_result LD [M] /root/Desktop/alsa-driver-unstable/pci/riptide/snd-riptide.o ........... No errors at end: LD [M] /root/Desktop/alsa-driver-unstable/usb/usx2y/snd-usb-usx2y.ko make[1]: Leaving directory `/usr/src/kernels/2.6.23.1-42.fc8-x86_64' utils/link-modules /root/Desktop/alsa-driver-unstable ALSA modules were successfully compiled. [root@localhost alsa-driver-unstable]#

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[alsa-devel] [PATCH] ASoC: remove duplicated code on pxa-ssp.c
by Daniel Ribeiro 11 Jun '09

11 Jun '09

* We don't need to write the registers twice, remove the first write. * DAIFMT_INV switch is duplicated inside DAIFMT_FORMAT switch, move it out. (This patch is for Mark's for-2.6.32 branch, I have not checked if the code is duplicated on current 2.6.30) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 50 ++++++++++++++-------------------------------- 1 files changed, 15 insertions(+), 35 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e22c5ce 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -457,31 +457,27 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - ssp_write_reg(ssp, SSCR0, sscr0); - ssp_write_reg(ssp, SSCR1, sscr1); - ssp_write_reg(ssp, SSPSP, sspsp); + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; + break; + default: + return -EINVAL; + } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; - /* See hw_params() */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } break; case SND_SOC_DAIFMT_DSP_A: @@ -489,22 +485,6 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } break; default: -- tg: (a3ba169..) asoc/remove-code-duplication (depends on: asoc/upstream-for-2.6.32) -- Daniel Ribeiro

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[alsa-devel] HDA Intel IDT 92HD71B7X - HP Pavilion DV7-1299EF - Subwoofer not Working
by Wael Nasreddine (a.k.a eMxyzptlk) 11 Jun '09

11 Jun '09

Hello, Last night I bought myself a new laptop, the HP Pavilion DV7-1299EF. This laptop has 2 frontal speakers and a subwoofer in the back, on Windows, when I play a sound the 3 speakers works perfectly, and since it has a subwoofer the spearkers hardly has any bass capabilities, On linux the subwoofer is not working, only the 2 speakers are, hence the sound doesn't have any bass and it's loud and annoying, very annoying... I'm using alsa-drivers compiled from today's git checkout.. Here's the snd-hda-intel module options I tried options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=hp-m4 options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=hp-dv5 options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=hp-hdx options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=dell-m4-1 options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=dell-m4-2 options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=dell-m4-3 Whatever the model option I try, I always end up with only 2 speakers... Here's the alsa-info.sh output -> http://www.alsa-project.org/db/?f=0fd7cb5749b12dd4322879bb3828f9391bfee487 I also attached some more informative files.. Thanks for any help, it's really appreciated -- Wael Nasreddine Blog : http://wael.nasreddine.com E-mail : wael.nasreddine(a)gmail.com gTalk : wael.nasreddine(a)gmail.com Tel : +33.6.32.94.70.13 Skype : eMxyzptlk Twitter : @eMxyzptlk Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 .: An infinite number of monkeys typing into GNU emacs, would never make a good program. (L. Torvalds 1995) :.

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[alsa-devel] x-fi drivers and ati hdmi sound
by Matthias Diehl 11 Jun '09

11 Jun '09

hey Takashi, i'm really looking forward to you newest x-fi "hack". i loaded the file >> ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… << and tried to install it but i think there are problems with my graphics adapter (ati radeon hd 4870) which has a hdmi-soundchip onboard (hda intel). i dont know how to deactivate the hdmi-soundchip on my graphics adapter. lspci | grep audio shows me when installing your driver next to the hdmi driver: 01:00.1 Audio device: ATI Technologies Inc HD48x0 audio 05:00.0 Multimedia audio controller: Creative Labs SB X-Fi but there's no device to select in the audio options. do you have any idea or help for me? kind regards from germany, matthias ____________________________________________________________ Text: GRATIS für alle WEB.DE-Nutzer: Die maxdome Movie-FLAT! Jetzt freischalten unter http://movieflat.web.de

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[alsa-devel] Want to test a patch
by Chris Schumann 11 Jun '09

11 Jun '09

Hello all, I was hoping to get involved to fix als bug #305 and related ones, but I see Takashi Iwai has beaten me to it. Awesome. I'd like to test his patch of 5-27 for snd_cs46xx but I haven't done driver development before. How do I test out this patch on my hardware (ThinkPad 600X)? Thanks, Chris

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[alsa-devel] [PATCH] alsa: lx6464es - support index module parameter
by Tim Blechmann 11 Jun '09

11 Jun '09

trivial patch to put the device to a specific index Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 870bfc5..dd0f12b 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -43,6 +43,9 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Digigram LX6464ES interface."); + static const char card_name[] = "LX6464ES"; -- 1.6.2.1

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[alsa-devel] AC97 atmel
by Sedji Gaouaou 11 Jun '09

11 Jun '09

Hi, I have a patch which adds support for ATMEL AT91. I would like to know on which branch should I base my patch (linux ketchup or http://www.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git). Regards, Sedji

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Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted:
by bill＠tellytopia.com 11 Jun '09

11 Jun '09

I am using the snd-ctxfi driver on a 64-bit version of fedora code 7. Is there anyone else working on a 64-bit system? There are some compile warnings during the compilation of the driver, but no errors. But no sound, either, so far. I have also taken the 1.0.20 library, utils and tools sources from the Alsa main downloads. In order to compile the utils I had to link the 64-bit sound libraries to /usr/lib/libasound.2.0.0, in other words: (cd /lib64; ln -s /usr/lib/libasound.2.0.0 libasound.2.0.0; ln -s /usr/lib/libasound.2.0.0 libasound.2) There is no /lib64/libasound.la Also had to install xmlto (yum install xmlto). So far I have added the snd-hda-intel driver to the blacklist but still the Soundcard Detection shows snd-hda-intel. Any ideas, anyone? Regards Bill

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[alsa-devel] [PATCH v2 1/2] ASoC: change set_tdm_slot api to allow slot_size override.
by Daniel Ribeiro 11 Jun '09

11 Jun '09

Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override, I will submit this as a separate patch after I figure how to handle the I2S case. (this series is meant for Mark's for-2.6.32 branch + "remove duplicated code" patch I sent yesterday) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc-dai.h | 5 +++-- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++++++++------- sound/soc/soc-core.c | 9 ++++++--- 5 files changed, 33 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..6692410 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e..c47d573 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1206,8 +1206,9 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +/* FIXME: Needs to handle slot_width */ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1219,7 +1220,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cd..34437d8 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e22c5ce..613cccb 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(7) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..c72a8e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } -- tg: (c787980..) asoc/set_tdm_slot (depends on: asoc/remove-code-duplication) -- Daniel Ribeiro

2 2

[alsa-devel] [PATCH v2 2/2] ASoC: pxa-ssp.c, Automatically set TDM when needed
by Daniel Ribeiro 11 Jun '09

11 Jun '09

* Automatically sets TDM mode for frame_width larger than 32 bits, if the user doesn't setup the TDM slots with set_tdm_slot(). * Reset SSCR0_EDSS and SSCR0_DSS on pxa_ssp_set_dai_fmt. * Makes SSCR0_MOD optional. * Clears SSCR1_RWOT case SSCR0_MOD is set. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 88 ++++++++++++++++++++++++++++------------------ 1 files changed, 53 insertions(+), 35 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 613cccb..7e72c41 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -452,8 +452,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | + SSCR0_ACS | SSCR0_DSS | SSCR0_EDSS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -496,7 +496,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; break; @@ -532,10 +532,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sscr1, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int slot_width, frame_width = 0; + + /* check if the user explicitly set a slot_width */ + sscr0 = ssp_read_reg(ssp, SSCR0); + + if (sscr0 & (SSCR0_EDSS | SSCR0_DSS)) + slot_width = (sscr0 & SSCR0_DSS) + + (sscr0 & SSCR0_EDSS ? 17 : 1); + else + frame_width = slot_width = width * chn; /* generate correct DMA params */ if (cpu_dai->dma_data) @@ -545,35 +553,53 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), + cpu_dai->dma_data = ssp_get_dma_params(ssp, slot_width > 16, substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (sscr0 & SSCR0_SSE) return 0; - /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; + if (width == 16 && cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; #endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; + + if (frame_width > 0) { + /* Not using network mode */ + if (frame_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); + else + sscr0 |= SSCR0_DataSize(frame_width); + + if (frame_width > 32) { + /* + * Network mode is needed to support this frame width. + * We assume the wire is not networked and setup a + * fake network mode here. Use as many slots as needed + * each with 32 bits. + */ + int slots = frame_width / 32; + + sscr0 |= SSCR0_MOD; + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* + * Set active slots. Only set an active TX slot + * if we are going to use it. + */ + ssp_write_reg(ssp, SSRSA, slots - 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssp_write_reg(ssp, SSTSA, slots - 1); + } } + + /* If SSCR0_MOD is set we can't use SSCR1_RWOT */ + if (sscr0 & SSCR0_MOD) { + sscr1 = ssp_read_reg(ssp, SSCR1); + ssp_write_reg(ssp, SSCR1, sscr1 & ~SSCR1_RWOT); + } + ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -620,14 +646,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - dump_registers(ssp); return 0; -- tg: (5231b18..) asoc/ssp-internals (depends on: asoc/set_tdm_slot) -- Daniel Ribeiro

1 0

[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: set mixername
by Daniel Mack 11 Jun '09

11 Jun '09

alsamixer and friends want the mixername to be set. Even though the driver does not exports a real mixer device, export the name doesn't harm. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/device.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index b9a2b31..2240624 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.15"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -426,6 +426,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) strlcpy(card->driver, MODNAME, sizeof(card->driver)); strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); + strlcpy(card->mixername, dev->product_name, sizeof(card->mixername)); /* if the id was not passed as module option, fill it with a shortened * version of the product string which does not contain any -- 1.6.3.1

2 1

[alsa-devel] no sound in ubuntu 8.10 on AD1986 sound card
by sandip gangakhedkar 10 Jun '09

10 Jun '09

Hi all, I am facing a very perplexing issue with sound in Ubuntu 8.10 which I have never encountered before with linux. I do not hear any sound from my speakers. The sound card is AD 1986 (Analog Devices) and it worked wonderfully without any hiccups before on 8.10 and other Ubuntu versions. My alsa-info script is here: http://www.alsa-project.org/db/?f=7f34b4e28d97178d7f709740f91ff66ea08a9468 I suspect pulseaudio is the culprit: Output of : /var/log/syslog | grep pulseaudio is here: http://paste.ubuntu.com/192629/ Running : speaker-test -c2 -D hw:0,0 -t wav -l1 gives no sound. My motherboard is ASUS A8N-VM on a AMD 64 Athlon 3200 Desktop PC.The sound card is detected so no problems there. I have reinstalled ALSA, pulseaudio and all the related packages but to no avail. I am pretty sure this is not a problem of inadeuate support because sound was working before without any issues. Any ideas/suggestions on how to fix this problem will be greatly appreciated. I am also raising an email bug report at alsa-devel so anyone can canter around there and provide expert help :) Thanks, Sandip

3 3

[alsa-devel] [PATCH] ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205)
by Simos Xenitellis 10 Jun '09

10 Jun '09

A quirk is required for 8086:284b (rev 03) [Subsystem: 161f:2073]. The following has been tested with Alsa 1.0.20 (git master). Background details can be found at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4561 http://forum.ubuntu-gr.org/viewtopic.php?f=38&t=5290 Tested-by: Theodora Iliopoulou <th30dr(a)gmail.com> Signed-off-by: Simos Xenitellis <simos(a)gnome.org> -- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 42f944b..104d739 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,6 +1590,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, + "SigmaTel",STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ @@ -5939,6 +5941,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 }, { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 }, { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 }, + { .id = 0x83847698, .name = "STAC9205", .patch = patch_stac9205 }, { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 }, { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 }, { .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 },

2 1

Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 10 Jun '09

10 Jun '09

At Wed, 10 Jun 2009 18:21:55 +0200, MonsieurWeller wrote: > > Thanks for reply. > (Very) last little thing: i've tried to use the x-fi external rack with > my audigy2ZS (alsa driver 1.0.20 with emu10k1). And it seems to work (i > have only tried with the headphone jack). Maybe it's the same thing? Hard to tell, but it's anyway good to know. Thanks for the information! Takashi

1 0

[alsa-devel] EMU 0404USB: Odd sample rate shown during playback.
by Klaus Schulz 10 Jun '09

10 Jun '09

Hi there. The EMU 0404 is running in async USB mode as you know. When looking at the stream parameters during playback the sample rate shown is always slightly below 44100 -- something like 44098 is shown. As you write in the driver, the 0404 is returning number of samples per packet instead of the number of samples per microframes. Has this something to do with it? Question: Are we looking at a wrong output or is the samplerate really lower than 44100? THX Klaus

3 4

[alsa-devel] [PATCH] ASoC: Automatically set TDM if needed on pxa-ssp and allow frame width override when using TDM.
by Daniel Ribeiro 10 Jun '09

10 Jun '09

* Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. * Reset SSCR0_EDSS and SSCR0_DSS on pxa_ssp_set_dai_fmt. * Makes SSCR0_MOD optional. * Automatically sets network mode when needed if set_tdm_slot was never called. * Clears SSCR1_RWOT case SSCR0_MOD is set. * Updates magician.c and wm9081.c for the new set_tdm_slot() (Patch is based on Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc-dai.h | 5 + sound/soc/codecs/wm9081.c | 4 - sound/soc/pxa/magician.c | 2 sound/soc/pxa/pxa-ssp.c | 119 ++++++++++++++++++++++++++++------------------ sound/soc/soc-core.c | 9 ++- 5 files changed, 85 insertions(+), 54 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..f96cc36 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int frame_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e..85c720a 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1207,7 +1207,7 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, } static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1219,7 +1219,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (tx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cd..2345869 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1, 1, 16); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 46d14f3..f0931e7 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(7) | SSCR0_EDSS | SSCR0_DSS); + + /* set frame width */ + if (frame_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); + else + sscr0 |= SSCR0_DataSize(frame_width); + + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } @@ -439,8 +452,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | + SSCR0_ACS | SSCR0_EDSS | SSCR0_DSS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -487,7 +500,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -537,48 +550,70 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sscr1, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int frame_width; + + /* check if the user explicitly set a frame_width */ + sscr0 = ssp_read_reg(ssp, SSCR0); + + if (sscr0 & (SSCR0_EDSS | SSCR0_DSS)) + frame_width = (sscr0 & SSCR0_DSS) + + (sscr0 & SSCR0_EDSS ? 17 : 1); + else + frame_width = width * chn; /* generate correct DMA params */ if (cpu_dai->dma_data) kfree(cpu_dai->dma_data); - /* Network mode with one active slot (ttsa == 1) can be used - * to force 16-bit frame width on the wire (for S16_LE), even - * with two channels. Use 16-bit DMA transfers for this case. - */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), + cpu_dai->dma_data = ssp_get_dma_params(ssp, frame_width > 16, substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (sscr0 & SSCR0_SSE) return 0; - /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + /* FIXME: What this is for? */ #ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; + if (width == 16 && cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; #endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; + + if (!(sscr0 & (SSCR0_EDSS | SSCR0_DSS))) { + /* Frame width not set yet, we are not using network mode */ + if (frame_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); + else + sscr0 |= SSCR0_DataSize(frame_width); + + if (frame_width > 32) { + /* + * Network mode is needed to support this frame_width + * We assume that the wire is not networked and setup + * a "fake" network mode here. + */ + int slots = frame_width / 32; + + sscr0 |= SSCR0_MOD; + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* + * Set active slots. Only set an active TX slot + * if we are going to use it. + */ + ssp_write_reg(ssp, SSRSA, slots - 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssp_write_reg(ssp, SSTSA, slots - 1); + } } + + /* If SSCR0_MOD is set we can't use SSCR1_RWOT */ + if (sscr0 & SSCR0_MOD) { + sscr1 = ssp_read_reg(ssp, SSCR1); + ssp_write_reg(ssp, SSCR1, sscr1 & ~SSCR1_RWOT); + } + ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -625,14 +660,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - dump_registers(ssp); return 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..69becf2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @frame_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, frame_width); else return -EINVAL; } -- Daniel Ribeiro

4 9

[alsa-devel] X-Fi suspend/resume support patch
by Takashi Iwai 10 Jun '09

10 Jun '09

Hi, can anyone test the patch below for supporting suspend/resume with X-Fi driver? Note that it's just compile tested but not really checked with any real hardware, so highly likely incomplete. Keep backup before trying this :) thanks, Takashi --- diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 80fb2ba..9861418 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1119,18 +1119,67 @@ static int atc_spdif_out_passthru(struct ct_atc *atc, unsigned char state) return err; } -static int ct_atc_destroy(struct ct_atc *atc) +static void ct_daio_clear_connections(struct ct_atc *atc) { struct daio_mgr *daio_mgr; struct dao *dao; struct dai *dai; struct daio *daio; + int i; + + daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; + for (i = 0; i < atc->n_daio; i++) { + daio = atc->daios[i]; + if (daio->type < LINEIM) { + dao = container_of(daio, struct dao, daio); + dao->ops->clear_left_input(dao); + dao->ops->clear_right_input(dao); + } else { + dai = container_of(daio, struct dai, daio); + /* some thing to do for dai ... */ + } + daio_mgr->put_daio(daio_mgr, daio); + } +} + +static void ct_sum_clear_connections(struct ct_atc *atc) +{ struct sum_mgr *sum_mgr; + int i; + + sum_mgr = atc->rsc_mgrs[SUM]; + for (i = 0; i < atc->n_pcm; i++) + sum_mgr->put_sum(sum_mgr, atc->pcm[i]); +} + +static void ct_src_clear_connections(struct ct_atc *atc) +{ struct src_mgr *src_mgr; + int i; + + src_mgr = atc->rsc_mgrs[SRC]; + for (i = 0; i < atc->n_src; i++) + src_mgr->put_src(src_mgr, atc->srcs[i]); +} + +static void ct_srcimp_clear_connections(struct ct_atc *atc) +{ struct srcimp_mgr *srcimp_mgr; struct srcimp *srcimp; + int i; + + srcimp_mgr = atc->rsc_mgrs[SRCIMP]; + for (i = 0; i < atc->n_srcimp; i++) { + srcimp = atc->srcimps[i]; + srcimp->ops->unmap(srcimp); + srcimp_mgr->put_srcimp(srcimp_mgr, atc->srcimps[i]); + } +} + +static int ct_atc_destroy(struct ct_atc *atc) +{ struct ct_mixer *mixer; - int i = 0; + int i; if (NULL == atc) return 0; @@ -1157,45 +1206,22 @@ static int ct_atc_destroy(struct ct_atc *atc) } if (NULL != atc->daios) { - daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; - for (i = 0; i < atc->n_daio; i++) { - daio = atc->daios[i]; - if (daio->type < LINEIM) { - dao = container_of(daio, struct dao, daio); - dao->ops->clear_left_input(dao); - dao->ops->clear_right_input(dao); - } else { - dai = container_of(daio, struct dai, daio); - /* some thing to do for dai ... */ - } - daio_mgr->put_daio(daio_mgr, daio); - } + ct_daio_clear_connections(atc); kfree(atc->daios); } if (NULL != atc->pcm) { - sum_mgr = atc->rsc_mgrs[SUM]; - for (i = 0; i < atc->n_pcm; i++) - sum_mgr->put_sum(sum_mgr, atc->pcm[i]); - + ct_sum_clear_connections(atc); kfree(atc->pcm); } if (NULL != atc->srcs) { - src_mgr = atc->rsc_mgrs[SRC]; - for (i = 0; i < atc->n_src; i++) - src_mgr->put_src(src_mgr, atc->srcs[i]); - + ct_src_clear_connections(atc); kfree(atc->srcs); } if (NULL != atc->srcimps) { - srcimp_mgr = atc->rsc_mgrs[SRCIMP]; - for (i = 0; i < atc->n_srcimp; i++) { - srcimp = atc->srcimps[i]; - srcimp->ops->unmap(srcimp); - srcimp_mgr->put_srcimp(srcimp_mgr, atc->srcimps[i]); - } + ct_srcimp_clear_connections(atc); kfree(atc->srcimps); } @@ -1314,37 +1340,15 @@ static int __devinit atc_create_hw_devs(struct ct_atc *atc) return 0; } -static int __devinit atc_get_resources(struct ct_atc *atc) +static int ct_daio_init_connections(struct ct_atc *atc) { struct daio_desc da_desc = {0}; struct daio_mgr *daio_mgr; - struct src_desc src_dsc = {0}; - struct src_mgr *src_mgr; - struct srcimp_desc srcimp_dsc = {0}; - struct srcimp_mgr *srcimp_mgr; - struct sum_desc sum_dsc = {0}; - struct sum_mgr *sum_mgr; - int err, i; - - atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL); - if (NULL == atc->daios) - return -ENOMEM; - - atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); - if (NULL == atc->srcs) - return -ENOMEM; - - atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); - if (NULL == atc->srcimps) - return -ENOMEM; - - atc->pcm = kzalloc(sizeof(void *)*(2*4), GFP_KERNEL); - if (NULL == atc->pcm) - return -ENOMEM; + int i, err; daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; da_desc.msr = atc->msr; - for (i = 0, atc->n_daio = 0; i < DAIONUM-1; i++) { + for (i = 0, atc->n_daio = 0; i < DAIONUM - 1; i++) { da_desc.type = i; err = daio_mgr->get_daio(daio_mgr, &da_desc, (struct daio **)&atc->daios[i]); @@ -1366,12 +1370,20 @@ static int __devinit atc_get_resources(struct ct_atc *atc) return err; } atc->n_daio++; + return 0; +} + +static int ct_src_init_connections(struct ct_atc *atc) +{ + struct src_desc src_dsc = {0}; + struct src_mgr *src_mgr; + int i, err; src_mgr = atc->rsc_mgrs[SRC]; src_dsc.multi = 1; src_dsc.msr = atc->msr; src_dsc.mode = ARCRW; - for (i = 0, atc->n_src = 0; i < (2*2); i++) { + for (i = 0, atc->n_src = 0; i < (2 * 2); i++) { err = src_mgr->get_src(src_mgr, &src_dsc, (struct src **)&atc->srcs[i]); if (err) @@ -1379,10 +1391,18 @@ static int __devinit atc_get_resources(struct ct_atc *atc) atc->n_src++; } + return 0; +} + +static int ct_srcimp_init_connections(struct ct_atc *atc) +{ + struct srcimp_desc srcimp_dsc = {0}; + struct srcimp_mgr *srcimp_mgr; + int i, err; srcimp_mgr = atc->rsc_mgrs[SRCIMP]; srcimp_dsc.msr = 8; /* SRCIMPs for S/PDIFIn SRT */ - for (i = 0, atc->n_srcimp = 0; i < (2*1); i++) { + for (i = 0, atc->n_srcimp = 0; i < (2 * 1); i++) { err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc, (struct srcimp **)&atc->srcimps[i]); if (err) @@ -1391,7 +1411,7 @@ static int __devinit atc_get_resources(struct ct_atc *atc) atc->n_srcimp++; } srcimp_dsc.msr = 8; /* SRCIMPs for LINE/MICIn SRT */ - for (i = 0; i < (2*1); i++) { + for (i = 0; i < (2 * 1); i++) { err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc, (struct srcimp **)&atc->srcimps[2*1+i]); if (err) @@ -1399,10 +1419,18 @@ static int __devinit atc_get_resources(struct ct_atc *atc) atc->n_srcimp++; } + return 0; +} + +static int ct_sum_init_connections(struct ct_atc *atc) +{ + struct sum_desc sum_dsc = {0}; + struct sum_mgr *sum_mgr; + int i, err; sum_mgr = atc->rsc_mgrs[SUM]; sum_dsc.msr = atc->msr; - for (i = 0, atc->n_pcm = 0; i < (2*4); i++) { + for (i = 0, atc->n_pcm = 0; i < (2 * 4); i++) { err = sum_mgr->get_sum(sum_mgr, &sum_dsc, (struct sum **)&atc->pcm[i]); if (err) @@ -1410,6 +1438,41 @@ static int __devinit atc_get_resources(struct ct_atc *atc) atc->n_pcm++; } + return 0; +} + +static int __devinit atc_get_resources(struct ct_atc *atc) +{ + int err; + + atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL); + if (NULL == atc->daios) + return -ENOMEM; + + atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); + if (NULL == atc->srcs) + return -ENOMEM; + + atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); + if (NULL == atc->srcimps) + return -ENOMEM; + + atc->pcm = kzalloc(sizeof(void *)*(2*4), GFP_KERNEL); + if (NULL == atc->pcm) + return -ENOMEM; + + err = ct_daio_init_connections(atc); + if (err < 0) + return err; + err = ct_src_init_connections(atc); + if (err < 0) + return err; + err = ct_srcimp_init_connections(atc); + if (err < 0) + return err; + err = ct_sum_init_connections(atc); + if (err < 0) + return err; err = ct_mixer_create(atc, (struct ct_mixer **)&atc->mixer); if (err) { @@ -1420,7 +1483,7 @@ static int __devinit atc_get_resources(struct ct_atc *atc) return 0; } -static void __devinit +static void atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai, struct src **srcs, struct srcimp **srcimps) { @@ -1459,7 +1522,7 @@ atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai, src_mgr->commit_write(src_mgr); /* Synchronously enable SRCs */ } -static void __devinit atc_connect_resources(struct ct_atc *atc) +static void atc_connect_resources(struct ct_atc *atc) { struct dai *dai; struct dao *dao; @@ -1617,3 +1680,51 @@ error1: printk(KERN_ERR "ctxfi: Something wrong!!!\n"); return err; } + +#ifdef CONFIG_PM +int ct_atc_suspend(struct ct_atc *atc) +{ + struct hw *hw = atc->hw; + int i; + + snd_power_change_state(atc->card, SNDRV_CTL_POWER_D3hot); + for (i = FRONT; i <= IEC958; i++) + ct_pcm_suspend(atc->pcm_devs[i]); + ct_mixer_suspend(atc->mixer); + + ct_daio_clear_connections(atc); + ct_sum_clear_connections(atc); + ct_src_clear_connections(atc); + ct_srcimp_clear_connections(atc); + + memset(atc->daios, 0, sizeof(void *) * DAIONUM); + memset(atc->pcm, 0, sizeof(void *) * (2 * 4)); + memset(atc->srcs, 0, sizeof(void *) * (2 *2)); + memset(atc->srcimps, 0, sizeof(void *) * (2*2)); + + hw->card_suspend(hw); + return 0; +} + +int ct_atc_resume(struct ct_atc *atc) +{ + struct hw *hw = atc->hw; + int err; + + err = hw->card_resume(hw); + if (err < 0) { + snd_card_disconnect(atc->card); + return -EIO; + } + + ct_daio_init_connections(atc); + ct_src_init_connections(atc); + ct_srcimp_init_connections(atc); + ct_sum_init_connections(atc); + + atc_connect_resources(atc); + ct_mixer_resume(atc->mixer); + snd_power_change_state(atc->card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index a033472..e8251ff 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -79,6 +79,7 @@ struct ct_atc { unsigned int rsr; /* reference sample rate in Hz */ unsigned int msr; /* master sample rate in rsr */ unsigned int pll_rate; /* current rate of Phase Lock Loop */ + struct snd_pcm *pcm_devs[NUM_CTALSADEVS]; int chip_type; int model; @@ -144,4 +145,9 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, struct ct_atc **ratc); int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc); +#ifdef CONFIG_PM +int ct_atc_suspend(struct ct_atc *atc); +int ct_atc_resume(struct ct_atc *atc); +#endif + #endif /* CTATC_H */ diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index 4a8e04f..6a3de6a 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -60,6 +60,10 @@ struct card_conf { struct hw { int (*card_init)(struct hw *hw, struct card_conf *info); int (*card_stop)(struct hw *hw); +#ifdef CONFIG_PM + int (*card_suspend)(struct hw *hw); + int (*card_resume)(struct hw *hw); +#endif int (*pll_init)(struct hw *hw, unsigned int rsr); int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source); int (*select_adc_source)(struct hw *hw, enum ADCSRC source); @@ -164,6 +168,8 @@ struct hw { void (*irq_callback)(void *data, unsigned int bit); void *irq_callback_data; + struct card_conf info; + struct pci_dev *pci; /* the pci kernel structure of this card */ int irq; unsigned long io_base; diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index cb69d9d..e1b6d8d 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1976,7 +1976,7 @@ static int hw_card_shutdown(struct hw *hw) return 0; } -static int hw_card_init(struct hw *hw, struct card_conf *info) +static int hw_init(struct hw *hw, struct card_conf *info) { int err; unsigned int gctl; @@ -2119,11 +2119,54 @@ static void hw_write_pci(struct hw *hw, u32 reg, u32 data) &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags); } +static int hw_card_init(struct hw *hw, struct card_conf *info) +{ + hw->info = *info; + return hw_init(hw, info); +} + +#ifdef CONFIG_PM +static int hw_card_suspend(struct hw *hw) +{ + hw_card_stop(hw); + pci_disable_device(hw->pci); + pci_save_state(hw->pci); + return 0; +} + +static int hw_card_resume(struct hw *hw) +{ + int err; + struct pci_dev *pci = hw->pci; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "ctxfi: pci_enable_device failed, " + "disabling device\n"); + return -EIO; + } + if (hw->model == CTUAA) { + err = uaa_to_xfi(pci); + if (err) + return err; + } + pci_set_master(pci); + + hw_init(hw, &hw->info); + return 0; +} +#endif + static struct hw ct20k1_preset __devinitdata = { .irq = -1, .card_init = hw_card_init, .card_stop = hw_card_stop, +#ifdef CONFIG_PM + .card_suspend = hw_card_suspend, + .card_resume = hw_card_resume, +#endif .pll_init = hw_pll_init, .is_adc_source_selected = hw_is_adc_input_selected, .select_adc_source = hw_adc_input_select, diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 4493a51..5db75ea 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1845,7 +1845,6 @@ static int hw_card_start(struct hw *hw) { int err = 0; struct pci_dev *pci = hw->pci; - unsigned int gctl; err = pci_enable_device(pci); if (err < 0) @@ -1872,11 +1871,6 @@ static int hw_card_start(struct hw *hw) goto error2; } - /* Switch to 20k2 mode from UAA mode. */ - gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL); - set_field(&gctl, GCTL_UAA, 0); - hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); - /*if ((err = request_irq(pci->irq, ct_atc_interrupt, IRQF_SHARED, atc->chip_details->nm_card, hw))) { goto error3; @@ -1927,7 +1921,7 @@ static int hw_card_shutdown(struct hw *hw) return 0; } -static int hw_card_init(struct hw *hw, struct card_conf *info) +static int hw_init(struct hw *hw, struct card_conf *info) { int err; unsigned int gctl; @@ -1945,6 +1939,11 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) return err; } + /* Switch to 20k2 mode from UAA mode. */ + gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL); + set_field(&gctl, GCTL_UAA, 0); + hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); + /* PLL init */ err = hw_pll_init(hw, info->rsr); if (err < 0) @@ -2006,6 +2005,39 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) return 0; } +static int hw_card_init(struct hw *hw, struct card_conf *info) +{ + hw->info = *info; + return hw_init(hw, info); +} + +#ifdef CONFIG_PM +static int hw_card_suspend(struct hw *hw) +{ + hw_card_stop(hw); + pci_disable_device(hw->pci); + pci_save_state(hw->pci); + return 0; +} + +static int hw_card_resume(struct hw *hw) +{ + struct pci_dev *pci = hw->pci; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "ctxfi: pci_enable_device failed, " + "disabling device\n"); + return -EIO; + } + pci_set_master(pci); + + hw_init(hw, &hw->info); + return 0; +} +#endif + static u32 hw_read_20kx(struct hw *hw, u32 reg) { return readl((void *)(hw->mem_base + reg)); @@ -2021,6 +2053,10 @@ static struct hw ct20k2_preset __devinitdata = { .card_init = hw_card_init, .card_stop = hw_card_stop, +#ifdef CONFIG_PM + .card_suspend = hw_card_suspend, + .card_resume = hw_card_resume, +#endif .pll_init = hw_pll_init, .is_adc_source_selected = hw_is_adc_input_selected, .select_adc_source = hw_adc_input_select, diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index 666722d..4e5547a 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -485,6 +485,57 @@ static int ct_alsa_mix_switch_get(struct snd_kcontrol *kcontrol, return 0; } +static void ct_mixer_switch_update(struct ct_atc *atc, + struct ct_mixer *mixer, + enum CTALSA_MIXER_CTL type, + int state) +{ + /* Do changes in mixer. */ + if (SWH_CAPTURE_START <= type && SWH_CAPTURE_END >= type) { + enum CT_AMIXER_CTL idx = get_amixer_index(type); + if (state) + ct_mixer_recording_select(mixer, idx); + else + ct_mixer_recording_unselect(mixer, idx); + return; + } + + /* Do changes out of mixer. */ + switch (type) { + case MIXER_LINEIN_C_S: + case MIXER_MIC_C_S: + if (state) + do_line_mic_switch(atc, type); + break; + case MIXER_WAVEF_P_S: + atc->line_front_unmute(atc, state); + break; + case MIXER_WAVES_P_S: + atc->line_surround_unmute(atc, state); + break; + case MIXER_WAVEC_P_S: + atc->line_clfe_unmute(atc, state); + break; + case MIXER_WAVER_P_S: + atc->line_rear_unmute(atc, state); + break; + case MIXER_LINEIN_P_S: + atc->line_in_unmute(atc, state); + break; + case MIXER_SPDIFO_P_S: + atc->spdif_out_unmute(atc, state); + break; + case MIXER_SPDIFI_P_S: + atc->spdif_in_unmute(atc, state); + break; + case MIXER_DIGITAL_IO_S: + do_digit_io_switch(atc, state); + break; + default: + break; + } +} + static int ct_alsa_mix_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -498,35 +549,7 @@ static int ct_alsa_mix_switch_put(struct snd_kcontrol *kcontrol, return 0; set_switch_state(mixer, type, state); - /* Do changes in mixer. */ - if ((SWH_CAPTURE_START <= type) && (SWH_CAPTURE_END >= type)) { - if (state) { - ct_mixer_recording_select(mixer, - get_amixer_index(type)); - } else { - ct_mixer_recording_unselect(mixer, - get_amixer_index(type)); - } - } - /* Do changes out of mixer. */ - if (state && (MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type)) - do_line_mic_switch(atc, type); - else if (MIXER_WAVEF_P_S == type) - atc->line_front_unmute(atc, state); - else if (MIXER_WAVES_P_S == type) - atc->line_surround_unmute(atc, state); - else if (MIXER_WAVEC_P_S == type) - atc->line_clfe_unmute(atc, state); - else if (MIXER_WAVER_P_S == type) - atc->line_rear_unmute(atc, state); - else if (MIXER_LINEIN_P_S == type) - atc->line_in_unmute(atc, state); - else if (MIXER_SPDIFO_P_S == type) - atc->spdif_out_unmute(atc, state); - else if (MIXER_SPDIFI_P_S == type) - atc->spdif_in_unmute(atc, state); - else if (MIXER_DIGITAL_IO_S == type) - do_digit_io_switch(atc, state); + ct_mixer_switch_update(atc, mixer, type, state); return 1; } @@ -823,7 +846,6 @@ error1: static int ct_mixer_get_mem(struct ct_mixer **rmixer) { struct ct_mixer *mixer; - int err; *rmixer = NULL; /* Allocate mem for mixer obj */ @@ -833,24 +855,27 @@ static int ct_mixer_get_mem(struct ct_mixer **rmixer) mixer->amixers = kzalloc(sizeof(void *)*(NUM_CT_AMIXERS*CHN_NUM), GFP_KERNEL); - if (NULL == mixer->amixers) { - err = -ENOMEM; - goto error1; - } + if (!mixer->amixers) + goto error; mixer->sums = kzalloc(sizeof(void *)*(NUM_CT_SUMS*CHN_NUM), GFP_KERNEL); - if (NULL == mixer->sums) { - err = -ENOMEM; - goto error2; - } + if (!mixer->sums) + goto error; + +#ifdef CONFIG_PM + mixer->saved_vol = kmalloc(sizeof(u32) * VOL_MIXER_NUM * CHN_NUM, + GFP_KERNEL); + if (!mixer->saved_vol) + goto error; +#endif *rmixer = mixer; return 0; -error2: +error: kfree(mixer->amixers); -error1: + kfree(mixer->sums); kfree(mixer); - return err; + return -ENOMEM; } static int ct_mixer_topology_build(struct ct_mixer *mixer) @@ -1039,13 +1064,13 @@ mixer_set_input_right(struct ct_mixer *mixer, return 0; } -int ct_mixer_destroy(struct ct_mixer *mixer) +static void ct_mixer_release_resources(struct ct_mixer *mixer) { struct sum_mgr *sum_mgr = (struct sum_mgr *)mixer->atc->rsc_mgrs[SUM]; struct amixer_mgr *amixer_mgr = (struct amixer_mgr *)mixer->atc->rsc_mgrs[AMIXER]; struct amixer *amixer; - int i = 0; + int i; /* Release amixer resources */ for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) { @@ -1060,8 +1085,15 @@ int ct_mixer_destroy(struct ct_mixer *mixer) if (NULL != mixer->sums[i]) sum_mgr->put_sum(sum_mgr, (struct sum *)mixer->sums[i]); } +} +int ct_mixer_destroy(struct ct_mixer *mixer) +{ + ct_mixer_release_resources(mixer); /* Release mem assigned to mixer object */ +#ifdef CONFIG_PM + kfree(mixer->saved_vol); +#endif kfree(mixer->sums); kfree(mixer->amixers); kfree(mixer); @@ -1121,3 +1153,49 @@ int ct_alsa_mix_create(struct ct_atc *atc, return 0; } + +#ifdef CONFIG_PM +void ct_mixer_suspend(struct ct_mixer *mixer) +{ + struct ct_atc *atc = mixer->atc; + int i, c, idx; + + idx = 0; + for (i = VOL_MIXER_START; i <= VOL_MIXER_END; i++) { + for (c = 0; c < CHN_NUM; c++) { + struct amixer *amixer = mixer->amixers[idx]; + mixer->saved_vol[idx] = amixer->ops->get_scale(amixer); + idx++; + } + } + + atc->spdif_out_get_status(atc, &mixer->saved_spdif_status); + ct_mixer_release_resources(mixer); +} + +void ct_mixer_resume(struct ct_mixer *mixer) +{ + struct ct_atc *atc = mixer->atc; + int i, c, idx; + + ct_mixer_topology_build(mixer); + + idx = 0; + for (i = VOL_MIXER_START; i <= VOL_MIXER_END; i++) { + for (c = 0; c < CHN_NUM; c++) { + struct amixer *amixer = mixer->amixers[idx]; + amixer->ops->set_scale(amixer, mixer->saved_vol[idx]); + idx++; + } + } + + for (i = SWH_MIXER_START; i <= SWH_MIXER_END; i++) { + if (i == MIXER_DIGITAL_IO_S && + !atc->have_digit_io_switch(atc)) + continue; + ct_mixer_switch_update(atc, mixer, i, + get_switch_state(mixer, i)); + } + atc->spdif_out_set_status(atc, mixer->saved_spdif_status); +} +#endif diff --git a/sound/pci/ctxfi/ctmixer.h b/sound/pci/ctxfi/ctmixer.h index e2d96eb..597a0c9 100644 --- a/sound/pci/ctxfi/ctmixer.h +++ b/sound/pci/ctxfi/ctmixer.h @@ -48,6 +48,10 @@ struct ct_mixer { void **amixers; /* amixer resources for volume control */ void **sums; /* sum resources for signal collection */ unsigned int switch_state; /* A bit-map to indicate state of switches */ +#ifdef CONFIG_PM + u32 *saved_vol; + unsigned int saved_spdif_status; +#endif int (*get_output_ports)(struct ct_mixer *mixer, enum MIXER_PORT_T type, struct rsc **rleft, struct rsc **rright); @@ -64,4 +68,9 @@ int ct_alsa_mix_create(struct ct_atc *atc, int ct_mixer_create(struct ct_atc *atc, struct ct_mixer **rmixer); int ct_mixer_destroy(struct ct_mixer *mixer); +#ifdef CONFIG_PM +void ct_mixer_suspend(struct ct_mixer *mixer); +void ct_mixer_resume(struct ct_mixer *mixer); +#endif + #endif /* CTMIXER_H */ diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 9e5c0c4..6f872ba 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -422,5 +422,31 @@ int ct_alsa_pcm_create(struct ct_atc *atc, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(atc->pci), 128*1024, 128*1024); + atc->pcm_devs[device] = pcm; + return 0; } + +#ifdef CONFIG_PM +void ct_pcm_suspend(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *s; + int i; + + if (!pcm) + return; + snd_pcm_suspend_all(pcm); + /* clear all PCM resources once */ + for (i = 0; i < 2; i++) { + for (s = pcm->streams[i].substream; s; s = s->next) { + struct ct_atc *atc; + struct ct_atc_pcm *apcm; + if (!s->runtime) + continue; + apcm = s->runtime->private_data; + atc = snd_pcm_substream_chip(s); + atc->pcm_release_resources(atc, apcm); + } + } +} +#endif diff --git a/sound/pci/ctxfi/ctpcm.h b/sound/pci/ctxfi/ctpcm.h index 178da0d..867139b 100644 --- a/sound/pci/ctxfi/ctpcm.h +++ b/sound/pci/ctxfi/ctpcm.h @@ -24,4 +24,8 @@ int ct_alsa_pcm_create(struct ct_atc *atc, enum CTALSADEVS device, const char *device_name); +#ifdef CONFIG_PM +void ct_pcm_suspend(struct snd_pcm *pcm); +#endif + #endif /* CTPCM_H */ diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 2d3dd89..59a0876 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -121,11 +121,31 @@ static void __devexit ct_card_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } +#ifdef CONFIG_PM +static int ct_card_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct ct_atc *atc = card->private_data; + return ct_atc_suspend(atc); +} + +static int ct_card_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct ct_atc *atc = card->private_data; + return ct_atc_resume(atc); +} +#endif + static struct pci_driver ct_driver = { .name = "SB-XFi", .id_table = ct_pci_dev_ids, .probe = ct_card_probe, .remove = __devexit_p(ct_card_remove), +#ifdef CONFIG_PM + .suspend = ct_card_suspend, + .resume = ct_card_resume, +#endif }; static int __init ct_card_init(void)

1 1

[alsa-devel] ASoC update for 2.6.31
by Mark Brown 10 Jun '09

10 Jun '09

The following changes since commit 74b8f955a73d20b1e22403fd1ef85834fbf38d98: Mark Brown (1): ASoC: Apostrophe patrol are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Mark Brown (1): ASoC: Fix lm4857 control sound/soc/s3c24xx/neo1973_wm8753.c | 16 ++++++++++------ 1 files changed, 10 insertions(+), 6 deletions(-)

2 1

[alsa-devel] Sound streaming application: which device to use?
by Lex Wassenberg 10 Jun '09

10 Jun '09

Hi all, I'm completely new to ALSA, so forgive me if these are newby questions. I'm working on an application which receives sound chunks via an IP connection, and it should use the local sound card to make this sound audible. Up to now it used OSS, but there were some problems with that, so I'm busy with rewriting the application to ALSA. The sound is 16-bit stereo, sampled at 8000 Hz, and it comes in chunks of 1024 frames. I managed to get things more or less working, but there are still some things unclear to me. 1. If I open the soundcard with device "default" or device "plughw" I get good sound, but the device "default" only accepts 1020 frames of every 1024 that I offer it (as can be seen from the return value of snd_pcm_writei() ). The device "plughw" nicely accepts all 1024 frames. Why is this? In both cases I have set the sample rate to 8000 via a call to snd_pcm_hw_params_set_rate_near() and the buffer size to 2048 frames via snd_pcm_hw_params_set_buffer_size_near(). 2. If I use device "default", there seems to be no obligation to set the buffer size nor the period size, I still get good sound at the output. If I use "plughw", I have to set either the buffer size (2048 frames) or the period size (1024 frames), otherwise I get distorted sound. If I set both (the buffer size being 2048 and the period size being 1024) I also get distorted sound. Why? 3. If I set the buffer size twice the period size (as above), and ask the sizes back, ALSA nicely reports the buffer size as being twice the period size. But if I ask the buffer and period TIMES, they are reported as being the same: 128000 us! Huh???? 4. The return value of snd_pcm_hw_params_set_rate_near() is sometimes 8000, sometimes 7999, from the very same program. I would expect ALSA to always return the same value, be it what I request or not, but at least being deterministic. How can this be? 5. There is another application running on the same target system, which will sometimes produce audible alarms. However, if my application is running that other application cannot generate the alarms, it produces a "resource busy" error. It is clear to me that my application is using the sound card exclusively, but how can I get it to allow other applications access to it too? I tried using the device "dmix", but then my sound is again very distorted. Hope that somebody can help me out. Thanks in advance. Lex

4 5

[alsa-devel] Codec 1708b
by Dark Dragon 10 Jun '09

10 Jun '09

I cant seem to get my m3873-vm asus motherboard to play Sorround Sound audio via digital (HDMI) Profile at http://www.alsa-project.org/db/?f=26d1122ea384bc64805e516610c1bab3bea4d75c Works when playing sterio audio but "file now found" errors when trying to play sorround sound audio

1 0

[alsa-devel] Problems with X-Fi driver
by Daniel Bankmann 09 Jun '09

09 Jun '09

Takashi Iwai schrieb: > At Fri, 22 May 2009 11:54:47 +0200, > Daniel Bankmann wrote: >> Takashi Iwai schrieb: >>> At Thu, 21 May 2009 09:53:23 +0200, >>> Daniel Bankmann wrote: >>>> Hi! >>>> >>>> I just tested the newest driver version with Open Suse 11.1 and >>>> pae-kernel. I seems not to work probably, although the latest "official" >>>> driver by Creative worked more or less good. >>>> >>>> The problem is, that always, when hearing some music or any other sound, >>>> the signal gets interrupted every few seconds. At the other side, the >>>> startup jingle for example is accompanied by a knock every few seconds. >>>> >>>> Any idea, or what information do you need to fix this? >>> If you are using pulseaudio, try to upgrade to the one in Factory. >>> PA on 11.1 is really a crappy version, unfortunately. >>> >>> >>> Takashi >>> >> Hi, >> >> are you sure, that this is a problem with PA? > > Depends on your system setup. As default, PA is used on KDE and GNOME, > regardless of h/w multi-stream functions. > >> Upgrading to the newest >> version or the one in Factory seems to be a bit tricky (because auf >> missing dependencies). > > You can upgrade the packages on OBS multimedia:libs repo. > >> Shouldn't the driver just work without PA and ESD installed? I tried >> that out, but the problem still resists. > > It's not the driver to decide whether to use PA or not. It's the > upper part. > > If the problem still occurs without PA (and you make sure that you > don't use it), the issue is likely the hwptr update timing of the > ctxfi driver. The recent PCM core has a more strict check, and > possibly ctxfi might not provide the correct DMA position. > > Anyway, please check the problem occurs with which apps more exactly. > At best, try to reproduce with aplay and speaker-test programs, i.e. > with less complexity. And, please report back to alsa-devel ML. > (Note that your post may appear later after manual approvals if you > aren't subscribed yet). > > > thanks, > > Takashi > Hi, so I tried reproducing this problem. Here's a link to a wav-file which includes tests with aplay and the starting/shutdown -jingle. I hope this can give you a better idea of my problem;) Daniel http://rapidshare.com/files/239574432/output_test.wav

2 4

[alsa-devel] Default enabling of mixer controls
by Aggarwal, Anuj 09 Jun '09

09 Jun '09

Hi, I want to by-default enable the following mixer controls for my platform: Analog Left Capture Route AUXL Analog Right Capture Route AUXR Right now, they both come disabled in the default configuration and I have to enable them explicitly by issuing the amixer cset command. How can I do that automatically? Thanks and Regards, Anuj Aggarwal

3 6

[alsa-devel] Setting gain/volume of PCM capture or playback
by erumusele Odigie 09 Jun '09

09 Jun '09

Hi, I am new to programming with ALSA, and I would like to find out how to change the gain (capture or playback) for my PCM device in my C code. I am aware I am supposed to use the mixer API, possibly functions like: snd_mixer_selem_set_playback_volume() but I don't know how this function can be connected to my PCM structure/device. Please whoever answers this should kindly attach some sample code, or a link to sample code because I can't find any on line. Thanks a lot. Erumusele

2 3

[alsa-devel] Testers wanted: New SB X-Fi driver
by Gernot Kohlhaas 09 Jun '09

09 Jun '09

Hi and thanks for this driver. It works great and even 6- and 8-Channel-Output is supported now. The only two little things I noticed are: 1. Most programs are looking for "surround51" to use for multichannel sound. But this driver only provides the "default"-Device (in it's description it says: aplay -L default:CARD=XFi Creative X-Fi, Front/WaveIn Default Audio Device ) 2. I can hear a gentle grumbling out of my speakers. But nothing serious. -- http://www.gnupp.de/verschluesselung/warum.html Key ID: 0x4C0B2250 Key: http://home.arcor.de/mr_maniac/ANDERES/public.gpg

3 3

[alsa-devel] [PATCH 4/7] Alchemy: DB1200 AC97+I2S audio support.
by Manuel Lauss 08 Jun '09

08 Jun '09

Replaces the sample Alchemy PSC AC97 machine code with a DB1200 machine driver with AC97 and I2S support. AC97/I2S can be selected at boot time by setting switch S6.7. Cc: alsa-devel(a)alsa-project.org Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- arch/mips/alchemy/devboards/db1200/platform.c | 12 ++ sound/soc/au1x/Kconfig | 10 +- sound/soc/au1x/Makefile | 4 +- sound/soc/au1x/db1200.c | 189 +++++++++++++++++++++++++ sound/soc/au1x/sample-ac97.c | 144 ------------------- 5 files changed, 209 insertions(+), 150 deletions(-) create mode 100644 sound/soc/au1x/db1200.c delete mode 100644 sound/soc/au1x/sample-ac97.c diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c index cc3f663..b3142e0 100644 --- a/arch/mips/alchemy/devboards/db1200/platform.c +++ b/arch/mips/alchemy/devboards/db1200/platform.c @@ -428,6 +428,7 @@ static int __init db1200_dev_init(void) ARRAY_SIZE(db1200_i2c_devs)); /* SWITCHES: S6.8 I2C/SPI selector (OFF=I2C ON=SPI) + * S6.7 AC97/I2S selector (OFF=AC97 ON=I2S) */ /* NOTE: GPIO215 controls OTG VBUS supply. In SPI mode however @@ -466,6 +467,17 @@ static int __init db1200_dev_init(void) au_writel(pfc, SYS_PINFUNC); au_sync(); + /* audio is handled in sound/soc/au1x/db1200.c; + * however we need I2C to talk to I2S codec. + */ + if ((bcsr->switches & 3) == BCSR_SWITCHES_DIP_8) { + bcsr->resets |= BCSR_RESETS_PSC1MUX; + printk(KERN_INFO " S6.7 ON : PSC1 mode I2S\n"); + } else { + bcsr->resets &= ~BCSR_RESETS_PSC1MUX; + printk(KERN_INFO " S6.7 OFF: PSC1 mode AC97\n"); + } + return platform_add_devices(db1200_devs, ARRAY_SIZE(db1200_devs)); } device_initcall(db1200_dev_init); diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 410a893..4b67140 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -22,11 +22,13 @@ config SND_SOC_AU1XPSC_AC97 ## ## Boards ## -config SND_SOC_SAMPLE_PSC_AC97 - tristate "Sample Au12x0/Au1550 PSC AC97 sound machine" +config SND_SOC_DB1200 + tristate "DB1200 AC97+I2S audio support" depends on SND_SOC_AU1XPSC select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC + select SND_SOC_AU1XPSC_I2S + select SND_SOC_WM8731 help - This is a sample AC97 sound machine for use in Au12x0/Au1550 - based systems which have audio on PSC1 (e.g. Db1200 demoboard). + Select this option to enable audio (AC97 or I2S) on the + Alchemy/AMD/RMI DB1200 demoboard. diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile index 6c6950b..1687307 100644 --- a/sound/soc/au1x/Makefile +++ b/sound/soc/au1x/Makefile @@ -8,6 +8,6 @@ obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o # Boards -snd-soc-sample-ac97-objs := sample-ac97.o +snd-soc-db1200-objs := db1200.o -obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o +obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c new file mode 100644 index 0000000..a2cb02c --- /dev/null +++ b/sound/soc/au1x/db1200.c @@ -0,0 +1,189 @@ +/* + * DB1200 ASoC audio support code. + * + * (c) 2008 Manuel Lauss <manuel.lauss(a)gmail.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx.h> +#include <asm/mach-au1x00/au1xxx_psc.h> +#include <asm/mach-au1x00/au1xxx_dbdma.h> + +/* it sucks that the ASoC headers are not under include/ */ +#include "../codecs/ac97.h" +#include "../codecs/wm8731.h" +#include "psc.h" + +/*------------------------- AC97 PART ---------------------------*/ + +static int db1200_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link db1200_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &au1xpsc_ac97_dai, + .codec_dai = &ac97_dai, + .init = db1200_ac97_init, +}; + +static struct snd_soc_card db1200_ac97_machine = { + .name = "DB1200 AC97 Audio", + .dai_link = &db1200_ac97_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_ac97_devdata = { + .card = &db1200_ac97_machine, + .codec_dev = &soc_codec_dev_ac97, +}; + +/*------------------------- I2S PART ---------------------------*/ + +static int db1200_i2s_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* WM8731 has its own 12MHz crystal */ + snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + + /* codec is bitclock and lrclk master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = 0; +out: + return ret; +} + +static struct snd_soc_ops db1200_i2s_wm8731_ops = { + .startup = db1200_i2s_startup, +}; + +static int db1200_i2s_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link db1200_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &au1xpsc_i2s_dai, + .codec_dai = &wm8731_dai, + .init = db1200_i2s_init, + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1200_i2s_machine = { + .name = "DB1200 I2S Audio", + .dai_link = &db1200_i2s_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_i2s_devdata = { + .card = &db1200_i2s_machine, + .codec_dev = &soc_codec_dev_wm8731, +}; + +/*------------------------- COMMON PART ---------------------------*/ + +static struct resource psc1_res[] = { + [0] = { + .start = CPHYSADDR(PSC1_BASE_ADDR), + .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = AU1200_PSC1_INT, + .end = AU1200_PSC1_INT, + .flags = IORESOURCE_IRQ, + }, + [2] = { + .start = DSCR_CMD0_PSC1_TX, + .end = DSCR_CMD0_PSC1_TX, + .flags = IORESOURCE_DMA, + }, + [3] = { + .start = DSCR_CMD0_PSC1_RX, + .end = DSCR_CMD0_PSC1_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device *db1200_asoc_dev; + +static int __init db1200_audio_load(void) +{ + int ret; + + /* DB1200: PSC clock is supplied externally */ + au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); + au_sync(); + + ret = -ENOMEM; + db1200_asoc_dev = platform_device_alloc("soc-audio", -1); + if (!db1200_asoc_dev) + goto out; + + db1200_asoc_dev->resource = + kmemdup(psc1_res, sizeof(struct resource) * + ARRAY_SIZE(psc1_res), GFP_KERNEL); + db1200_asoc_dev->num_resources = ARRAY_SIZE(psc1_res); + db1200_asoc_dev->id = 1; + + /* DB1200 board setup set PSC1MUX to preferred audio device */ + if (bcsr->resets & BCSR_RESETS_PSC1MUX) + platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_devdata); + else + platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_devdata); + + db1200_ac97_devdata.dev = &db1200_asoc_dev->dev; + db1200_i2s_devdata.dev = &db1200_asoc_dev->dev; + ret = platform_device_add(db1200_asoc_dev); + + if (ret) { + platform_device_put(db1200_asoc_dev); + db1200_asoc_dev = NULL; + } +out: + return ret; +} + +static void __exit db1200_audio_unload(void) +{ + platform_device_unregister(db1200_asoc_dev); +} + +module_init(db1200_audio_load); +module_exit(db1200_audio_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("DB1200 ASoC audio support"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c deleted file mode 100644 index 27683eb..0000000 --- a/sound/soc/au1x/sample-ac97.c +++ /dev/null @@ -1,144 +0,0 @@ -/* - * Sample Au12x0/Au1550 PSC AC97 sound machine. - * - * Copyright (c) 2007-2008 Manuel Lauss <mano(a)roarinelk.homelinux.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms outlined in the file COPYING at the root of this - * source archive. - * - * This is a very generic AC97 sound machine driver for boards which - * have (AC97) audio at PSC1 (e.g. DB1200 demoboards). - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <asm/mach-au1x00/au1000.h> -#include <asm/mach-au1x00/au1xxx_psc.h> -#include <asm/mach-au1x00/au1xxx_dbdma.h> - -#include "../codecs/ac97.h" -#include "psc.h" - -static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_sync(codec); - return 0; -} - -static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */ - .codec_dai = &ac97_dai, /* see codecs/ac97.c */ - .init = au1xpsc_sample_ac97_init, - .ops = NULL, -}; - -static struct snd_soc_card au1xpsc_sample_ac97_machine = { - .name = "Au1xxx PSC AC97 Audio", - .dai_link = &au1xpsc_sample_ac97_dai, - .num_links = 1, -}; - -static struct snd_soc_device au1xpsc_sample_ac97_devdata = { - .card = &au1xpsc_sample_ac97_machine, - .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct resource au1xpsc_psc1_res[] = { - [0] = { - .start = CPHYSADDR(PSC1_BASE_ADDR), - .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, - .flags = IORESOURCE_MEM, - }, - [1] = { -#ifdef CONFIG_SOC_AU1200 - .start = AU1200_PSC1_INT, - .end = AU1200_PSC1_INT, -#elif defined(CONFIG_SOC_AU1550) - .start = AU1550_PSC1_INT, - .end = AU1550_PSC1_INT, -#endif - .flags = IORESOURCE_IRQ, - }, - [2] = { - .start = DSCR_CMD0_PSC1_TX, - .end = DSCR_CMD0_PSC1_TX, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = DSCR_CMD0_PSC1_RX, - .end = DSCR_CMD0_PSC1_RX, - .flags = IORESOURCE_DMA, - }, -}; - -static struct platform_device *au1xpsc_sample_ac97_dev; - -static int __init au1xpsc_sample_ac97_load(void) -{ - int ret; - -#ifdef CONFIG_SOC_AU1200 - unsigned long io; - - /* modify sys_pinfunc for AC97 on PSC1 */ - io = au_readl(SYS_PINFUNC); - io |= SYS_PINFUNC_P1C; - io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B); - au_writel(io, SYS_PINFUNC); - au_sync(); -#endif - - ret = -ENOMEM; - - /* setup PSC clock source for AC97 part: external clock provided - * by codec. The psc-ac97.c driver depends on this setting! - */ - au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); - au_sync(); - - au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); - if (!au1xpsc_sample_ac97_dev) - goto out; - - au1xpsc_sample_ac97_dev->resource = - kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * - ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); - au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); - au1xpsc_sample_ac97_dev->id = 1; - - platform_set_drvdata(au1xpsc_sample_ac97_dev, - &au1xpsc_sample_ac97_devdata); - au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; - ret = platform_device_add(au1xpsc_sample_ac97_dev); - - if (ret) { - platform_device_put(au1xpsc_sample_ac97_dev); - au1xpsc_sample_ac97_dev = NULL; - } - -out: - return ret; -} - -static void __exit au1xpsc_sample_ac97_exit(void) -{ - platform_device_unregister(au1xpsc_sample_ac97_dev); -} - -module_init(au1xpsc_sample_ac97_load); -module_exit(au1xpsc_sample_ac97_exit); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); -- 1.6.3.1

2 12

[alsa-devel] PATCH: C-Media 6206 USB Audio
by Dan Allongo 08 Jun '09

08 Jun '09

I've been posting on the alsa-users list and I think I finally have a working patch now that I'd like to submit. The C-Media 6206 device is based off of CM106 and just needed a boot quirk added to set the registers properly for output. At least one other user can confirm that this patch works and that the SPDIF-out is also functioning now with this change. This is diff'd against today's snapshot so it should be good to go. Let me know if there's anything else that I need to do or change with this. Thanks, ---Dan --- alsa-driver/alsa-kernel/usb/usbaudio.c 2009-06-07 06:20:32.000000000 -0400 +++ alsa-driver/alsa-kernel/usb/usbaudio.c 2009-06-07 22:24:44.000000000 -0400 @@ -3291,6 +3291,21 @@ return snd_usb_cm106_write_int_reg(dev, 2, 0x8004); } +static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) +{ + /* + * C-Media CM6206 is based on CM106 with two additional + * registers that are not documented in the data sheet. + * Values here are chosen based on sniffing USB traffic + * under Windows. + */ + return snd_usb_cm106_write_int_reg(dev, 0, 0x200c) + + snd_usb_cm106_write_int_reg(dev, 1, 0x3000) + + snd_usb_cm106_write_int_reg(dev, 2, 0xf800) + + snd_usb_cm106_write_int_reg(dev, 3, 0x143f) + + snd_usb_cm106_write_int_reg(dev, 4, 0x0000) + + snd_usb_cm106_write_int_reg(dev, 5, 0x3000); +} /* * Setup quirks @@ -3577,6 +3592,12 @@ goto __err_val; } + /* C-Media CM6206 / CM106-Like Sound Device */ + if (id == USB_ID(0x0d8c, 0x0102)) { + if (snd_usb_cm6206_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */

2 2

[PATCH - CM6206 1/1] modified: sound/usb/usbaudio.c
by Dan Allongo 08 Jun '09

08 Jun '09

Added boot quirk for C-Media CM6206 device in snd_usb_audio_probe. The function snd_usb_cm6206_boot_quirk sets up six internal 16-bit registers in order to initialize the device. Values for the registers came from sniffing USB traffic under Windows since only four of the six are documented in the datasheet for CM106 and some reserved bits were also being set. Signed-off-by: Dan Allongo <gongo2k1(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index a8ef2cb..94e43ee 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3291,6 +3291,24 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) return snd_usb_cm106_write_int_reg(dev, 2, 0x8004); } +/* + * C-Media CM6206 is based on CM106 with two additional + * registers that are not documented in the data sheet. + * Values here are chosen based on sniffing USB traffic + * under Windows. + */ +static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) +{ + int err, reg; + int val[] = {0x200c,0x3000,0xf800,0x143f,0x0000,0x3000}; + + for (reg = 0; reg < 6; reg++) { + if ((err = snd_usb_cm106_write_int_reg(dev, reg, val[reg])) < 0) + return err; + } + + return err; +} /* * Setup quirks @@ -3577,6 +3595,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* C-Media CM6206 / CM106-Like Sound Device */ + if (id == USB_ID(0x0d8c, 0x0102)) { + if (snd_usb_cm6206_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ -- 1.5.6.5 On Mon, Jun 8, 2009 at 1:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sun, 7 Jun 2009 22:46:51 -0400, > Dan Allongo wrote: > > > > I've been posting on the alsa-users list and I think I finally have a > > working patch now that I'd like to submit. > > The C-Media 6206 device is based off of CM106 and just needed a boot > quirk > > added to set the registers properly for output. > > At least one other user can confirm that this patch works and that the > > SPDIF-out is also functioning now with this change. > > This is diff'd against today's snapshot so it should be good to go. > > Let me know if there's anything else that I need to do or change with > this. > > Thanks for the patch! I'd like certainly to apply it, however... > > > + return snd_usb_cm106_write_int_reg(dev, 0, 0x200c) + > > + snd_usb_cm106_write_int_reg(dev, 1, 0x3000) + > > + snd_usb_cm106_write_int_reg(dev, 2, 0xf800) + > > + snd_usb_cm106_write_int_reg(dev, 3, 0x143f) + > > + snd_usb_cm106_write_int_reg(dev, 4, 0x0000) + > > + snd_usb_cm106_write_int_reg(dev, 5, 0x3000); > > Summing the return values here look weird. > You want to return an error code, right? Then use a simple if > sequence, or write a loop there. > > Also, run $LINUX/scripts/checkpatch.pl to your patch and fix errors / > warnings suggested there. Then please repost the patch with your > sign-off. > > > thanks, > > Takashi >

1 0

[alsa-devel] [PATCH 0/7] DAPM power sequencing changes
by Mark Brown 08 Jun '09

08 Jun '09

I've implemented what I hope are some improvements to the DAPM power sequence application. Currently when it is implementing power changes DAPM will do one register write per widget. Normally this leads to multiple writes to the same register since power control tends to be clustered in a few registers, slowing things down. This can get noticable on slow or heavily contended buses, especially around resume. The overall effect of these changes is that DAPM will try to minimise the number of register writes it does when implementing power sequences. As with any substantial change in the power sequencing there is a risk that there may have been some performance regressions on some systems - my biggest concern is that some thing were being helped by the delays introduced by the repeated register writes and may require special handling to fix. My hope is that this won't happen if anything it should fix some problems by ensuring that both left and right channels of stereo paths get powered up together but without wide testing that's hard to guarantee. Any testing that you can do on your systems would be greatly appreciated. Obviously this is quite an invasive change so I don't intend to submit to ALSA until after the 2.6.31 merge window has closed. The patches can be found here for the time being: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git dapm Mark Brown (7): ASoC: Sort DAPM power sequences while building lists ASoC: Factor out DAPM sequence execution ASoC: Coalesce register writes for DAPM sequences ASoC: Sort specialised mixers and muxes together ASoC: Coalesce power updates for DAPM widgets with events ASoC: Coalesce power updates for PGAs ASoC: Make DAPM power sequence lists local variables include/sound/soc.h | 2 - sound/soc/soc-dapm.c | 302 +++++++++++++++++++++++++++++++++++++------------- 2 files changed, 227 insertions(+), 77 deletions(-)

1 7

[alsa-devel] [PATCH] ARM: DaVinci: ASoc: Minor fixes to probe function
by chaithrika＠ti.com 08 Jun '09

08 Jun '09

From: Chaithrika U S <chaithrika(a)ti.com> Assign proper errors when platform resource claims fail. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to branch 'davinci' of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 500d2f5..2f11970 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -469,7 +469,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; - int ret = 0; + int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { @@ -511,6 +511,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; @@ -518,6 +519,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; -- 1.5.6

2 1

[alsa-devel] M-Audio Transit: alsamixer shows 100% and -0.5db at the same time !?
by Klaus Schulz 08 Jun '09

08 Jun '09

Hi folks. My M-Audio Transit shows at 100% and 0.5db attentuation in alsamixer at the same time. See below printout. What could be the explanation for this behaviour? THX \Klaus amixer -Dhw:0 Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 254 Mono: Front Left: Playback 254 [100%] [0.50dB] [on] Front Right: Playback 254 [100%] [0.50dB] [on]

3 2

[alsa-devel] [PATCH v4 2/4] ARM: DaVinci: ASoC: Introduce platform driver model for dm644x, dm355
by Chaithrika U S 08 Jun '09

08 Jun '09

Introduce the platform driver model to get platform data for dm355 and dm644x. Register platform driver and acquire the resources in the probe function Since the platform specific code had been moved from machine driver to dm<soc>.c Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 117 ++++++++++++++++++++++----------------- 1 files changed, 67 insertions(+), 50 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index b1ea52f..500d2f5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -436,16 +436,40 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_i2s_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea; - struct evm_snd_platform_data *pdata; - int ret; + struct resource *mem, *ioarea, *res; + int ret = 0; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { @@ -466,9 +490,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, goto err_release_region; } - cpu_dai->private_data = dev; - - dev->clk = clk_get(&pdev->dev, NULL); + dev->clk = clk_get(&pdev->dev, pdata->clk_name); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -476,18 +498,35 @@ static int davinci_i2s_probe(struct platform_device *pdev, clk_enable(dev->clk); dev->base = (void __iomem *)IO_ADDRESS(mem->start); - pdata = pdev->dev.platform_data; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_i2s_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_i2s_dai); + if (ret != 0) + goto err_free_mem; + return 0; err_free_mem: @@ -498,62 +537,40 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int davinci_i2s_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = davinci_i2s_dai.private_data; struct resource *mem; + snd_soc_unregister_dai(&davinci_i2s_dai); clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; - kfree(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); -} -#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 - -static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, -}; + return 0; +} -struct snd_soc_dai davinci_i2s_dai = { - .name = "davinci-i2s", - .id = 0, - .probe = davinci_i2s_probe, - .remove = davinci_i2s_remove, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &davinci_i2s_dai_ops, +static struct platform_driver davinci_mcbsp_driver = { + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .driver = { + .name = "davinci-asp", + .owner = THIS_MODULE, + }, }; -EXPORT_SYMBOL_GPL(davinci_i2s_dai); static int __init davinci_i2s_init(void) { - return snd_soc_register_dai(&davinci_i2s_dai); + return platform_driver_register(&davinci_mcbsp_driver); } module_init(davinci_i2s_init); static void __exit davinci_i2s_exit(void) { - snd_soc_unregister_dai(&davinci_i2s_dai); + platform_driver_unregister(&davinci_mcbsp_driver); } module_exit(davinci_i2s_exit); -- 1.5.6

3 2

[alsa-devel] [PATCH] ASoC: Allow 32bits registers
by Daniel Ribeiro 08 Jun '09

08 Jun '09

Replace the remaining unsigned shorts with unsigned ints. Tested with pcap2 codec (25 bits registers). Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 28 ++++++++++++++-------------- sound/soc/soc-dapm.c | 16 ++++++++-------- 3 files changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a40bc6f..33b7de1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -212,9 +212,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1cd149b..c9f19d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1262,10 +1262,10 @@ EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); * Returns 1 for change else 0. */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1292,10 +1292,10 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * Returns 1 for change else 0. */ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1580,7 +1580,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1609,8 +1609,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask, bitmask; + unsigned int val; + unsigned int mask, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1646,7 +1646,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1685,8 +1685,8 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask; + unsigned int val; + unsigned int mask; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -1846,7 +1846,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); if (invert) @@ -1912,7 +1912,7 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; int max = mc->max; - unsigned int mask = (1<<fls(max))-1; + unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; ucontrol->value.integer.value[0] = @@ -1952,7 +1952,7 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; int err; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val_mask = mask << shift; val = (ucontrol->value.integer.value[0] & mask); @@ -2044,7 +2044,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; int min = mc->min; - unsigned short val; + unsigned int val; val = (ucontrol->value.integer.value[0]+min) & 0xff; val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 735903a..75c372f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -227,7 +227,7 @@ static int dapm_connect_mixer(struct snd_soc_codec *codec, static int dapm_update_bits(struct snd_soc_dapm_widget *widget) { int change, power; - unsigned short old, new; + unsigned int old, new; struct snd_soc_codec *codec = widget->codec; /* check for valid widgets */ @@ -1205,7 +1205,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; int ret; val = (ucontrol->value.integer.value[0] & mask); @@ -1269,7 +1269,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1297,8 +1297,8 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask, bitmask; + unsigned int val, mux; + unsigned int mask, bitmask; int ret = 0; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) @@ -1356,7 +1356,7 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(widget->codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1396,8 +1396,8 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask; + unsigned int val, mux; + unsigned int mask; int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) -- tg: (81ee1ba..) asoc/allow-32bits-registers (depends on: master) -- Daniel Ribeiro

2 2

[alsa-devel] ASoC update for 2.6.31
by Mark Brown 08 Jun '09

08 Jun '09

The following changes since commit e3509ff0fb9df53e45cd68488e3b463a80455db7: Daniel Mack (1): ASoC: fix NULL pointer dereference in soc_suspend() are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Mark Brown (1): ASoC: Apostrophe patrol Troy Kisky (1): ASoC: codec tlv320aic23 fix bogus divide by 0 message sound/soc/codecs/tlv320aic23.c | 10 +++++----- sound/soc/soc-dapm.c | 10 +++++----- 2 files changed, 10 insertions(+), 10 deletions(-)

2 1

[alsa-devel] Audio driver for dbAu1250 Development board
by Manjunath A M 08 Jun '09

08 Jun '09

Hi I am trying to make audio up in dbAu1250 Development board, I downloaded linux-2.6.29.4 kernel from the site "http://www.linux-mips.org/pub/linux/mips/kernel/v2.6/ " and applied the patch "asoc-v2.6.29-rc8-2009-03-17.patch"downloaded from "http://opensource.wolfsonmicro.com/~git/asoc/" but while compiling i am getting the bellow errors ********Linux compilation errors************* sound/soc/au1x/exmmb-ac97.c:24:38: error: asm/mach-exm1200/exm1200.h: No such file or directory sound/soc/au1x/exmmb-ac97.c:26:25: error: linux/exm32.h: No such file or directory sound/soc/au1x/exmmb-ac97.c:58: error: subscripted value is neither array nor pointer sound/soc/au1x/exmmb-ac97.c:65: error: variable âsnd_soc_machine_exm97â has initializer but incomplete type sound/soc/au1x/exmmb-ac97.c:66: error: unknown field ânameâ specified in initializer sound/soc/au1x/exmmb-ac97.c:66: warning: excess elements in struct initializer sound/soc/au1x/exmmb-ac97.c:66: warning: (near initialization for âsnd_soc_machine_exm97â) sound/soc/au1x/exmmb-ac97.c:67: error: unknown field âdai_linkâ specified in initializer sound/soc/au1x/exmmb-ac97.c:67: warning: excess elements in struct initializer sound/soc/au1x/exmmb-ac97.c:67: warning: (near initialization for âsnd_soc_machine_exm97â) sound/soc/au1x/exmmb-ac97.c:68: error: unknown field ânum_linksâ specified in initializer sound/soc/au1x/exmmb-ac97.c:68: warning: excess elements in struct initializer sound/soc/au1x/exmmb-ac97.c:68: warning: (near initialization for âsnd_soc_machine_exm97â) sound/soc/au1x/exmmb-ac97.c:73: error: unknown field âmachineâ specified in initializer sound/soc/au1x/exmmb-ac97.c:73: warning: initialization from incompatible pointer type sound/soc/au1x/exmmb-ac97.c:74: error: unknown field âplatformâ specified in initializer sound/soc/au1x/exmmb-ac97.c:74: warning: initialization from incompatible pointer type sound/soc/au1x/exmmb-ac97.c:80: error: âPSC1_BASEâ undeclared here (not in a function) sound/soc/au1x/exmmb-ac97.c: In function âpsc_to_ac97â: sound/soc/au1x/exmmb-ac97.c:107: error: implicit declaration of function âau_readlâ sound/soc/au1x/exmmb-ac97.c:110: error: implicit declaration of function âau_writelâ sound/soc/au1x/exmmb-ac97.c:111: error: implicit declaration of function âau_syncâ sound/soc/au1x/exmmb-ac97.c:115: error: âPSC_CTLâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:115: error: (Each undeclared identifier is reported only once sound/soc/au1x/exmmb-ac97.c:115: error: for each function it appears in.) sound/soc/au1x/exmmb-ac97.c:117: error: âPSC_SELâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:122: error: âPSC_AC97RSTâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:124: error: implicit declaration of function âmsleepâ sound/soc/au1x/exmmb-ac97.c:133: error: âPSC_AC97STATâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:140: error: âPSC_AC97CFGâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c: In function âexm1200_mobo_ac97_initâ: sound/soc/au1x/exmmb-ac97.c:160: error: implicit declaration of function âau_readwâ sound/soc/au1x/exmmb-ac97.c:160: error: âEXM1200_FPGA_BRDCTRLâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:162: error: implicit declaration of function âau_writewâ make[3]: *** [sound/soc/au1x/exmmb-ac97.o] Error 1 make[2]: *** [sound/soc/au1x] Error 2 **************************************************** before applying the patch the board was working but the audio was not working.. can you suggest me which kernel source you used and how you got the audio working for your board. can you send me the sample kernel image "vmlinux.srec" image for testing.. Thanks Manjunath.AM The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it.

3 6

Re: [alsa-devel] ALSA patch Nvidia 8 channel hdmi audio
by Takashi Iwai 08 Jun '09

08 Jun '09

At Thu, 4 Jun 2009 08:18:23 +0200, Alan Alan wrote: > > 2009/6/4 Takashi Iwai <tiwai(a)suse.de> > > At Thu, 4 Jun 2009 01:28:24 +0200, > Alan Alan wrote: > > > > Hello Takashi ! > > > > I tried to apply the patch that removes restrictions to 16bits 48khz > format. > > Well now we have the following: > > 16bit 48khz works > > 32bit 48khz works > > 32bit 44khz, 96khz, 192khz don't work > > 24bit 44khz, 48khz, 96khz, 192khz don't work > > 16bit 44khz, 96khz, 192khz don't work > > > > I only get > > Sample format not available for playback: > > Setting of hwparams failed: > > As you can find in alsa-info output, it seems that the codec supports > only 48kHz and 88.2kHz, strangely. I don't know whether it's a > (hardware) bug or not. > > The codec surely supports up to 192khz as it works on Windows. > > I think this line is completely wrong as here it says: It is not "wrong". These are values that the codec reports, so it's more or less correct as the h/w information. > rates [0xc0]: 48000 88200 Wei, is it right? Or is anything missing here? > bits [0xf]: 8 16 20 24 > > 8, 20, 24bits are supported, but they all give an error message. 24bit here means 24bit in 4 bytes, thus it's essentially 32bit format. 24bit in 3 bytes format isn't supported on HD-audio. > 32 bits are not supposed to be supported but it works with 48 and 88.2khz. > > How can we change the software side to accept 96 or 192khz spample formats and > 24 bits ? As long as the codec reports so, it's no "software" problem. > I think it is not hardware limited and we just need to change the allowed > formats to be played. If it's just a hardware problem that reports wrong values, you can override the information, simply replacing the bits my patch disabled to the right values. Takashi > > > > > I enclosed to pastebin my alsainfo script output. > > > > http://pastebin.com/m77dcfb3f > > > > Thanks for the help ! > > > > 2009/6/3 Takashi Iwai <tiwai(a)suse.de> > > > > At Wed, 3 Jun 2009 11:34:00 +0200, > > Alan Alan wrote: > > > > > > Hello Takashi ! > > > > > > What should this patch do ? > > > > The patch removes the restriction patch_nvhdmi.c sets. Instead, > > the driver tries to detect the supported format, rate, etc from the > > codec widget information. > > > > > Is it a cure for the 96, 192khz or 24bit sample playback problem ? > > > > Possibly. > > > > > If so, will it resample or natievly pass the stream to the > amplifier ? > > > > It just enables the native support, possibly. > > > > And, don't forget to give back the alsa-info.sh output (run with > > --no-upload option). > > > > Takashi > > > > > > > > Thank you > > > > > > Alan > > > > > > 2009/6/3 Takashi Iwai <tiwai(a)suse.de> > > > > > > At Wed, 3 Jun 2009 10:00:24 +0200, > > > Alan Alan wrote: > > > > > > > > Hello Wei ! > > > > > > > > Thanks for replying. > > > > I tested it with speaker-test and xbmc babylon. Both had > problems > > with > > > > channel mapping. > > > > I used this command: > > > > > > > > speaker-test -Dhdmi -c6 > > > > or > > > > speaker-test -Dhdmi -c8 > > > > > > > > alternatively you could try > > > > > > > > speaker-test -Dhw:0,3 -c6 > > > > > > > > > > > > I had to rund Xorg first and have the command from an xterm, > > becasue > > > from a > > > > normal terminal it would not work. > > > > I think that is normal as a limitation of hdmi. > > > > > > > > What about the support for 24bits 192khz sound ? > > > > > > > > speaker-test -Dhdmi -c6 -r192000 does not work. > > > > > > How about the patch below? > > > > > > Also, could you give alsa-info.sh output? > > > > > > Takashi > > > > > > --- > > > diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/ > > patch_nvhdmi.c > > > index f5792e2..48443de 100644 > > > --- a/sound/pci/hda/patch_nvhdmi.c > > > +++ b/sound/pci/hda/patch_nvhdmi.c > > > @@ -258,9 +258,11 @@ static struct hda_pcm_stream > > > nvhdmi_pcm_digital_playback_8ch = { > > > .channels_min = 2, > > > .channels_max = 8, > > > .nid = Nv_Master_Convert_nid, > > > +#if 0 > > > .rates = SNDRV_PCM_RATE_48000, > > > .maxbps = 16, > > > .formats = SNDRV_PCM_FMTBIT_S16_LE, > > > +#endif > > > .ops = { > > > .open = nvhdmi_dig_playback_pcm_open, > > > .close = nvhdmi_dig_playback_pcm_close_8ch, > > > @@ -273,9 +275,11 @@ static struct hda_pcm_stream > > > nvhdmi_pcm_digital_playback_2ch = { > > > .channels_min = 2, > > > .channels_max = 2, > > > .nid = Nv_Master_Convert_nid, > > > +#if 0 > > > .rates = SNDRV_PCM_RATE_48000, > > > .maxbps = 16, > > > .formats = SNDRV_PCM_FMTBIT_S16_LE, > > > +#endif > > > .ops = { > > > .open = nvhdmi_dig_playback_pcm_open, > > > .close = nvhdmi_dig_playback_pcm_close_2ch, > > > > > > > > > > > >

3 7

[alsa-devel] [PATCH v4 0/4] ARM: DaVinci: ASoC: Audio support for DM646x
by Chaithrika U S 07 Jun '09

07 Jun '09

From: Chaithrika U S <chaithrika(a)ti.com> Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch set adds audio support for DM646x This updated version has included most of the review comments received for the previous version of the patch set. Patch 1: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 2: Introduces platform driver model for dm644x and dm355 Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 3: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 4: Add machine driver support for DM646x Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> and Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> on Monstavista 2.6.18 kernel.

2 1

[alsa-devel] Testers wanted: New SB X-Fi driver
by Gernot Kohlhaas 07 Jun '09

07 Jun '09

Hi! I found a little strange thing: Recording from microphone is only possible, when you unmute the "Line-In" playback-device (and set microphone as capture device, of course). Oh, and by the way: Is there any possibility to apply a low pass filter to the subwoofer-channel without using ladspa (xmms and audacious won't play songs with my low pass filter. Mplayer, xine and aplay are working fine). Thanks again for this driver :) -- http://www.gnupp.de/verschluesselung/warum.html Key ID: 0x4C0B2250 Key: http://home.arcor.de/mr_maniac/ANDERES/public.gpg

1 0

[alsa-devel] [RFC/PATCH] asoc: pxa-ssp.c -> enchance the I2S enchancement :)
by Daniel Ribeiro 06 Jun '09

06 Jun '09

Hi, this patch is what i need to use Paul's pxa-ssp.c I2S enchancements on my board. Basically, it doesn't enforce the "fake" network mode, as i dont need it. And i need _real_ network mode, so i prefer to set SSCR0_MOD only when really using network mode. And it uses PXA_SSP_FRM_??FS to actually set SSCR0_DataSize() too. This allows me to use 32bit frames for 16 bit stereo audio. (No, I don't want to fake network mode for this, and SSCR0_MOD doesn't work as expected when PXA is slave). This patch is incremental to the recent patch from Paul Shen. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ed0906e..762cb93 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -579,7 +579,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: - sscr0 |= SSCR0_PSP | SSCR0_MOD; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; /* See hw_params() */ @@ -655,7 +655,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, int dma = 0, chn = params_channels(params); u32 sscr0; u32 sspsp; - int frame_width; + int frame_width = 0; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; @@ -698,7 +698,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; } - ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -742,6 +741,15 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } + if (frame_width) { + sscr0 &= ~(SSCR0_EDSS | SSCR0_DataSize(16)); + if (frame_width % 48 == 0) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(8); + else + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(16); + } + ssp_write_reg(ssp, SSCR0, sscr0); + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ -- Daniel Ribeiro

1 0

[alsa-devel] Testers wanted: New SB X-Fi driver
by MonsieurWeller 06 Jun '09

06 Jun '09

Hello. Works fine, but the external rack of the "x-fi fatality" didn't seems to work. I plug my headphone on it and i have no sound. Thanks.

2 1

[alsa-devel] [PATCH] ASoc: codec tlv320aic23 fix bogus divide by 0 message
by Troy Kisky 06 Jun '09

06 Jun '09

Some code analyzer software mistakenly gives divide by 0 error messages for these lines. This patch will end its confusion. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/codecs/tlv320aic23.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 9fcbb9c..0b8dcb5 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -273,14 +273,14 @@ static const unsigned short sr_valid_mask[] = { * Every divisor is a factor of 11*12 */ #define SR_MULT (11*12) -#define A(x) (x) ? (SR_MULT/x) : 0 +#define A(x) (SR_MULT/x) static const unsigned char sr_adc_mult_table[] = { - A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), - A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) + A(2), A(2), A(12), A(12), 0, 0, A(3), A(1), + A(2), A(2), A(11), A(11), 0, 0, 0, A(1) }; static const unsigned char sr_dac_mult_table[] = { - A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), - A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) + A(2), A(12), A(2), A(12), 0, 0, A(3), A(1), + A(2), A(11), A(2), A(11), 0, 0, 0, A(1) }; static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, -- 1.5.4.3

2 1

Re: [alsa-devel] [PATCH 1/4] ASoC: pxa-ssp: enhance I2S and add Left_J support
by Paul Shen 06 Jun '09

06 Jun '09

Hi Daniel, This is Paul Shen , I will substitute boshen9(a)gmail.com for bshen9(a)marvell.com to answer open source mails. > From: Daniel Mack <daniel(a)caiaq.de> > Date: Wed, Jun 3, 2009 at 9:18 PM > Subject: Re: [alsa-devel] [PATCH 1/4] ASoC: pxa-ssp: enhance I2S and add Left_J support > To: Eric Miao <eric.y.miao(a)gmail.com> > Cc: alsa-devel(a)alsa-project.org, linux-arm-kernel <linux-arm-kernel(a)lists.arm.linux.org.uk>, Mark Brown <broonie(a)sirena.org.uk> > > > On Wed, Jun 03, 2009 at 08:33:42PM +0800, Eric Miao wrote: > > Make the pxa I2S configuration generic, add support for Left_J, add > > support for variable frame width like 32fs, 48fs, 64fs and 96fs > > > > Signed-off-by: Paul Shen <bshen9(a)marvell.com> > > Signed-off-by: Eric Miao <eric.miao(a)marvell.com> > > Cc: Daniel Mack <daniel(a)caiaq.de> > > --- > > arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 +++--- > > sound/soc/pxa/pxa-ssp.c | 62 ++++++++++++++-------------- > > sound/soc/pxa/pxa-ssp.h | 9 ++++ > > 3 files changed, 47 insertions(+), 38 deletions(-) > > Ok, I tried that code on my board and this is what I had to change > there: > > The tdm time slot configuration needs to be set again in my board support > code just like in your example: snd_soc_set_tdm_slot(cpu_dai, 3, 2). And > the PXA_SSP_DIV_SCR value needed to be doubled from 4 to 8. > I tested with below codes to set the cpu_dai : format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_64FS; snd_soc_dai_set_fmt(cpu_dai, format); snd_soc_dai_set_tdm_slot(cpu_dai, 3, 2); snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 0); snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); It is no need to "snd_soc_dai_set_pll(cpu_dai, 0, 0, clk)" , for your case ,no ssp-on-chip pll configuration is needed. On my littleton platform,it is ok, and I did not modify the PXA_SSP_DIV_SCR from 4 to 8. I dumped the ssp register: SSCR0 0xa10003ff SSCR1 0x00e01d80 SSTO 0x00000000 SSPSP 0x31a00084 SSSR 0x0000f0fc SSACD 0x00000000 SSACDD 0x00000000 How about your registers ? > With that changes, LRCLK is 44100Hz when configured to 44100Hz. But the > bitclk is not 64fs anymore but 32fs only (1.41Mhz). Is there any > implementation details I miss? What does your codec clock config look > like? > Do you mean it only down to 32fs when LRCLK is 44100HZ ? And when LRCLK is 48000HZ, all is OK ? > Another small thing: > > CC sound/soc/pxa/pxa-ssp.o > sound/soc/pxa/pxa-ssp.c:186: warning: 'ssp_get_scr' defined but not used > > Daniel > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

3 2

[alsa-devel] [PATCH] Palm interrupt driven touchscreen driver
by Marek Vasut 05 Jun '09

05 Jun '09

Hi, this patch adds interrupt driven touchscreen driver. It's an addition to mainstone-wm97xx, but does some reorganization of the code there. Mark, it's the same patch we discussed on IRC, so it should be OK. Please consider applying, it's against Eric's tree. Thanks

7 17

[alsa-devel] [PATCH v4 4/4] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Chaithrika U S 05 Jun '09

05 Jun '09

This patch does the following: (1) Add support for the DM646x machine (2) Modifications required to introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-evm.c | 102 ++++++++++++++++++-------------------- 1 files changed, 48 insertions(+), 54 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 58fd1cb..f3bb6f6 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -27,9 +27,10 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" - +#include "davinci-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -43,7 +44,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -144,6 +145,24 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "McASP", + .stream_name = "spdif", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_DIT_DAI], + .codec_dai = &dit_stub_dai, + .ops = &evm_ops, + }, +}; + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -152,12 +171,26 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = ARRAY_SIZE(dm6467_evm_dai), +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -165,60 +198,30 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_ASP0_BASE, - .end = DAVINCI_ASP0_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP0_TX, - .rx_dma_ch = DAVINCI_DMA_ASP0_RX, -}; - -/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ -static struct resource dm335evm_snd_resources[] = { - { - .start = DAVINCI_ASP1_BASE, - .end = DAVINCI_ASP1_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data dm335evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP1_TX, - .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { - struct resource *resources; - unsigned num_resources; - struct evm_snd_platform_data *data; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; if (machine_is_davinci_evm()) { - davinci_cfg_reg(DM644X_MCBSP); - - resources = evm_snd_resources; - num_resources = ARRAY_SIZE(evm_snd_resources); - data = &evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { - /* we don't use ASP1 IRQs, or we'd need to mux them ... */ - davinci_cfg_reg(DM355_EVT8_ASP1_TX); - davinci_cfg_reg(DM355_EVT9_ASP1_RX); - - resources = dm335evm_snd_resources; - num_resources = ARRAY_SIZE(dm335evm_snd_resources); - data = &dm335evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; } else return -EINVAL; @@ -226,17 +229,8 @@ static int __init evm_init(void) if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, data, sizeof(*data)); - - ret = platform_device_add_resources(evm_snd_device, resources, - num_resources); - if (ret) { - platform_device_put(evm_snd_device); - return ret; - } - + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); -- 1.5.6

1 0

[alsa-devel] M-Audio Transit: Running asynchronous USB?
by Klaus Schulz 05 Jun '09

05 Jun '09

Hi there. I would like to get a confirmation on below. One of my USB soundcards is a M-Audio Transit. cat /proc/asound/card0/stream0 lists below: M-Audio Transit USB at usb-0000:00:1d.2-1.3, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 3 URBs = 2 [ 1 1 ] Packet Size = 196 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous) Interface 1 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (NONE) Rates: 8000 - 48000 (continuous) Interface 1 Altset 3 Format: 0x2 (16 bits) Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Do I interprete it right that the Transit runs in asynchronous USB mode up to 48khz and 16bit? Does above also mean that <48/24 is not supported? THX Klaus

1 0

[alsa-devel] How to send control codes to a USB device?
by Klaus Schulz 05 Jun '09

05 Jun '09

Hi there. Is there a way to sent control codes to a USB device, without using a specific driver . I would like to try to set latency and output modes of my M-Audio Transit by sending a control code as additional part of the UDEV madfuload firmware initialzation. How can I do something like that? If this works in general I just need to figure out the required control codes. THX for your support Klaus

1 0

[alsa-devel] [PATCH] Fix possibly incorrect offset in wm97xx-core
by Marek Vasut 05 Jun '09

05 Jun '09

Hi, I just hit a problem with the accelerated (interrupt driven) ts driver. It stop responding after a while (sometimes right after you release a stylus off the touchscreen, it doesn't generate any more interrupts). I made the following patch that fixes the problem for me (and if I read the 9712 and 9713 manuals correctly, the patch should be OK) Thanks for reply. diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c index 69af838..4f13a7f 100644 --- a/drivers/input/touchscreen/wm97xx-core.c +++ b/drivers/input/touchscreen/wm97xx-core.c @@ -307,11 +307,7 @@ static void wm97xx_pen_irq_worker(struct work_struct *work) WM97XX_GPIO_13); } - if (wm->id == WM9712_ID2) - wm97xx_reg_write(wm, AC97_GPIO_STATUS, (status & - ~WM97XX_GPIO_13) << 1); - else - wm97xx_reg_write(wm, AC97_GPIO_STATUS, status & + wm97xx_reg_write(wm, AC97_GPIO_STATUS, status & ~WM97XX_GPIO_13); mutex_unlock(&wm->codec_mutex); }

2 5

[alsa-devel] ASoC update for 2.6.31
by Mark Brown 05 Jun '09

05 Jun '09

The following changes since commit d08664fdb50795b29cf70b0269ea02f7248e76c3: Takashi Iwai (1): ASoC: Fix build error in twl4030.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Daniel Mack (1): ASoC: fix NULL pointer dereference in soc_suspend() sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-)

2 1

[alsa-devel] ALSA 1.0.20 Speex PCM Plugin
by Robert Krakora 04 Jun '09

04 Jun '09

Hello, Has anyone successfully employed the ALSA 1.0.20 Speex PCM Plugin? I followed the "speexdsp.txt" document under the 'doc' directory but the result was the following error: [root@vizioroom105 ~]# arecord -Dplug:mic poopy.wav Recording WAVE 'poopy.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono ALSA lib pcm_params.c:2135:(snd1_pcm_hw_refine_slave) Slave PCM not usable arecord: set_params:957: Broken configuration for this PCM: no configurations available Here is my asound.conf: # MessageNet Systems ALSA sound configuration file # Allowing multiple playback and capture pcm.!default { type plug slave.pcm "duplex" } pcm.duplex { type asym playback.pcm "dmixer" capture.pcm "analog_input" } pcm.dmixer { type dmix ipc_key 1024 ipc_perm 0666 slave { pcm "hw:2,3" period_time 0 period_size 1024 buffer_size 8192 } bindings { 0 0 1 1 } } pcm.mic { type speex slave.pcm { type hw card 1 } agc 1 agc_level 8000 } pcm.analog_input { type dsnoop ipc_key 2048 ipc_perm 0666 slave{ pcm "hw:1,2" } } pcm.dsp0 { type plug slave.pcm "dmixer" } ctl.mixer0 { type hw card 0 } Here are the results of alsa-info.sh: http://www.alsa-project.org/db/?f=358e4d5d828dda3c44220382c8b480fd6019ab2a Thanks in advance. Best Regards, -- Rob Krakora Senior Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

2 1

[alsa-devel] [PATCH 3/4] ASoC: pxa add Marvell Littleton platform audio support
by Eric Miao 04 Jun '09

04 Jun '09

Signed-off-by: Paul Shen <bshen9(a)marvell.com> Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/littleton.c | 228 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/littleton.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ad8a10f..a4ad5ba 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -134,3 +134,12 @@ config SND_PXA2XX_SOC_MIOA701 help Say Y if you want to add support for SoC audio on the MIO A701. + +config SND_PXA2XX_SOC_LITTLETON + tristate "SoC Audio support for Marvell littleton" + depends on SND_PXA2XX_SOC && MACH_LITTLETON + select SND_PXA_SOC_SSP + select SND_SOC_DA9034 + help + Say Y if you want to add support for SoC audio on the + Marvell Littleton. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 4b90c3c..94c719d 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o +snd-soc-littleton-objs := littleton.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -34,4 +35,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o +obj-$(CONFIG_SND_PXA2XX_SOC_LITTLETON) += snd-soc-littleton.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/littleton.c b/sound/soc/pxa/littleton.c new file mode 100644 index 0000000..911a5ef --- /dev/null +++ b/sound/soc/pxa/littleton.c @@ -0,0 +1,228 @@ +/* + * littleton.c -- SoC audio for Littleton + * + * Copyright (C) 2009 Marvell International Ltd. + * + * Author: Paul Shen <bshen9(a)marvell.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation; either version 2 of the + * License, or (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> +#include <mach/ssp.h> +#include <mach/regs-ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +static int hifi_mode; + +extern struct snd_soc_dai da9034_dais[2]; +extern struct snd_soc_codec_device soc_codec_dev_da9034; + +static const char *hifi_mode_select[] = { + "HIFI 64FS LEFT_J, SSP CLK Master FRM Master", + "HIFI 64FS I2S , SSP CLK Master FRM Master", + "HIFI 32FS LEFT_J, SSP CLK Master FRM Master", + "HIFI 32FS I2S , SSP CLK Master FRM Master", +}; + +static const struct soc_enum littleton_enum[] = { + SOC_ENUM_SINGLE_EXT(4, hifi_mode_select), +}; + +static int littleton_get_hifi_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = hifi_mode; + return 0; +} + +static int littleton_set_hifi_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + hifi_mode = ucontrol->value.integer.value[0]; + return 1; +} + +static const struct snd_kcontrol_new littleton_da9034_controls[] = { + SOC_ENUM_EXT("Hifi Mode Select", littleton_enum[0], + littleton_get_hifi_mode, littleton_set_hifi_mode), +}; + +static int littleton_da9034_init(struct snd_soc_codec *codec) +{ + snd_soc_add_controls(codec, littleton_da9034_controls, + ARRAY_SIZE(littleton_da9034_controls)); + return snd_soc_dapm_sync(codec); +} + +struct ssp_clk_div { + unsigned int rate; + unsigned int pll; + unsigned int scdb; + unsigned int acds; +}; + +static struct ssp_clk_div littleton_hifi_clk_div_64fs[] = { + { 48000, 12288000, PXA_SSP_CLK_SCDB_4, 0 }, + { 44100, 11289600, PXA_SSP_CLK_SCDB_4, 0 }, + { 32000, 4096000, PXA_SSP_CLK_SCDB_1, 1 }, + { 22050, 11289600, PXA_SSP_CLK_SCDB_8, 0 }, + { 16000, 4096000, PXA_SSP_CLK_SCDB_4, 0 }, + { 11025, 11289600, PXA_SSP_CLK_SCDB_4, 2 }, + { 8000, 4096000, PXA_SSP_CLK_SCDB_4, 1 }, + {}, +}; + +static struct ssp_clk_div littleton_hifi_clk_div_32fs[] = { + { 48000, 12288000, PXA_SSP_CLK_SCDB_4, 1 }, + { 44100, 11289600, PXA_SSP_CLK_SCDB_4, 1 }, + { 32000, 4096000, PXA_SSP_CLK_SCDB_1, 2 }, + { 22050, 11289600, PXA_SSP_CLK_SCDB_8, 1 }, + { 16000, 4096000, PXA_SSP_CLK_SCDB_4, 1 }, + { 11025, 11289600, PXA_SSP_CLK_SCDB_8, 2 }, + { 8000, 4096000, PXA_SSP_CLK_SCDB_8, 1 }, + {}, +}; + +static int littleton_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_clk_div *clkdiv; + struct clk *clk_pout; + int format; + + clk_pout = clk_get(NULL, "CLK_POUT"); + if (IS_ERR(clk_pout)) { + pr_err("failed to get POUT clock\n"); + return PTR_ERR(clk_pout); + } + + clk_enable(clk_pout); + + switch (hifi_mode) { + case 0: + format = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_64FS; + clkdiv = littleton_hifi_clk_div_64fs; + break; + case 1: + format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_64FS; + clkdiv = littleton_hifi_clk_div_64fs; + break; + case 2: + format = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_32FS; + clkdiv = littleton_hifi_clk_div_32fs; + break; + case 3: + format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_32FS; + clkdiv = littleton_hifi_clk_div_32fs; + break; + default: + return -EINVAL; + } + + snd_soc_dai_set_fmt(codec_dai, format); + snd_soc_dai_set_fmt(cpu_dai, format); + snd_soc_dai_set_tdm_slot(cpu_dai, 3, 2); + + for (; clkdiv->rate; clkdiv++) { + if (clkdiv->rate == params_rate(params)) + break; + } + + if (!clkdiv->rate) + return -EINVAL; + + snd_soc_dai_set_pll(cpu_dai, 0, 0, clkdiv->pll); + snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); + snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_SCDB, clkdiv->scdb); + snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, clkdiv->acds); + return 0; +} + +static struct snd_soc_ops littleton_hifi_ops = { + .hw_params = littleton_hifi_hw_params, +}; + +static struct snd_soc_dai_link littleton_dai[] = { + /* HiFi */ + { + .name = "DA9034 HiFi", + .stream_name = "DA9034 HiFi", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3], + .codec_dai = &da9034_dais[0], + .init = littleton_da9034_init, + .ops = &littleton_hifi_ops, + }, +}; + +static struct snd_soc_card littleton = { + .name = "Littleton", + .platform = &pxa2xx_soc_platform, + .dai_link = littleton_dai, + .num_links = ARRAY_SIZE(littleton_dai), +}; + +static struct snd_soc_device littleton_snd_devdata = { + .card = &littleton, + .codec_dev = &soc_codec_dev_da9034, +}; + +static struct platform_device *littleton_snd_device; + +static int __init littleton_audio_init(void) +{ + int ret; + + if (!machine_is_littleton()) { + pr_err("Only Littleton hardware supported by ASoC driver\n"); + return -ENODEV; + } + + littleton_snd_device = platform_device_alloc("soc-audio", -1); + if (!littleton_snd_device) + return -ENOMEM; + + platform_set_drvdata(littleton_snd_device, &littleton_snd_devdata); + littleton_snd_devdata.dev = &littleton_snd_device->dev; + + ret = platform_device_add(littleton_snd_device); + if (ret != 0) + platform_device_put(littleton_snd_device); + + return ret; +} + +static void __exit littleton_audio_exit(void) +{ + platform_device_unregister(littleton_snd_device); +} + +module_init(littleton_audio_init); +module_exit(littleton_audio_exit); + +MODULE_AUTHOR("Paul Shen <bshen9(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC Da9034 Littleton"); +MODULE_LICENSE("GPL"); -- 1.6.0.4

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[alsa-devel] [PATCH] ALSA: ASoC: fix NULL pointer dereference in soc_suspend()
by Daniel Mack 04 Jun '09

04 Jun '09

In case the initalization of an soc_device failed, there is no codec associated with it. soc_suspend() will still dereference the pointer and cause an Ooops when entering the sleep mode. This happens on our board with a multi-target kernel image when booted on a machine without audio circuits. This patch makes the code bail out very early in this special case. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1cd149b..4545a15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -632,6 +632,12 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = card->codec; int i; + /* If the initialization of this soc device failed, there is no codec + * associated with it. Just bail out in this case. + */ + if (!codec) + return 0; + /* Due to the resume being scheduled into a workqueue we could * suspend before that's finished - wait for it to complete. */ -- 1.6.3.1

2 1

[alsa-devel] [PATCH] hda_intel: fix build error when !PM
by Alexander Beregalov 04 Jun '09

04 Jun '09

Fix this build error when CONFIG_PM is not set: ound/pci/hda/hda_intel.c: In function 'azx_bus_reset': sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all' sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend' sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume' Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/hda/hda_intel.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f44645..4e9ea70 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1260,17 +1260,20 @@ static void azx_stop_chip(struct azx *chip); static void azx_bus_reset(struct hda_bus *bus) { struct azx *chip = bus->private_data; - int i; bus->in_reset = 1; azx_stop_chip(chip); azx_init_chip(chip); +#ifdef CONFIG_PM if (chip->initialized) { + int i; + for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus); snd_hda_resume(chip->bus); } +#endif bus->in_reset = 0; } -- 1.6.3.1

2 1

[alsa-devel] Wonder if VT1708A has Mic Boost or not
by Konstantin Kletschke 04 Jun '09

04 Jun '09

Hi Folks! I have a mini-PC here which should record Voice over Microphone. The Level is not really loud enough and the alsa-driver-1.0.19 doesn't reveal a mic boost setting in its mixer. This of course may be of the driver not yet being adapted to the special motherboard AFAIK. What I wonder is, is the Mic Boost setting/function integrated in this chip itself or does hardware vendor need to implement external opamps for this? If it is integrated into the chip itself I could try to modifiy the code to make the setting public. Is there a source for a datasheet anywhere for the VT1708A? King regards, Konsti -- GPG KeyID EF62FCEF Fingerprint: 13C9 B16B 9844 EC15 CC2E A080 1E69 3FDA EF62 FCEF

2 2

[alsa-devel] [PATCH v3 5/6] ARM: DaVinci: ASoC: Add mcasp support for DM646x
by Chaithrika U S 04 Jun '09

04 Jun '09

Adds driver support for the two instances of McASP on TI's DM646x. The multichannel audio serial port (McASP) functions as a general-purpose audio serial port optimized for the needs of multichannel audio application. (http://www.ti.com/litv/pdf/spruer1b). There are two instances of McASP on DM646x. The McASP0 module includes up to 4 serializers that can be individually enabled to either transmit or receive in different modes. The McASP1 module is limited with only 1 pinned-out serializer that can be enabled to only transmit in DIT mode (neither receiving in any mode nor transmitting in either Burst or TDM mode is supported). The codec on the DM646x EVM is AIC32 which is connected to Serializer0 and Serializer1 of McASP0. The McASP0 consists of transmit and receive sections that may operate synchronized, or completely independently with separate master clocks, bit clocks, and frame syncs, and using different transmit modes with different bit-stream formats. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/Kconfig | 13 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-i2s-mcasp.c | 908 +++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s-mcasp.h | 58 +++ sound/soc/davinci/davinci-pcm.h | 17 +- 5 files changed, 991 insertions(+), 8 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.c create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 411a710..7d10a8c 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_I2S_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" depends on SND_DAVINCI_SOC @@ -19,6 +22,16 @@ config SND_DAVINCI_SOC_EVM Say Y if you want to add support for SoC audio on TI DaVinci DM6446 or DM355 EVM platforms. +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM + select SND_DAVINCI_SOC_I2S_MCASP + select SND_SOC_TLV320AIC3X + select SND_SOC_SPDIF + + help + Say Y if you want to add support for SoC audio on TI + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1..152d59c 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,16 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-i2s-mcasp-objs:= davinci-i2s-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S_MCASP) += snd-soc-davinci-i2s-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-i2s-mcasp.c b/sound/soc/davinci/davinci-i2s-mcasp.c new file mode 100644 index 0000000..6ea75eb --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.c @@ -0,0 +1,908 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s-mcasp.h" + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left (even TDM Slot) Channel Status Register File*/ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right (odd TDM slot) Channel Status RegisterFile*/ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left (even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right (odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define MCASP_FREE BIT(0) +#define MCASP_SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define DISMOD (val)(val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + int i = 0; + + mcasp_set_bits(regs, val); + + /* programming GBLCTL needs to read back from GBLCTL and verfiy */ + /* loop count is to avoid the lock-up */ + for (i = 0; i < 1000; i++) { + if ((mcasp_get_reg(regs) & val) == val) + break; + } + + if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + printk(KERN_ERR "GBLCTL write error\n"); +} + +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +static void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +static void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +static void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_i2s_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct snd_pcm_substream *substream, + int channel_size) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F; + break; + + default: + return -1; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + return 0; +} + +static void davinci_hw_common_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} + +static void davinci_hw_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i, active_slots; + u32 mask = 0; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + } else { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_i2s_mcasp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(substream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(substream); + else + davinci_hw_param(substream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(substream, word_length); + + return 0; +} + +static int davinci_i2s_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(substream); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_iis_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &davinci_i2s_dai_ops, + + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &davinci_i2s_dai_ops, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_iis_mcasp_dai); + +static int davinci_i2s_mcasp_probe(struct platform_device *pdev) +{ + struct evm_snd_platform_data *parray = pdev->dev.platform_data; + struct davinci_pcm_dma_params *dma_data; + struct resource *mem, *ioarea, *res; + struct evm_snd_platform_data *pdata; + int tmp, link_cnt, num_links = 2; + struct davinci_audio_dev *dev; + int count = 0; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev) * num_links, + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * + (num_links << 1), GFP_KERNEL); + + if (!dma_data) { + ret = -ENOMEM; + goto err_release_dev; + } + + for (link_cnt = 0; link_cnt < num_links; link_cnt++) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, link_cnt); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_data; + } + + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + goto err_release_data; + } + + pdata = &parray[link_cnt]; + dev[link_cnt].clk = clk_get(&pdev->dev, pdata->clk_name); + + if (IS_ERR(dev[link_cnt].clk)) { + ret = -ENODEV; + goto err_release_region; + } + + clk_enable(dev[link_cnt].clk); + + dev[link_cnt].base = (void __iomem *)IO_ADDRESS(mem->start); + dev[link_cnt].op_mode = pdata->op_mode; + dev[link_cnt].tdm_slots = pdata->tdm_slots; + dev[link_cnt].num_serializer = pdata->num_serializer; + dev[link_cnt].serial_dir = pdata->serial_dir; + dev[link_cnt].codec_fmt = pdata->codec_fmt; + + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_PLAYBACK] = + &dma_data[count]; + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, + 2 * link_cnt + 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_CAPTURE] = + &dma_data[count]; + + res = platform_get_resource(pdev, IORESOURCE_DMA, + 2 * link_cnt + 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + count++; + davinci_iis_mcasp_dai[link_cnt].private_data = &dev[link_cnt]; + } + ret = snd_soc_register_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + if (ret != 0) + goto err_release_region; + return 0; + +err_release_region: + for (tmp = link_cnt; tmp >= 0; tmp--) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, tmp); + release_mem_region(mem->start, + (mem->end - mem->start) + 1); + } +err_release_data: + kfree(dma_data); +err_release_dev: + kfree(dev); + + return ret; +} + +static int davinci_i2s_mcasp_remove(struct platform_device *pdev) +{ + int i, num_links = 2; + struct davinci_audio_dev *dev; + struct resource *mem; + struct davinci_pcm_dma_params *dma_data; + + snd_soc_unregister_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + + for (i = 0; i < num_links; i++) { + dev = davinci_iis_mcasp_dai[i].private_data; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + } + + dev = davinci_iis_mcasp_dai[0].private_data; + dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(dma_data); + kfree(dev); + return 0; +} + +static struct platform_driver davinci_mcasp_driver = { + .probe = davinci_i2s_mcasp_probe, + .remove = davinci_i2s_mcasp_remove, + .driver = { + .name = "davinci-mcasp", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_i2s_mcasp_init(void) +{ + return platform_driver_register(&davinci_mcasp_driver); +} +module_init(davinci_i2s_mcasp_init); + +static void __exit davinci_i2s_mcasp_exit(void) +{ + platform_driver_unregister(&davinci_mcasp_driver); +} +module_exit(davinci_i2s_mcasp_exit); + + + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McASP) SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/davinci/davinci-i2s-mcasp.h b/sound/soc/davinci/davinci-i2s-mcasp.h new file mode 100644 index 0000000..f881e54 --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.h @@ -0,0 +1,58 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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 DAVINCI_I2S_MCASP_H +#define DAVINCI_I2S_MCASP_H + +#include <linux/io.h> +#include <mach/asp.h> +#include "davinci-pcm.h" + +extern struct snd_soc_dai davinci_iis_mcasp_dai[]; +extern struct snd_soc_dai davinci_dit_mcasp_dai; + + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 +#define DAVINCI_MCASP_I2S_DAI 0 +#define DAVINCI_MCASP_DIT_DAI 1 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + + + +#endif /* DAVINCI_I2S_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..3cbe281 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,17 +12,18 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> +#include <mach/asp.h> + + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ + enum dma_event_q eventq_no; /* event queue number */ }; -struct evm_snd_platform_data { - int tx_dma_ch; - int rx_dma_ch; -}; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

3 5

[alsa-devel] [PATCH] ALSA: Add missing __devexit_p() markers
by Jean Delvare 04 Jun '09

04 Jun '09

3 ISA sound drivers lack their __devexit_p() markers, which would cause build failures when the kernel is built without hotplug support. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Kyle McMartin <kyle(a)mcmartin.ca> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/es1688/es1688.c | 2 +- sound/isa/gus/gusextreme.c | 2 +- sound/parisc/harmony.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) --- linux-2.6.30-rc8.orig/sound/isa/es1688/es1688.c 2009-04-08 08:54:02.000000000 +0200 +++ linux-2.6.30-rc8/sound/isa/es1688/es1688.c 2009-06-04 10:37:06.000000000 +0200 @@ -193,7 +193,7 @@ static int __devexit snd_es1688_remove(s static struct isa_driver snd_es1688_driver = { .match = snd_es1688_match, .probe = snd_es1688_probe, - .remove = snd_es1688_remove, + .remove = __devexit_p(snd_es1688_remove), #if 0 /* FIXME */ .suspend = snd_es1688_suspend, .resume = snd_es1688_resume, --- linux-2.6.30-rc8.orig/sound/isa/gus/gusextreme.c 2009-04-08 08:54:02.000000000 +0200 +++ linux-2.6.30-rc8/sound/isa/gus/gusextreme.c 2009-06-04 10:37:13.000000000 +0200 @@ -348,7 +348,7 @@ static int __devexit snd_gusextreme_remo static struct isa_driver snd_gusextreme_driver = { .match = snd_gusextreme_match, .probe = snd_gusextreme_probe, - .remove = snd_gusextreme_remove, + .remove = __devexit_p(snd_gusextreme_remove), #if 0 /* FIXME */ .suspend = snd_gusextreme_suspend, .resume = snd_gusextreme_resume, --- linux-2.6.30-rc8.orig/sound/parisc/harmony.c 2009-04-08 08:54:03.000000000 +0200 +++ linux-2.6.30-rc8/sound/parisc/harmony.c 2009-06-04 10:37:25.000000000 +0200 @@ -1020,7 +1020,7 @@ static struct parisc_driver snd_harmony_ .name = "harmony", .id_table = snd_harmony_devtable, .probe = snd_harmony_probe, - .remove = snd_harmony_remove, + .remove = __devexit_p(snd_harmony_remove), }; static int __init -- Jean Delvare

2 1

[alsa-devel] snd-ctxfi
by The Source 04 Jun '09

04 Jun '09

Hello. I tested the snd-ctxfi driver and found no problems (except no support for 5.1 and external console, but original ctxfi does not support them either). A minor problem: kernel oops reporter periodically pops up with message about using a timer in a wrong place. I'll post here this message when I get it next time.

3 7

[alsa-devel] [HELP] snd-aloop with dmix?
by Santo Chow 04 Jun '09

04 Jun '09

HI all, I'm trying to use snd-aloop as my default soundard. I have the .asoundrc that look like this. pcm.sndcard{ type hw card 1 #snd-aloop soundcard device 0 } pcm.dmixer { type dmix ipc_key 1025 ipc_perm 0660 slave { pcm sndcard period_time 0 period_size 881 buffer_size 3524 rate 44100 } bindings { 0 0 1 1 } } pcm.dsnooper { type dsnoop ipc_key 1026 ipc_perm 0660 slave { pcm sndcard period_time 0 period_size 881 buffer_size 3524 rate 44100 } bindings { 0 0 1 1 } } pcm.duplexasym { type asym playback.pcm "dmixer" # playback capture.pcm "dsnooper" # capture } pcm.!default { type plug slave { pcm duplexasym } } ctl.duplex { type hw card 1 } Right now my problem is, i can only get 1 sound from an application at a time. I can't run several applications at the same time and recorded all the sounds. This is what I did : My default sound card is indexed at 0, so the snd-aloop will be at index 1. To test this, I first run an MP3 player. Then I run this in terminal: arecord -D hw:1,1 -f cd | aplay -D hw:0 -f cd The sound come out good. Then when I open a browser, go to youtube and play a song over there, the sound won't come out. I have to first close the browser and the currently playing MP3 player, then re-open a browser, go to youtube, and play a song, the sound will come out. So, my question is.. is there something missing in my .asoundrc file? can the snd-aloop support multiple playbacks and captures at the same time? is there a way for me to achieve this?

1 0

[alsa-devel] Requesting opinions and experiences concerning laptops and external microphones
by komputes 04 Jun '09

04 Jun '09

I am trying to gather opinions and experiences from the ALSA developer community to understand what they expect to happen when they plug in an analog microphone into their laptops (which already has a built in mic). I am looking for more of an official ALSA document which states how capture should change when an analogue microphone is plugged in; should it takeover from the internal microphone or not? Is there specification, standard, or "correct" behavior which is applied to this process? If I am not mistaking, the computer can detect a signal/message when the jack goes in, right? The expected behavior when plugging in headphones is that the headphones are detected, the speakers are muted and the audio is channeled to the headphones. Is this not applicable to external microphone and the internal microphone as well? -komputes

2 1

Re: [alsa-devel] ALSA patch Nvidia 8 channel hdmi audio
by Marek Vasut 04 Jun '09

04 Jun '09

On Wednesday 03 of June 2009 00:30:04 Alan Alan wrote: > Thank's for the info. > How can i contact him ? He and alsa-devel mailing list are both in CC already, please preserve them there or fix your mailer. > > > 2009/6/3 Marek Vasut <marek.vasut(a)gmail.com> > > > I'm not the author of the patch, CCing alsa-devel and the original > > author. > > > > On Tuesday 02 of June 2009 23:58:26 Alan Alan wrote: > > > Hi Marek ! > > > > > > I just read your emailing with nvidia devs about alsa supporting 8 > > > > channel > > > > > lpcm over hdmi. > > > > > > I checked it out an i have to tell it is working awsome, except for a > > > few bugs: > > > > > > The channels are wrong mapped. Running a speakertest -Dhdmi -c6 gives > > > the following result (first where it should sound) > > > FL - FL > > > C - RL > > > FR - FR > > > RR - LFE > > > RL - C > > > > > > The other thing i realized that playback is limited to 16 bits 48 khz. > > > I think the minimum which is supported by nvidia hardware is 24bits 96 > > > (or 192) khz. > > > Why not ? > > > > > > Could you contact the devs as i don't know how to reach them. > > > > > > ps.: Mybe my build methot was not 100% ok ? I used 1.020 Alsa with the > > > oatched nvidia.c file. Anyway the amplifier shows multichannel LPCM and > > > that's great news.

6 17

[alsa-devel] [PATCH] ALSA: Fix double locking of card list in snd_card_register()
by Mark Brown 04 Jun '09

04 Jun '09

The introduction of snd_card_set_id() added a lock on the card list to the old choose_default_id() function when using it to implement the new API call. This lock is needed to allow us to walk the list and check to see if our new name is a duplicate. Unfortunately this causes a lockup when called from snd_card_register() (in cases where no ID is supplied for the card) since the card list is already locked there. Fix this fairly hideously by factoring out the implementation and using a flag to indicate if the lock should be held. A better fix would probably be to refactor snd_card_register() to move the _set_id() outside the locking region but I can't immediately see anything I can convince myself is safe. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This patch isn't terribly nice but I figured it was a more useful report than just the deadlock. sound/core/init.c | 31 +++++++++++++++++++------------ 1 files changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 6557dd8..a578d05 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -476,15 +476,8 @@ int snd_card_free(struct snd_card *card) EXPORT_SYMBOL(snd_card_free); -/** - * snd_card_set_id - set card identification name - * @card: soundcard structure - * @nid: new identification string - * - * This function sets the card identification and checks for name - * collisions. - */ -void snd_card_set_id(struct snd_card *card, const char *nid) +static void snd_card_set_id_internal(struct snd_card *card, const char *nid, + int do_locking) { int i, len, idx_flag = 0, loops = SNDRV_CARDS; const char *spos, *src; @@ -529,14 +522,16 @@ void snd_card_set_id(struct snd_card *card, const char *nid) } if (!snd_info_check_reserved_words(id)) goto __change; - mutex_lock(&snd_card_mutex); + if (do_locking) + mutex_lock(&snd_card_mutex); for (i = 0; i < snd_ecards_limit; i++) { if (snd_cards[i] && !strcmp(snd_cards[i]->id, id)) { mutex_unlock(&snd_card_mutex); goto __change; } } - mutex_unlock(&snd_card_mutex); + if (do_locking) + mutex_unlock(&snd_card_mutex); break; __change: @@ -561,6 +556,18 @@ void snd_card_set_id(struct snd_card *card, const char *nid) } } +/** + * snd_card_set_id - set card identification name + * @card: soundcard structure + * @nid: new identification string + * + * This function sets the card identification and checks for name + * collisions. + */ +void snd_card_set_id(struct snd_card *card, const char *nid) +{ + snd_card_set_id_internal(card, nid, 1); +} EXPORT_SYMBOL(snd_card_set_id); #ifndef CONFIG_SYSFS_DEPRECATED @@ -657,7 +664,7 @@ int snd_card_register(struct snd_card *card) return 0; } if (card->id[0] == '\0') - snd_card_set_id(card, NULL); + snd_card_set_id_internal(card, NULL, 0); snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -- 1.6.3.1

3 3

[alsa-devel] [PATCH] More Aspire 8930G fixes
by Hector Martin 04 Jun '09

04 Jun '09

Enable all three capture channels, including the missing nid 7 which is the only one capable of capturing DMIC input Enable Headphone amp for the HP jack. This causes a volume boost for headphones, but does not cause any noticeable effect for light loads like other amps, so there is no need to make it configurable. Add Input Mix capture mux setting to capture the output of the playback input mux (that is, what goes out the speakers except for PCM) Hack another coef register because the stereo DMIC for some reason produces a nonstandard sum/difference signal. I found a bit to make it just use the sum signal for both channels, which makes it behave like a standard mono microphone. The stereo is useless anyway (they're 1cm apart). Tested working: Three capture channels, mic in, line in, DMIC. Tested not working: CD. Not sure why, might be unconnected in the actual hardware or a CD drive issue. Also looked at SPDIF. It appears to work (emitter lights up inside the HP out jack) but I lack a proper miniTOSLINK cable to test it. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- alsa-driver-1.0.20.10.g955ee.233.gf65b3/sound/pci/hda/patch_realtek.c.orig 2009-06-03 23:17:55.000000000 +0200 +++ alsa-driver-1.0.20.10.g955ee.233.gf65b3/sound/pci/hda/patch_realtek.c 2009-06-03 23:45:40.000000000 +0200 @@ -1469,10 +1469,10 @@ }; /* - * ALC888 Acer Aspire 8930G model + * ALC889 Acer Aspire 8930G model */ -static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +static struct hda_verb alc889_acer_aspire_8930g_verbs[] = { /* Front Mic: set to PIN_IN (empty by default) */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Unselect Front Mic by default in input mixer 3 */ @@ -1492,7 +1492,7 @@ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect HP out to Front */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Enable all DACs */ @@ -1507,6 +1507,17 @@ /* Enable amplifiers */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, +/* DMIC fix + * This laptop has a stereo digital microphone. The mics are only 1cm apart + * which makes the stereo useless. However, either the mic or the ALC889 + * makes the signal become a difference/sum signal instead of standard + * stereo, which is annoying. So instead we flip this bit which makes the + * codec replicate the sum signal to both channels, turning it into a + * normal mono mic. + */ +/* DMIC_CONTROL? Init value = 0x0001 */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x0b}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0003}, { } }; @@ -1531,6 +1542,38 @@ } }; +static struct hda_input_mux alc889_capture_sources[3] = { + /* Digital mic only available on first "ADC" */ + { + .num_items = 5, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Front Mic", 0xb }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + } +}; + static struct snd_kcontrol_new alc888_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -1562,7 +1605,7 @@ alc_automute_amp(codec); } -static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9081,22 +9124,22 @@ .mixers = { alc888_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc888_acer_aspire_8930g_verbs }, + alc889_acer_aspire_8930g_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), - .adc_nids = alc883_adc_nids_rev, - .capsrc_nids = alc883_capsrc_nids_rev, + .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), + .adc_nids = alc889_adc_nids, + .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, .const_channel_count = 6, .num_mux_defs = - ARRAY_SIZE(alc888_2_capture_sources), - .input_mux = alc888_2_capture_sources, + ARRAY_SIZE(alc889_capture_sources), + .input_mux = alc889_capture_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_acer_aspire_8930g_init_hook, + .init_hook = alc889_acer_aspire_8930g_init_hook, }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer,

2 1

[alsa-devel] [PATCH] burgundy: timeout message is off by one.
by Roel Kluin 04 Jun '09

04 Jun '09

Timeout message is off by one. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 45a7629..d2c0a4e 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -46,12 +46,12 @@ snd_pmac_burgundy_extend_wait(struct snd_pmac *chip) timeout = 50; while (!(in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_extend_wait: timeout #1\n"); timeout = 50; while ((in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_extend_wait: timeout #2\n"); }

2 1

[alsa-devel] [PATCH - quirk for USB Aureon cards 1/1] Add quirk to provide proper naming of the Terratec Aureon 5.1 MkII USB card
by Andrea Borgia 03 Jun '09

03 Jun '09

Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 5d955aa..d84d6f3 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1951,6 +1951,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0028), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "TerraTec", + .product_name = "Aureon 5.1 MkII", + .ifnum = QUIRK_NO_INTERFACE + } +}, +{ USB_DEVICE(0x0ccd, 0x0035), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Miditech", -- 1.6.0.4

2 8

[alsa-devel] Sequencer timestamps for soft synth app
by Glyn Adgie 03 Jun '09

03 Jun '09

I have written a soft synth, which uses ALSA to get MIDI events, and Jack to do audio output. MIDI events are fetched and processed at the start of the Jack audio callback, then the Jack output buffers are filled by the synth engine. The problem I have is that, if the Jack buffer size is too large, the time resolution for MIDI events is too coarse, and this has a severe effect on rhythms. I currently get around this by setting up Jack for a buffer size of 64 frames, and a sample rate of 48 kHz, giving a timing resolution of 1.33 msec, which appears to be adequate. However, I might not always be able to get the desired buffer size. If the Jack buffer is some multiple of 64 frames, the synth renders multiple 64 frame chunks until the buffer is filled. This takes care of audio quality. However, MIDI timing resolution suffers. I am looking at using the event's timestamp, to decide when to process an event. For example, I might have a Jack buffer of 256 frames, so I want to work out whether to process an event right away, or after one 64 frame chunk, and so on. It would appear that, in order to make this work, I need an initial timestamp, which I get at the start of the audio callback. I then compare each event's timestamp, to determine where it should be processed. I cannot find an ALSA function for getting a timestamp. Perhaps I am missing something, or trying to do the job in the wrong way. Regards Glyn

2 1

[alsa-devel] Xonar DX spdif advice request
by Dmitri Seletski 03 Jun '09

03 Jun '09

Hello guys. I have perfectly working asus xonar card. it works fine with ordinary outputs, but I would like to know, did anyone try those cards with spdif? Is it worth buying something spdif compatible, if you intend to use this card? Thank you in advance. Dmitri

1 0

[alsa-devel] [PATCH 2/4] ASoC: add dialog da9034 (aka micco) audio codec support
by Eric Miao 03 Jun '09

03 Jun '09

Signed-off-by: Paul Shen <bshen9(a)marvell.com> Signed-off-by: Bin Yang <bin.yang(a)marvell.com> Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9034-audio.c | 696 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 702 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/da9034-audio.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 121d63f..d779f40 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS + select SND_SOC_DA9034 if PMIC_DA903X help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -160,3 +161,6 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + +config SND_SOC_DA9034 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d8e15a4..53bb3d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,7 @@ snd-soc-wm8990-objs := wm8990.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o +snd-soc-da9034-objs := da9034-audio.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o @@ -64,3 +65,4 @@ obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o +obj-$(CONFIG_SND_SOC_DA9034) += snd-soc-da9034.o diff --git a/sound/soc/codecs/da9034-audio.c b/sound/soc/codecs/da9034-audio.c new file mode 100644 index 0000000..159a883 --- /dev/null +++ b/sound/soc/codecs/da9034-audio.c @@ -0,0 +1,696 @@ +/* + * linux/sound/soc/codecs/micco.c + * + * Copyright (C) 2007-2009 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/mfd/da903x.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#define DA9034_MUX_MONO 0x70 +#define DA9034_MUX_BEAR 0x71 +#define DA9034_MUX_LINE 0x72 +#define DA9034_MUX_STEREO_CH1 0x73 +#define DA9034_MUX_STEREO_CH2 0x74 +#define DA9034_AUDIO_LINE_AMP 0x75 +#define DA9034_STEREO_AMP_CH1 0x76 +#define DA9034_STEREO_AMP_CH2 0x77 +#define DA9034_STEREO_DAC 0x78 +#define DA9034_MONO_VOL 0x79 +#define DA9034_BEAR_VOL 0x7a +#define DA9034_I2S_CONTROL 0x7b +#define DA9034_TX_PGA 0x7c +#define DA9034_MIC_PGA 0x7d +#define DA9034_TX_PGA_MUX 0x7e +#define DA9034_VOICE_ADC 0x7f +#define DA9034_VOICE_DAC 0x80 +#define DA9034_SIDE_TONE 0x81 +#define DA9034_PGA_AUX12 0x82 +#define DA9034_PGA_AUX3 0x83 +#define DA9034_PGA_DACS 0x84 + +/* NOTE: the ES1 revision of the chip has the following two issues + * regarding audio: + * 1. The MONO and BEAR are incorrectly exchanged + * 2. The Stereo DAC has to be enabled for Voice Codec + */ +#define DA9034_ERRATA_ES1 1 + +#define ARRAY_AND_SIZE(x) x, ARRAY_SIZE(x) + +/* initial values for the audio registers */ +static const uint8_t da9034_initregs[] = { + 0x00, 0x00, 0x00, 0x00, 0x00, 0x82, 0x00, 0x40, /* 0x70 ~ 0x77 */ + 0xe0, 0x80, 0x80, 0x40, 0x00, 0x40, 0x00, 0x02, /* 0x78 ~ 0x7f */ + 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x80 ~ 0x84 */ +}; + +/* Mono Out Mixer */ +static const struct snd_kcontrol_new da9034_mixer_mono_out_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_MONO, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_MONO, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_MONO, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_MONO, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_MONO, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_MONO, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_MONO, 6, 1, 0), +}; + +static const struct snd_kcontrol_new da9034_mixer_bear_out_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_MONO, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_MONO, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_MONO, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_MONO, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_MONO, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_MONO, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_MONO, 6, 1, 0), +}; + +/* Line Out Mixer */ +static const struct snd_kcontrol_new da9034_mixer_line_out_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_LINE, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_LINE, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_LINE, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_LINE, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_LINE, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_LINE, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_LINE, 6, 1, 0), + SOC_DAPM_SINGLE("MIC", DA9034_MUX_LINE, 7, 1, 0), +}; + +/* Stereo Out Mixer */ +static const struct snd_kcontrol_new da9034_mixer_stereo_ch1_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_STEREO_CH1, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_STEREO_CH1, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_STEREO_CH1, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_STEREO_CH1, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_STEREO_CH1, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_STEREO_CH1, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_STEREO_CH1, 6, 1, 0), +}; + +static const struct snd_kcontrol_new da9034_mixer_stereo_ch2_controls[] = { + SOC_DAPM_SINGLE("AUX1", DA9034_MUX_STEREO_CH2, 0, 1, 0), + SOC_DAPM_SINGLE("AUX2", DA9034_MUX_STEREO_CH2, 1, 1, 0), + SOC_DAPM_SINGLE("AUX3", DA9034_MUX_STEREO_CH2, 2, 1, 0), + SOC_DAPM_SINGLE("DAC1", DA9034_MUX_STEREO_CH2, 3, 1, 0), + SOC_DAPM_SINGLE("DAC2", DA9034_MUX_STEREO_CH2, 4, 1, 0), + SOC_DAPM_SINGLE("DAC3", DA9034_MUX_STEREO_CH2, 5, 1, 0), + SOC_DAPM_SINGLE("AUX2inv", DA9034_MUX_STEREO_CH2, 6, 1, 0), +}; + + +/* default to differential mode for MIC and AUX2 */ +static const char *mux_tx_pga_texts[] = { + "NONE", "MIC", "AUX1", "AUX2", "AUX3", "DAC1", "DAC2", +}; + +static const unsigned int mux_tx_pga_values[] = { + 0x00, 0x0c, 0x02, 0x11, 0x20, 0x40, 0x80, +}; + +static const char *mux_mic_texts[] = { "MIC1", "MIC2" }; + +static const struct soc_enum da9034_mux_enum[] = { + SOC_VALUE_ENUM_SINGLE(DA9034_TX_PGA_MUX, 0, 0xff, 7, + mux_tx_pga_texts, mux_tx_pga_values), + SOC_ENUM_SINGLE(DA9034_MIC_PGA, 4, 2, mux_mic_texts), +}; + +static const struct snd_kcontrol_new mux_ctrls[] = { + SOC_DAPM_VALUE_ENUM("Route", da9034_mux_enum[0]), + SOC_DAPM_ENUM("Route", da9034_mux_enum[1]), +}; + +static const struct snd_kcontrol_new pga_ctrls[] = { + /* PGAs for input */ + SOC_DAPM_SINGLE("MIC Volume", DA9034_MIC_PGA, 0, 0x07, 0), + SOC_DAPM_SINGLE("AUX1 Volume", DA9034_PGA_AUX12, 0, 0x03, 0), + SOC_DAPM_SINGLE("AUX2 Volume", DA9034_PGA_AUX12, 4, 0x03, 0), + SOC_DAPM_SINGLE("AUX3 Volume", DA9034_PGA_AUX3, 0, 0x03, 0), + SOC_DAPM_SINGLE("TX Volume", DA9034_TX_PGA, 1, 0x0f, 0), + + /* PGAs for output */ + SOC_DAPM_SINGLE("MONO Volume", DA9034_MONO_VOL, 0, 0x3f, 0), + SOC_DAPM_SINGLE("BEAR Volume", DA9034_BEAR_VOL, 0, 0x3f, 0), + SOC_DAPM_SINGLE("LINE_OUT Volume", DA9034_AUDIO_LINE_AMP, 0, 0xf, 0), + SOC_DAPM_SINGLE("STEREO_CH1 Volume", DA9034_STEREO_AMP_CH1, 0, 0x3f, 0), + SOC_DAPM_SINGLE("STEREO_CH2 Volume", DA9034_STEREO_AMP_CH2, 0, 0x3f, 0), +}; + +static const struct snd_soc_dapm_widget da9034_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("AUX1"), + SND_SOC_DAPM_INPUT("AUX2"), + SND_SOC_DAPM_INPUT("AUX3"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AUX2inv"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("BEAR"), + SND_SOC_DAPM_OUTPUT("MONO"), + SND_SOC_DAPM_OUTPUT("LINE_OUT"), + SND_SOC_DAPM_OUTPUT("STEREO_CH1"), + SND_SOC_DAPM_OUTPUT("STEREO_CH2"), + + /* muxers */ + SND_SOC_DAPM_VALUE_MUX("TX-PGA Mux", SND_SOC_NOPM, 0, 0, &mux_ctrls[0]), + SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mux_ctrls[1]), + + /* mixers */ + SND_SOC_DAPM_MIXER("LINE_OUT Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_line_out_controls, 8), + SND_SOC_DAPM_MIXER("MONO Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_mono_out_controls, 7), + SND_SOC_DAPM_MIXER("BEAR Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_bear_out_controls, 7), + SND_SOC_DAPM_MIXER("STEREO_CH1 Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_stereo_ch1_controls, 7), + SND_SOC_DAPM_MIXER("STEREO_CH2 Mixer", SND_SOC_NOPM, 0, 0, + da9034_mixer_stereo_ch2_controls, 7), + + /* DACs and ADCs */ +#ifdef DA9034_ERRATA_ES1 + SND_SOC_DAPM_DAC("DAC1", "HiFi Playback CH1", -1, 0, 0), + SND_SOC_DAPM_DAC("DAC2", "HiFi Playback CH2", -1, 0, 0), +#else + SND_SOC_DAPM_DAC("DAC1", "HiFi Playback CH1", DA9034_STEREO_DAC, 7, 0), + SND_SOC_DAPM_DAC("DAC2", "HiFi Playback CH2", DA9034_STEREO_DAC, 7, 0), +#endif + SND_SOC_DAPM_DAC("DAC3", "Voice Playback", DA9034_VOICE_DAC, 3, 0), + SND_SOC_DAPM_ADC("VADC", "Voice Capture", DA9034_VOICE_ADC, 0, 0), + + /* PGAs */ + SND_SOC_DAPM_PGA("MIC PGA", DA9034_MIC_PGA, 3, 0, &pga_ctrls[0], 1), + SND_SOC_DAPM_PGA("AUX1 PGA", DA9034_PGA_AUX12, 3, 0, &pga_ctrls[1], 1), + SND_SOC_DAPM_PGA("AUX2 PGA", DA9034_PGA_AUX12, 7, 0, &pga_ctrls[2], 1), + SND_SOC_DAPM_PGA("AUX3 PGA", DA9034_PGA_AUX3, 3, 0, &pga_ctrls[3], 1), + SND_SOC_DAPM_PGA("TX PGA", SND_SOC_NOPM, 0, 0, &pga_ctrls[4], 1), + + SND_SOC_DAPM_PGA("MONO PGA", DA9034_MONO_VOL, 6, 0, &pga_ctrls[5], 1), + SND_SOC_DAPM_PGA("BEAR PGA", DA9034_BEAR_VOL, 6, 0, &pga_ctrls[6], 1), + SND_SOC_DAPM_PGA("LINE_OUT PGA", DA9034_AUDIO_LINE_AMP, 4, 0, + &pga_ctrls[7], 1), + SND_SOC_DAPM_PGA("STEREO_CH1 PGA", DA9034_STEREO_AMP_CH1, 7, 0, + &pga_ctrls[8], 1), + SND_SOC_DAPM_PGA("STEREO_CH2 PGA", DA9034_STEREO_AMP_CH2, 7, 0, + &pga_ctrls[9], 1), + + /* Mic Bias */ + SND_SOC_DAPM_MICBIAS("MIC Int Bias", DA9034_MIC_PGA, 6, 0), + SND_SOC_DAPM_MICBIAS("MIC Ext Bias", DA9034_MIC_PGA, 7, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* MIC PGA */ + { "MIC PGA", NULL, "MIC1" }, + { "MIC PGA", NULL, "MIC2" }, + + /* MONO Mixer */ + { "MONO Mixer", "AUX1", "AUX1" }, + { "MONO Mixer", "AUX2", "AUX2" }, + { "MONO Mixer", "AUX3", "AUX3" }, + { "MONO Mixer", "DAC1", "DAC1" }, + { "MONO Mixer", "DAC2", "DAC2" }, + { "MONO Mixer", "DAC3", "DAC3" }, + { "MONO Mixer", "AUX2inv", "AUX2inv" }, + + { "MONO PGA", NULL, "MONO Mixer" }, +#ifdef DA9034_ERRATA_ES1 + { "BEAR PGA", NULL, "MONO Mixer" }, +#endif + { "MONO", NULL, "MONO PGA" }, + + /* BEAR Mixer */ + { "BEAR Mixer", "AUX1", "AUX1" }, + { "BEAR Mixer", "AUX2", "AUX2" }, + { "BEAR Mixer", "AUX3", "AUX3" }, + { "BEAR Mixer", "DAC1", "DAC1" }, + { "BEAR Mixer", "DAC2", "DAC2" }, + { "BEAR Mixer", "DAC3", "DAC3" }, + { "BEAR Mixer", "AUX2inv", "AUX2inv" }, + +#ifdef DA9034_ERRATA_ES1 + { "MONO PGA", NULL, "BEAR Mixer" }, +#endif + { "BEAR PGA", NULL, "BEAR Mixer" }, + { "BEAR", NULL, "BEAR PGA" }, + + /* LINE_OUT Mixer */ + { "LINE_OUT Mixer", "AUX1", "AUX1" }, + { "LINE_OUT Mixer", "AUX2", "AUX2" }, + { "LINE_OUT Mixer", "AUX3", "AUX3" }, + { "LINE_OUT Mixer", "DAC1", "DAC1" }, + { "LINE_OUT Mixer", "DAC2", "DAC2" }, + { "LINE_OUT Mixer", "DAC3", "DAC3" }, + { "LINE_OUT Mixer", NULL, "MIC PGA" }, + { "LINE_OUT Mixer", "AUX2inv", "AUX2inv" }, + + { "LINE_OUT PGA", NULL, "LINE_OUT Mixer" }, + { "LINE_OUT", NULL, "LINE_OUT PGA" }, + + /* STEREO_CH1 Mixer */ + { "STEREO_CH1 Mixer", "AUX1", "AUX1" }, + { "STEREO_CH1 Mixer", "AUX2", "AUX2" }, + { "STEREO_CH1 Mixer", "AUX3", "AUX3" }, + { "STEREO_CH1 Mixer", "DAC1", "DAC1" }, + { "STEREO_CH1 Mixer", "DAC2", "DAC2" }, + { "STEREO_CH1 Mixer", "DAC3", "DAC3" }, + { "STEREO_CH1 Mixer", "AUX2inv", "AUX2inv" }, + + { "STEREO_CH1 PGA", NULL, "STEREO_CH1 Mixer" }, + { "STEREO_CH1", NULL, "STEREO_CH1 PGA" }, + + /* STEREO_CH2 Mixer */ + { "STEREO_CH2 Mixer", "AUX1", "AUX1" }, + { "STEREO_CH2 Mixer", "AUX2", "AUX2" }, + { "STEREO_CH2 Mixer", "AUX3", "AUX3" }, + { "STEREO_CH2 Mixer", "DAC1", "DAC1" }, + { "STEREO_CH2 Mixer", "DAC2", "DAC2" }, + { "STEREO_CH2 Mixer", "DAC3", "DAC3" }, + { "STEREO_CH2 Mixer", "AUX2inv", "AUX2inv" }, + + { "STEREO_CH2 PGA", NULL, "STEREO_CH2 Mixer" }, + { "STEREO_CH2", NULL, "STEREO_CH2 PGA" }, + + /* TX-PGA P Mux */ + { "TX-PGA Mux", "MIC", "MIC PGA" }, + { "TX-PGA Mux", "AUX1", "AUX1 PGA" }, + { "TX-PGA Mux", "AUX2", "AUX2 PGA" }, + { "TX-PGA Mux", "AUX3", "AUX3 PGA" }, + { "TX-PGA Mux", "DAC1", "DAC1" }, + { "TX-PGA Mux", "DAC2", "DAC2" }, + + { "VADC", NULL, "TX-PGA Mux" }, +}; + +static int da9034_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(da9034_dapm_widgets)); + snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static unsigned int da9034_audio_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + uint8_t val; + + da903x_read(codec->dev, reg, &val); + return val; +} + +static int da9034_audio_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val) +{ + return da903x_write(codec->dev, reg, val); +} + +static int da9034_reset(struct snd_soc_codec *codec, int try_warm) +{ + int reg; + + for (reg = 0; reg < ARRAY_SIZE(da9034_initregs); reg++) + da903x_write(codec->dev, 0x70 + reg, da9034_initregs[reg]); + +#ifdef DA9034_ERRATA_ES1 + da903x_set_bits(codec->dev, DA9034_STEREO_DAC, 1 << 7); +#endif + return 0; +} + +static int da9034_hifi_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + uint8_t i2s_control; + + da903x_read(codec->dev, DA9034_I2S_CONTROL, &i2s_control); + + i2s_control &= 0x0f; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: break; + case SND_SOC_DAIFMT_CBM_CFM: i2s_control |= 0xa0; break; + case SND_SOC_DAIFMT_CBM_CFS: i2s_control |= 0x20; break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + i2s_control |= 0x10; + break; + default: + return -EINVAL; + } + + return da903x_write(codec->dev, DA9034_I2S_CONTROL, i2s_control); +} + +static int da9034_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + uint8_t i2s_control; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + return -EINVAL; + } + + da903x_read(codec->dev, DA9034_I2S_CONTROL, &i2s_control); + + i2s_control &= 0xf0; + + switch (params_rate(params)) { + case 8000: break; + case 11025: i2s_control |= 0x01; break; + case 12000: i2s_control |= 0x02; break; + case 16000: i2s_control |= 0x03; break; + case 22050: i2s_control |= 0x04; break; + case 24000: i2s_control |= 0x05; break; + case 32000: i2s_control |= 0x06; break; + case 44100: i2s_control |= 0x07; break; + case 48000: i2s_control |= 0x0f; break; + default: + return -EINVAL; + } + + return da903x_write(codec->dev, DA9034_I2S_CONTROL, i2s_control); +} + +static int da9034_voice_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + return 0; +} + +static int da9034_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + uint8_t val; + + da903x_read(codec->dev, DA9034_VOICE_ADC, &val); + + val |= 0x18; + + switch ((params_rate(params))) { + case 8000: break; + case 16000: val |= 0x01 << 3; break; + case 32000: val |= 0x02 << 3; break; + default: + return -EINVAL; + } + + return da903x_write(codec->dev, DA9034_VOICE_ADC, val); +} + +#define DA9034_HIFI_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define DA9034_VOICE_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000) + +#define DA9034_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops da9034_dai_ops_hifi = { + .hw_params = da9034_hifi_hw_params, + .set_fmt = da9034_hifi_set_fmt, +}; + +static struct snd_soc_dai_ops da9034_dai_ops_voice = { + .hw_params = da9034_voice_hw_params, + .set_fmt = da9034_voice_set_fmt, +}; + +struct snd_soc_dai da9034_dais[] = { + [0] = { + .name = "I2S HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = DA9034_HIFI_RATES, + .formats = DA9034_PCM_FORMATS, + }, + .ops = &da9034_dai_ops_hifi, + }, + [1] = { + .name = "PCM Voice", + .playback = { + .stream_name = "Voice Playback", + .channels_min = 1, + .channels_max = 1, + .rates = DA9034_VOICE_RATES, + .formats = DA9034_PCM_FORMATS, + }, + .capture = { + .stream_name = "Voice Capture", + .channels_min = 1, + .channels_max = 1, + .rates = DA9034_VOICE_RATES, + .formats = DA9034_PCM_FORMATS, + }, + .ops = &da9034_dai_ops_voice, + }, +}; +EXPORT_SYMBOL_GPL(da9034_dais); + +#ifdef CONFIG_DEBUG_FS +static struct { + const char *reg_name; + int reg_addr; +} da9034_regs[] = { + { "MUX_MONO", 0x70 }, + { "MUX_BEAR", 0x71 }, + { "MUX_LINE", 0x72 }, + { "MUX_STEREO_CH1", 0x73 }, + { "MUX_STEREO_CH2", 0x74 }, + { "AUDIO_LINE_AMP", 0x75 }, + { "STEREO_AMP_CH1", 0x76 }, + { "STEREO_AMP_CH2", 0x77 }, + { "STEREO_DAC", 0x78 }, + { "MONO_VOL", 0x79 }, + { "BEAR_VOL", 0x7a }, + { "I2S_CONTROL", 0x7b }, + { "TX_PGA", 0x7c }, + { "MIC_AMP", 0x7d }, + { "TX_PGA_MUX", 0x7e }, + { "VOICE_ADC", 0x7f }, + { "VOICE_DAC", 0x80 }, + { "SIDE_TONE", 0x81 }, + { "PGA_AUXI12", 0x82 }, + { "PGA_AUXI3", 0x83 }, + { "PGA_DACS", 0x84 }, +}; + +static int da9034_debugfs_show(struct seq_file *s, void *data) +{ + struct snd_soc_codec *codec = data; + struct device *dev = codec->dev; + uint8_t v; + int i; + + for (i = 0; i < ARRAY_SIZE(da9034_regs); i++) { + da903x_read(codec->dev, da9034_regs[i].reg_addr, &v); + seq_printf(s, "%20s: 0x%02x\n", da9034_regs[i].reg_name, v); + } + return 0; +} + +static int da9034_debugfs_open(struct inode *inode, struct file *file) +{ + return single_open(file, da9034_debugfs_show, inode->i_private); +} + +static const struct file_operations da9034_debugfs_ops = { + .owner = THIS_MODULE, + .open = da9034_debugfs_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void __init da9034_init_debugfs(struct snd_soc_codec *codec) +{ + debugfs_create_file("da9034", S_IFREG | S_IRUGO, NULL, + codec, da9034_debugfs_ops); +} +#else +static inline void da9034_init_debugfs(struct snd_soc_codec *codec) {} +#endif /* CONFIG_DEBUG_FS */ + +static struct snd_soc_codec *da9034_codec; + +static int da9034_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret = 0; + + BUG_ON(!da9034_codec); + + socdev->card->codec = da9034_codec; + socdev->dev = &pdev->dev; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + da9034_add_widgets(da9034_codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + + return 0; + +reset_err: + snd_soc_free_pcms(socdev); + socdev->card->codec = NULL; + return ret; +} + +static int da9034_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_da9034 = { + .probe = da9034_soc_probe, + .remove = da9034_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_da9034); + +static int __devinit da9034_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(*codec), GFP_KERNEL); + if (codec == NULL) { + dev_err(&pdev->dev, "failed to allocate codec\n"); + return -ENOMEM; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = pdev->dev.parent; + codec->name = "DA9034"; + codec->owner = THIS_MODULE; + codec->read = da9034_audio_read; + codec->write = da9034_audio_write; + codec->dai = da9034_dais; + codec->num_dai = ARRAY_SIZE(da9034_dais); + + ret = da9034_reset(codec, 0); + if (ret < 0) { + dev_err(&pdev->dev, "codec reset error\n"); + goto err_free_codec; + } + + ret = snd_soc_register_codec(codec); + if (ret) + goto err_free_codec; + + ret = snd_soc_register_dais(ARRAY_AND_SIZE(da9034_dais)); + if (ret) + goto err_unregister; + + da9034_init_debugfs(codec); + da9034_codec = codec; + platform_set_drvdata(pdev, codec); + + dev_info(&pdev->dev, "da9034 (aka micco) audio codec registered\n"); + return 0; + +err_unregister: + snd_soc_unregister_codec(codec); +err_free_codec: + kfree(codec); + return ret; +} + +static int __devexit da9034_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_codec *codec = platform_get_drvdata(pdev); + + snd_soc_unregister_dais(ARRAY_AND_SIZE(da9034_dais)); + snd_soc_unregister_codec(codec); + kfree(codec); + return 0; +} + +static struct platform_driver da9034_audio_driver = { + .driver = { + .name = "da9034-audio", + .owner = THIS_MODULE, + }, + .probe = da9034_audio_probe, + .remove = da9034_audio_remove, +}; + +static int da9034_audio_init(void) +{ + return platform_driver_register(&da9034_audio_driver); +} +module_init(da9034_audio_init); + +static void da9034_audio_exit(void) +{ + platform_driver_unregister(&da9034_audio_driver); +} +module_exit(da9034_audio_exit); + +MODULE_DESCRIPTION("ASoC DA9034 Audio Codec Driver"); +MODULE_AUTHOR("Bin Yang <bin.yang(a)marvell.com>" + "Paul Shen <bshen9(a)marvell.com>"); +MODULE_LICENSE("GPL"); -- 1.6.0.4

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[alsa-devel] [PATCH 0/2] USB audio gadget driver
by Mike Frysinger 03 Jun '09

03 Jun '09

This is largely a PoC in case people are looking for this kind of thing and want to build upon something. Just throwing it out there at this point. Bryan Wu (2): USB: audio: add USB audio class definitions USB: gadget: add USB Audio Gadget driver drivers/usb/gadget/Kconfig | 14 + drivers/usb/gadget/Makefile | 2 + drivers/usb/gadget/audio.c | 302 ++++++++++++++++++ drivers/usb/gadget/f_audio.c | 707 ++++++++++++++++++++++++++++++++++++++++++ drivers/usb/gadget/u_audio.c | 319 +++++++++++++++++++ drivers/usb/gadget/u_audio.h | 56 ++++ include/linux/usb/audio.h | 265 ++++++++++++++++- 7 files changed, 1663 insertions(+), 2 deletions(-) create mode 100644 drivers/usb/gadget/audio.c create mode 100644 drivers/usb/gadget/f_audio.c create mode 100644 drivers/usb/gadget/u_audio.c create mode 100644 drivers/usb/gadget/u_audio.h

1 2

[alsa-devel] [PATCH 0/4] add da9034 audio support for Littleton and generalize SSP I2S/LeftJ
by Eric Miao 03 Jun '09

03 Jun '09

Thanks for Paul's good work, and finally, we worked together to figure out the generalized I2S/LeftJ support in current pxa-ssp.c, and added the audio support for Littleton: Here's the summary: Paul Shen (4): ASoC: pxa-ssp: enhance I2S and add Left_J support ASoC: add dialog da9034 (aka micco) audio codec support ASoC: pxa add Marvell Littleton platform audio support [ARM] pxa/littleton: add support for audio (da9034) arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 +- arch/arm/mach-pxa/littleton.c | 20 +- include/linux/mfd/da903x.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9034-audio.c | 696 +++++++++++++++++++++++++++++ sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/littleton.c | 228 ++++++++++ sound/soc/pxa/pxa-ssp.c | 62 ++-- sound/soc/pxa/pxa-ssp.h | 9 + 11 files changed, 1008 insertions(+), 39 deletions(-) -- Cheers - eric

1 1

[alsa-devel] [PATCH 4/4] [ARM] pxa/littleton: add support for audio (da9034)
by Eric Miao 03 Jun '09

03 Jun '09

Signed-off-by: Paul Shen <bshen9(a)marvell.com> Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- arch/arm/mach-pxa/littleton.c | 20 +++++++++++++++++++- include/linux/mfd/da903x.h | 1 + 2 files changed, 20 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-pxa/littleton.c b/arch/arm/mach-pxa/littleton.c index e13f6a8..b26b3cb 100644 --- a/arch/arm/mach-pxa/littleton.c +++ b/arch/arm/mach-pxa/littleton.c @@ -81,6 +81,21 @@ static mfp_cfg_t littleton_mfp_cfg[] __initdata = { GPIO27_SSP2_TXD, GPIO17_GPIO, /* SFRM as chip-select */ + /* SSP3 */ + GPIO91_SSP3_SCLK, + GPIO92_SSP3_FRM, + GPIO93_SSP3_TXD, + GPIO94_SSP3_RXD, + /* SSP3 NETWORK CLK */ + GPIO126_EXT_CLK, + + /* SSP4 */ + GPIO95_SSP4_SCLK, + GPIO96_SSP4_FRM, + GPIO97_SSP4_TXD, + GPIO98_SSP4_RXD, + + /* Debug Ethernet */ GPIO90_GPIO, @@ -350,7 +365,10 @@ static struct da903x_subdev_info littleton_da9034_subdevs[] = { }, { .name = "da903x-backlight", .id = DA9034_ID_WLED, - }, + }, { + .name = "da9034-audio", + .id = DA9034_ID_AUDIO, + } }; static struct da903x_platform_data littleton_da9034_info = { diff --git a/include/linux/mfd/da903x.h b/include/linux/mfd/da903x.h index 115dbe9..ba645ff 100644 --- a/include/linux/mfd/da903x.h +++ b/include/linux/mfd/da903x.h @@ -39,6 +39,7 @@ enum { DA9034_ID_VIBRA, DA9034_ID_WLED, DA9034_ID_TOUCH, + DA9034_ID_AUDIO, DA9034_ID_BUCK1, DA9034_ID_BUCK2, -- 1.6.0.4

1 0

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 03 Jun '09

03 Jun '09

The following changes since commit 203350c1a8e23adf17fd9a96d8bfc7adf63c1ff6: Mark Brown (1): ASoC: Initialise dev for the dummy S/PDIF DAI are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Cliff Cai (3): ASoC: SSM2602: remove unsupported sample rates ASoC: Blackfin: set the transfer size according the ac97_frame size ASoC: SSM2602: assign last substream to the master when shutting down Peter Ujfalusi (2): ASoC: TWL4030: Use reg_cache in twl4030_init_chip ASoC: TWL4030: Check the interface format for 4 channel mode Sonic Zhang (1): ASoC: Blackfin: document how anomaly 05000250 is handled sound/soc/blackfin/bf5xx-ac97.c | 9 +++++++++ sound/soc/blackfin/bf5xx-sport.c | 4 ++-- sound/soc/codecs/ssm2602.c | 33 +++++++++++++++++++-------------- sound/soc/codecs/twl4030.c | 16 +++++++++++----- 4 files changed, 41 insertions(+), 21 deletions(-)

2 1

[alsa-devel] [PATCH - errata corrige for quirk 1/1]
by me＠borgia.bo.it 03 Jun '09

03 Jun '09

From: Andrea Borgia <andrea(a)borgia.bo.it> Cut'n'paste mistake, whose likely result was nothing at all. Correct version is "USB_DEVICE", not "USB_DEVICE_VENDOR_SPEC". Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 58f24f5..f0f7624 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1986,7 +1986,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0028), + USB_DEVICE(0x0ccd, 0x0028), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "TerraTec", .product_name = "Aureon 5.1 MkII", -- 1.6.0.4

2 1

Re: [alsa-devel] ALSA patch Nvidia 8 channel hdmi audio
by Marek Vasut 03 Jun '09

03 Jun '09

I'm not the author of the patch, CCing alsa-devel and the original author. On Tuesday 02 of June 2009 23:58:26 Alan Alan wrote: > Hi Marek ! > > I just read your emailing with nvidia devs about alsa supporting 8 channel > lpcm over hdmi. > > I checked it out an i have to tell it is working awsome, except for a few > bugs: > > The channels are wrong mapped. Running a speakertest -Dhdmi -c6 gives the > following result (first where it should sound) > FL - FL > C - RL > FR - FR > RR - LFE > RL - C > > The other thing i realized that playback is limited to 16 bits 48 khz. I > think the minimum which is supported by nvidia hardware is 24bits 96 (or > 192) khz. > Why not ? > > Could you contact the devs as i don't know how to reach them. > > ps.: Mybe my build methot was not 100% ok ? I used 1.020 Alsa with the > oatched nvidia.c file. Anyway the amplifier shows multichannel LPCM and > that's great news.

1 0

[alsa-devel] problem with snd_hda_intel for toshiba nb200 (no audio in ubuntu remix)
by sbuonann＠libero.it 02 Jun '09

02 Jun '09

the output of lspci -nv is the fellow: 00:00.0 0600: 8086:27ac (rev 03) Subsystem: 1179:ff60 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information <?> Kernel driver in use: agpgart-intel Kernel modules: intel-agp 00:02.0 0300: 8086:27ae (rev 03) Subsystem: 1179: ff60 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at f0000000 (32- bit, non-prefetchable) [size=512K] I/O ports at 1800 [size=8] Memory at d0000000 (32-bit, prefetchable) [size=256M] Memory at f0200000 (32-bit, non- prefetchable) [size=256K] Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [d0] Power Management version 2 Kernel modules: intelfb 00:02.1 0380: 8086:27a6 (rev 03) Subsystem: 1179:ff60 Flags: bus master, fast devsel, latency 0 Memory at f0080000 (32-bit, non- prefetchable) [size=512K] Capabilities: [d0] Power Management version 2 00:1b. 0 0403: 8086:27d8 (rev 02) Subsystem: 1179:ff6e Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0440000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel 00:1c.0 0604: 8086:27d0 (rev 02) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 Capabilities: [40] Express Root Port (Slot+), MSI 00 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 1179:ff60 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> Kernel driver in use: pcieport-driver Kernel modules: shpchp 00:1c.1 0604: 8086:27d2 (rev 02) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 Memory behind bridge: f0100000-f01fffff Capabilities: [40] Express Root Port (Slot+), MSI 00 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 1179:ff60 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> Kernel driver in use: pcieport- driver Kernel modules: shpchp 00:1c.2 0604: 8086:27d4 (rev 02) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=04, subordinate=04, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: 50000000-500fffff Prefetchable memory behind bridge: 00000000f0500000- 00000000f05fffff Capabilities: [40] Express Root Port (Slot+), MSI 00 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 1179:ff60 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> Kernel driver in use: pcieport-driver Kernel modules: shpchp 00:1c.3 0604: 8086:27d6 (rev 02) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=05, subordinate=05, sec-latency=0 Capabilities: [40] Express Root Port (Slot+), MSI 00 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 1179: ff60 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> Kernel driver in use: pcieport-driver Kernel modules: shpchp 00:1d.0 0c03: 8086:27c8 (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 23 I/O ports at 1820 [size=32] Kernel driver in use: uhci_hcd 00:1d.1 0c03: 8086: 27c9 (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 1840 [size=32] Kernel driver in use: uhci_hcd 00:1d.2 0c03: 8086:27ca (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1860 [size=32] Kernel driver in use: uhci_hcd 00:1d.3 0c03: 8086:27cb (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 16 I/O ports at 1880 [size=32] Kernel driver in use: uhci_hcd 00:1d.7 0c03: 8086:27cc (rev 02) (prog-if 20) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 23 Memory at f0444000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port: BAR=1 offset=00a0 Kernel driver in use: ehci_hcd 00:1e.0 0604: 8086:2448 (rev e2) (prog-if 01) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=06, subordinate=06, sec-latency=0 Capabilities: [50] Subsystem: 1179:ff60 00:1f.0 0601: 8086:27b9 (rev 02) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0 Capabilities: [e0] Vendor Specific Information <?> Kernel modules: iTCO_wdt, intel-rng 00:1f.1 0101: 8086:27df (rev 02) (prog-if 8a [Master SecP PriP]) Subsystem: 1179:ff60 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 1810 [size=16] Kernel driver in use: ata_piix 00:1f.2 0106: 8086:27c5 (rev 02) (prog-if 01) Subsystem: 1179:ff60 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 2299 I/O ports at 18c8 [size=8] I/O ports at 18c0 [size=4] I/O ports at 18a8 [size=8] I/O ports at 180c [size=4] I/O ports at 18b0 [size=16] Memory at f0444400 (32-bit, non-prefetchable) [size=1K] Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [70] Power Management version 2 Kernel driver in use: ahci 00:1f.3 0c05: 8086:27da (rev 02) Subsystem: 1179:ff60 Flags: medium devsel, IRQ 10 I/O ports at 18e0 [size=32] Kernel modules: i2c-i801 03:00.0 0280: 168c:002b (rev 01) Subsystem: 1113:e811 Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at f0100000 (64-bit, non-prefetchable) [size=64K] Capabilities: [40] Power Management version 3 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [60] Express Legacy Endpoint, MSI 00 Capabilities: [100] Advanced Error Reporting <?> Capabilities: [140] Virtual Channel <?> Capabilities: [160] Device Serial Number 12-14-24-ff-ff-17-15-00 Capabilities: [170] Power Budgeting <?> Kernel driver in use: ath9k Kernel modules: ath9k 04:00.0 0200: 10ec:8136 (rev 02) Subsystem: 1179:ff60 Flags: bus master, fast devsel, latency 0, IRQ 2298 I/O ports at 2000 [size=256] Memory at f0510000 (64-bit, prefetchable) [size=4K] Memory at f0500000 (64- bit, prefetchable) [size=64K] [virtual] Expansion ROM at f0520000 [disabled] [size=128K] Capabilities: [40] Power Management version 3 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Capabilities: [70] Express Endpoint, MSI 01 Capabilities: [ac] MSI-X: Enable- Mask- TabSize=2 Capabilities: [cc] Vital Product Data <?> Capabilities: [100] Advanced Error Reporting <?> Capabilities: [140] Virtual Channel <?> Capabilities: [160] Device Serial Number 00-00-00-00-00-00-00-01 Kernel driver in use: r8169 Kernel modules: r8169

1 0

[alsa-devel] How can i solve the alsa underrun issue
by Ren Brian 02 Jun '09

02 Jun '09

Hi, I'm trying to develop one application using alsa lib. The application receives encoded sound data from a udp socket and decodes the data and write it to device. Is it possible to set the buffer length for alsa?. In my application i have a short array of size 6000 to keep the decoded data and decoded data length is 5120. What period size i should set. It always set 940 frames. When i used the OSS calls, it works fine. When i used the alsa lib it gives underrun. How can i resolve this? Thanks, -Renju

1 1

[alsa-devel] HDAudio volume/widget control?
by pl bossart 02 Jun '09

02 Jun '09

Hi all, Bear with my ignorance here. It's not clear to me how an application or an audio server can precisely control volumes if the hierarchy of HDAudio widgets is not exposed. I checked in the HDaudio code, and it seems the connection list and connectivity entries are only showed in sysfs, not really directly usable. My findings are backed by the 2008 LPC report at http://lwn.net/Articles/299093/ which asserts that "The new audio standard, HD-Audio, is commonly found on new systems, and it's well-supported at the kernel level. However, it's based on "widgets" with vendor-configurable I/O pins. A driver can't tell how the HD-Audio part is connected, so some Linux plumbing work is required to identify which of the many exposed level controls is the right one to show the user. An audience member pointed out the need to tweak multiple level settings on his hardware, to get the right level without distortion. Linux will need more information on how each machine has its HD-Audio hardware hooked up in order to reliably give the user a useful volume control." Am I missing something? Or is there already a plan to change this? Thanks for your feedback. Cheers, Pierre

1 0

[alsa-devel] unknown card works
by Oscon 02 Jun '09

02 Jun '09

Hello! I have got a "retro" :) PCI analog tuner card. id: unknown card 0x878-0x144f:0x3000 board name: Askey Computer Corp. MagicTView'99 digital audio rate: 48000 Hz (?) ------------------ Other informations: / dmesg, modules.conf, lspci, system information (version of kernel+alsa-utils) dmesg: May 30 14:19:51 osconsfortress kernel: unknown card 0x878-0x144f:0x3000 May 30 14:19:51 osconsfortress kernel: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> May 30 14:19:51 osconsfortress kernel: Bt87x 0000:03:06.1: PCI INT A -> GSI 21 (level, low) -> IRQ 21 May 30 14:19:51 osconsfortress kernel: bt87x0: Using board 0, analog, digital (rate 48000 Hz) -------------------- modules.conf: options snd-bt87x index=2 enable=1 load_all=1 digital_rate=48000 -------------------- lspci -vvv 03:06.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 02) Subsystem: Askey Computer Corp. MagicTView CPH060 - Audio Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (1000ns min, 63750ns max) Interrupt: pin A routed to IRQ 21 Region 0: Memory at f8ffe000 (32-bit, prefetchable) [size=4K] Kernel driver in use: Bt87x --------------------------- System information: Debian Lenny kernel: root@osconsfortress:~# uname -a Linux osconsfortress 2.6.27-limbo #1 SMP Thu Mar 5 11:18:25 CET 2009 i686 GNU/Linux root@osconsfortress:~# (based on 2.6.27.10+grsec+layer7) root@osconsfortress:~# uptime 14:32:12 up 48 days, 19:41, 3 users, load average: 0.28, 0.30, 0.35 alsa-base :Version: 1.0.17.dfsg-4 -------------------- Cheers, Oscon

2 1

[alsa-devel] Playback sound is only audible starting at 50% on Dell Inspiron 1420 (Ubuntu 32-bit)
by komputes 02 Jun '09

02 Jun '09

Hello ALSA Developers, I would like to know if anyone can help me correct the following issue. Playback sound is only audible starting at 50% on Dell Inspiron 1420 running Ubuntu 9.04 Desktop x86. This is true with 9.04, 8.10, 8.04 (all x86) but does not affect Ubuntu 9.04 AMD64 on which sound is audible at ~4%. Both of these volume levels refer to the "Master" level, all other playback levels are at 100% (even the ones that are invisible/disabled by default). I hang out on #alsa where I was proposed to start alsa with the options "options snd-hda-intel model=dell-3stack" then "options snd-hda-intel model=3stack" none of which corrected this 50% volume issue. This issue as well as all technical details can be found here (includes alsa-info output): https://bugs.edge.launchpad.net/ubuntu/+source/alsa-driver/+bug/382488 Thank you for your time, -komputes

2 1

[alsa-devel] [PATCH 1/2] ALSA: snd_usb_caiaq: give better shortname
by Daniel Mack 02 Jun '09

02 Jun '09

If not passed as module option, provide an own card ID with the newly introduced snd_set_card_id() call. This will prevent ALSA from calling choose_default_name() which only takes the last part of a name containing whitespaces. This for example caused 'Audio 4 DJ' to be shortened to 'DJ', which was not very descriptive. The implementation now takes the short name and removes all whitespaces from it which is much nicer. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/device.c | 21 +++++++++++++++++++-- 1 files changed, 19 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 15052e4..43bd0db 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -382,10 +382,10 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) static int __devinit init_card(struct snd_usb_caiaqdev *dev) { - char usbpath[32]; + char *c, usbpath[32]; struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; - int err; + int err, len; if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); @@ -427,6 +427,23 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) strlcpy(card->driver, MODNAME, sizeof(card->driver)); strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); + /* if the id was not passed as module option, fill it with a shortened + * version of the product string which does not contain any + * whitespaces */ + + if (*card->id == '\0') { + char id[sizeof(card->id)]; + + memset(id, 0, sizeof(id)); + + for (c = card->shortname, len = 0; + *c && len < sizeof(card->id); c++) + if (*c != ' ') + id[len++] = *c; + + snd_card_set_id(card, id); + } + usb_make_path(usb_dev, usbpath, sizeof(usbpath)); snprintf(card->longname, sizeof(card->longname), "%s %s (%s)", -- 1.6.3.1

2 4

[alsa-devel] [PATCH 1/5] ALSA: snd_usb_caiaq: clean whitespaces
by Daniel Mack 02 Jun '09

02 Jun '09

Cosmetic changes only, no code change. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 88 +++++++++++++++++++++++----------------------- sound/usb/caiaq/device.c | 60 ++++++++++++++++---------------- sound/usb/caiaq/midi.c | 24 ++++++------ 3 files changed, 86 insertions(+), 86 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b13ce76..b144513 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -42,10 +42,10 @@ (stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1) static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S24_3BE, - .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 0, /* will overwrite later */ @@ -68,7 +68,7 @@ activate_substream(struct snd_usb_caiaqdev *dev, dev->sub_capture[sub->number] = sub; } -static void +static void deactivate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { @@ -118,7 +118,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) return -EPIPE; } } - + return 0; } @@ -129,7 +129,7 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) debug("%s(%p)\n", __func__, dev); if (!dev->streaming) return; - + dev->streaming = 0; for (i = 0; i < N_URBS; i++) { @@ -154,7 +154,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); if (all_substreams_zero(dev->sub_playback) && all_substreams_zero(dev->sub_capture)) { - /* when the last client has stopped streaming, + /* when the last client has stopped streaming, * all sample rates are allowed again */ stream_stop(dev); dev->pcm_info.rates = dev->samplerates; @@ -194,7 +194,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; debug("%s(%p)\n", __func__, substream); - + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; @@ -205,19 +205,19 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) if (dev->streaming) return 0; - + /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ for (i=0; i < ARRAY_SIZE(rates); i++) if (runtime->rate == rates[i]) dev->pcm_info.rates = 1 << i; - + snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; if (dev->spec.data_alignment == 2) bytes_per_sample++; - + bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams; @@ -232,7 +232,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) ret = stream_start(dev); if (ret) return ret; - + dev->output_running = 0; wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ); if (!dev->output_running) { @@ -273,7 +273,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) return SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + return bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else return bytes_to_frames(sub->runtime, @@ -291,7 +291,7 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer }; - + static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream **subs) { @@ -333,7 +333,7 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] + audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) @@ -354,14 +354,14 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, for (i = 0; i < iso->actual_length;) { if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { - for (stream = 0; - stream < dev->n_streams; + for (stream = 0; + stream < dev->n_streams; stream++, i++) { if (dev->first_packet) continue; check_byte = MAKE_CHECKBYTE(dev, stream, i); - + if ((usb_buf[i] & 0x3f) != check_byte) dev->input_panic = 1; @@ -410,21 +410,21 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, } if ((dev->input_panic || dev->output_panic) && !dev->warned) { - debug("streaming error detected %s %s\n", + debug("streaming error detected %s %s\n", dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); dev->warned = 1; } } -static void fill_out_urb(struct snd_usb_caiaqdev *dev, - struct urb *urb, +static void fill_out_urb(struct snd_usb_caiaqdev *dev, + struct urb *urb, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; int stream, i; - + for (i = 0; i < iso->length;) { for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; @@ -444,7 +444,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && - i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == + i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); @@ -453,7 +453,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, static void read_completed(struct urb *urb) { - struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; @@ -478,7 +478,7 @@ static void read_completed(struct urb *urb) out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; - + if (len > 0) { spin_lock(&dev->spinlock); fill_out_urb(dev, out, &out->iso_frame_desc[outframe]); @@ -497,14 +497,14 @@ static void read_completed(struct urb *urb) out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } - + /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; urb->iso_frame_desc[frame].length = BYTES_PER_FRAME; urb->iso_frame_desc[frame].actual_length = 0; } - + urb->number_of_packets = FRAMES_PER_URB; urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); @@ -528,7 +528,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) struct usb_device *usb_dev = dev->chip.dev; unsigned int pipe; - pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? + pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) : usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE); @@ -547,25 +547,25 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) return urbs; } - urbs[i]->transfer_buffer = + urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { log("unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } - + for (frame = 0; frame < FRAMES_PER_URB; frame++) { - struct usb_iso_packet_descriptor *iso = + struct usb_iso_packet_descriptor *iso = &urbs[i]->iso_frame_desc[frame]; - + iso->offset = BYTES_PER_FRAME * frame; iso->length = BYTES_PER_FRAME; } - + urbs[i]->dev = usb_dev; urbs[i]->pipe = pipe; - urbs[i]->transfer_buffer_length = FRAMES_PER_URB + urbs[i]->transfer_buffer_length = FRAMES_PER_URB * BYTES_PER_FRAME; urbs[i]->context = &dev->data_cb_info[i]; urbs[i]->interval = 1; @@ -589,7 +589,7 @@ static void free_urbs(struct urb **urbs) for (i = 0; i < N_URBS; i++) { if (!urbs[i]) continue; - + usb_kill_urb(urbs[i]); kfree(urbs[i]->transfer_buffer); usb_free_urb(urbs[i]); @@ -602,11 +602,11 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) { int i, ret; - dev->n_audio_in = max(dev->spec.num_analog_audio_in, - dev->spec.num_digital_audio_in) / + dev->n_audio_in = max(dev->spec.num_analog_audio_in, + dev->spec.num_digital_audio_in) / CHANNELS_PER_STREAM; dev->n_audio_out = max(dev->spec.num_analog_audio_out, - dev->spec.num_digital_audio_out) / + dev->spec.num_digital_audio_out) / CHANNELS_PER_STREAM; dev->n_streams = max(dev->n_audio_in, dev->n_audio_out); @@ -619,7 +619,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) return -EINVAL; } - ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, + ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, dev->n_audio_out, dev->n_audio_in, &dev->pcm); if (ret < 0) { @@ -632,7 +632,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); - + memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware, sizeof(snd_usb_caiaq_pcm_hardware)); @@ -651,9 +651,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) break; } - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usb_caiaq_ops); - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); snd_pcm_lib_preallocate_pages_for_all(dev->pcm, @@ -662,7 +662,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); dev->data_cb_info = - kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, + kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); if (!dev->data_cb_info) @@ -672,14 +672,14 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; } - + dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret); if (ret < 0) { kfree(dev->data_cb_info); free_urbs(dev->data_urbs_in); return ret; } - + dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret); if (ret < 0) { kfree(dev->data_cb_info); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 515de1c..9be0f2e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -79,7 +79,7 @@ static struct usb_device_id snd_usb_id_table[] = { { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = USB_VID_NATIVEINSTRUMENTS, - .idProduct = USB_PID_RIGKONTROL2 + .idProduct = USB_PID_RIGKONTROL2 }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, @@ -197,7 +197,7 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, if (buffer && len > 0) memcpy(dev->ep1_out_buf+1, buffer, len); - + dev->ep1_out_buf[0] = command; return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), dev->ep1_out_buf, len+1, &actual_len, 200); @@ -208,7 +208,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, { int ret; char tmp[5]; - + switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; case 48000: tmp[0] = SAMPLERATE_48000; break; @@ -237,12 +237,12 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, if (ret) return ret; - - if (!wait_event_timeout(dev->ep1_wait_queue, + + if (!wait_event_timeout(dev->ep1_wait_queue, dev->audio_parm_answer >= 0, HZ)) return -EPIPE; - - if (dev->audio_parm_answer != 1) + + if (dev->audio_parm_answer != 1) debug("unable to set the device's audio params\n"); else dev->bpp = bpp; @@ -250,8 +250,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, return dev->audio_parm_answer == 1 ? 0 : -EINVAL; } -int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, - int digital, int analog, int erp) +int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, + int digital, int analog, int erp) { char tmp[3] = { digital, analog, erp }; return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG, @@ -262,7 +262,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) { int ret; char val[4]; - + /* device-specific startup specials */ switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): @@ -314,7 +314,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) dev->control_state, 1); break; } - + if (dev->spec.num_analog_audio_out + dev->spec.num_analog_audio_in + dev->spec.num_digital_audio_out + @@ -323,7 +323,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) if (ret < 0) log("Unable to set up audio system (ret=%d)\n", ret); } - + if (dev->spec.num_midi_in + dev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(dev); @@ -363,7 +363,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) if (devnum >= SNDRV_CARDS) return -ENODEV; - err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, + err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, sizeof(struct snd_usb_caiaqdev), &card); if (err < 0) return err; @@ -386,7 +386,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; int err, len; - + if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); return -EIO; @@ -395,19 +395,19 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_init_urb(&dev->ep1_in_urb); usb_init_urb(&dev->midi_out_urb); - usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, + usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x1), - dev->ep1_in_buf, EP1_BUFSIZE, + dev->ep1_in_buf, EP1_BUFSIZE, usb_ep1_command_reply_dispatch, dev); - usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, + usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, usb_sndbulkpipe(usb_dev, 0x1), - dev->midi_out_buf, EP1_BUFSIZE, + dev->midi_out_buf, EP1_BUFSIZE, snd_usb_caiaq_midi_output_done, dev); - + init_waitqueue_head(&dev->ep1_wait_queue); init_waitqueue_head(&dev->prepare_wait_queue); - + if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0) return -EIO; @@ -420,10 +420,10 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_string(usb_dev, usb_dev->descriptor.iManufacturer, dev->vendor_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iProduct, dev->product_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iSerialNumber, dev->serial, CAIAQ_USB_STR_LEN); @@ -431,7 +431,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) c = strchr(dev->serial, ' '); if (c) *c = '\0'; - + strcpy(card->driver, MODNAME); strcpy(card->shortname, dev->product_name); @@ -449,18 +449,18 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) return 0; } -static int __devinit snd_probe(struct usb_interface *intf, +static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - + ret = create_card(device, &card); - + if (ret < 0) return ret; - + usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { @@ -468,7 +468,7 @@ static int __devinit snd_probe(struct usb_interface *intf, snd_card_free(card); return ret; } - + return 0; } @@ -489,10 +489,10 @@ static void snd_disconnect(struct usb_interface *intf) snd_usb_caiaq_input_free(dev); #endif snd_usb_caiaq_audio_free(dev); - + usb_kill_urb(&dev->ep1_in_urb); usb_kill_urb(&dev->midi_out_urb); - + snd_card_free(card); usb_reset_device(interface_to_usbdev(intf)); } diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 8fa8cd8..538e8c0 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -40,7 +40,7 @@ static void snd_usb_caiaq_midi_input_trigger(struct snd_rawmidi_substream *subst if (!dev) return; - + dev->midi_receive_substream = up ? substream : NULL; } @@ -64,18 +64,18 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, struct snd_rawmidi_substream *substream) { int len, ret; - + dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; dev->midi_out_buf[1] = 0; /* port */ len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, EP1_BUFSIZE - 3); - + if (len <= 0) return; - + dev->midi_out_buf[2] = len; dev->midi_out_urb.transfer_buffer_length = len+3; - + ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); if (ret < 0) log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," @@ -88,7 +88,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - + if (up) { dev->midi_out_substream = substream; if (!dev->midi_out_active) @@ -113,12 +113,12 @@ static struct snd_rawmidi_ops snd_usb_caiaq_midi_input = .trigger = snd_usb_caiaq_midi_input_trigger, }; -void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, +void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len) { if (!dev->midi_receive_substream) return; - + snd_rawmidi_receive(dev->midi_receive_substream, buf, len); } @@ -142,16 +142,16 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (device->spec.num_midi_out > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usb_caiaq_midi_output); } if (device->spec.num_midi_in > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usb_caiaq_midi_input); } - + device->rmidi = rmidi; return 0; @@ -160,7 +160,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { struct snd_usb_caiaqdev *dev = urb->context; - + dev->midi_out_active = 0; if (urb->status != 0) return; -- 1.6.3.1

3 23

Re: [alsa-devel] ASoC: TWL4030 driver support multi channel playback
by Naveen Krishna Ch 02 Jun '09

02 Jun '09

2009/6/2 Mark Brown <broonie(a)opensource.wolfsonmicro.com> > On Tue, Jun 02, 2009 at 03:24:11PM +0530, Naveen Krishna Ch wrote: > > > I am done with the entire back porting of ASoC. > > > > I can see the dapm widgets getting ON/OFF and codec registers being > changed. > > > > But, there is no audio out. Am i missing somthing > > > > A direct aplay should play the audio on the headset, right? > > I am not familiar with the TWL4030, you should ask these questions on > the list. I hit reply (Instead of replyall) Sorry, > As a first step I would check that there is data being > transferred over the digital link to the CODEC, then look at the > audio routing to ensure that the signal is routed to the output. Thanks for suggestions.. -- Cheers (: Naveen Krishna Ch :)

2 1

[alsa-devel] Palm27x-asoc: fix it to work without jack gpio specified
by Marek Vasut 02 Jun '09

02 Jun '09

Hi, the patch follows. The problem was with palmz72 which doesn't have the jack detection.

2 3

[alsa-devel] ASoC: TWL4030 driver support multi channel playback
by Naveen Krishna Ch 02 Jun '09

02 Jun '09

Hi Peter, I am using linux 2.6.28 kernel, there is a requirement to support multi (4) channel play back . I guess TWL4030 ASoC driver now has full fledged support for both the DAC1(TX1) and DAC2(TX2) paths in option 1 of codec operation. I am looking forward to back port the ASoC stuff drivers from 2.6.30 rc7 to 2.6.28. Is there any better way of meeting the requirement with out back porting the stuff Suggestions please. Sorry for the english,, -- Cheers (: Naveen Krishna Ch :)

3 4

[PATCH] ALSA: introduce snd_card_make_id_unique()
by Daniel Mack 02 Jun '09

02 Jun '09

Sound card drivers may choose their own way to provide a nice sound card ID, especially when the default behaviour taken by choose_default_name() is not appropriate. ALSA's IDs must be unique, so there is need for some core logic to ensure that. This patch adds the function snd_card_make_id_unique() to the ALSA core which lowlevel drivers can call to make sure the computed string isn't already taken. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/core.h | 1 + sound/core/init.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..01ea816 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -319,6 +319,7 @@ int snd_card_info_done(void); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); +void snd_card_make_id_unique(struct snd_card *card); #ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/sound/core/init.c b/sound/core/init.c index fd56afe..8503a01 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -547,6 +547,51 @@ static void choose_default_id(struct snd_card *card) } } +static int +card_id_is_ambiguous(struct snd_card *card, const char *id) +{ + int i; + + for (i = 0; i < snd_ecards_limit; i++) { + /* only run for exisiting cards but not for the + * one in question */ + if (i == card->number || !snd_cards[i]) + continue; + + if (strcmp(snd_cards[i]->id, id) == 0) + return 1; + } + + return 0; +} + +void +snd_card_make_id_unique(struct snd_card *card) +{ + int count = 1; + char id[sizeof(card->id)]; + + if (*card->id == '\0') + return; + + strlcpy(id, card->id, sizeof(id)); + + while (card_id_is_ambiguous(card, id)) { + /* As we need to add some characters to the id, make + * sure there's enough room for it */ + int maxlen = sizeof(card->id) - (2 + (count / 10)); + + if (strlen(card->id) > maxlen) + card->id[maxlen] = '\0'; + + snprintf(id, sizeof(id), "%s_%d", card->id, count); + count++; + } + + strlcpy(card->id, id, sizeof(id)); +} +EXPORT_SYMBOL_GPL(snd_card_make_id_unique); + #ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, -- 1.6.3.1

1 0

[PATCH] ALSA: introduce snd_card_make_id_unique()
by Daniel Mack 02 Jun '09

02 Jun '09

Sound card drivers may choose their own way to provide a nice sound card ID, especially when the default behaviour taken by choose_default_name() is not appropriate. ALSA's IDs must be unique, so there is need for some core logic to ensure that. This patch adds the function snd_card_make_id_unique() to the ALSA core which lowlevel drivers can call to make sure the computed string isn't already taken. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/core.h | 1 + sound/core/init.c | 53 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 54 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..01ea816 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -319,6 +319,7 @@ int snd_card_info_done(void); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); +void snd_card_make_id_unique(struct snd_card *card); #ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/sound/core/init.c b/sound/core/init.c index fd56afe..bce2982 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -547,6 +547,59 @@ static void choose_default_id(struct snd_card *card) } } +static int +card_id_is_ambiguous(struct snd_card *card, const char *id) +{ + int i; + + for (i = 0; i < snd_ecards_limit; i++) { + /* only run for exisiting cards but not for the + * one in question */ + if (i == card->number || !snd_cards[i]) + continue; + + if (strcmp(snd_cards[i]->id, id) == 0) + return 1; + } + + return 0; +} + +void +snd_card_make_id_unique(struct snd_card *card) +{ + int count = 1; + char id[sizeof(card->id)]; + + if (*card->id == '\0') + return; + + /* if there's a reserved word in the card's chosen id, reset it + * to empty. choose_default_id() will then take care to generate + * an id for the card at a later point */ + if (!snd_info_check_reserved_words(card->id)) { + *card->id = '\0'; + return; + } + + strlcpy(id, card->id, sizeof(id)); + + while (card_id_is_ambiguous(card, id)) { + /* As we need to add some characters to the id, make + * sure there's enough room for it */ + int maxlen = sizeof(card->id) - (2 + (count / 10)); + + if (strlen(card->id) > maxlen) + card->id[maxlen] = '\0'; + + snprintf(id, sizeof(id), "%s_%d", card->id, count); + count++; + } + + strlcpy(card->id, id, sizeof(id)); +} +EXPORT_SYMBOL_GPL(snd_card_make_id_unique); + #ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, -- 1.6.3.1

1 0

[alsa-devel] [PATCH] Acer Aspire 8930G support
by Hector Martin 02 Jun '09

02 Jun '09

Getting this one to work has annoyed me to no end. Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- alsa-driver-1.0.20.8.g6254b.219.g5b2b1/alsa-kernel/pci/hda/patch_realtek_orig.c 2009-06-02 07:48:47.000000000 +0200 +++ alsa-driver-1.0.20.8.g6254b.219.g5b2b1/alsa-kernel/pci/hda/patch_realtek.c 2009-06-02 08:32:29.000000000 +0200 @@ -223,6 +223,7 @@ ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, @@ -313,6 +314,8 @@ const struct hda_channel_mode *channel_mode; int num_channel_mode; int need_dac_fix; + int const_channel_count; + int ext_channel_count; /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ @@ -368,6 +371,7 @@ unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; int need_dac_fix; + int const_channel_count; unsigned int num_mux_defs; const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); @@ -462,7 +466,7 @@ struct alc_spec *spec = codec->spec; return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - spec->multiout.max_channels); + spec->ext_channel_count); } static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, @@ -472,9 +476,12 @@ struct alc_spec *spec = codec->spec; int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; + &spec->ext_channel_count); + if(err >= 0 && spec->const_channel_count == 0) { + spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; + } return err; } @@ -854,8 +861,13 @@ spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; spec->need_dac_fix = preset->need_dac_fix; + spec->const_channel_count = preset->const_channel_count; - spec->multiout.max_channels = spec->channel_mode[0].channels; + if(preset->const_channel_count == 0) + spec->multiout.max_channels = preset->const_channel_count; + else + spec->multiout.max_channels = spec->channel_mode[0].channels; + spec->ext_channel_count = spec->channel_mode[0].channels; spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; @@ -1456,6 +1468,44 @@ { } }; +/* + * ALC888 Acer Aspire 8930G model + */ + +static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Unselect Front Mic by default in input mixer 3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Connect Internal Front to Front */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Internal Rear to Rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect Internal CLFE to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Connect HP out to Front */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Enable all DACs */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x03}, // DAC DISABLE/MUTE 1? + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, // bits 1-5, when set, disable DAC nids 0x02-0x06 apparently. Init value 0x38 + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, // DAC DISABLE/MUTE 2? + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, // some bit here disables the other DACs. Init value 0x4900 +/* Enable amplifiers */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + { } +}; + static struct hda_input_mux alc888_2_capture_sources[2] = { /* Front mic only available on one ADC */ { @@ -1508,6 +1558,17 @@ alc_automute_amp(codec); } +static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x1b; + alc_automute_amp(codec); +} + /* * ALC880 3-stack model * @@ -8708,6 +8769,7 @@ [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", + [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", [ALC883_LAPTOP_EAPD] = "laptop-eapd", @@ -8746,6 +8808,8 @@ ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", + ALC888_ACER_ASPIRE_8930G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -8959,6 +9023,27 @@ .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_8930G] = { + .mixers = { alc888_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_8930g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .const_channel_count = 6, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_2_capture_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_8930g_init_hook, + }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, alc883_chmode_mixer },

2 5

[alsa-devel] SPDIF/AES input sample rate handling?
by Robert Hancock 02 Jun '09

02 Jun '09

A question about how ALSA drivers should handle SPDIF or AES digital audio inputs: I may be eventually be writing an ALSA driver for an audio card which has multiple AES digital audio inputs. The card would support multiple sample rates on input with the provided input data being fed directly to software with no sample rate conversion. This raises the question of how to handle requests for a particular sample rate. If the user app requests recording at a sample rate of say 32KHz and we say "OK", and then they proceed to connect an input running at 44.1KHz sampling rate, the card will produce and the app will receive samples at a faster than expected rate. Presumably this is something the app just has to deal with, but will anything in the ALSA core/library puke because of it? Or is there a better way to deal with this kind of situation?

2 1

Re: [alsa-devel] Macbook[Pro] 5 6ch support
by Takashi Iwai 02 Jun '09

02 Jun '09

At Mon, 1 Jun 2009 20:21:27 +0200, Kacper Szczesniak wrote: > > On Mon, Jun 1, 2009 at 6:54 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 1 Jun 2009 15:36:35 +0200, > > Kacper Szczesniak wrote: > >> > >> On Mon, Jun 1, 2009 at 10:54 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > Hm, this patch changes the channel mapping completely. > >> > Is it intentional? > >> yes, 0xc/0xd/0xe for Front/LFE/Surr > >> > >> >> static struct snd_kcontrol_new alc885_mb5_mixer[] = { > >> >> - HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x00, HDA_OUTPUT), > >> >> - HDA_BIND_MUTE ("Front Playback Switch", 0x0d, 0x02, HDA_INPUT), > >> >> - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), > >> >> - HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), > >> >> + HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), > >> >> + HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), > >> >> HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), > >> >> HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), > >> > > >> > > >> > I'm wondering whether this breaks MB5, not MB5.1. If it's MB5.1 > >> > specific, please create another model instead of breaking the old > >> > one. > >> it's confirmed to work on MB5.1 MBP5.1 MBP52 > > > > Hrm, then I wonder why the old mb5 model works as is although the > > mapping is totally different... > > /* Front mixer */ > - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > - /* LineOut mixer */ > {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > - /* Front Pin: output 0 (0x0d) */ > + /* Surround mixer */ > + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > + /* LFE mixer */ > + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > + /* HP mixer */ > + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, > + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > + /* Front Pin (0x0c) */ > {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, > {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, > - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, > - /* HP Pin: output 0 (0x0c) */ > + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, > + /* LFE Pin (0x0e) */ > + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, > + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, > + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, > + /* HP Pin (0x0f) */ > {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, > {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, > - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, > + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, > > connections are different also... I've changed mixer names and connected PINs to > the right mixer nodes. Fair enough. I merged the patch now to sound git tree. Let me know if you see still any problem. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - fix audio on LG R510
by Ozan Çağlayan 02 Jun '09

02 Jun '09

Currently, LG R510 is only able to produce sound on headphones, the internal speakers are not working. The user tested and confirmed that with model=Dell headphones, internal speakers and the microphone are working flawlessly. Tested-by: Serdar Soytetir <tulliana(a)gmail.com> Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0fd258e..a2133ac 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12063,6 +12063,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), + SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), -- 1.6.2

2 2

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Additional check for 4 channel mode
by Peter Ujfalusi 02 Jun '09

02 Jun '09

Hello, the following patch adds additional check in case of four channel mode. The codec has to be configured for TDM (DSP_A) mode when using 4 channel audio. Without the patch boards which does not prepared for the four channel mode could try to play four channel audio, but configure the interface to be in I2S mode, which is not valid. --- Peter Ujfalusi (1): ASoC: TWL4030: Check the interface format for 4 channel mode sound/soc/codecs/twl4030.c | 12 +++++++++--- 1 files changed, 9 insertions(+), 3 deletions(-)

2 2

[alsa-devel] [PATCH] alsa-info.sh: Provide system manufacturer and product name from DMI
by Ozan Çağlayan 02 Jun '09

02 Jun '09

This commit adds system manufacturer and product name informations acquired using dmidecode to the output of the alsa-info script. Note that those informations will only be available when dmidecode utility is installed and alsa-info is run with root privileges. Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- utils/alsa-info.sh | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index f1660f8..4be4e05 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -400,6 +400,12 @@ ESDINST=$(which esd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) PAINST=$(which pulseaudio 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) ARTSINST=$(which artsd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) JACKINST=$(which jackd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) +DMIDECODE=$(which dmidecode 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) + +if [ -x $DMIDECODE ]; then + DMI_SYSTEM_MANUFACTURER=$($DMIDECODE -s system-manufacturer 2>/dev/null) + DMI_SYSTEM_PRODUCT_NAME=$($DMIDECODE -s system-product-name 2>/dev/null) +fi cat /proc/asound/modules 2>/dev/null|awk {'print $2'}>$TEMPDIR/alsamodules.tmp cat /proc/asound/cards >$TEMPDIR/alsacards.tmp @@ -431,6 +437,13 @@ echo "" >> $FILE echo $DISTRO >> $FILE echo "" >> $FILE echo "" >> $FILE +echo "!!DMI Information" >> $FILE +echo "!!---------------" >> $FILE +echo "" >> $FILE +echo "Manufacturer: $DMI_SYSTEM_MANUFACTURER" >> $FILE +echo "Product Name: $DMI_SYSTEM_PRODUCT_NAME" >> $FILE +echo "" >> $FILE +echo "" >> $FILE echo "!!Kernel Information" >> $FILE echo "!!------------------" >> $FILE echo "" >> $FILE -- 1.6.3

2 3

Re: [alsa-devel] Macbook[Pro] 5 6ch support
by Takashi Iwai 01 Jun '09

01 Jun '09

At Mon, 1 Jun 2009 15:36:35 +0200, Kacper Szczesniak wrote: > > On Mon, Jun 1, 2009 at 10:54 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > Hm, this patch changes the channel mapping completely. > > Is it intentional? > yes, 0xc/0xd/0xe for Front/LFE/Surr > > >> static struct snd_kcontrol_new alc885_mb5_mixer[] = { > >> - HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x00, HDA_OUTPUT), > >> - HDA_BIND_MUTE ("Front Playback Switch", 0x0d, 0x02, HDA_INPUT), > >> - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), > >> - HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), > >> + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), > >> + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), > >> + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), > >> + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), > >> + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), > >> + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), > >> + HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), > >> + HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), > >> HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), > >> HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), > > > > > > I'm wondering whether this breaks MB5, not MB5.1. If it's MB5.1 > > specific, please create another model instead of breaking the old > > one. > it's confirmed to work on MB5.1 MBP5.1 MBP52 Hrm, then I wonder why the old mb5 model works as is although the mapping is totally different... Takashi

1 0

[alsa-devel] [PATCH/RFC] alsa-info: add DMI data to the alsa-info output
by Ozan Çağlayan 01 Jun '09

01 Jun '09

This is a working but not a clean/good patch for doing that because I'm not very good at shell scripting but I'm posting it just to give a possible idea. dmidecode should be run as root because it reads /dev/mem. If alsa-info is run as root, providing at least the system-manufacturer and system-product-name in the output will be very useful for distro maintainers. Thanks, Ozan Caglayan Index: alsa-info.sh =================================================================== --- alsa-info.sh (revision 69253) +++ alsa-info.sh (working copy) @@ -361,6 +361,18 @@ ARTSINST=$(which artsd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) JACKINST=$(which jackd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) +DMIDECODE=$(which dmidecode) +DMI_SYSTEM_MANUFACTURER="Not provided" +DMI_SYSTEM_PRODUCT_NAME="Not provided" + +if [ -f $DMIDECODE ]; then + $DMIDECODE -q &> /dev/null + if [ x$? != "x1" ]; then + DMI_SYSTEM_MANUFACTURER=$($DMIDECODE -s system-manufacturer) + DMI_SYSTEM_PRODUCT_NAME=$($DMIDECODE -s system-product-name) + fi +fi + cat /proc/asound/modules 2>/dev/null|awk {'print $2'}>$TEMPDIR/alsamodules.tmp cat /proc/asound/cards >$TEMPDIR/alsacards.tmp lspci |grep -i "multi\|audio">$TEMPDIR/lspci.tmp @@ -391,6 +403,13 @@ echo $DISTRO >> $FILE echo "" >> $FILE echo "" >> $FILE +echo "!!DMI Information" >> $FILE +echo "!!---------------" >> $FILE +echo "" >> $FILE +echo "Manufacturer: $DMI_SYSTEM_MANUFACTURER" >> $FILE +echo "Product Name: $DMI_SYSTEM_PRODUCT_NAME" >> $FILE +echo "" >> $FILE +echo "" >> $FILE echo "!!Kernel Information" >> $FILE echo "!!------------------" >> $FILE echo "" >> $FILE

3 3

[alsa-devel] [PATCH] snd-hda-intel: Jack Mode changes for Sigmatel boards
by Nickolas Lloyd 01 Jun '09

01 Jun '09

This patch changes Line In as Out Switch and Mic In as Out Switch to enums for consistency, and causes all mic ports to be probed and controls to be added appropriately. Signed-off-by: Nickolas Lloyd <ultrageek.lloyd(a)gmail.com> Note: diffed against alsa-driver-20090520 ---- --- alsa-driver/sound/pci/hda/patch_sigmatel.c 2009-05-19 22:05:02.000000000 +0000 +++ alsa-driver.a/sound/pci/hda/patch_sigmatel.c 2009-05-20 12:17:49.000000000 +0000 @@ -668,62 +668,6 @@ static unsigned int stac92xx_vref_get(st return vref; } -static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int new_vref; - unsigned int error; - - if (ucontrol->value.enumerated.item[0] == 0) - new_vref = AC_PINCTL_VREF_80; - else if (ucontrol->value.enumerated.item[0] == 1) - new_vref = AC_PINCTL_VREF_GRD; - else - new_vref = AC_PINCTL_VREF_HIZ; - - if (new_vref != stac92xx_vref_get(codec, kcontrol->private_value)) { - error = stac92xx_vref_set(codec, - kcontrol->private_value, new_vref); - return error; - } - - return 0; -} - -static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int vref = stac92xx_vref_get(codec, kcontrol->private_value); - if (vref == AC_PINCTL_VREF_80) - ucontrol->value.enumerated.item[0] = 0; - else if (vref == AC_PINCTL_VREF_GRD) - ucontrol->value.enumerated.item[0] = 1; - else if (vref == AC_PINCTL_VREF_HIZ) - ucontrol->value.enumerated.item[0] = 2; - - return 0; -} - -static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static char *texts[] = { - "Mic In", "Line In", "Line Out" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = 3; - uinfo->count = 1; - if (uinfo->value.enumerated.item >= 3) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; -} - static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -2695,7 +2639,93 @@ static int stac92xx_hp_switch_put(struct return 1; } -#define stac92xx_io_switch_info snd_ctl_boolean_mono_info +static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int i; + static char *texts[] = { + "Mic In", "Line In", "Line Out" + }; + + if (kcontrol->private_value & 0x100) + i = 3; + else + i = 2; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = i; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= i) + uinfo->value.enumerated.item = i-1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + +static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0x0ff; + unsigned int vref = stac92xx_vref_get(codec, nid); + + if (vref == stac92xx_get_vref(codec, nid)) + ucontrol->value.enumerated.item[0] = 0; + else if (vref == AC_PINCTL_VREF_GRD) + ucontrol->value.enumerated.item[0] = 1; + else if (vref == AC_PINCTL_VREF_HIZ) + ucontrol->value.enumerated.item[0] = 2; + + return 0; +} + +static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int new_vref = 0; + unsigned int error; + hda_nid_t nid = (kcontrol->private_value & 0x0ff); + + if (ucontrol->value.enumerated.item[0] == 0) + new_vref = stac92xx_get_vref(codec, nid); + else if (ucontrol->value.enumerated.item[0] == 1) + new_vref = AC_PINCTL_VREF_GRD; + else if (ucontrol->value.enumerated.item[0] == 2) + new_vref = AC_PINCTL_VREF_HIZ; + else + return 1; + + if (new_vref != stac92xx_vref_get(codec, nid)) { + error = stac92xx_vref_set(codec, nid, new_vref); + return error; + } + + return 0; +} + + + +static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2]; + if (kcontrol->private_value & 0xff) + texts[0] = "Mic In"; + else + texts[0] = "Line In"; + texts[1] = "Line Out"; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = 2; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= 2) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2703,7 +2733,7 @@ static int stac92xx_io_switch_get(struct struct sigmatel_spec *spec = codec->spec; int io_idx = kcontrol-> private_value & 0xff; - ucontrol->value.integer.value[0] = spec->io_switch[io_idx]; + ucontrol->value.enumerated.item[0] = spec->io_switch[io_idx]; return 0; } @@ -2713,7 +2743,7 @@ static int stac92xx_io_switch_put(struct struct sigmatel_spec *spec = codec->spec; hda_nid_t nid = kcontrol->private_value >> 8; int io_idx = kcontrol-> private_value & 0xff; - unsigned short val = !!ucontrol->value.integer.value[0]; + unsigned short val = !!ucontrol->value.enumerated.item[0]; spec->io_switch[io_idx] = val; @@ -3242,7 +3272,10 @@ static int stac92xx_auto_create_multi_ou const struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid; + int def_conf; int err; + int idx; err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, spec->multiout.dac_nids, @@ -3261,18 +3294,34 @@ static int stac92xx_auto_create_multi_ou if (spec->line_switch) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, - "Line In as Output Switch", + "Line In Jack Mode", spec->line_switch << 8); if (err < 0) return err; } - if (spec->mic_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_DC_BIAS, - "Mic Jack Mode", - spec->mic_switch); - if (err < 0) - return err; + nid = cfg->input_pins[AUTO_PIN_MIC]; + idx = 0; +again: + if (nid) { + def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (!((get_defcfg_connect(def_conf)) & AC_JACK_PORT_FIXED)) { + if (snd_hda_query_pin_caps(codec, nid) + & (AC_PINCAP_VREF_GRD << AC_PINCAP_VREF_SHIFT)) + stac92xx_add_control_idx(spec, + STAC_CTL_WIDGET_DC_BIAS, idx++, + "Mic Jack Mode", nid + | ((nid == spec->mic_switch) + ? 0x100 : 0)); + else if (nid == spec->mic_switch) + stac92xx_add_control_idx(spec, + STAC_CTL_WIDGET_IO_SWITCH, idx++, + "Mic Jack Mode", ((nid << 8) | 1)); + } + } + if (nid == cfg->input_pins[AUTO_PIN_MIC]) { + nid = cfg->input_pins[AUTO_PIN_FRONT_MIC]; + goto again; } return 0; --- alsa-driver/sound/pci/hda/patch_sigmatel.c 2009-05-19 22:05:02.000000000 +0000 +++ alsa-driver.a/sound/pci/hda/patch_sigmatel.c 2009-05-20 12:17:49.000000000 +0000 @@ -668,62 +668,6 @@ static unsigned int stac92xx_vref_get(st return vref; } -static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int new_vref; - unsigned int error; - - if (ucontrol->value.enumerated.item[0] == 0) - new_vref = AC_PINCTL_VREF_80; - else if (ucontrol->value.enumerated.item[0] == 1) - new_vref = AC_PINCTL_VREF_GRD; - else - new_vref = AC_PINCTL_VREF_HIZ; - - if (new_vref != stac92xx_vref_get(codec, kcontrol->private_value)) { - error = stac92xx_vref_set(codec, - kcontrol->private_value, new_vref); - return error; - } - - return 0; -} - -static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int vref = stac92xx_vref_get(codec, kcontrol->private_value); - if (vref == AC_PINCTL_VREF_80) - ucontrol->value.enumerated.item[0] = 0; - else if (vref == AC_PINCTL_VREF_GRD) - ucontrol->value.enumerated.item[0] = 1; - else if (vref == AC_PINCTL_VREF_HIZ) - ucontrol->value.enumerated.item[0] = 2; - - return 0; -} - -static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static char *texts[] = { - "Mic In", "Line In", "Line Out" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = 3; - uinfo->count = 1; - if (uinfo->value.enumerated.item >= 3) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; -} - static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -2695,7 +2639,93 @@ static int stac92xx_hp_switch_put(struct return 1; } -#define stac92xx_io_switch_info snd_ctl_boolean_mono_info +static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int i; + static char *texts[] = { + "Mic In", "Line In", "Line Out" + }; + + if (kcontrol->private_value & 0x100) + i = 3; + else + i = 2; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = i; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= i) + uinfo->value.enumerated.item = i-1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + +static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0x0ff; + unsigned int vref = stac92xx_vref_get(codec, nid); + + if (vref == stac92xx_get_vref(codec, nid)) + ucontrol->value.enumerated.item[0] = 0; + else if (vref == AC_PINCTL_VREF_GRD) + ucontrol->value.enumerated.item[0] = 1; + else if (vref == AC_PINCTL_VREF_HIZ) + ucontrol->value.enumerated.item[0] = 2; + + return 0; +} + +static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int new_vref = 0; + unsigned int error; + hda_nid_t nid = (kcontrol->private_value & 0x0ff); + + if (ucontrol->value.enumerated.item[0] == 0) + new_vref = stac92xx_get_vref(codec, nid); + else if (ucontrol->value.enumerated.item[0] == 1) + new_vref = AC_PINCTL_VREF_GRD; + else if (ucontrol->value.enumerated.item[0] == 2) + new_vref = AC_PINCTL_VREF_HIZ; + else + return 1; + + if (new_vref != stac92xx_vref_get(codec, nid)) { + error = stac92xx_vref_set(codec, nid, new_vref); + return error; + } + + return 0; +} + + + +static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2]; + if (kcontrol->private_value & 0xff) + texts[0] = "Mic In"; + else + texts[0] = "Line In"; + texts[1] = "Line Out"; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = 2; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= 2) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2703,7 +2733,7 @@ static int stac92xx_io_switch_get(struct struct sigmatel_spec *spec = codec->spec; int io_idx = kcontrol-> private_value & 0xff; - ucontrol->value.integer.value[0] = spec->io_switch[io_idx]; + ucontrol->value.enumerated.item[0] = spec->io_switch[io_idx]; return 0; } @@ -2713,7 +2743,7 @@ static int stac92xx_io_switch_put(struct struct sigmatel_spec *spec = codec->spec; hda_nid_t nid = kcontrol->private_value >> 8; int io_idx = kcontrol-> private_value & 0xff; - unsigned short val = !!ucontrol->value.integer.value[0]; + unsigned short val = !!ucontrol->value.enumerated.item[0]; spec->io_switch[io_idx] = val; @@ -3242,7 +3272,10 @@ static int stac92xx_auto_create_multi_ou const struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid; + int def_conf; int err; + int idx; err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, spec->multiout.dac_nids, @@ -3261,18 +3294,34 @@ static int stac92xx_auto_create_multi_ou if (spec->line_switch) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, - "Line In as Output Switch", + "Line In Jack Mode", spec->line_switch << 8); if (err < 0) return err; } - if (spec->mic_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_DC_BIAS, - "Mic Jack Mode", - spec->mic_switch); - if (err < 0) - return err; + nid = cfg->input_pins[AUTO_PIN_MIC]; + idx = 0; +again: + if (nid) { + def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (!((get_defcfg_connect(def_conf)) & AC_JACK_PORT_FIXED)) { + if (snd_hda_query_pin_caps(codec, nid) + & (AC_PINCAP_VREF_GRD << AC_PINCAP_VREF_SHIFT)) + stac92xx_add_control_idx(spec, + STAC_CTL_WIDGET_DC_BIAS, idx++, + "Mic Jack Mode", nid + | ((nid == spec->mic_switch) + ? 0x100 : 0)); + else if (nid == spec->mic_switch) + stac92xx_add_control_idx(spec, + STAC_CTL_WIDGET_IO_SWITCH, idx++, + "Mic Jack Mode", ((nid << 8) | 1)); + } + } + if (nid == cfg->input_pins[AUTO_PIN_MIC]) { + nid = cfg->input_pins[AUTO_PIN_FRONT_MIC]; + goto again; } return 0;

2 9

[alsa-devel] Roland/Edirol M-16DX
by Lasse Kärkkäinen 01 Jun '09

01 Jun '09

This device doesn't seem to be supported yet. Does Roland make the specifications available, etc? The device is not compatible with USB Audio, but rather uses Vendor Specific Class. lsusb -vvv is attached below and if you need more information, I can debug the device on Linux and on Vista. Bus 007 Device 025: ID 0582:00c4 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x00c4 bcdDevice 0.00 iManufacturer 1 EDIROL iProduct 2 M-16DX iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 12 04 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x01f8 1x 504 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 01 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1

4 10

[alsa-devel] [PATCH] Support NVIDIA 8 channel HDMI audio
by Wei Ni 01 Jun '09

01 Jun '09

Hi, Takashi I'm a Nvidia engineer, we wish to support 8 channel HDMI audio for MCP78/7A. I attached the patch file, it based on the latest sound git tree, please check it. <<0001-Support-NVIDIA-8-channel-HDMI-audio.patch>> Thanks Wei. ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

3 4

[alsa-devel] Macbook[Pro] 5 6ch support
by Kacper Szczesniak 01 Jun '09

01 Jun '09

Hi, this is a patch against current snapshot that adds: 6 channels support for the MB5 model Kacper Szczesniak <kacper(a)qwe.pl>

2 1

Re: [alsa-devel] Recent breakages (was: (no subject))
by James Gardiner 31 May '09

31 May '09

--- On Sun, 5/31/09, Takashi Iwai <tiwai(a)suse.de> wrote: > OK, I found out the culprit. It was a typo in my last > patch. > Fixed now. > > Sorry for inconvenience. > > > Takashi > Working now! Thank you! James Gardiner

1 0

[alsa-devel] Recent breakages (was: (no subject))
by renidragsemaj＠yahoo.com 31 May '09

31 May '09

--- On Sat, 5/30/09, Takashi Iwai <tiwai(a)suse.de> wrote: >> Hi, someone recently submitted changes to the patch_sigmatel.c codec >> source. Whatever was done is breaking my patch for the HP dv4-1222nr >> laptop. I also strongly suspect that other code in the stac92hd71bxx >> function has been affected. There is no autodetect, there is no >> option for modprobe that will load that codec currently, to the best >> of my knowledge. I've tracked the difference to snapshot >> 20090529. Before that, I can get it to work, there and after refuses >> to work properly. > >Could you try to copy hda_intel.c from 1.0.20 and check whether this >fixes the problem? Most likely I broke something. You seem to have that right. I changed out that file like you said and it went back to working. Now, only to find out what happened... >> >> Also, there was a recent change to the >> stac92xx_hp_check_power_status function: While the change looks good >> for its intended purpose of localizing board changes to the >> switch/case statements in the stac92hd71bxx function, the >> implementor reversed the if/then selector: > >Fixed now. Thanks for checking! Thank you. It seems that I had two big booboos with this part of my post. 1) Sorry about all those question marks: It looks like my cut and paste job didn't go the way I planned. 2) I messed up and switched the old and new code in my post, but it looks like you caught my screw-up. The recent snapshot (20090531) has it right. *Don't change it back!* :) Thank you, James Gardiner

2 1

[alsa-devel] [PATCH 1/5] ALSA: snd_usb_caiaq: clean whitespaces
by Daniel Mack 31 May '09

31 May '09

Cosmetic changes only, no code change. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 88 +++++++++++++++++++++++----------------------- sound/usb/caiaq/device.c | 60 ++++++++++++++++---------------- sound/usb/caiaq/midi.c | 24 ++++++------ 3 files changed, 86 insertions(+), 86 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b13ce76..b144513 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -42,10 +42,10 @@ (stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1) static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S24_3BE, - .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 0, /* will overwrite later */ @@ -68,7 +68,7 @@ activate_substream(struct snd_usb_caiaqdev *dev, dev->sub_capture[sub->number] = sub; } -static void +static void deactivate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { @@ -118,7 +118,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) return -EPIPE; } } - + return 0; } @@ -129,7 +129,7 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) debug("%s(%p)\n", __func__, dev); if (!dev->streaming) return; - + dev->streaming = 0; for (i = 0; i < N_URBS; i++) { @@ -154,7 +154,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); if (all_substreams_zero(dev->sub_playback) && all_substreams_zero(dev->sub_capture)) { - /* when the last client has stopped streaming, + /* when the last client has stopped streaming, * all sample rates are allowed again */ stream_stop(dev); dev->pcm_info.rates = dev->samplerates; @@ -194,7 +194,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; debug("%s(%p)\n", __func__, substream); - + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; @@ -205,19 +205,19 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) if (dev->streaming) return 0; - + /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ for (i=0; i < ARRAY_SIZE(rates); i++) if (runtime->rate == rates[i]) dev->pcm_info.rates = 1 << i; - + snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; if (dev->spec.data_alignment == 2) bytes_per_sample++; - + bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams; @@ -232,7 +232,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) ret = stream_start(dev); if (ret) return ret; - + dev->output_running = 0; wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ); if (!dev->output_running) { @@ -273,7 +273,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) return SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + return bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else return bytes_to_frames(sub->runtime, @@ -291,7 +291,7 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer }; - + static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream **subs) { @@ -333,7 +333,7 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] + audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) @@ -354,14 +354,14 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, for (i = 0; i < iso->actual_length;) { if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { - for (stream = 0; - stream < dev->n_streams; + for (stream = 0; + stream < dev->n_streams; stream++, i++) { if (dev->first_packet) continue; check_byte = MAKE_CHECKBYTE(dev, stream, i); - + if ((usb_buf[i] & 0x3f) != check_byte) dev->input_panic = 1; @@ -410,21 +410,21 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, } if ((dev->input_panic || dev->output_panic) && !dev->warned) { - debug("streaming error detected %s %s\n", + debug("streaming error detected %s %s\n", dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); dev->warned = 1; } } -static void fill_out_urb(struct snd_usb_caiaqdev *dev, - struct urb *urb, +static void fill_out_urb(struct snd_usb_caiaqdev *dev, + struct urb *urb, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; int stream, i; - + for (i = 0; i < iso->length;) { for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; @@ -444,7 +444,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && - i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == + i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); @@ -453,7 +453,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, static void read_completed(struct urb *urb) { - struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; @@ -478,7 +478,7 @@ static void read_completed(struct urb *urb) out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; - + if (len > 0) { spin_lock(&dev->spinlock); fill_out_urb(dev, out, &out->iso_frame_desc[outframe]); @@ -497,14 +497,14 @@ static void read_completed(struct urb *urb) out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } - + /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; urb->iso_frame_desc[frame].length = BYTES_PER_FRAME; urb->iso_frame_desc[frame].actual_length = 0; } - + urb->number_of_packets = FRAMES_PER_URB; urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); @@ -528,7 +528,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) struct usb_device *usb_dev = dev->chip.dev; unsigned int pipe; - pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? + pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) : usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE); @@ -547,25 +547,25 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) return urbs; } - urbs[i]->transfer_buffer = + urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { log("unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } - + for (frame = 0; frame < FRAMES_PER_URB; frame++) { - struct usb_iso_packet_descriptor *iso = + struct usb_iso_packet_descriptor *iso = &urbs[i]->iso_frame_desc[frame]; - + iso->offset = BYTES_PER_FRAME * frame; iso->length = BYTES_PER_FRAME; } - + urbs[i]->dev = usb_dev; urbs[i]->pipe = pipe; - urbs[i]->transfer_buffer_length = FRAMES_PER_URB + urbs[i]->transfer_buffer_length = FRAMES_PER_URB * BYTES_PER_FRAME; urbs[i]->context = &dev->data_cb_info[i]; urbs[i]->interval = 1; @@ -589,7 +589,7 @@ static void free_urbs(struct urb **urbs) for (i = 0; i < N_URBS; i++) { if (!urbs[i]) continue; - + usb_kill_urb(urbs[i]); kfree(urbs[i]->transfer_buffer); usb_free_urb(urbs[i]); @@ -602,11 +602,11 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) { int i, ret; - dev->n_audio_in = max(dev->spec.num_analog_audio_in, - dev->spec.num_digital_audio_in) / + dev->n_audio_in = max(dev->spec.num_analog_audio_in, + dev->spec.num_digital_audio_in) / CHANNELS_PER_STREAM; dev->n_audio_out = max(dev->spec.num_analog_audio_out, - dev->spec.num_digital_audio_out) / + dev->spec.num_digital_audio_out) / CHANNELS_PER_STREAM; dev->n_streams = max(dev->n_audio_in, dev->n_audio_out); @@ -619,7 +619,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) return -EINVAL; } - ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, + ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, dev->n_audio_out, dev->n_audio_in, &dev->pcm); if (ret < 0) { @@ -632,7 +632,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); - + memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware, sizeof(snd_usb_caiaq_pcm_hardware)); @@ -651,9 +651,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) break; } - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usb_caiaq_ops); - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); snd_pcm_lib_preallocate_pages_for_all(dev->pcm, @@ -662,7 +662,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); dev->data_cb_info = - kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, + kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); if (!dev->data_cb_info) @@ -672,14 +672,14 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; } - + dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret); if (ret < 0) { kfree(dev->data_cb_info); free_urbs(dev->data_urbs_in); return ret; } - + dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret); if (ret < 0) { kfree(dev->data_cb_info); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 515de1c..9be0f2e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -79,7 +79,7 @@ static struct usb_device_id snd_usb_id_table[] = { { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = USB_VID_NATIVEINSTRUMENTS, - .idProduct = USB_PID_RIGKONTROL2 + .idProduct = USB_PID_RIGKONTROL2 }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, @@ -197,7 +197,7 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, if (buffer && len > 0) memcpy(dev->ep1_out_buf+1, buffer, len); - + dev->ep1_out_buf[0] = command; return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), dev->ep1_out_buf, len+1, &actual_len, 200); @@ -208,7 +208,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, { int ret; char tmp[5]; - + switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; case 48000: tmp[0] = SAMPLERATE_48000; break; @@ -237,12 +237,12 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, if (ret) return ret; - - if (!wait_event_timeout(dev->ep1_wait_queue, + + if (!wait_event_timeout(dev->ep1_wait_queue, dev->audio_parm_answer >= 0, HZ)) return -EPIPE; - - if (dev->audio_parm_answer != 1) + + if (dev->audio_parm_answer != 1) debug("unable to set the device's audio params\n"); else dev->bpp = bpp; @@ -250,8 +250,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, return dev->audio_parm_answer == 1 ? 0 : -EINVAL; } -int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, - int digital, int analog, int erp) +int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, + int digital, int analog, int erp) { char tmp[3] = { digital, analog, erp }; return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG, @@ -262,7 +262,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) { int ret; char val[4]; - + /* device-specific startup specials */ switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): @@ -314,7 +314,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) dev->control_state, 1); break; } - + if (dev->spec.num_analog_audio_out + dev->spec.num_analog_audio_in + dev->spec.num_digital_audio_out + @@ -323,7 +323,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) if (ret < 0) log("Unable to set up audio system (ret=%d)\n", ret); } - + if (dev->spec.num_midi_in + dev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(dev); @@ -363,7 +363,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) if (devnum >= SNDRV_CARDS) return -ENODEV; - err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, + err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, sizeof(struct snd_usb_caiaqdev), &card); if (err < 0) return err; @@ -386,7 +386,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; int err, len; - + if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); return -EIO; @@ -395,19 +395,19 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_init_urb(&dev->ep1_in_urb); usb_init_urb(&dev->midi_out_urb); - usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, + usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x1), - dev->ep1_in_buf, EP1_BUFSIZE, + dev->ep1_in_buf, EP1_BUFSIZE, usb_ep1_command_reply_dispatch, dev); - usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, + usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, usb_sndbulkpipe(usb_dev, 0x1), - dev->midi_out_buf, EP1_BUFSIZE, + dev->midi_out_buf, EP1_BUFSIZE, snd_usb_caiaq_midi_output_done, dev); - + init_waitqueue_head(&dev->ep1_wait_queue); init_waitqueue_head(&dev->prepare_wait_queue); - + if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0) return -EIO; @@ -420,10 +420,10 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_string(usb_dev, usb_dev->descriptor.iManufacturer, dev->vendor_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iProduct, dev->product_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iSerialNumber, dev->serial, CAIAQ_USB_STR_LEN); @@ -431,7 +431,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) c = strchr(dev->serial, ' '); if (c) *c = '\0'; - + strcpy(card->driver, MODNAME); strcpy(card->shortname, dev->product_name); @@ -449,18 +449,18 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) return 0; } -static int __devinit snd_probe(struct usb_interface *intf, +static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - + ret = create_card(device, &card); - + if (ret < 0) return ret; - + usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { @@ -468,7 +468,7 @@ static int __devinit snd_probe(struct usb_interface *intf, snd_card_free(card); return ret; } - + return 0; } @@ -489,10 +489,10 @@ static void snd_disconnect(struct usb_interface *intf) snd_usb_caiaq_input_free(dev); #endif snd_usb_caiaq_audio_free(dev); - + usb_kill_urb(&dev->ep1_in_urb); usb_kill_urb(&dev->midi_out_urb); - + snd_card_free(card); usb_reset_device(interface_to_usbdev(intf)); } diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 8fa8cd8..538e8c0 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -40,7 +40,7 @@ static void snd_usb_caiaq_midi_input_trigger(struct snd_rawmidi_substream *subst if (!dev) return; - + dev->midi_receive_substream = up ? substream : NULL; } @@ -64,18 +64,18 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, struct snd_rawmidi_substream *substream) { int len, ret; - + dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; dev->midi_out_buf[1] = 0; /* port */ len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, EP1_BUFSIZE - 3); - + if (len <= 0) return; - + dev->midi_out_buf[2] = len; dev->midi_out_urb.transfer_buffer_length = len+3; - + ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); if (ret < 0) log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," @@ -88,7 +88,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - + if (up) { dev->midi_out_substream = substream; if (!dev->midi_out_active) @@ -113,12 +113,12 @@ static struct snd_rawmidi_ops snd_usb_caiaq_midi_input = .trigger = snd_usb_caiaq_midi_input_trigger, }; -void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, +void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len) { if (!dev->midi_receive_substream) return; - + snd_rawmidi_receive(dev->midi_receive_substream, buf, len); } @@ -142,16 +142,16 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (device->spec.num_midi_out > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usb_caiaq_midi_output); } if (device->spec.num_midi_in > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usb_caiaq_midi_input); } - + device->rmidi = rmidi; return 0; @@ -160,7 +160,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { struct snd_usb_caiaqdev *dev = urb->context; - + dev->midi_out_active = 0; if (urb->status != 0) return; -- 1.6.3.1

2 6

[alsa-devel] [PATCH 1/5] ALSA: snd_usb_caiaq: clean whitespaces
by Daniel Mack 30 May '09

30 May '09

Cosmetic changes only, no code change. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 88 +++++++++++++++++++++++----------------------- sound/usb/caiaq/device.c | 60 ++++++++++++++++---------------- sound/usb/caiaq/midi.c | 24 ++++++------ 3 files changed, 86 insertions(+), 86 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b13ce76..b144513 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -42,10 +42,10 @@ (stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1) static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S24_3BE, - .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 0, /* will overwrite later */ @@ -68,7 +68,7 @@ activate_substream(struct snd_usb_caiaqdev *dev, dev->sub_capture[sub->number] = sub; } -static void +static void deactivate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { @@ -118,7 +118,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) return -EPIPE; } } - + return 0; } @@ -129,7 +129,7 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) debug("%s(%p)\n", __func__, dev); if (!dev->streaming) return; - + dev->streaming = 0; for (i = 0; i < N_URBS; i++) { @@ -154,7 +154,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); if (all_substreams_zero(dev->sub_playback) && all_substreams_zero(dev->sub_capture)) { - /* when the last client has stopped streaming, + /* when the last client has stopped streaming, * all sample rates are allowed again */ stream_stop(dev); dev->pcm_info.rates = dev->samplerates; @@ -194,7 +194,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; debug("%s(%p)\n", __func__, substream); - + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; @@ -205,19 +205,19 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) if (dev->streaming) return 0; - + /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ for (i=0; i < ARRAY_SIZE(rates); i++) if (runtime->rate == rates[i]) dev->pcm_info.rates = 1 << i; - + snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; if (dev->spec.data_alignment == 2) bytes_per_sample++; - + bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams; @@ -232,7 +232,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) ret = stream_start(dev); if (ret) return ret; - + dev->output_running = 0; wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ); if (!dev->output_running) { @@ -273,7 +273,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) return SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + return bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else return bytes_to_frames(sub->runtime, @@ -291,7 +291,7 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer }; - + static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream **subs) { @@ -333,7 +333,7 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] + audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) @@ -354,14 +354,14 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, for (i = 0; i < iso->actual_length;) { if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { - for (stream = 0; - stream < dev->n_streams; + for (stream = 0; + stream < dev->n_streams; stream++, i++) { if (dev->first_packet) continue; check_byte = MAKE_CHECKBYTE(dev, stream, i); - + if ((usb_buf[i] & 0x3f) != check_byte) dev->input_panic = 1; @@ -410,21 +410,21 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, } if ((dev->input_panic || dev->output_panic) && !dev->warned) { - debug("streaming error detected %s %s\n", + debug("streaming error detected %s %s\n", dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); dev->warned = 1; } } -static void fill_out_urb(struct snd_usb_caiaqdev *dev, - struct urb *urb, +static void fill_out_urb(struct snd_usb_caiaqdev *dev, + struct urb *urb, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; int stream, i; - + for (i = 0; i < iso->length;) { for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; @@ -444,7 +444,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && - i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == + i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); @@ -453,7 +453,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, static void read_completed(struct urb *urb) { - struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; @@ -478,7 +478,7 @@ static void read_completed(struct urb *urb) out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; - + if (len > 0) { spin_lock(&dev->spinlock); fill_out_urb(dev, out, &out->iso_frame_desc[outframe]); @@ -497,14 +497,14 @@ static void read_completed(struct urb *urb) out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } - + /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; urb->iso_frame_desc[frame].length = BYTES_PER_FRAME; urb->iso_frame_desc[frame].actual_length = 0; } - + urb->number_of_packets = FRAMES_PER_URB; urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); @@ -528,7 +528,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) struct usb_device *usb_dev = dev->chip.dev; unsigned int pipe; - pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? + pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) : usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE); @@ -547,25 +547,25 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) return urbs; } - urbs[i]->transfer_buffer = + urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { log("unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } - + for (frame = 0; frame < FRAMES_PER_URB; frame++) { - struct usb_iso_packet_descriptor *iso = + struct usb_iso_packet_descriptor *iso = &urbs[i]->iso_frame_desc[frame]; - + iso->offset = BYTES_PER_FRAME * frame; iso->length = BYTES_PER_FRAME; } - + urbs[i]->dev = usb_dev; urbs[i]->pipe = pipe; - urbs[i]->transfer_buffer_length = FRAMES_PER_URB + urbs[i]->transfer_buffer_length = FRAMES_PER_URB * BYTES_PER_FRAME; urbs[i]->context = &dev->data_cb_info[i]; urbs[i]->interval = 1; @@ -589,7 +589,7 @@ static void free_urbs(struct urb **urbs) for (i = 0; i < N_URBS; i++) { if (!urbs[i]) continue; - + usb_kill_urb(urbs[i]); kfree(urbs[i]->transfer_buffer); usb_free_urb(urbs[i]); @@ -602,11 +602,11 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) { int i, ret; - dev->n_audio_in = max(dev->spec.num_analog_audio_in, - dev->spec.num_digital_audio_in) / + dev->n_audio_in = max(dev->spec.num_analog_audio_in, + dev->spec.num_digital_audio_in) / CHANNELS_PER_STREAM; dev->n_audio_out = max(dev->spec.num_analog_audio_out, - dev->spec.num_digital_audio_out) / + dev->spec.num_digital_audio_out) / CHANNELS_PER_STREAM; dev->n_streams = max(dev->n_audio_in, dev->n_audio_out); @@ -619,7 +619,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) return -EINVAL; } - ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, + ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, dev->n_audio_out, dev->n_audio_in, &dev->pcm); if (ret < 0) { @@ -632,7 +632,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); - + memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware, sizeof(snd_usb_caiaq_pcm_hardware)); @@ -651,9 +651,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) break; } - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usb_caiaq_ops); - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); snd_pcm_lib_preallocate_pages_for_all(dev->pcm, @@ -662,7 +662,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); dev->data_cb_info = - kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, + kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); if (!dev->data_cb_info) @@ -672,14 +672,14 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; } - + dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret); if (ret < 0) { kfree(dev->data_cb_info); free_urbs(dev->data_urbs_in); return ret; } - + dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret); if (ret < 0) { kfree(dev->data_cb_info); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 515de1c..9be0f2e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -79,7 +79,7 @@ static struct usb_device_id snd_usb_id_table[] = { { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = USB_VID_NATIVEINSTRUMENTS, - .idProduct = USB_PID_RIGKONTROL2 + .idProduct = USB_PID_RIGKONTROL2 }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, @@ -197,7 +197,7 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, if (buffer && len > 0) memcpy(dev->ep1_out_buf+1, buffer, len); - + dev->ep1_out_buf[0] = command; return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), dev->ep1_out_buf, len+1, &actual_len, 200); @@ -208,7 +208,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, { int ret; char tmp[5]; - + switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; case 48000: tmp[0] = SAMPLERATE_48000; break; @@ -237,12 +237,12 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, if (ret) return ret; - - if (!wait_event_timeout(dev->ep1_wait_queue, + + if (!wait_event_timeout(dev->ep1_wait_queue, dev->audio_parm_answer >= 0, HZ)) return -EPIPE; - - if (dev->audio_parm_answer != 1) + + if (dev->audio_parm_answer != 1) debug("unable to set the device's audio params\n"); else dev->bpp = bpp; @@ -250,8 +250,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, return dev->audio_parm_answer == 1 ? 0 : -EINVAL; } -int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, - int digital, int analog, int erp) +int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, + int digital, int analog, int erp) { char tmp[3] = { digital, analog, erp }; return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG, @@ -262,7 +262,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) { int ret; char val[4]; - + /* device-specific startup specials */ switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): @@ -314,7 +314,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) dev->control_state, 1); break; } - + if (dev->spec.num_analog_audio_out + dev->spec.num_analog_audio_in + dev->spec.num_digital_audio_out + @@ -323,7 +323,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) if (ret < 0) log("Unable to set up audio system (ret=%d)\n", ret); } - + if (dev->spec.num_midi_in + dev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(dev); @@ -363,7 +363,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) if (devnum >= SNDRV_CARDS) return -ENODEV; - err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, + err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, sizeof(struct snd_usb_caiaqdev), &card); if (err < 0) return err; @@ -386,7 +386,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; int err, len; - + if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); return -EIO; @@ -395,19 +395,19 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_init_urb(&dev->ep1_in_urb); usb_init_urb(&dev->midi_out_urb); - usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, + usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x1), - dev->ep1_in_buf, EP1_BUFSIZE, + dev->ep1_in_buf, EP1_BUFSIZE, usb_ep1_command_reply_dispatch, dev); - usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, + usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, usb_sndbulkpipe(usb_dev, 0x1), - dev->midi_out_buf, EP1_BUFSIZE, + dev->midi_out_buf, EP1_BUFSIZE, snd_usb_caiaq_midi_output_done, dev); - + init_waitqueue_head(&dev->ep1_wait_queue); init_waitqueue_head(&dev->prepare_wait_queue); - + if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0) return -EIO; @@ -420,10 +420,10 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_string(usb_dev, usb_dev->descriptor.iManufacturer, dev->vendor_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iProduct, dev->product_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iSerialNumber, dev->serial, CAIAQ_USB_STR_LEN); @@ -431,7 +431,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) c = strchr(dev->serial, ' '); if (c) *c = '\0'; - + strcpy(card->driver, MODNAME); strcpy(card->shortname, dev->product_name); @@ -449,18 +449,18 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) return 0; } -static int __devinit snd_probe(struct usb_interface *intf, +static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - + ret = create_card(device, &card); - + if (ret < 0) return ret; - + usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { @@ -468,7 +468,7 @@ static int __devinit snd_probe(struct usb_interface *intf, snd_card_free(card); return ret; } - + return 0; } @@ -489,10 +489,10 @@ static void snd_disconnect(struct usb_interface *intf) snd_usb_caiaq_input_free(dev); #endif snd_usb_caiaq_audio_free(dev); - + usb_kill_urb(&dev->ep1_in_urb); usb_kill_urb(&dev->midi_out_urb); - + snd_card_free(card); usb_reset_device(interface_to_usbdev(intf)); } diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 8fa8cd8..538e8c0 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -40,7 +40,7 @@ static void snd_usb_caiaq_midi_input_trigger(struct snd_rawmidi_substream *subst if (!dev) return; - + dev->midi_receive_substream = up ? substream : NULL; } @@ -64,18 +64,18 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, struct snd_rawmidi_substream *substream) { int len, ret; - + dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; dev->midi_out_buf[1] = 0; /* port */ len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, EP1_BUFSIZE - 3); - + if (len <= 0) return; - + dev->midi_out_buf[2] = len; dev->midi_out_urb.transfer_buffer_length = len+3; - + ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); if (ret < 0) log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," @@ -88,7 +88,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - + if (up) { dev->midi_out_substream = substream; if (!dev->midi_out_active) @@ -113,12 +113,12 @@ static struct snd_rawmidi_ops snd_usb_caiaq_midi_input = .trigger = snd_usb_caiaq_midi_input_trigger, }; -void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, +void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len) { if (!dev->midi_receive_substream) return; - + snd_rawmidi_receive(dev->midi_receive_substream, buf, len); } @@ -142,16 +142,16 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (device->spec.num_midi_out > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usb_caiaq_midi_output); } if (device->spec.num_midi_in > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usb_caiaq_midi_input); } - + device->rmidi = rmidi; return 0; @@ -160,7 +160,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { struct snd_usb_caiaqdev *dev = urb->context; - + dev->midi_out_active = 0; if (urb->status != 0) return; -- 1.6.3.1

1 5

[alsa-devel] Testers wanted: New SB X-Fi driver
by gantonop＠cs.uoi.gr 30 May '09

30 May '09

I have an X-FI Platinum.Though I hear sound, it is periodically interrupted. More specifically every half a second an interrupt occurs producing a pause and a "tsik" :) which of course is annoying. So this has happened for every sound file I've tried to play.

1 0

[alsa-devel] (no subject)
by James Gardiner 30 May '09

30 May '09

Hi, someone recently submitted changes to the patch_sigmatel.c codec source. Whatever was done is breaking my patch for the HP dv4-1222nr laptop. I also strongly suspect that other code in the stac92hd71bxx function has been affected. There is no autodetect, there is no option for modprobe that will load that codec currently, to the best of my knowledge. I've tracked the difference to snapshot 20090529. Before that, I can get it to work, there and after refuses to work properly. Also, there was a recent change to the stac92xx_hp_check_power_status function: While the change looks good for its intended purpose of localizing board changes to the switch/case statements in the stac92hd71bxx function, the implementor reversed the if/then selector: if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) spec->gpio_data &= ~spec->gpio_led; /* white */ else spec->gpio_data |= spec->gpio_led; /* orange */ it should be: if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) spec->gpio_data |= spec->gpio_led; /* white */ else spec->gpio_data &= ~spec->gpio_led; /* orange */ The current implementation does not work-in fact sound is completely muted 100% of the time. I'll attempt to track down the 1st problem I mentioned, but I don't know if only patch_sigmatel.c was affected, or even the source of the problem. Thanks, James Gardiner

2 1

[alsa-devel] ALC883 features
by Michael Leies 30 May '09

30 May '09

Hi, I'm new here. I just bought a new computer from Dell with a Realtek ALC1200 (883?). After messing with alsa configurations and looking into the driver code a bit, I noted that a few features i'd like don't seem to be supported by the current driver. 1) Use of all 10 channels, only 8 are wired out to the 7.1 pcm device (forgive me if i lack the proper terminology for the moment). From the 883 datasheet, and comments in 1.0.20 code it seems the last 2 channels could be wired out as a separate PCM device. 2) Some more controls, specifically the ability to prevent the headphone jack from automuting the speakers. And if its possible to wire out the missing two channels then perhaps some more controls for the new channels. I was wondering if there someone who might point me in the right direction to get started with fixing up these things in the driver. Are my ambitions realistic, or am I barking up the tree of having to rewrite the whole HDA driver? Thanks, Michael

2 1

[alsa-devel] [PATCH v3 1/6] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by Chaithrika U S 30 May '09

30 May '09

Adds McASP clock support for the two instances of mcasp (mcasp0,mcasp1). This patch is part of the audio support for dm646x series. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/dm646x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index f95b5e9..07c8d88 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -227,6 +227,18 @@ static struct clk gpio_clk = { .lpsc = DM646X_LPSC_GPIO, }; +static struct clk mcasp0_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP0, +}; + +static struct clk mcasp1_clk = { + .name = "mcasp1", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP1, +}; + static struct clk aemif_clk = { .name = "aemif", .parent = &pll1_sysclk3, @@ -314,6 +326,8 @@ struct davinci_clk dm646x_clks[] = { CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK(NULL, "gpio", &gpio_clk), + CLK(NULL, "mcasp0", &mcasp0_clk), + CLK(NULL, "mcasp1", &mcasp1_clk), CLK(NULL, "aemif", &aemif_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK(NULL, "pwm0", &pwm0_clk), -- 1.5.6

2 1

[alsa-devel] Add info in my sound-issue with the front Jack.
by Kristoffer Grundström 29 May '09

29 May '09

I added some possible info in my earlier bugreport about the front jack. Please visit this adress <https://qa.mandriva.com/show_bug.cgi?id=49048> and tell me if I've missed to add something. /Kristoffer Grundström P.S. I changed my emailadress since I got spammessages in my mailbox. D.S.

2 1

[alsa-devel] [PATCH v3 2/6] ARM: DaVinci: ASoC: Add the platform devices for ASP
by Chaithrika U S 29 May '09

29 May '09

1) Registers the platform devices for ASP on dm355, dm644x and dm646x so that the machine driver can probe to get ASP related platform data. 2) Move towards definition of the asp clocks using phyisical name(for dm355 and dm644x) 3) Add platform data to board specific files. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/board-dm355-evm.c | 5 ++ arch/arm/mach-davinci/board-dm644x-evm.c | 5 ++ arch/arm/mach-davinci/board-dm646x-evm.c | 32 +++++++++++++++ arch/arm/mach-davinci/dm355.c | 41 ++++++++++++++++++- arch/arm/mach-davinci/dm644x.c | 36 ++++++++++++++++- arch/arm/mach-davinci/dm646x.c | 58 +++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 35 +++++++++++++++- arch/arm/mach-davinci/include/mach/dm355.h | 2 + arch/arm/mach-davinci/include/mach/dm644x.h | 2 + arch/arm/mach-davinci/include/mach/dm646x.h | 2 + 10 files changed, 213 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm355-evm.c b/arch/arm/mach-davinci/board-dm355-evm.c index f32e3d8..5bf1e13 100644 --- a/arch/arm/mach-davinci/board-dm355-evm.c +++ b/arch/arm/mach-davinci/board-dm355-evm.c @@ -118,6 +118,10 @@ static struct davinci_i2c_platform_data i2c_pdata = { .bus_delay = 0 /* usec */, }; +static struct evm_snd_platform_data dm355_evm_snd_data = { + .clk_name = "asp1", +}; + static int dm355evm_mmc_gpios = -EINVAL; static void dm355evm_mmcsd_gpios(unsigned gpio) @@ -278,6 +282,7 @@ static __init void dm355_evm_init(void) dm355_init_spi0(BIT(0), dm355_evm_spi_info, ARRAY_SIZE(dm355_evm_spi_info)); + dm355_init_asp(&dm355_evm_snd_data); } static __init void dm355_evm_irq_init(void) diff --git a/arch/arm/mach-davinci/board-dm644x-evm.c b/arch/arm/mach-davinci/board-dm644x-evm.c index e97216c..bdfbffe 100644 --- a/arch/arm/mach-davinci/board-dm644x-evm.c +++ b/arch/arm/mach-davinci/board-dm644x-evm.c @@ -226,6 +226,10 @@ static struct platform_device ide_dev = { }, }; +static struct evm_snd_platform_data dm644x_evm_snd_data = { + .clk_name = "asp0", +}; + /*----------------------------------------------------------------------*/ /* @@ -667,6 +671,7 @@ static __init void davinci_evm_init(void) davinci_setup_mmc(0, &dm6446evm_mmc_config); davinci_serial_init(&uart_config); + dm644x_init_asp(&dm644x_evm_snd_data); soc_info->emac_pdata->phy_mask = DM644X_EVM_PHY_MASK; soc_info->emac_pdata->mdio_max_freq = DM644X_EVM_MDIO_FREQUENCY; diff --git a/arch/arm/mach-davinci/board-dm646x-evm.c b/arch/arm/mach-davinci/board-dm646x-evm.c index a55f45c..774497b 100644 --- a/arch/arm/mach-davinci/board-dm646x-evm.c +++ b/arch/arm/mach-davinci/board-dm646x-evm.c @@ -206,6 +206,37 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; +static u8 dm646x_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm646x_dit_serializer_direction[] = { + TX_MODE, +}; + +static struct evm_snd_platform_data dm646x_evm_snd_data[] = { + { + .clk_name = "mcasp0", + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(dm646x_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = dm646x_iis_serializer_direction, + .eventq_no = EVENTQ_0, + }, + { + .clk_name = "mcasp1", + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = ARRAY_SIZE(dm646x_dit_serializer_direction), + .tdm_slots = 32, + .serial_dir = dm646x_dit_serializer_direction, + .eventq_no = EVENTQ_0, + }, +}; + static struct i2c_board_info __initdata i2c_info[] = { { I2C_BOARD_INFO("24c256", 0x50), @@ -239,6 +270,7 @@ static __init void evm_init(void) evm_init_i2c(); davinci_serial_init(&uart_config); + dm646x_init_mcasp(dm646x_evm_snd_data); soc_info->emac_pdata->phy_mask = DM646X_EVM_PHY_MASK; soc_info->emac_pdata->mdio_max_freq = DM646X_EVM_MDIO_FREQUENCY; diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index 737ba50..dca38e2 100644 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -14,6 +14,7 @@ #include <linux/serial_8250.h> #include <linux/platform_device.h> #include <linux/dma-mapping.h> +#include <mach/asp.h> #include <linux/spi/spi.h> @@ -360,8 +361,8 @@ static struct davinci_clk dm355_clks[] = { CLK(NULL, "uart1", &uart1_clk), CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), - CLK("soc-audio.0", NULL, &asp0_clk), - CLK("soc-audio.1", NULL, &asp1_clk), + CLK(NULL, "asp0", &asp0_clk), + CLK(NULL, "asp1", &asp1_clk), CLK("davinci_mmc.0", NULL, &mmcsd0_clk), CLK("davinci_mmc.1", NULL, &mmcsd1_clk), CLK(NULL, "spi0", &spi0_clk), @@ -624,6 +625,32 @@ static struct platform_device dm355_edma_device = { .resource = edma_resources, }; +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm355_asp_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP1_TX, + .end = DAVINCI_DMA_ASP1_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP1_RX, + .end = DAVINCI_DMA_ASP1_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm355_asp_device = { + .name = "davinci-asp", + .id = -1, + .num_resources = ARRAY_SIZE(dm355_asp_resources), + .resource = dm355_asp_resources, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm355_io_desc[] = { @@ -733,6 +760,16 @@ static struct davinci_soc_info davinci_soc_info_dm355 = { .sram_len = SZ_32K, }; +void __init dm355_init_asp(struct evm_snd_platform_data *pdata) +{ + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + + dm355_asp_device.dev.platform_data = pdata; + platform_device_register(&dm355_asp_device); +} + void __init dm355_init(void) { davinci_common_init(&davinci_soc_info_dm355); diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index 7cea25b..8c6c23b 100644 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -27,6 +27,7 @@ #include <mach/gpio.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -303,7 +304,7 @@ struct davinci_clk dm644x_clks[] = { CLK("davinci_emac.1", NULL, &emac_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK("palm_bk3710", NULL, &ide_clk), - CLK("soc-audio.0", NULL, &asp_clk), + CLK(NULL, "asp0", &asp_clk), CLK("davinci_mmc.0", NULL, &mmcsd_clk), CLK(NULL, "spi", &spi_clk), CLK(NULL, "gpio", &gpio_clk), @@ -550,6 +551,32 @@ static struct platform_device dm644x_edma_device = { .resource = edma_resources, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ +static struct resource dm644x_asp_resources[] = { + { + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP0_TX, + .end = DAVINCI_DMA_ASP0_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP0_RX, + .end = DAVINCI_DMA_ASP0_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm644x_asp_device = { + .name = "davinci-asp", + .id = -1, + .num_resources = ARRAY_SIZE(dm644x_asp_resources), + .resource = dm644x_asp_resources, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm644x_io_desc[] = { @@ -660,6 +687,13 @@ static struct davinci_soc_info davinci_soc_info_dm644x = { .sram_len = SZ_16K, }; +void __init dm644x_init_asp(struct evm_snd_platform_data *pdata) +{ + davinci_cfg_reg(DM644X_MCBSP); + dm644x_asp_device.dev.platform_data = pdata; + platform_device_register(&dm644x_asp_device); +} + void __init dm644x_init(void) { davinci_common_init(&davinci_soc_info_dm644x); diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index 07c8d88..ad5f479 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -27,6 +27,7 @@ #include <mach/gpio.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -587,6 +588,56 @@ static struct platform_device dm646x_edma_device = { .resource = edma_resources, }; +static struct resource dm646x_mcasp_resources[] = { + { + .name = "mcasp0", + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + { + .name = "mcasp1", + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + /* first TX, then RX */ + { + .start = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .end = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .end = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .flags = IORESOURCE_DMA, + }, + /* DIT mode, only TX event */ + { + .start = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .end = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .flags = IORESOURCE_DMA, + }, + /* DIT mode, dummy entry */ + { + .start = -1, + .end = -1, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm646x_mcasp_device = { + .name = "davinci-mcasp", + .id = -1, + .num_resources = ARRAY_SIZE(dm646x_mcasp_resources), + .resource = dm646x_mcasp_resources, +}; + +static struct platform_device dm646x_dit_device = { + .name = "spdif-dit", + .id = -1, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm646x_io_desc[] = { @@ -697,6 +748,13 @@ static struct davinci_soc_info davinci_soc_info_dm646x = { .sram_len = SZ_32K, }; +void __init dm646x_init_mcasp(struct evm_snd_platform_data *pdata) +{ + dm646x_mcasp_device.dev.platform_data = pdata; + platform_device_register(&dm646x_mcasp_device); + platform_device_register(&dm646x_dit_device); +} + void __init dm646x_init(void) { davinci_common_init(&davinci_soc_info_dm646x); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..cffc55c 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -5,21 +5,52 @@ #define __ASM_ARCH_DAVINCI_ASP_H #include <mach/irqs.h> +#include <mach/edma.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 + /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT #define DAVINCI_ASP1_RX_INT IRQ_MBRINT #define DAVINCI_ASP1_TX_INT IRQ_MBXINT +struct evm_snd_platform_data { + char *clk_name; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + #endif /* __ASM_ARCH_DAVINCI_ASP_H */ diff --git a/arch/arm/mach-davinci/include/mach/dm355.h b/arch/arm/mach-davinci/include/mach/dm355.h index 54903b7..b8f9c50 100644 --- a/arch/arm/mach-davinci/include/mach/dm355.h +++ b/arch/arm/mach-davinci/include/mach/dm355.h @@ -12,11 +12,13 @@ #define __ASM_ARCH_DM355_H #include <mach/hardware.h> +#include <mach/asp.h> struct spi_board_info; void __init dm355_init(void); void dm355_init_spi0(unsigned chipselect_mask, struct spi_board_info *info, unsigned len); +void __init dm355_init_asp(struct evm_snd_platform_data *pdata); #endif /* __ASM_ARCH_DM355_H */ diff --git a/arch/arm/mach-davinci/include/mach/dm644x.h b/arch/arm/mach-davinci/include/mach/dm644x.h index 15d42b9..c508b5e 100644 --- a/arch/arm/mach-davinci/include/mach/dm644x.h +++ b/arch/arm/mach-davinci/include/mach/dm644x.h @@ -25,6 +25,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM644X_EMAC_BASE (0x01C80000) #define DM644X_EMAC_CNTRL_OFFSET (0x0000) @@ -34,5 +35,6 @@ #define DM644X_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm644x_init(void); +void __init dm644x_init_asp(struct evm_snd_platform_data *pdata); #endif /* __ASM_ARCH_DM644X_H */ diff --git a/arch/arm/mach-davinci/include/mach/dm646x.h b/arch/arm/mach-davinci/include/mach/dm646x.h index 7afc613..47b3423 100644 --- a/arch/arm/mach-davinci/include/mach/dm646x.h +++ b/arch/arm/mach-davinci/include/mach/dm646x.h @@ -14,6 +14,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM646X_EMAC_BASE (0x01C80000) #define DM646X_EMAC_CNTRL_OFFSET (0x0000) @@ -23,5 +24,6 @@ #define DM646X_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm646x_init(void); +void __init dm646x_init_mcasp(struct evm_snd_platform_data *pdata); #endif /* __ASM_ARCH_DM646X_H */ -- 1.5.6

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[alsa-devel] Couple of alsa-tools patches from Gentoo
by Diego Elio 'Flameeyes' Pettenò 29 May '09

29 May '09

I already sent a link to the git repository where I pushed this last week, but given I'm actually going to remove it (need to reduce the number of repos on the server), I thought it would be easier just to send them as they are. One fixes build with glibc-2.10, the other fixes --as-needed in general. HTH, -- Diego E. Pettenò — Flameeyes

2 3

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Using codec_regcache in twl4030_init_chip
by Peter Ujfalusi 29 May '09

29 May '09

Hello, The twl4030_init_chip used the twl4030_reg array directly to set the default register values. In other places the defaults has been modified in the codec->reg_cache (prime example is the Headset ramp delay configuration). Switch using the codec->reg_cache in the twl4030_init_chip function, so updated defaults will be actually used. --- Peter Ujfalusi (1): ASoC: TWL4030: Use reg_cache in twl4030_init_chip sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

2 2

Re: [alsa-devel] Acer Aspire One codec: Realtek ALC272 not found in ALSA-Configuration.txt Mic does not capture in alsamixer
by Takashi Iwai 29 May '09

29 May '09

At Sat, 23 May 2009 14:17:16 -0400, wirechief wrote: > > upon further review attempts have been un-successfull in getting the mic to work > with arecord -Ddefault:1 -f cd -vv /dev/null > also am finding this in dmesg: [ 11.684245] hda_codec: Unknown model > for ALC272, trying auto-probe from BIOS.. > my alsa http://www.alsa-project.org/db/?f=754f6ed51b0e45bf78662e50bde8f26546fb2bb4 Did you choose "Mic" as the input source? The default "model=auto" has no automatic mic-switching yet, so you'll have to choose the recording source manually. Takashi

1 0

[alsa-devel] [PATCH] ALSA: clean up the logic for building sequencer modules
by Michal Marek 29 May '09

29 May '09

Instead of mangling the CONFIG_* variables in the makefiles over and over, set a few helper variables in Kconfig. Signed-off-by: Michal Marek <mmarek(a)suse.cz> --- sound/core/Kconfig | 2 ++ sound/core/seq/Kconfig | 16 ++++++++++++++++ sound/core/seq/Makefile | 18 +++++------------- sound/drivers/opl3/Makefile | 10 +--------- sound/drivers/opl4/Makefile | 10 +--------- sound/isa/sb/Makefile | 10 +--------- sound/pci/emu10k1/Makefile | 10 +--------- sound/synth/Makefile | 12 ++---------- sound/synth/emux/Makefile | 12 ++---------- 9 files changed, 31 insertions(+), 69 deletions(-) create mode 100644 sound/core/seq/Kconfig diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 7bbdda0..6061fb5 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -205,3 +205,5 @@ config SND_PCM_XRUN_DEBUG config SND_VMASTER bool + +source "sound/core/seq/Kconfig" diff --git a/sound/core/seq/Kconfig b/sound/core/seq/Kconfig new file mode 100644 index 0000000..b851fd8 --- /dev/null +++ b/sound/core/seq/Kconfig @@ -0,0 +1,16 @@ +# define SND_XXX_SEQ to min(SND_SEQUENCER,SND_XXX) + +config SND_RAWMIDI_SEQ + def_tristate SND_SEQUENCER && SND_RAWMIDI + +config SND_OPL3_LIB_SEQ + def_tristate SND_SEQUENCER && SND_OPL3_LIB + +config SND_OPL4_LIB_SEQ + def_tristate SND_SEQUENCER && SND_OPL4_LIB + +config SND_SBAWE_SEQ + def_tristate SND_SEQUENCER && SND_SBAWE + +config SND_EMU10K1_SEQ + def_tristate SND_SEQUENCER && SND_EMU10K1 diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index 0695937..1bcb360 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -17,14 +17,6 @@ snd-seq-midi-event-objs := seq_midi_event.o snd-seq-dummy-objs := seq_dummy.o snd-seq-virmidi-objs := seq_virmidi.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o ifeq ($(CONFIG_SND_SEQUENCER_OSS),y) obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o @@ -33,8 +25,8 @@ obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o # Toplevel Module Dependency obj-$(CONFIG_SND_VIRMIDI) += snd-seq-virmidi.o snd-seq-midi-event.o -obj-$(call sequencer,$(CONFIG_SND_RAWMIDI)) += snd-seq-midi.o snd-seq-midi-event.o -obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o -obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-seq-midi-emul.o snd-seq-virmidi.o -obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-seq-midi-emul.o snd-seq-virmidi.o +obj-$(CONFIG_SND_RAWMIDI_SEQ) += snd-seq-midi.o snd-seq-midi-event.o +obj-$(CONFIG_SND_OPL3_LIB_SEQ) += snd-seq-midi-event.o snd-seq-midi-emul.o +obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-seq-midi-event.o snd-seq-midi-emul.o +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-seq-midi-emul.o snd-seq-virmidi.o +obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-seq-midi-emul.o snd-seq-virmidi.o diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile index 19767a6..7f2c2a1 100644 --- a/sound/drivers/opl3/Makefile +++ b/sound/drivers/opl3/Makefile @@ -7,14 +7,6 @@ snd-opl3-lib-objs := opl3_lib.o opl3_synth.o snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - obj-$(CONFIG_SND_OPL3_LIB) += snd-opl3-lib.o obj-$(CONFIG_SND_OPL4_LIB) += snd-opl3-lib.o -obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-opl3-synth.o +obj-$(CONFIG_SND_OPL3_LIB_SEQ) += snd-opl3-synth.o diff --git a/sound/drivers/opl4/Makefile b/sound/drivers/opl4/Makefile index d178b39..b94009b 100644 --- a/sound/drivers/opl4/Makefile +++ b/sound/drivers/opl4/Makefile @@ -6,13 +6,5 @@ snd-opl4-lib-objs := opl4_lib.o opl4_mixer.o opl4_proc.o snd-opl4-synth-objs := opl4_seq.o opl4_synth.o yrw801.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - obj-$(CONFIG_SND_OPL4_LIB) += snd-opl4-lib.o -obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-opl4-synth.o +obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-opl4-synth.o diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index 1098a56..faeffce 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -13,14 +13,6 @@ snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o obj-$(CONFIG_SND_SB16_DSP) += snd-sb16-dsp.o @@ -33,4 +25,4 @@ ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o endif -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emu8000-synth.o +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-emu8000-synth.o diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile index cf2d563..fc5591e 100644 --- a/sound/pci/emu10k1/Makefile +++ b/sound/pci/emu10k1/Makefile @@ -9,15 +9,7 @@ snd-emu10k1-objs := emu10k1.o emu10k1_main.o \ snd-emu10k1-synth-objs := emu10k1_synth.o emu10k1_callback.o emu10k1_patch.o snd-emu10k1x-objs := emu10k1x.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_EMU10K1) += snd-emu10k1.o -obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emu10k1-synth.o +obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-emu10k1-synth.o obj-$(CONFIG_SND_EMU10K1X) += snd-emu10k1x.o diff --git a/sound/synth/Makefile b/sound/synth/Makefile index e99fd76..11eb06a 100644 --- a/sound/synth/Makefile +++ b/sound/synth/Makefile @@ -5,16 +5,8 @@ snd-util-mem-objs := util_mem.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_EMU10K1) += snd-util-mem.o obj-$(CONFIG_SND_TRIDENT) += snd-util-mem.o -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-util-mem.o -obj-$(call sequencer,$(CONFIG_SND)) += emux/ +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-util-mem.o +obj-$(CONFIG_SND_SEQUENCER) += emux/ diff --git a/sound/synth/emux/Makefile b/sound/synth/emux/Makefile index b690352..328594e 100644 --- a/sound/synth/emux/Makefile +++ b/sound/synth/emux/Makefile @@ -7,14 +7,6 @@ snd-emux-synth-objs := emux.o emux_synth.o emux_seq.o emux_nrpn.o \ emux_effect.o emux_proc.o emux_hwdep.o soundfont.o \ $(if $(CONFIG_SND_SEQUENCER_OSS),emux_oss.o) -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependencies -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emux-synth.o -obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emux-synth.o +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-emux-synth.o +obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-emux-synth.o -- 1.6.3

2 2

[alsa-devel] Increasing maximum sampling frequency?
by Cameron Stone 29 May '09

29 May '09

I've got a USB microphone application that I want to sample at 256000 8 bit samples per second so I can listen to bats. I know that my device can get close to that rate, but alsa seems to restrict sampling rates to 192kHz. Is it possible to push the maximum sampling frequency higher? If so, where should I start looking? I tried hacking aplay.c so arecord would accept command line requests up to 256k, but it seems the alsa driver is also limiting the frequency to a maximum of 192k. However, the files generated by arecord *do* have the right number of samples in them. $ arecord -v -Dplughw:1,0 test.wav -d 1 -r 256 Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 256000 Hz, Mono Plug PCM: Linear conversion PCM (S8) Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 256000 exact rate : 256000 (256000/1) msbits : 8 buffer_size : 128000 period_size : 32000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 32000 period_event : 0 start_threshold : 1 stop_threshold : 128000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 1 '8-Mic AVR Adaptor' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S8 subformat : STD channels : 1 rate : 256000 exact rate : 256000 (256000/1) msbits : 8 buffer_size : 128000 period_size : 32000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 32000 period_event : 0 start_threshold : 1 stop_threshold : 128000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 $ aplay -r 256 test.wav Playing WAVE 'test.wav' : Unsigned 8 bit, Rate 256000 Hz, Mono Warning: rate is not accurate (requested = 256000Hz, got = 192000Hz) please, try the plug plugin (-Dplug:default) $ ls -l test.wav -rw-r--r-- 1 camerons camerons 256044 2009-05-28 11:29 test.wav $ file test.wav test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 8 bit, mono 256000 Hz $

5 7

Re: [alsa-devel] Increasing maximum sampling frequency?
by Cameron Stone 29 May '09

29 May '09

The A/D chip is an Analog Devices 7928. It can do 2 mega-samples per second, multiplexed across 8 inputs. The USB audio interface is provided by my firmware code, assisted by the AVR AT90USB647 micro which has some low-level USB support. HTH. Cameron. Alex Austin wrote: > By Codec, I think he's asking about the A/D chip. For example, the TI > PCM2902E is a stereo USB audio codec. > > - Alex Austin > (651) 238-9273 > > "...and then I visited Wikipedia ...and the next 8 hours are a blur." > > > On Thu, May 28, 2009 at 8:30 AM, Cameron Stone > <camerons.lists(a)cse.unsw.edu.au <mailto:camerons.lists@cse.unsw.edu.au>> > wrote: > > Daniel Mack wrote: > > On Thu, May 28, 2009 at 11:47:34AM +1000, Cameron Stone wrote: > >> I've got a USB microphone application that I want to sample at > 256000 8 > >> bit samples per second so I can listen to bats. I know that my device > >> can get close to that rate, but alsa seems to restrict sampling > rates to > >> 192kHz. > > > > Just out of curiosity - which microphone and what kind of codec > are you > > using for such frequencies? > > I don't know which microphone but I can find out if you're interested. I > do know that its sensitivity starts at 10kHz, if that helps. > > There is no codec used. I'm using Type I (uncompressed PCM) because it's > being powered by an AVR running at 16MHz and there's not enough clock > cycles to encode the samples. > > Cameron. > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org <mailto:Alsa-devel@alsa-project.org> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

1 0

[alsa-devel] [PATCH V3 0/4] mpc5200 audio rework for AC97
by Jon Smirl 28 May '09

28 May '09

The following series implements audio support for the mpc5200. It adds an AC97 driver and STAC9766 codec driver. Board support for the Efika and Phytec pcm030 are also included. I've tried to implement the feedback received on the previous two versions. based on commit 0bc53a67ac831ec84f730a657dbcadd80a589ef5 on broonie/for-2.6.31 at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git --- Jon Smirl (4): Fabric bindings for STAC9766 on the Efika Support for AC97 on Phytec pmc030 base board. AC97 driver for mpc5200 Main rewite of the mpc5200 audio DMA code sound/soc/fsl/Kconfig | 27 ++ sound/soc/fsl/Makefile | 5 sound/soc/fsl/efika-audio-fabric.c | 95 +++++++ sound/soc/fsl/mpc5200_dma.c | 504 +++++++++++++++++++++++------------ sound/soc/fsl/mpc5200_dma.h | 33 +- sound/soc/fsl/mpc5200_psc_ac97.c | 392 +++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + sound/soc/fsl/mpc5200_psc_i2s.c | 247 +++-------------- sound/soc/fsl/mpc5200_psc_i2s.h | 12 + sound/soc/fsl/pcm030-audio-fabric.c | 95 +++++++ 10 files changed, 1036 insertions(+), 389 deletions(-) create mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- Jon Smirl jonsmirl(a)gmail.com

4 21

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 28 May '09

28 May '09

The following changes since commit 08d15f034e94251606479d7ca9070994c2e2fcf0: Mark Brown (1): ASoC: Switch FSL SSI DAI over to symmetric_rates are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Chaithrika U S (1): ASoC: Add dummy S/PDIF codec support Jon Smirl (1): ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout() Mark Brown (1): ASoC: Initialise dev for the dummy S/PDIF DAI Roel Kluin (1): ASoC: correct print specifiers for unsigneds sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/spdif_transciever.c | 71 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/spdif_transciever.h | 17 ++++++++ sound/soc/codecs/tlv320aic23.c | 4 +- sound/soc/codecs/uda134x.c | 4 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 4 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 4 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 6 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 30 +++++++-------- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 4 +- sound/soc/sh/ssi.c | 2 +- 19 files changed, 130 insertions(+), 38 deletions(-) create mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transciever.h

2 1

[alsa-devel] [PATCH v3 4/6] ARM: DaVinci: ASoC: Add dummy codec support
by Chaithrika U S 28 May '09

28 May '09

McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed. This patch provides stub codec that can be used in these configurations. On DM646x EVM the McASP1 is connected to the S/PDIF out. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/spdif_transciever.c | 69 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/spdif_transciever.h | 17 ++++++++ 4 files changed, 92 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transciever.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb07d9b..bbc97fd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 + select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C @@ -91,6 +92,9 @@ config SND_SOC_L3 config SND_SOC_PCM3008 tristate +config SND_SOC_SPDIF + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 46c007c..8b75305 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(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 obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c new file mode 100644 index 0000000..118e976 --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC SPDIF DIT driver + * + * This driver is used by controllers which can operate in DIT (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. TI DaVinci Audio controller uses this driver. + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * 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 <sound/soc.h> +#include <sound/pcm.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE + + +struct snd_soc_dai dit_stub_dai = { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int spdif_dit_probe(struct platform_device *pdev) +{ + return snd_soc_register_dai(&dit_stub_dai); +} + +static int spdif_dit_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&dit_stub_dai); + return 0; +} + +static struct platform_driver spdif_dit_driver = { + .probe = spdif_dit_probe, + .remove = spdif_dit_remove, + .driver = { + .name = "spdif-dit", + .owner = THIS_MODULE, + }, +}; + +static int __init dit_modinit(void) +{ + return platform_driver_register(&spdif_dit_driver); +} + +static void __exit dit_exit(void) +{ + platform_driver_unregister(&spdif_dit_driver); +} + +module_init(dit_modinit); +module_exit(dit_exit); + diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h new file mode 100644 index 0000000..296f2eb --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef CODEC_STUBS_H +#define CODEC_STUBS_H + +extern struct snd_soc_dai dit_stub_dai; + +#endif /* CODEC_STUBS_H */ -- 1.5.6

2 1

[alsa-devel] [PATCH v3 6/6] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Chaithrika U S 28 May '09

28 May '09

This patch does the following: (1) Add support for the DM646x machine (2)Modifications required to introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-evm.c | 103 ++++++++++++++++++-------------------- 1 files changed, 49 insertions(+), 54 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 58fd1cb..d66fbd3 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -25,11 +25,13 @@ #include <mach/asp.h> #include <mach/edma.h> #include <mach/mux.h> +#include <mach/cpu.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" - +#include "davinci-i2s-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -43,7 +45,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -144,6 +146,24 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_iis_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "McASP", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[DAVINCI_MCASP_DIT_DAI], + .codec_dai = &dit_stub_dai, + .ops = &evm_ops, + }, +}; + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -152,12 +172,26 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = ARRAY_SIZE(dm6467_evm_dai), +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -165,60 +199,30 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_ASP0_BASE, - .end = DAVINCI_ASP0_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP0_TX, - .rx_dma_ch = DAVINCI_DMA_ASP0_RX, -}; - -/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ -static struct resource dm335evm_snd_resources[] = { - { - .start = DAVINCI_ASP1_BASE, - .end = DAVINCI_ASP1_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data dm335evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP1_TX, - .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { - struct resource *resources; - unsigned num_resources; - struct evm_snd_platform_data *data; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; if (machine_is_davinci_evm()) { - davinci_cfg_reg(DM644X_MCBSP); - - resources = evm_snd_resources; - num_resources = ARRAY_SIZE(evm_snd_resources); - data = &evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { - /* we don't use ASP1 IRQs, or we'd need to mux them ... */ - davinci_cfg_reg(DM355_EVT8_ASP1_TX); - davinci_cfg_reg(DM355_EVT9_ASP1_RX); - - resources = dm335evm_snd_resources; - num_resources = ARRAY_SIZE(dm335evm_snd_resources); - data = &dm335evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; } else return -EINVAL; @@ -226,17 +230,8 @@ static int __init evm_init(void) if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, data, sizeof(*data)); - - ret = platform_device_add_resources(evm_snd_device, resources, - num_resources); - if (ret) { - platform_device_put(evm_snd_device); - return ret; - } - + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); -- 1.5.6

1 0

[alsa-devel] [PATCH v3 3/6] ARM: DaVinci: ASoC: Introduce platform driver model for dm644x, dm355
by Chaithrika U S 28 May '09

28 May '09

Introduce the platform driver model to get platform data for dm355 and dm644x. Register platform driver and acquire the resources in the probe function Since the platform specific code had been moved from machine driver to dm<soc>.c Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 115 ++++++++++++++++++++++----------------- 1 files changed, 66 insertions(+), 49 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index b1ea52f..dec7611 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -436,15 +436,39 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_i2s_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct evm_snd_platform_data *pdata = pdev->dev.platform_data;; struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea; - struct evm_snd_platform_data *pdata; + struct resource *mem, *ioarea, *res; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -466,9 +490,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, goto err_release_region; } - cpu_dai->private_data = dev; - - dev->clk = clk_get(&pdev->dev, NULL); + dev->clk = clk_get(&pdev->dev, pdata->clk_name); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -476,18 +498,35 @@ static int davinci_i2s_probe(struct platform_device *pdev, clk_enable(dev->clk); dev->base = (void __iomem *)IO_ADDRESS(mem->start); - pdata = pdev->dev.platform_data; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_i2s_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_i2s_dai); + if (ret != 0) + goto err_free_mem; + return 0; err_free_mem: @@ -498,62 +537,40 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int davinci_i2s_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = davinci_i2s_dai.private_data; struct resource *mem; + snd_soc_unregister_dai(&davinci_i2s_dai); clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; - kfree(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); -} -#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + return 0; +} -static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, +static struct platform_driver davinci_mcbsp_driver = { + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .driver = { + .name = "davinci-asp", + .owner = THIS_MODULE, + }, }; -struct snd_soc_dai davinci_i2s_dai = { - .name = "davinci-i2s", - .id = 0, - .probe = davinci_i2s_probe, - .remove = davinci_i2s_remove, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &davinci_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(davinci_i2s_dai); - static int __init davinci_i2s_init(void) { - return snd_soc_register_dai(&davinci_i2s_dai); + return platform_driver_register(&davinci_mcbsp_driver); } module_init(davinci_i2s_init); static void __exit davinci_i2s_exit(void) { - snd_soc_unregister_dai(&davinci_i2s_dai); + platform_driver_unregister(&davinci_mcbsp_driver); } module_exit(davinci_i2s_exit); -- 1.5.6

1 0

[alsa-devel] [PATCH v3 0/6] ARM: DaVinci: Audio support for DM646x
by Chaithrika U S 28 May '09

28 May '09

From: Chaithrika U S <chaithrika(a)ti.com> Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch set adds audio support for DM646x This updated version has included most of the review comments received for the previous version of the patch set. This series is dependent on the following patches submitted to the DaVinci mailing list: [PATCH v2 1/2] ARM: DaVinci: EDMA: Move queue related mappings to dm<soc>.c [PATCH v2 2/2] ARM: DaVinci: EDMA: Enable EDMA support for dm646x Patch 1: Adds McASP clock support for the two instances of mcasp(mcasp0,mcasp1) This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 2: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 3: Introduces platform driver model for dm644x and dm355 Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 4: Add dummy codec support Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 5: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 6: Add machine driver support for DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> and Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> on Monstavista 2.6.18 kernel.

1 0

[alsa-devel] [PATCH] Compaq Presario CQ60 patching for Conexant
by Tony Vroon 28 May '09

28 May '09

A docking mic control is shown by default. The Compaq Presario CQ60 laptop has no docking connector, so designate it as a CXT5051_HP model. This makes the phantom mixer slider disappear. Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56ce19e..4fcbe21 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1848,6 +1848,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), + SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),

2 1

[alsa-devel] Testers wanted: New SB X-Fi driver
by jimi o'beirne 28 May '09

28 May '09

Takashi Thanks so much for making this driver! Unfortunately, I'm a bit of a newbie, how would I actually install the driver? I am using Ubuntu Jaunty with a SB X-Fi card. Thanks for your help, Jim

1 0

Re: [alsa-devel] [Pkg-alsa-devel] Bug#529940: lib32asound2: symbol versioning changed in an incompatible way
by Elimar Riesebieter 28 May '09

28 May '09

tags 529940 pending forwarded 529940 alsa-devel(a)alsa-project.org thanks * Adeodato Simó [090527 13:53 +0200] > severity 529940 serious > retitle 529940 lib32asound2: symbol versioning changed in an incompatible way (ALSA_0.9.0rc4/8 dropped) > thanks > > Hello, > > it seems that between 1.0.19-1 and 1.0.20-1 the list of symbols in > lib32asound2 changed. More precisely, the list of symbols remained but > their associated versioning changed: all symbols associated with > ALSA_0.9.0rc4 and ALSA_0.9.0rc8 were changed to live under ALSA_0.9 > instead, making applications compiled against <= 1.0.19 fail to start > due to missing symbols. > > Note that this problem affects the 32-bit library in amd64, but not the > native libraries nor in i386 nor amd64. Don't we have to verify it at ppc64 as well? CC'd to debian-ppc. > > Also note that we've started having applications like WINE 1.1.19 in > experimental rebuilt against 1.0.20, and which don't work against > alsa-lib 1.0.19 either. And we'll have no easy way to detect these, > since shlibs are not bumped. > > Will you be able to upload fixed packages in a timely manner? According to [0] I created svn version 2249 on alioth [1]. I've tested it on amd64, were rc4 symbols appeared by checking via readelf. But maybe alsa-devel rechecks the buildsystem of alsa-libs to avoid that dirty hacks? --with-versioned appended to ./configure opts didn't help. [0] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016083.html [1] http://svn.debian.org/wsvn/pkg-alsa/?op=comp&compare[]=%2F@2248&compare[]=%… Jordi, please recheck and upload. Index: debian/rules =================================================================== --- debian/rules (Revision 2248) +++ debian/rules (Arbeitskopie) @@ -39,7 +39,7 @@ endif endif -CFLAGS = -g -Wall +CFLAGS = -g -Wall -DUSE_VERSIONED_SYMBOLS ifneq (,$(findstring noopt,$(DEB_BUILD_OPTIONS))) CFLAGS += -O0 -- "Talking much about oneself can also be a means to conceal oneself." -Friedrich Nietzsche

2 1

[alsa-devel] skype and old alsa RC4 symbols
by Jon Smirl 27 May '09

27 May '09

I would like to run pulseaudio 9.15 contained in this ubuntu repository since it fixes my problems with S/PDIF https://launchpad.net/~themuso/+archive/ppa However, the 32b compatibility library in this build don't work with skype since it is missing some RC4 obsolete symbols. jonsmirl@terra:/home/apps/alsa-lib-1.0.19/debian/lib32asound2/usr/lib32$ readelf -W -s libasound.so.2.0.0 | grep snd_pcm_hw_params_get_channels_min 174: 0004a970 58 FUNC WEAK DEFAULT 12 snd_pcm_hw_params_get_channels_min@@ALSA_0.9 454: 0004a970 58 FUNC GLOBAL DEFAULT 12 __snd_pcm_hw_params_get_channels_min@@ALSA_0.9 The 64b library has the missing RC4 symbols. jonsmirl@terra:/home/apps/alsa-lib-1.0.19/debian/libasound2/usr/lib$ readelf -W -s libasound.so.2.0.0 | grep snd_pcm_hw_params_get_channels_min 173: 000000000004ff10 15 FUNC GLOBAL DEFAULT 12 snd_pcm_hw_params_get_channels_min@@ALSA_0.9.0rc4 177: 000000000004ff20 31 FUNC GLOBAL DEFAULT 12 snd_pcm_hw_params_get_channels_min(a)ALSA_0.9 462: 000000000004ff10 15 FUNC GLOBAL DEFAULT 12 __snd_pcm_hw_params_get_channels_min@@ALSA_0.9 jonsmirl@terra:/home/apps/alsa-lib-1.0.19/debian/libasound2/usr/lib$ How do I get the RC4 symbols into the 32 library? I can pull down the source for this and build without problem. I just can't identify what needs to be changed to make the symbols appear in the 32b compat lib. -- Jon Smirl jonsmirl(a)gmail.com

2 2

Re: [alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by Takashi Iwai 27 May '09

27 May '09

At Wed, 27 May 2009 12:35:49 -0400, Sentinel wrote: > > Yeah it didn't work. > > I am lost and generally I can find the answers quickly. I cant use the > "kernel" alsa due to the spdif out and for some reason those undefined > messages just wont go away. I looked into the kernel source and they both > do exist in sound core. I can't imagine what could be wrong. If it's really an ALSA module (not soundcore), it's often because you didn't unload some old modules before loading a new one. Also, if the old modules are also update modules located in /lib/modules/$VERSION/update or so, then the old modules will still win over the newly installed one in /lib/modules/$VERSION/kernel/sound/*. IIRC, Ubuntu or some others install the update modules there. In such a case, you'll need to specify the installation path explicitly via configure option, too. If it's the sound core, you need to make sure that this is really built in, either into the kernel or as a module. And, the kernel tree really points to the version you are running. There are symbols to be referred, so it's not only the source you need to build external modules. At best, you should consult on user forum (or such) for your own distro, since this topic is really distro-specific. Takashi > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, May 27, 2009 1:35 AM > To: Sentinel > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] spdif/iec958 outputs at VT1708S not recognized > > At Tue, 26 May 2009 22:57:31 -0400, > Sentinel wrote: > > > > Okay I figured out my own problem. > > > > > > > > This is specific to those who run 64bit OS. > > > > > > > > for anyone who is getting register_sound_special_device error when > > attempting to compile ALSA, on a 64 bit OS make sure your running your > > configure properly. Most likely your distro has ALSA libraries located in > > the /usr/lib64 path not in /usr/lib and most likely the ld.so.conf is set > to > > look in /usr/lib64 before it searches /usr/lib which is the default for > the > > configure script. As a warning I'd not try to change that order > especially > > if you have built any 32bit compatibility drivers into /usr/lib > > I guess you are looking at a wrong place. > The fix must be coincident :) > > Basically alsa-driver (and kernel) compilation doesn't use any > user-space library, so libpath is irrelevant. The only important > location is the kernel source tree and its build stub. If these > aren't set up properly, such an error (version mismatch or so) may > happen. > > Note that sound_register_special() and co aren't defined in the ALSA > driver, but it's defined in sound_core.c in the kernel. > > > Takashi > >

1 0

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 27 May '09

27 May '09

The following changes since commit 0f89bdcac61536c5cb2a095a514657019573afb4: Peter Ujfalusi (1): ASoC: TWL4030: HandsfreeL/R mute DAPM switch are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Jon Smirl (4): ASoC: Main rewite of the mpc5200 audio DMA code ASoC: AC97 driver for mpc5200 ASoC: Support for AC97 on Phytec pmc030 base board. ASoC: Fabric bindings for STAC9766 on the Efika Mark Brown (2): ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved ASoC: Switch FSL SSI DAI over to symmetric_rates sound/soc/fsl/Kconfig | 28 +++- sound/soc/fsl/Makefile | 5 + sound/soc/fsl/efika-audio-fabric.c | 90 +++++++ sound/soc/fsl/fsl_ssi.c | 9 +- sound/soc/fsl/mpc5200_dma.c | 442 ++++++++++++++++++++++------------- sound/soc/fsl/mpc5200_dma.h | 33 ++-- sound/soc/fsl/mpc5200_psc_ac97.c | 331 ++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 ++ sound/soc/fsl/mpc5200_psc_i2s.c | 247 ++++---------------- sound/soc/fsl/mpc5200_psc_i2s.h | 12 + sound/soc/fsl/pcm030-audio-fabric.c | 90 +++++++ 11 files changed, 905 insertions(+), 397 deletions(-) create mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c

2 1

Re: [alsa-devel] 2.6.30: xruns caused by "ALSA: pcm - Safer boundary checks"
by Takashi Iwai 27 May '09

27 May '09

At Tue, 26 May 2009 13:16:49 -0400, Robert Krakora wrote: > > On Tue, May 26, 2009 at 11:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 26 May 2009 10:28:26 -0400, > > Robert Krakora wrote: > >> > >> On Tue, May 26, 2009 at 4:03 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Sun, 24 May 2009 23:56:40 -0700, > >> > Nathan Grennan wrote: > >> >> > >> >> Takashi Iwai wrote: > >> >> > Testing the patch would be helpful :) > >> >> > > >> >> > >> >> Will do > >> > > >> > Did you get any positive / negative result? > >> > > >> > We are in the late rc stage, so a fix should be merged ASAP... > >> > > >> > > >> > thanks, > >> > > >> > Takashi > >> > _______________________________________________ > >> > Alsa-devel mailing list > >> > Alsa-devel(a)alsa-project.org > >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >> > > >> > > >> > >> Takashi: > >> > >> I cannot speak for the other person experiencing this problem. > >> However, neither your snapshot or Jarloslav's patch corrected the > >> problem which I am experiencing. > > > > So, did you get the same kernel messages when xrun_debug is set? > > > >> I have not looked at the code much > >> in pcm_lib.c or pcm_native.c but it seems as though you are trying to > >> manage FIFO overflow and underflow scenarios. Is my assumption > >> correct? ;-) > > > > No. The problem appears like an invalid check of jiffies in > > snd_pcm_hw_ptr_update*(). After the pause, jiffies isn't recorded > > properly, thus it gets screwed up. > > > > To be sure, could you try the patch below? This will disable the > > jiffies check completely. If the problem still happens, then it's > > somewhere else where we need to check. > > > > > > thanks, > > > > Takashi > > > > --- > > diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c > > index a2a792c..47837a6 100644 > > --- a/sound/core/pcm_lib.c > > +++ b/sound/core/pcm_lib.c > > @@ -249,6 +249,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) > > new_hw_ptr = hw_base + pos; > > } > > } > > +#if 0 /* no jiffies check */ > > /* Skip the jiffies check for hardwares with BATCH flag. > > * Such hardware usually just increases the position at each IRQ, > > * thus it can't give any strange position. > > @@ -279,6 +280,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) > > delta = 0; > > } > > no_jiffies_check: > > +#endif /* no jiffies check */ > > if (delta > runtime->period_size + runtime->period_size / 2) { > > hw_ptr_error(substream, > > "Lost interrupts? " > > @@ -336,6 +338,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) > > hw_base = 0; > > new_hw_ptr = hw_base + pos; > > } > > +#if 0 /* no jiffies check */ > > if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { > > hw_ptr_error(substream, > > "hw_ptr skipping! " > > @@ -345,6 +348,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) > > ((delta * HZ) / runtime->rate)); > > return 0; > > } > > +#endif /* no jiffies check */ > > if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && > > runtime->silence_size > 0) > > snd_pcm_playback_silence(substream, new_hw_ptr); > > > > > > Takashi: > > Your patch worked. jiffies seems to be the cause. xrun_debug is on > and I do not see any of the error messages. [Re-added some dropped addresses to Cc] Thanks for testing! I have an impression that the jiffies check can behave pretty buggy in cases the hardware doesn't give a smooth pointer update. It's supposed to filter out spontaneous invalid pointer values. That'll work indeed (although I don't think the check in snd_pcm_period_elapsed() is worth). But what if a hardware returns the pointer value that is not always updated smoothly? Assume a hardware returning only spontaneously updated values, 0 at t=0, 1000 at t=100, 2000 at t=200, and so on. If snd_pcm_update_hw_ptr() is called at t=99, it gets still position 0. Then when it's called at t=100, it suddenly increases to position 1000 while jiffies delta is only 1. The current code would filter out this update because it appears as if too much. The problem continues, if snd_pcm_update_hw_ptr() is constantly called at each t=100, 101, ... At each call, jiffies delta is 1, but the position delta is still 1000 because it wasn't updated. That is, as long as the app keeps calling snd_pcm_update_hw_ptr(), the position won't be updated. This is pretty possible especially with an app like pulseaudio. I have a feeling that we should disable the jiffies check in that part, at least for 2.6.30, until a better solution is found. The jiffies check is already done and works pretty well in the driver side (hda-intel and intel8x0). But this check in the PCM core can be (not always, though) buggy, and actually gives problems sometimes. Takashi

3 3

[alsa-devel] [PATCH V2] Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
by Jon Smirl 27 May '09

27 May '09

The function signature for spin_event_timeout() has changed in version V9. Adjust the mpc5200 AC97 driver to use the new function. v2 - switch back to udelay() for fixed timeouts. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/mpc5200_psc_ac97.c | 30 ++++++++++++++---------------- 1 files changed, 14 insertions(+), 16 deletions(-) diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 9f2df15..4342b51 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -31,13 +31,13 @@ static struct psc_dma *psc_dma; static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - int rc; + int status; unsigned int val; /* Wait for command send status zero = ready */ - spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); - if (rc == 0) { + status= spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); return -ENODEV; } @@ -45,9 +45,9 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); /* Wait for the answer */ - spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_DATA_VAL), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL), 100, 0); + if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); return -ENODEV; @@ -66,12 +66,12 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - int rc; + int status; /* Wait for command status zero = ready */ - spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); return; } @@ -82,24 +82,22 @@ static void psc_ac97_write(struct snd_ac97 *ac97, static void psc_ac97_warm_reset(struct snd_ac97 *ac97) { - int rc; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); - spin_event_timeout(0, 3, 0, rc); + udelay(3); out_be32(&regs->sicr, psc_dma->sicr); } static void psc_ac97_cold_reset(struct snd_ac97 *ac97) { - int rc; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; /* Do a cold reset */ out_8(&regs->op1, MPC52xx_PSC_OP_RES); - spin_event_timeout(0, 10, 0, rc); + udelay(10); out_8(&regs->op0, MPC52xx_PSC_OP_RES); - spin_event_timeout(0, 50, 0, rc); + udelat(50); psc_ac97_warm_reset(ac97); }

2 1

[alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by Sentinel 27 May '09

27 May '09

Okay I figured out my own problem. This is specific to those who run 64bit OS. for anyone who is getting register_sound_special_device error when attempting to compile ALSA, on a 64 bit OS make sure your running your configure properly. Most likely your distro has ALSA libraries located in the /usr/lib64 path not in /usr/lib and most likely the ld.so.conf is set to look in /usr/lib64 before it searches /usr/lib which is the default for the configure script. As a warning I'd not try to change that order especially if you have built any 32bit compatibility drivers into /usr/lib ./configure -prefix=/usr/lib64 Although it wouldn't be a bad idea for the configure to detect that if possible.

2 1

[alsa-devel] [PATCH V3] Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
by Jon Smirl 27 May '09

27 May '09

The function signature for spin_event_timeout() has changed in version V9. Adjust the mpc5200 AC97 driver to use the new function. v2 - switch back to udelay() for fixed timeouts. v3 - forgot about driver being market broken, fix typo Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/mpc5200_psc_ac97.c | 30 ++++++++++++++---------------- 1 files changed, 14 insertions(+), 16 deletions(-) diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 9f2df15..794a247 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -31,13 +31,13 @@ static struct psc_dma *psc_dma; static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - int rc; + int status; unsigned int val; /* Wait for command send status zero = ready */ - spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); return -ENODEV; } @@ -45,9 +45,9 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); /* Wait for the answer */ - spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_DATA_VAL), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL), 100, 0); + if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); return -ENODEV; @@ -66,12 +66,12 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - int rc; + int status; /* Wait for command status zero = ready */ - spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); return; } @@ -82,24 +82,22 @@ static void psc_ac97_write(struct snd_ac97 *ac97, static void psc_ac97_warm_reset(struct snd_ac97 *ac97) { - int rc; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); - spin_event_timeout(0, 3, 0, rc); + udelay(3); out_be32(&regs->sicr, psc_dma->sicr); } static void psc_ac97_cold_reset(struct snd_ac97 *ac97) { - int rc; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; /* Do a cold reset */ out_8(&regs->op1, MPC52xx_PSC_OP_RES); - spin_event_timeout(0, 10, 0, rc); + udelay(10); out_8(&regs->op0, MPC52xx_PSC_OP_RES); - spin_event_timeout(0, 50, 0, rc); + udelay(50); psc_ac97_warm_reset(ac97); }

1 0

[alsa-devel] [PATCH] Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
by Jon Smirl 27 May '09

27 May '09

The function signature for spin_event_timeout() has changed in version V9. Adjust the mpc5200 AC97 driver to use the new function. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/mpc5200_psc_ac97.c | 30 ++++++++++++++---------------- 1 files changed, 14 insertions(+), 16 deletions(-) diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 9f2df15..bab80ac 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -31,13 +31,13 @@ static struct psc_dma *psc_dma; static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - int rc; + int status; unsigned int val; /* Wait for command send status zero = ready */ - spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); - if (rc == 0) { + status= spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); return -ENODEV; } @@ -45,9 +45,9 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); /* Wait for the answer */ - spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_DATA_VAL), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL), 100, 0); + if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); return -ENODEV; @@ -66,12 +66,12 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - int rc; + int status; /* Wait for command status zero = ready */ - spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); return; } @@ -82,24 +82,22 @@ static void psc_ac97_write(struct snd_ac97 *ac97, static void psc_ac97_warm_reset(struct snd_ac97 *ac97) { - int rc; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); - spin_event_timeout(0, 3, 0, rc); + spin_event_timeout(0, 3, 0); out_be32(&regs->sicr, psc_dma->sicr); } static void psc_ac97_cold_reset(struct snd_ac97 *ac97) { - int rc; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; /* Do a cold reset */ out_8(&regs->op1, MPC52xx_PSC_OP_RES); - spin_event_timeout(0, 10, 0, rc); + spin_event_timeout(0, 10, 0); out_8(&regs->op0, MPC52xx_PSC_OP_RES); - spin_event_timeout(0, 50, 0, rc); + spin_event_timeout(0, 50, 0); psc_ac97_warm_reset(ac97); }

3 8

[alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by Sentinel 27 May '09

27 May '09

Well I guess not it still didn't work and its not -prefix= its -libdir=

1 0

[alsa-devel] [PATCH] ASoC: Switch FSL SSI DAI over to symmetric_rates
by Mark Brown 27 May '09

27 May '09

The effect of symmetric_constraints should provide a standard way to enforce the use of the same sample rate for both directions. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/fsl/fsl_ssi.c | 9 +++------ 1 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 47afaa9..93f0f38 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -375,18 +375,14 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_pcm_runtime *first_runtime = ssi_private->first_stream->runtime; - if (!first_runtime->rate || !first_runtime->sample_bits) { + if (!first_runtime->sample_bits) { dev_err(substream->pcm->card->dev, - "set sample rate and size in %s stream first\n", + "set sample size in %s stream first\n", substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "capture" : "playback"); return -EAGAIN; } - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - first_runtime->rate, first_runtime->rate); - /* If we're in synchronous mode, then we need to constrain * the sample size as well. We don't support independent sample * rates in asynchronous mode. @@ -693,6 +689,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) fsl_ssi_dai->name = ssi_private->name; fsl_ssi_dai->id = ssi_info->id; fsl_ssi_dai->dev = ssi_info->dev; + fsl_ssi_dai->symmetric_rates = 1; ret = snd_soc_register_dai(fsl_ssi_dai); if (ret != 0) { -- 1.6.3.1

2 1

[alsa-devel] [PATCH V5 0/5] AC97 driver for mpc5200
by Jon Smirl 26 May '09

26 May '09

Version 5. Modified to use spin_event_timeout() as originally written. --- Jon Smirl (5): Fabric bindings for STAC9766 on the Efika Support for AC97 on Phytec pmc030 base board. AC97 driver for mpc5200 Main rewite of the mpc5200 audio DMA code powerpc: introduce macro spin_event_timeout() arch/powerpc/include/asm/delay.h | 33 +++ sound/soc/fsl/Kconfig | 27 ++ sound/soc/fsl/Makefile | 5 sound/soc/fsl/efika-audio-fabric.c | 90 +++++++ sound/soc/fsl/mpc5200_dma.c | 442 ++++++++++++++++++++++------------- sound/soc/fsl/mpc5200_dma.h | 33 +-- sound/soc/fsl/mpc5200_psc_ac97.c | 331 ++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + sound/soc/fsl/mpc5200_psc_i2s.c | 247 +++----------------- sound/soc/fsl/mpc5200_psc_i2s.h | 12 + sound/soc/fsl/pcm030-audio-fabric.c | 90 +++++++ 11 files changed, 934 insertions(+), 391 deletions(-) create mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- Jon Smirl jonsmirl(a)gmail.com

3 9

[alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by Sentinel 26 May '09

26 May '09

I tried using the snapshot because I was having the same problem as Peter was. No spdif out on my vt1708s. Unfortunately for me my build didn't go quite as well, for some reason I had 2 undefined which make the snd.ko fail to load. Is there something that I'm missing? I am running slamd64 distro current, with the latest kernel along with the alsa kernel patch "2.6.29.4-rt15". Please help. Building modules, stage 2. MODPOST 131 modules WARNING: "register_sound_special_device" [/usr/src/alsa-driver/acore/snd.ko] undefined! WARNING: "unregister_sound_special" [/usr/src/alsa-driver/acore/snd.ko] undefined! make[1]: Leaving directory `/usr/src/linux-2.6.29.4' utils/link-modules /usr/src/alsa-driver

1 0

[alsa-devel] [PATCH V4 0/5] AC97 driver for mpc5200
by Jon Smirl 26 May '09

26 May '09

Version 4. Changed timeouts to use relax_cpu() instead of udelay(). Fixed locking to lock bestcomm instead of registers. Code reorg as requested in the various comments. Removed sysfs over/underrun reporting. Removed reset retries. Removing the retries makes the driver fail to load about 5% of the time. An oscilliscope is needed to figure out what is going on with failed resets. Both pcm030 and Efika ocasionally fail to reset and they use different codec chips. --- Jon Smirl (5): Fabric bindings for STAC9766 on the Efika Support for AC97 on Phytec pmc030 base board. AC97 driver for mpc5200 Main rewite of the mpc5200 audio DMA code The macro spin_event_timeout() takes a condition and timeout value arch/powerpc/include/asm/delay.h | 33 +++ sound/soc/fsl/Kconfig | 27 ++ sound/soc/fsl/Makefile | 5 sound/soc/fsl/efika-audio-fabric.c | 90 +++++++ sound/soc/fsl/mpc5200_dma.c | 442 ++++++++++++++++++++++------------- sound/soc/fsl/mpc5200_dma.h | 43 ++- sound/soc/fsl/mpc5200_psc_ac97.c | 331 ++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + sound/soc/fsl/mpc5200_psc_i2s.c | 247 +++----------------- sound/soc/fsl/mpc5200_psc_i2s.h | 12 + sound/soc/fsl/pcm030-audio-fabric.c | 90 +++++++ 11 files changed, 946 insertions(+), 389 deletions(-) create mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- Jon Smirl jonsmirl(a)gmail.com

7 24

[alsa-devel] Fixes in intel8x0.c
by Pavel Hofman 26 May '09

26 May '09

Hi, while trying to dive into complexities of intel8x0.c I found the following: static struct snd_pci_quirk intel8x0_clock_list[] __devinitdata = { SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000), SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100), ..... Is it really supposed to be 41000, instead of 44100? In static int snd_intel8x0_pcm_open(struct snd_pcm_substream *substream, struct ichdev *ichdev) ... runtime->hw.rates = ichdev->pcm->rates; The pcm field in ichdev is snd_ac97, while the rates field in snd_ac97 is defined unsigned int rates[6]; /* see AC97_RATES_* defines */ I guess this way always rates[0] from the list is taken, unlike e.g. in via82xx.c: runtime->hw.rates = chip->ac97->rates[AC97_RATES_SPDIF]; It seems a bug to me, though I may be wrong of course. Thanks and regards, Pavel

2 5

[alsa-devel] period_size and .asoundrc
by Canella Matteo 26 May '09

26 May '09

Hi everybody, I'm writing a little software that play a tone on a pcm device. In the hw param initialization I'm hardcoding the "periods" and "period_size" values: unsigned int periods = 4; if (snd_pcm_hw_params_set_periods_near(buzzer_device.pcm_handle, buzzer_device.hwparams, &periods, 0) < 0) { fprintf(stderr, "Error setting periods.\n"); return 1; } ///* set the period time */ period_size = 8192; err = snd_pcm_hw_params_set_period_size_near(buzzer_device.pcm_handle, buzzer_device.hwparams, &period_size, 0); if (err < 0) { printf("Unable to get period size for playback: %s\n", snd_strerror(err)); return err; } I'd like to read this parameter form the .asoundrc file, so I'm wondering if there were API that does this task. Thanks Matteo Canella

2 1

[alsa-devel] 2.6.30: xruns caused by "ALSA: pcm - Safer boundary checks"
by Chuck Ebbert 26 May '09

26 May '09

Bug report: https://bugzilla.redhat.com/show_bug.cgi?id=498858 To reproduce, pause a video in mplayer and then hit play: ALSA sound/core/pcm_lib.c:263: hda_codec: hw_ptr skipping! [Q] (pos=11263, delta=17400, period=1024, jdelta=21/362/0) ALSA sound/core/pcm_lib.c:337: hda_codec: hw_ptr skipping! (pos=11730, delta=1490, period=1024, jdelta=10/31) ALSA sound/core/pcm_lib.c:337: hda_codec: hw_ptr skipping! (pos=11734, delta=1494, period=1024, jdelta=10/31) ALSA sound/core/pcm_lib.c:337: hda_codec: hw_ptr skipping! (pos=11734, delta=1494, period=1024, jdelta=10/31) ALSA sound/core/pcm_lib.c:337: hda_codec: hw_ptr skipping! (pos=11735, delta=1495, period=1024, jdelta=10/31) ALSA sound/core/pcm_lib.c:263: hda_codec: hw_ptr skipping! [Q] (pos=12287, delta=2047, period=1024, jdelta=21/42/0) ALSA sound/core/pcm_lib.c:337: hda_codec: hw_ptr skipping! (pos=12449, delta=2209, period=1024, jdelta=4/46)

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[alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by Sentinel 26 May '09

26 May '09

I tried using the snapshot because I was having the same problem as Peter was. No spdif out on my vt1708s. Unfortunately for me my build didn't go quite as well, for some reason I had 2 undefined which make the snd.ko fail to load. Is there something that I'm missing? I am running slamd64 distro current, with the latest kernel along with the alsa kernel patch "2.6.29.4-rt15". Please help. Building modules, stage 2. MODPOST 131 modules WARNING: "register_sound_special_device" [/usr/src/alsa-driver/acore/snd.ko] undefined! WARNING: "unregister_sound_special" [/usr/src/alsa-driver/acore/snd.ko] undefined! make[1]: Leaving directory `/usr/src/linux-2.6.29.4' utils/link-modules /usr/src/alsa-driver

1 0

Re: [alsa-devel] Help on writing an ALSA ASoC Driver for WM8985
by Leung Chi Tat 26 May '09

26 May '09

Mark, I got it. Once we set those volume update bits in wm8990_init(), those bits in the wm8985 register cache array, wm8985_reg[], will then be set too. So, whenever we write a new value to a volume register, wm8985_write() will then write the new value to the volume register and update the corresponding volume update bit also. This is the trick. A Great Idea! Thank you so much! Chitat On Mon, May 25, 2009 at 08:47:31PM +0800, Leung Chi Tat wrote: > I'm right now studying the driver code for wm8980 codec (from the main > trunk of linux-2.6-asoc) as it seems to be more close to wm8985. I > can't find any code for triggering those volume update bits after > modified those volume registers. But, the wm8980 datasheet states > that we have to do so to make those volume updates properly. Is there > some black magic in the code? The normal approach for volume update bits is to set them when initialising the CODEC at startup - this is what the wm8990 driver is doing when it writes to the volume registers in wm8990_init(). This means that any further register writes which only change the volume field in the registers will also have the volume update bit set.

1 0

[alsa-devel] Question about snd_dma_buffer bytes field
by Troy Kisky 26 May '09

26 May '09

Does (substream->dma_buffer.bytes) need to equal (snd_pcm_lib_period_bytes(substream) * runtime->periods) ? When this is not true, it seems like the buffer is not being filled correctly. for testing, this works .buffer_bytes_max = 16 * 7 * 512, .period_bytes_min = 32, .period_bytes_max = 7 * 512, /* This is size of ping + pong buffer*/ and this doesn't. .buffer_bytes_max = 16 * 7 * 512 + 512, .period_bytes_min = 32, .period_bytes_max = 7 * 512, /* This is size of ping + pong buffer*/ When playing back a sine wave, I can hear breaks. Thanks Troy

2 2

[alsa-devel] [snd-ctxfi] Test EMU20K2
by Sylvain Loulergue 25 May '09

25 May '09

Hi, Modules snd-ctxfi works fine for me (http://www.alsa-project.org/db/?f=e90356f82f257f499e8028d8a28c98775fd765e7). If you need some specific test just say it. thanks for this devel

1 0

[alsa-devel] Alsamixer isn't working properly with HDA VIA VT82xx with Realtek ALC888 Codec
by Stephan Jänecke 25 May '09

25 May '09

Dear developers, alsamixer doesn't work on my fedora machine. When I type in 'alsamixer -c 0', I get "hda_codec: num-steps=0 for NID=0xf (ctl=Side Playback Volume)" "alsamixer: function snd_mixer_load failed: Invalid argument" Although Pulseaudio and OSS Mixers seem to be present, I don't get any sound. Please have a look at http://www.alsa- project.org/db/?f=15c9f665eb141dee755a533d804456e81442f270 or check the attachment. I hope you can help me solving my problem. Regards, Stephan Jänecke PS: I had the same mixer problem with Mandriva 2009.0 before I installed Fedora 10 to check whether it's an distribution specific bug. Whereas alsamixer didn't work properly as well, I could enjoy working sound. -- Powered by openSUSE 11.1 and KMail 1.10.3

2 5

[alsa-devel] pb with AC'97 codec on atmel at91 boards
by Sedji Gaouaou 25 May '09

25 May '09

Hi all, I am currently working on an AC'97 driver for ATMEL boards(AT91), and I encounter some problems. On the board we have an AD1981b, and looking at the file ac97_patch.c I was wondering in the function patch_ad1881 why is there 3 codecs? I've got the AD1981b datasheet, but the registers description is not really clear to me(vendor id register). My problem is: when I try to read this registers(vendor id) during the probe(in snd_ac97_mixer), it will fail, and from time to time I will even not be able to recognize the codec... So my question is: is that normal that in the function patch_ad1881 the code fails to read these lines: codecs[1] = patch_ad1881_unchained(ac97, 1, (1<<14)); codecs[2] = patch_ad1881_unchained(ac97, 2, (1<<13)); Regards, Sedji

1 0

[alsa-devel] usb audio alsa driver
by wei zhang 25 May '09

25 May '09

hi I was for our devices under linux audio alsa driver, I encountered a lot of trouble now, please help me. My audio device is this: usb interface types, transmission of data through the INT endpoint, each 64 bytes, read out the PCM audio data stream, the sampling rate is 22,050, a single channel, I have the debugging information data crawlers , but I am under no voice in skype, you can tell me the possible reason? Or give me some recommendations, I am very anxious, I would like to thank.I sent my driver, please help me show you, thank you

1 0

Re: [alsa-devel] Help on writing an ALSA ASoC Driver for WM8985
by Leung Chi Tat 25 May '09

25 May '09

Dear Mark, Thank you so much! I right now have a more clear picture. I'm right now studying the driver code for wm8980 codec (from the main trunk of linux-2.6-asoc) as it seems to be more close to wm8985. I can't find any code for triggering those volume update bits after modified those volume registers. But, the wm8980 datasheet states that we have to do so to make those volume updates properly. Is there some black magic in the code? Chitat On Mon, May 25, 2009 at 11:31:48AM +0800, Leung Chi Tat wrote: > 1. describing those important structures, e.g. struct soc_enum, struct snd_kcontrol_new, struct snd_soc_dapm_widget; An enum is used to define an enumerated ALSA control. A kcontrol is a generic ALSA control - there's nothing ASoC specific about it - and a DAPM widget is a thing for which there is power control. > 2. what is/are the relationships among those important structures; > 3. what is the differences between those controls for struct snd_kcontrol_new and struct snd_soc_dapm_widget; It'd be easier to answer this if you had some more specific questions? In general the answer is that things only go into DAPM if they have some relevance to power - that is, if they're a thing for which there is power control or if they control the routing of audio through the chip (and therefore let the core decide which parts of the chip need to be powered on). As has already been suggested the easiest thing is probably to look at other CODEC drivers and compare what their drivers do with the datasheets. > 4. Should those sinks, sources, paths be defined in the arrays of struct snd_dapm_widget; Yes, all the things you have power control for should be there. > 5. How can I select those defined paths through ALSA user-mode library as I can't find any examples in those ALSA tutorials; The controls are all visible as normal ALSA controls - there's nothing ASoC specific in the userspace API. Look with an application such as amixer or alsamixer to see what's there. > Indeed, I'm right now trying to write a WM8985 driver for my s3c6400 based platform based on the kernel source v2.6.21 from Samsung. Note that this is rather old and there were a few issues in the example code with the way that the clocking was set up for the CPU - mostly due to direct access to the system clocking registers which could interfere with other drivers. It might be worth looking at the very latest linux-next tree where there's a mostly complete implementation of audio support.

2 1

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 25 May '09

25 May '09

The following changes since commit 86ed3669f068b514ab85ffd548456a342b3fb8d3: Mark Brown (1): ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Jon Smirl (4): ASoC: Codec for STAC9766 used on the Efika ASoC: Basic split of mpc5200 DMA code out of mpc5200_psc_i2s ASoC: Rename the PSC functions to DMA ASoC: Add a few more mpc5200 PSC defines Mark Brown (2): ASoC: Fix WM9081 PowerPC compiler issues ASoC: Fix minor issues in STAC9766 driver Peter Ujfalusi (3): ASoC: TWL4030: Handsfree pop removal redesign ASoC: TWL4030: Add shadow register ASoC: TWL4030: HandsfreeL/R mute DAPM switch arch/powerpc/include/asm/mpc52xx_psc.h | 11 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/stac9766.c | 463 +++++++++++++++++++++++ sound/soc/codecs/stac9766.h | 21 + sound/soc/codecs/twl4030.c | 99 ++++- sound/soc/codecs/twl4030.h | 7 +- sound/soc/codecs/wm9081.c | 6 +- sound/soc/fsl/Kconfig | 4 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/mpc5200_dma.c | 458 ++++++++++++++++++++++ sound/soc/fsl/mpc5200_dma.h | 81 ++++ sound/soc/fsl/mpc5200_psc_i2s.c | 645 ++++---------------------------- 13 files changed, 1217 insertions(+), 586 deletions(-) create mode 100644 sound/soc/codecs/stac9766.c create mode 100644 sound/soc/codecs/stac9766.h create mode 100644 sound/soc/fsl/mpc5200_dma.c create mode 100644 sound/soc/fsl/mpc5200_dma.h

2 1

[alsa-devel] Help on writing an ALSA ASoC Driver for WM8985
by Leung Chi Tat 25 May '09

25 May '09

Hi all, I'm new to ALSA ASoC and ALSA. I'd like get some advice where I can get more information about writing an ASoC driver. I've browsed the internet couples of days and I can only find links on ALSA driver api's and writing an ALSA driver for PCI devices. It seems there is not much information on how to writing an ALSA ASoC driver. I'm stuck on the following questions: 1. describing those important structures, e.g. struct soc_enum, struct snd_kcontrol_new, struct snd_soc_dapm_widget; 2. what is/are the relationships among those important structures; 3. what is the differences between those controls for struct snd_kcontrol_new and struct snd_soc_dapm_widget; 4. Should those sinks, sources, paths be defined in the arrays of struct snd_dapm_widget; 5. How can I select those defined paths through ALSA user-mode library as I can't find any examples in those ALSA tutorials; Indeed, I'm right now trying to write a WM8985 driver for my s3c6400 based platform based on the kernel source v2.6.21 from Samsung. Thank you so much for your valuable advice and suggestions in advance. Best Regards, LEUNG, Chi Tat Senior Software Engineer CCT Tech Advanced Products Limited 18/F, CCT Telecom Building, 11 Wo Shing Street Fo Tan, Shatin, N.T., Hong Kong Tel: +852 26005276 Fax: +852 26948660 Email: ctleung(a)cct.com.hk Website: http://www.cct-tech.com.hk

3 2

[alsa-devel] [PATCH 0/3 take2] ASoC: TWL4030: Handsfree pop removal and mute
by Peter Ujfalusi 25 May '09

25 May '09

Hello, Changes: REG_SW_SHADOW = 0x4A (instead of 0x50) The following series fixes the HandsfreeL/R pop removal sequence: The pop removal is moved from the MUX_E to PGA_E DAPM control and also the sequence is modified to match with the sequence described in the TRM. Note, that the power-down sequence is not described in the TRM, it has been implemented in a way that it should make sense. HandsetL/R mute: Since all bits are needed for the pop removal and they have to modified in a strict order/timing: I have added SW_SHADOW non HW register. This register at the moment only handles the HandsfreeL/R mute. --- Peter Ujfalusi (3): ASoC: TWL4030: Handsfree pop removal redesign ASoC: TWL4030: Add shadow register ASoC: TWL4030: HandsfreeL/R mute DAPM switch sound/soc/codecs/twl4030.c | 99 +++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/twl4030.h | 7 +++- 2 files changed, 85 insertions(+), 21 deletions(-)

2 4

[alsa-devel] [PATCH V2 0/9] mpc5200 audio rework for AC97
by Jon Smirl 25 May '09

25 May '09

The following series implements audio support for the mpc5200. It adds an AC97 driver and STAC9766 codec driver. Board support for the Efika and Phytec pcm030 are also included. Mark is not enthused about soc-of-simple.c so rather than extend it for AC97 I altered the drivers to not use it. Instead they use the old way of manually binding everything. Mark would like to see OF binding more closely integrated to the core. Once a proper solution for OF binding is agreed upon it is easy to convert the existing drivers. A first step would be converting the existing codec drivers so that they can be dynamically loaded. Grant, please check over the spin locking on register access. I'm not clear on why and when the registers have to be protected. Once these basic drivers are in-kernel and more people are testing them, I can add more features like pause/resume, power management, etc based on feedback. I2S will get examined in more detail for the next kernel cycle. Our multi-channel prototype I2S hardware is just about working. Once I get the multi-channel hardware I will implement and heavily test a lot more I2S capability. --- Jon Smirl (9): Support for AC97 on Phytec pmc030 base board. Fabric bindings for STAC9766 on the Efika AC97 driver for mpc5200 Codec for STAC9766 used on the Efika Main rewite of the mpc5200 audio DMA code Add a few more mpc5200 PSC defines Rename the PSC functions to DMA Basic split of mpc5200 DMA code out from mpc5200_psc_i2s Register the wm9712 DAIs on module load sound/soc/fsl/Kconfig | 31 + sound/soc/fsl/Makefile | 7 sound/soc/fsl/efika-audio-fabric.c | 94 ++++ sound/soc/fsl/mpc5200_dma.c | 635 ++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_dma.h | 80 ++++ sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + sound/soc/fsl/mpc5200_psc_i2s.c | 750 ++--------------------------------- sound/soc/fsl/mpc5200_psc_i2s.h | 12 + sound/soc/fsl/pcm030-audio-fabric.c | 94 ++++ 10 files changed, 1414 insertions(+), 698 deletions(-) create mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_dma.c create mode 100644 sound/soc/fsl/mpc5200_dma.h create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] [PATCH 1/2] pxa2xx-ac97-lib: move set_resetgpio to pxa27x.c
by Dmitry Eremin-Solenikov 25 May '09

25 May '09

The code related to set_resetgpio is pxa27x specific and should not be present in sound driver lib. Move it to appropriate place. The code is moved to arch/arm as it will be converted from (exported) pxa_gpio_mode() calls to use MFP which are unexported to modules. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> Cc: Eric Miao <eric.miao(a)marvell.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- arch/arm/mach-pxa/include/mach/audio.h | 24 ++++++++++++++++ arch/arm/mach-pxa/pxa27x.c | 37 +++++++++++++++++++++++++ sound/arm/pxa2xx-ac97-lib.c | 47 +++----------------------------- 3 files changed, 65 insertions(+), 43 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/audio.h b/arch/arm/mach-pxa/include/mach/audio.h index 16eb025..517c868 100644 --- a/arch/arm/mach-pxa/include/mach/audio.h +++ b/arch/arm/mach-pxa/include/mach/audio.h @@ -24,4 +24,28 @@ typedef struct { extern void pxa_set_ac97_info(pxa2xx_audio_ops_t *ops); +/* + * Beware PXA27x bugs: + * + * o Slot 12 read from modem space will hang controller. + * o CDONE, SDONE interrupt fails after any slot 12 IO. + * + * We therefore have an hybrid approach for waiting on SDONE (interrupt or + * 1 jiffy timeout if interrupt never comes). + */ + +enum { + RESETGPIO_FORCE_HIGH, + RESETGPIO_FORCE_LOW, + RESETGPIO_NORMAL_ALTFUNC +}; + +#if defined(CONFIG_PXA27x) +extern void pxa27x_ac97_reset(int reset_gpio, int resetgpio_action); +#else +static inline void pxa27x_ac97_reset(int reset_gpio, int resetgpio_action) +{ +} +#endif + #endif diff --git a/arch/arm/mach-pxa/pxa27x.c b/arch/arm/mach-pxa/pxa27x.c index a425ec7..a0e787e 100644 --- a/arch/arm/mach-pxa/pxa27x.c +++ b/arch/arm/mach-pxa/pxa27x.c @@ -28,6 +28,7 @@ #include <mach/pm.h> #include <mach/dma.h> #include <mach/i2c.h> +#include <mach/audio.h> #include "generic.h" #include "devices.h" @@ -345,6 +346,42 @@ void __init pxa27x_set_i2c_power_info(struct i2c_pxa_platform_data *info) pxa_register_device(&pxa27x_device_i2c_power, info); } +/** + * pxa27x_ac97_reset - computes and sets the AC97_RESET gpio mode on PXA + * @mode: chosen action + * + * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line + * must be done to insure proper work of AC97 reset line. This function + * computes the correct gpio_mode for further use by reset functions, and + * applied the change through pxa_gpio_mode. + */ +/* temporary include */ +#include <mach/pxa2xx-gpio.h> +void pxa27x_ac97_reset(int reset_gpio, int resetgpio_action) +{ + int mode = 0; + + if (reset_gpio) + switch (resetgpio_action) { + case RESETGPIO_NORMAL_ALTFUNC: + if (reset_gpio == 113) + mode = 113 | GPIO_ALT_FN_2_OUT; + if (reset_gpio == 95) + mode = 95 | GPIO_ALT_FN_1_OUT; + break; + case RESETGPIO_FORCE_LOW: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; + break; + }; + + if (mode) + pxa_gpio_mode(mode); +} +EXPORT_SYMBOL(pxa27x_ac97_reset); + static struct platform_device *devices[] __initdata = { &pxa27x_device_udc, &pxa_device_ffuart, diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 6fdca97..2227314 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -42,45 +42,6 @@ static int reset_gpio; * 1 jiffy timeout if interrupt never comes). */ -enum { - RESETGPIO_FORCE_HIGH, - RESETGPIO_FORCE_LOW, - RESETGPIO_NORMAL_ALTFUNC -}; - -/** - * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA - * @mode: chosen action - * - * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line - * must be done to insure proper work of AC97 reset line. This function - * computes the correct gpio_mode for further use by reset functions, and - * applied the change through pxa_gpio_mode. - */ -static void set_resetgpio_mode(int resetgpio_action) -{ - int mode = 0; - - if (reset_gpio) - switch (resetgpio_action) { - case RESETGPIO_NORMAL_ALTFUNC: - if (reset_gpio == 113) - mode = 113 | GPIO_ALT_FN_2_OUT; - if (reset_gpio == 95) - mode = 95 | GPIO_ALT_FN_1_OUT; - break; - case RESETGPIO_FORCE_LOW: - mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; - break; - case RESETGPIO_FORCE_HIGH: - mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; - break; - }; - - if (mode) - pxa_gpio_mode(mode); -} - unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -176,10 +137,10 @@ static inline void pxa_ac97_warm_pxa27x(void) /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - set_resetgpio_mode(RESETGPIO_FORCE_HIGH); + pxa27x_ac97_reset(reset_gpio, RESETGPIO_FORCE_HIGH); udelay(10); GCR |= GCR_WARM_RST; - set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); + pxa27x_ac97_reset(reset_gpio, RESETGPIO_NORMAL_ALTFUNC); udelay(500); } @@ -353,7 +314,7 @@ int pxa2xx_ac97_hw_resume(void) } if (cpu_is_pxa27x()) { /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ - set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); + pxa27x_ac97_reset(reset_gpio, RESETGPIO_NORMAL_ALTFUNC); } clk_enable(ac97_clk); return 0; @@ -395,7 +356,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); + pxa27x_ac97_reset(reset_gpio, RESETGPIO_NORMAL_ALTFUNC); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); -- 1.6.2.4

3 3

[alsa-devel] Testers wanted: New SB X-Fi driver
by Mario Contreras 25 May '09

25 May '09

Hello, I have this card. Will the unstable version work with my card? I have read it has a different chipset. I installed it a few days ago, but there wasn't any sound. I have Ubuntu 9.04. When I test it in sound preferences it freezes. I would like to test it. If I can be helpful, I'd appreciate you help me. Than you.

3 11

[alsa-devel] [PATCH 0/3] ASoC: TWL4030: Handsfree pop removal and mute
by Peter Ujfalusi 25 May '09

25 May '09

Hello, The following series fixes the HandsfreeL/R pop removal sequence: The pop removal is moved from the MUX_E to PGA_E DAPM control and also the sequence is modified to match with the sequence described in the TRM. Note, that the power-down sequence is not described in the TRM, it has been implemented in a way that it should make sense. HandsetL/R mute: Since all bits are needed for the pop removal and they have to modified in a strict order/timing: I have added SW_SHADOW non HW register. This register at the moment only handles the HandsfreeL/R mute. --- Peter Ujfalusi (3): ASoC: TWL4030: Handsfree pop removal redesign ASoC: TWL4030: Add shadow register ASoC: TWL4030: HandsfreeL/R mute DAPM switch sound/soc/codecs/twl4030.c | 99 +++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/twl4030.h | 7 +++- 2 files changed, 85 insertions(+), 21 deletions(-)

2 5

[alsa-devel] snd_pcm_writei() return value
by Cristian Morales Vega 25 May '09

25 May '09

Hi, I have an stupid doubt... The snd_pcm_writei() documentation says: "If the blocking behaviour is selected, then routine waits until all requested bytes are played or put to the playback ring buffer. The count of bytes can be less only if a signal or underrun occurred." and at the same time it says "-EPIPE an underrun occurred" is a possible return value. Then snd_pcm_recover() documentation says: "This functions handles -EINTR (interrupted system call), -EPIPE (overrun or underrun) and -ESTRPIPE (stream is suspended) error codes trying to prepare given stream for next I/O." So if I want to write the full content of a buffer, would this be valid?: while(buffer.length > 0) { snd_pcm_sframes_t written = snd_pcm_writei(device.handle, buffer.data, buffer.length); if(written < 0) snd_pcm_recover(device.handle, written, 1); else buffer.length = 0; } "The count of bytes can be less only if a signal ... occurred." makes me think that if a signal interrups snd_pcm_writei() it will return a value between 0 and buffer.length, but "This functions handles -EINTR (interrupted system call)" makes me think that it will always return -EINTR. "The count of bytes can be less only if a ... underrun occurred.", even if technically correct, also seems to be against the "-EPIPE an underrun occurred" return value. So, in the previous code, "if(written < 0)" should be changed by "if(written < buffer.length)"? Anyway, how snd_pcm_recover() handles an -EINTR? If only half the data has been written it can't modify buffer.data to point to the last sample written+1. Perhaps I need a full: uint32_t *buffer_ptr = buffer.data; while(buffer.length > 0) { snd_pcm_sframes_t written = snd_pcm_writei(device.handle, buffer_ptr, buffer.length); if(written < 0) { snd_pcm_recover(device.handle, written, 1); } else if(written <= buffer.length) { buffer.length -= written; buffer_ptr += written; } } ? And still, in this case an interrupt would be handled by snd_pcm_recover() (written < 0) or by the "else" code? Thanks.

3 3

[alsa-devel] [PATCH] ALSA: hda - Add forced codec-slots for ASUS W5Fm
by Ozan Çağlayan 25 May '09

25 May '09

ASUS W5Fm needs the fixed codec-slots to probe to override the BIOS problem like W5F. Tested-by: Alp Kılıç <kilic.alp(a)gmail.com> Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21e99cf..4419e11 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2142,6 +2142,7 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), /* forced codec slots */ SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), + SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103), {} }; -- 1.6.2

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Re: [alsa-devel] Help on writing an ALSA ASoC Driver for WM8985
by Leung Chi Tat 25 May '09

25 May '09

Thank you so much for your suggestion. I've already copied a codec source for wm8980 from linux-2.6-ASoC main trunk. I have already to modify it back to the old api's and old structures for kernel v2.6.21. What remains I can't get through are those points I listed in my previous email. Could you give me some hints? Best Regards, LEUNG, Chi Tat Senior Software Engineer CCT Tech Advanced Products Limited 18/F, CCT Telecom Building, 11 Wo Shing Street Fo Tan, Shatin, N.T., Hong Kong Tel: +852 26005276 Fax: +852 26948660 Email: ctleung(a)cct.com.hk Website: http://www.cct-tech.com.hk -----Original Message----- From: Jon Smirl [mailto:jonsmirl@gmail.com] Sent: Monday, May 25, 2009 11:45 AM To: Leung Chi Tat Cc: alsa-devel(a)alsa-project.org Subject: Re: [alsa-devel] Help on writing an ALSA ASoC Driver for WM8985 On Sun, May 24, 2009 at 11:31 PM, Leung Chi Tat <ctleung(a)cct.com.hk> wrote: > Hi all, > > I'm new to ALSA ASoC and ALSA. I'd like get some advice where I can get more information about writing an ASoC driver. I've browsed the internet couples of days and I can only find links on ALSA driver api's and writing an ALSA driver for PCI devices. > > It seems there is not much information on how to writing an ALSA ASoC driver. I'm stuck on the following questions: > 1. describing those important structures, e.g. struct soc_enum, struct snd_kcontrol_new, struct snd_soc_dapm_widget; > 2. what is/are the relationships among those important structures; > 3. what is the differences between those controls for struct snd_kcontrol_new and struct snd_soc_dapm_widget; > 4. Should those sinks, sources, paths be defined in the arrays of struct snd_dapm_widget; > 5. How can I select those defined paths through ALSA user-mode library as I can't find any examples in those ALSA tutorials; > > Indeed, I'm right now trying to write a WM8985 driver for my s3c6400 based platform based on the kernel source v2.6.21 from Samsung. Look in sound/soc/codecs. There are implementations for many similar Wolfson chips. Just cut and paste them together to make the wm8985. > > Thank you so much for your valuable advice and suggestions in advance. > > Best Regards, > LEUNG, Chi Tat > Senior Software Engineer > CCT Tech Advanced Products Limited > 18/F, CCT Telecom Building, 11 Wo Shing Street > Fo Tan, Shatin, N.T., Hong Kong > Tel: +852 26005276 Fax: +852 26948660 > Email: ctleung(a)cct.com.hk > Website: http://www.cct-tech.com.hk > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by linux user 24 May '09

24 May '09

Hi I am running 2.6.28-11-generic (Kubunti install) with M4A78T-E Motherboard and also have no digital spdif output. aplay -L default:CARD=SB HDA ATI SB, VT1708S Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture) hdmi:CARD=HDMI HDA ATI HDMI, ATI HDMI HDMI Audio Output I have read this thread ( http://mailman.alsa-project.org/pipermail/alsa-devel/2009-May/016958.html ) and gather that it is a problem with the BIOS and that it was fixable by patching alsa source. As a newbie and not wanting to destroy my alsa source files can you explain or refer me to a good howto of the steps needed? Thanks http://mailman.alsa-project.org/pipermail/alsa-devel/2009-May/016958.html

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Re: [alsa-devel] Alsa-devel Digest, Vol 27, Issue 124
by Demian Martin 24 May '09

24 May '09

Demian Martin PDS 209 613 6990 -----Original Message----- From: alsa-devel-request(a)alsa-project.org Date: Sun, 24 May 2009 01:13:19 To: <alsa-devel(a)alsa-project.org> Subject: Alsa-devel Digest, Vol 27, Issue 124 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. [PATCH V2 4/9] Add a few more mpc5200 PSC defines (Jon Smirl) 2. [PATCH V2 6/9] Codec for STAC9766 used on the Efika (Jon Smirl) 3. [PATCH V2 7/9] AC97 driver for mpc5200 (Jon Smirl) ---------------------------------------------------------------------- Message: 1 Date: Sat, 23 May 2009 19:13:03 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 4/9] Add a few more mpc5200 PSC defines To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231303.17919.35877.stgit@terra> Content-Type: text/plain; charset="utf-8" Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. This patch is going in via Grant's tree. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..fb84120 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6 /* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001 /* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400 @@ -117,6 +127,7 @@ #define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24) +#define MPC52xx_PSC_SICR_AWR (1 << 30) #define MPC52xx_PSC_SICR_GENCLK (1 << 23) #define MPC52xx_PSC_SICR_I2S (1 << 22) #define MPC52xx_PSC_SICR_CLKPOL (1 << 21) ------------------------------ Message: 2 Date: Sat, 23 May 2009 19:13:07 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 6/9] Codec for STAC9766 used on the Efika To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231307.17919.5277.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 codec for STAC9766 used on the Efika. Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65..cb07d9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa..46c007c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..7740cd5 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-of-simple.h> + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif ------------------------------ Message: 3 Date: Sat, 23 May 2009 19:13:09 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 7/9] AC97 driver for mpc5200 To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231309.17919.83073.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 driver for mpc5200 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 1 sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + 4 files changed, 421 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1918c78..3bce952 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..fa1bb9a --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,394 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int timeout; + unsigned int val; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 bus (rdy)\n"); + return 0xffff; + } + + /* Do the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + timeout = 1000; + while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + return 0xffff; + } + + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if ( ((val>>24) & 0x7f) != reg ) { + pr_err("reg echo error on ac97 read\n"); + return 0xffff; + } + val = (val >> 8) & 0xffff; + + spin_unlock(&psc_dma->lock); + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + int timeout; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 write\n"); + return; + } + + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + spin_unlock(&psc_dma->lock); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(&regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* PSC recover from cold reset (cfr user manual, not sure if useful) */ + out_be32(&regs->sicr, in_be32(&regs->sicr)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + udelay(3); + out_be32(&regs->sicr, psc_dma->sicr); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +#ifdef CONFIG_PM +static int psc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int psc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define psc_ac97_suspend NULL +#define psc_ac97_resume NULL +#endif + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + params_channels(params), params_rate(params), params_format(params)); + + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (params_channels(params) == 1) + psc_dma->slots |= 0x00000100; + else + psc_dma->slots |= 0x00000300; + } else { + if (params_channels(params) == 1) + psc_dma->slots |= 0x01000000; + else + psc_dma->slots |= 0x03000000; + } + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + spin_lock(&psc_dma->lock); + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + break; + } + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .suspend = psc_ac97_suspend, + .resume = psc_ac97_resume, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +}}; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i, id1, id2, timeout, max_reset; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(&regs->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(&regs->ac97_slots, 0x00000000); + + /* AC97 clock is generated by the codec. + * Ensure that it starts ticking after codec reset. + */ + max_reset = 0; +reset: + if (max_reset++ > 5) { + dev_err(&op->dev, "AC97 codec failed to reset\n"); + mpc5200_audio_dma_destroy(op); + return -ENODEV; + } + + psc_ac97_cold_reset(&ac97); + psc_ac97_warm_reset(&ac97); + + /* first make sure it is low */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) { + udelay(1); + if (timeout++ > 1000) + goto reset; + } + /* then wait for the transition to high */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) { + udelay(1); + if (timeout++ > 1000) + psc_ac97_warm_reset(&ac97); + } + + /* Go */ + out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + + id1 = psc_ac97_read(&ac97, AC97_VENDOR_ID1); + id2 = psc_ac97_read(&ac97, AC97_VENDOR_ID2); + + dev_info(&op->dev, "Codec ID is %04x %04x\n", id1, id2); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 27, Issue 124 *******************************************

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 27, Issue 124
by Demian Martin 24 May '09

24 May '09

Demian Martin PDS 209 613 6990 -----Original Message----- From: alsa-devel-request(a)alsa-project.org Date: Sun, 24 May 2009 01:13:19 To: <alsa-devel(a)alsa-project.org> Subject: Alsa-devel Digest, Vol 27, Issue 124 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. [PATCH V2 4/9] Add a few more mpc5200 PSC defines (Jon Smirl) 2. [PATCH V2 6/9] Codec for STAC9766 used on the Efika (Jon Smirl) 3. [PATCH V2 7/9] AC97 driver for mpc5200 (Jon Smirl) ---------------------------------------------------------------------- Message: 1 Date: Sat, 23 May 2009 19:13:03 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 4/9] Add a few more mpc5200 PSC defines To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231303.17919.35877.stgit@terra> Content-Type: text/plain; charset="utf-8" Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. This patch is going in via Grant's tree. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..fb84120 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6 /* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001 /* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400 @@ -117,6 +127,7 @@ #define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24) +#define MPC52xx_PSC_SICR_AWR (1 << 30) #define MPC52xx_PSC_SICR_GENCLK (1 << 23) #define MPC52xx_PSC_SICR_I2S (1 << 22) #define MPC52xx_PSC_SICR_CLKPOL (1 << 21) ------------------------------ Message: 2 Date: Sat, 23 May 2009 19:13:07 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 6/9] Codec for STAC9766 used on the Efika To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231307.17919.5277.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 codec for STAC9766 used on the Efika. Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65..cb07d9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa..46c007c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..7740cd5 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-of-simple.h> + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif ------------------------------ Message: 3 Date: Sat, 23 May 2009 19:13:09 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 7/9] AC97 driver for mpc5200 To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231309.17919.83073.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 driver for mpc5200 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 1 sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + 4 files changed, 421 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1918c78..3bce952 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..fa1bb9a --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,394 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int timeout; + unsigned int val; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 bus (rdy)\n"); + return 0xffff; + } + + /* Do the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + timeout = 1000; + while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + return 0xffff; + } + + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if ( ((val>>24) & 0x7f) != reg ) { + pr_err("reg echo error on ac97 read\n"); + return 0xffff; + } + val = (val >> 8) & 0xffff; + + spin_unlock(&psc_dma->lock); + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + int timeout; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 write\n"); + return; + } + + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + spin_unlock(&psc_dma->lock); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(&regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* PSC recover from cold reset (cfr user manual, not sure if useful) */ + out_be32(&regs->sicr, in_be32(&regs->sicr)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + udelay(3); + out_be32(&regs->sicr, psc_dma->sicr); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +#ifdef CONFIG_PM +static int psc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int psc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define psc_ac97_suspend NULL +#define psc_ac97_resume NULL +#endif + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + params_channels(params), params_rate(params), params_format(params)); + + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (params_channels(params) == 1) + psc_dma->slots |= 0x00000100; + else + psc_dma->slots |= 0x00000300; + } else { + if (params_channels(params) == 1) + psc_dma->slots |= 0x01000000; + else + psc_dma->slots |= 0x03000000; + } + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + spin_lock(&psc_dma->lock); + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + break; + } + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .suspend = psc_ac97_suspend, + .resume = psc_ac97_resume, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +}}; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i, id1, id2, timeout, max_reset; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(&regs->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(&regs->ac97_slots, 0x00000000); + + /* AC97 clock is generated by the codec. + * Ensure that it starts ticking after codec reset. + */ + max_reset = 0; +reset: + if (max_reset++ > 5) { + dev_err(&op->dev, "AC97 codec failed to reset\n"); + mpc5200_audio_dma_destroy(op); + return -ENODEV; + } + + psc_ac97_cold_reset(&ac97); + psc_ac97_warm_reset(&ac97); + + /* first make sure it is low */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) { + udelay(1); + if (timeout++ > 1000) + goto reset; + } + /* then wait for the transition to high */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) { + udelay(1); + if (timeout++ > 1000) + psc_ac97_warm_reset(&ac97); + } + + /* Go */ + out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + + id1 = psc_ac97_read(&ac97, AC97_VENDOR_ID1); + id2 = psc_ac97_read(&ac97, AC97_VENDOR_ID2); + + dev_info(&op->dev, "Codec ID is %04x %04x\n", id1, id2); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 27, Issue 124 *******************************************

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 27, Issue 124
by Demian Martin 24 May '09

24 May '09

Demian Martin PDS 209 613 6990 -----Original Message----- From: alsa-devel-request(a)alsa-project.org Date: Sun, 24 May 2009 01:13:19 To: <alsa-devel(a)alsa-project.org> Subject: Alsa-devel Digest, Vol 27, Issue 124 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. [PATCH V2 4/9] Add a few more mpc5200 PSC defines (Jon Smirl) 2. [PATCH V2 6/9] Codec for STAC9766 used on the Efika (Jon Smirl) 3. [PATCH V2 7/9] AC97 driver for mpc5200 (Jon Smirl) ---------------------------------------------------------------------- Message: 1 Date: Sat, 23 May 2009 19:13:03 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 4/9] Add a few more mpc5200 PSC defines To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231303.17919.35877.stgit@terra> Content-Type: text/plain; charset="utf-8" Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. This patch is going in via Grant's tree. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..fb84120 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6 /* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001 /* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400 @@ -117,6 +127,7 @@ #define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24) +#define MPC52xx_PSC_SICR_AWR (1 << 30) #define MPC52xx_PSC_SICR_GENCLK (1 << 23) #define MPC52xx_PSC_SICR_I2S (1 << 22) #define MPC52xx_PSC_SICR_CLKPOL (1 << 21) ------------------------------ Message: 2 Date: Sat, 23 May 2009 19:13:07 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 6/9] Codec for STAC9766 used on the Efika To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231307.17919.5277.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 codec for STAC9766 used on the Efika. Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65..cb07d9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa..46c007c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..7740cd5 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-of-simple.h> + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif ------------------------------ Message: 3 Date: Sat, 23 May 2009 19:13:09 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 7/9] AC97 driver for mpc5200 To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231309.17919.83073.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 driver for mpc5200 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 1 sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + 4 files changed, 421 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1918c78..3bce952 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..fa1bb9a --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,394 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int timeout; + unsigned int val; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 bus (rdy)\n"); + return 0xffff; + } + + /* Do the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + timeout = 1000; + while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + return 0xffff; + } + + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if ( ((val>>24) & 0x7f) != reg ) { + pr_err("reg echo error on ac97 read\n"); + return 0xffff; + } + val = (val >> 8) & 0xffff; + + spin_unlock(&psc_dma->lock); + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + int timeout; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 write\n"); + return; + } + + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + spin_unlock(&psc_dma->lock); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(&regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* PSC recover from cold reset (cfr user manual, not sure if useful) */ + out_be32(&regs->sicr, in_be32(&regs->sicr)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + udelay(3); + out_be32(&regs->sicr, psc_dma->sicr); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +#ifdef CONFIG_PM +static int psc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int psc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define psc_ac97_suspend NULL +#define psc_ac97_resume NULL +#endif + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + params_channels(params), params_rate(params), params_format(params)); + + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (params_channels(params) == 1) + psc_dma->slots |= 0x00000100; + else + psc_dma->slots |= 0x00000300; + } else { + if (params_channels(params) == 1) + psc_dma->slots |= 0x01000000; + else + psc_dma->slots |= 0x03000000; + } + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + spin_lock(&psc_dma->lock); + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + break; + } + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .suspend = psc_ac97_suspend, + .resume = psc_ac97_resume, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +}}; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i, id1, id2, timeout, max_reset; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(&regs->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(&regs->ac97_slots, 0x00000000); + + /* AC97 clock is generated by the codec. + * Ensure that it starts ticking after codec reset. + */ + max_reset = 0; +reset: + if (max_reset++ > 5) { + dev_err(&op->dev, "AC97 codec failed to reset\n"); + mpc5200_audio_dma_destroy(op); + return -ENODEV; + } + + psc_ac97_cold_reset(&ac97); + psc_ac97_warm_reset(&ac97); + + /* first make sure it is low */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) { + udelay(1); + if (timeout++ > 1000) + goto reset; + } + /* then wait for the transition to high */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) { + udelay(1); + if (timeout++ > 1000) + psc_ac97_warm_reset(&ac97); + } + + /* Go */ + out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + + id1 = psc_ac97_read(&ac97, AC97_VENDOR_ID1); + id2 = psc_ac97_read(&ac97, AC97_VENDOR_ID2); + + dev_info(&op->dev, "Codec ID is %04x %04x\n", id1, id2); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 27, Issue 124 *******************************************

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 27, Issue 124
by Demian Martin 24 May '09

24 May '09

Demian Martin PDS 209 613 6990 -----Original Message----- From: alsa-devel-request(a)alsa-project.org Date: Sun, 24 May 2009 01:13:19 To: <alsa-devel(a)alsa-project.org> Subject: Alsa-devel Digest, Vol 27, Issue 124 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. [PATCH V2 4/9] Add a few more mpc5200 PSC defines (Jon Smirl) 2. [PATCH V2 6/9] Codec for STAC9766 used on the Efika (Jon Smirl) 3. [PATCH V2 7/9] AC97 driver for mpc5200 (Jon Smirl) ---------------------------------------------------------------------- Message: 1 Date: Sat, 23 May 2009 19:13:03 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 4/9] Add a few more mpc5200 PSC defines To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231303.17919.35877.stgit@terra> Content-Type: text/plain; charset="utf-8" Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. This patch is going in via Grant's tree. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..fb84120 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6 /* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001 /* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400 @@ -117,6 +127,7 @@ #define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24) +#define MPC52xx_PSC_SICR_AWR (1 << 30) #define MPC52xx_PSC_SICR_GENCLK (1 << 23) #define MPC52xx_PSC_SICR_I2S (1 << 22) #define MPC52xx_PSC_SICR_CLKPOL (1 << 21) ------------------------------ Message: 2 Date: Sat, 23 May 2009 19:13:07 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 6/9] Codec for STAC9766 used on the Efika To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231307.17919.5277.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 codec for STAC9766 used on the Efika. Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65..cb07d9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa..46c007c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..7740cd5 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-of-simple.h> + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif ------------------------------ Message: 3 Date: Sat, 23 May 2009 19:13:09 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 7/9] AC97 driver for mpc5200 To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231309.17919.83073.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 driver for mpc5200 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 1 sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + 4 files changed, 421 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1918c78..3bce952 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..fa1bb9a --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,394 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int timeout; + unsigned int val; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 bus (rdy)\n"); + return 0xffff; + } + + /* Do the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + timeout = 1000; + while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + return 0xffff; + } + + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if ( ((val>>24) & 0x7f) != reg ) { + pr_err("reg echo error on ac97 read\n"); + return 0xffff; + } + val = (val >> 8) & 0xffff; + + spin_unlock(&psc_dma->lock); + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + int timeout; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 write\n"); + return; + } + + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + spin_unlock(&psc_dma->lock); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(&regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* PSC recover from cold reset (cfr user manual, not sure if useful) */ + out_be32(&regs->sicr, in_be32(&regs->sicr)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + udelay(3); + out_be32(&regs->sicr, psc_dma->sicr); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +#ifdef CONFIG_PM +static int psc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int psc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define psc_ac97_suspend NULL +#define psc_ac97_resume NULL +#endif + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + params_channels(params), params_rate(params), params_format(params)); + + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (params_channels(params) == 1) + psc_dma->slots |= 0x00000100; + else + psc_dma->slots |= 0x00000300; + } else { + if (params_channels(params) == 1) + psc_dma->slots |= 0x01000000; + else + psc_dma->slots |= 0x03000000; + } + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + spin_lock(&psc_dma->lock); + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + break; + } + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .suspend = psc_ac97_suspend, + .resume = psc_ac97_resume, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +}}; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i, id1, id2, timeout, max_reset; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(&regs->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(&regs->ac97_slots, 0x00000000); + + /* AC97 clock is generated by the codec. + * Ensure that it starts ticking after codec reset. + */ + max_reset = 0; +reset: + if (max_reset++ > 5) { + dev_err(&op->dev, "AC97 codec failed to reset\n"); + mpc5200_audio_dma_destroy(op); + return -ENODEV; + } + + psc_ac97_cold_reset(&ac97); + psc_ac97_warm_reset(&ac97); + + /* first make sure it is low */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) { + udelay(1); + if (timeout++ > 1000) + goto reset; + } + /* then wait for the transition to high */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) { + udelay(1); + if (timeout++ > 1000) + psc_ac97_warm_reset(&ac97); + } + + /* Go */ + out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + + id1 = psc_ac97_read(&ac97, AC97_VENDOR_ID1); + id2 = psc_ac97_read(&ac97, AC97_VENDOR_ID2); + + dev_info(&op->dev, "Codec ID is %04x %04x\n", id1, id2); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 27, Issue 124 *******************************************

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Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G
by Karthik Ramgopal 23 May '09

23 May '09

Hi, Even after enabling DEBUG using --with-debug option with configure when compiling the alsa-driver snapshot leaves no more error messages in the dmesg logs. The only message relating to HDA is: [ 41.104817] hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. I just got the line-out output to work by toggling the surround volume. I still think it's a mapping problem. The headphone switch is turning the subwoofer on and off, and the surround volume is controlling the line-out volume. None of the line-in devices function and all of the amp-ins 0x0b to 0x0f are muted. Cannot find any verb to unmute them. hda-verb with SET_EAPD_BTLENABLE 2 at 0x15 is still needed at startup to fire up the amps. Please help. Regards, Karthik

3 21

[alsa-devel] Testers wanted: New SB X-Fi driver
by Ge-org Brohammer 23 May '09

23 May '09

Good day I compiled and installed alsa-lib-1.0.20 and alsa-utils-1.0.20 and then your driver on ubuntu Jaunty 64 bit. I have the X-FI Music PCI card and it seems to split the outputs into separate cards? This is the output from aplay -l **** List of PLAYBACK Hardware Devices **** card 0: XFi [Creative X-Fi], device 0: Front/WaveIn [Front/WaveIn] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 1: Rear [Rear] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 2: Center/LFE [Center/LFE] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 3: Surround [Surround] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 4: IEC958 Non-audio [IEC958 Non-audio] Subdevices: 1/1 Subdevice #0: subdevice #0 This a problem for me because I cannot play all the channels at once either front side rear, etc... Let me know if any extra info is required. Regards

1 0

[alsa-devel] force byte-swap at soc level before codec?
by Brad Parker 23 May '09

23 May '09

Hi I'm an ALSA SOC newbie. I created a new SOC driver for a new ARM chip but I'm having endian issues at the codec. I started with the pxa soc code since I am failure with the pxa* i2s hardware. The new code is working and data flows fine but due to hardware constraints the bytes arrive at the codec swapped. The codec claims S16_LE. The i2s layer claims S16_LE as does the pcm layer inside soc. The source data in user land is S16_LE. But when the i2s hardware (which I can't change) does dma the bytes go to the codec in reverse order. So, I need a byte swap somewhere. I verified this by sending constant data from user land via the oss compat layer (i.e. /dev/dsp). If I swap them in userland they come out on the scope at the codec in the correct order. Any know the simplest way to do this? I've tried a bunch of ways but nothing seems to work correctly. It seems like ALSA already handles this as long as I tell the right thing from below. -brad

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[alsa-devel] [PATCH 1/1] hdsp: right firmware path
by Raphaël Doursenaud 23 May '09

23 May '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 From: Raphaël Doursenaud <rdoursenaud(a)free.fr> Correct firmware path to reflect actual location. This was causing hotplug to fail. Inspired by the emu10k1 module. Signed-off-by: Raphael Doursenaud <rdoursenaud(a)free.fr> - --- sound/pci/rme9652/hdsp.c | 22 ++++++++++++++-------- 1 files changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 314e735..48f5434 100644 - --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -59,11 +59,17 @@ MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP}," "{RME HDSP-9652}," "{RME HDSP-9632}}"); + +#define MULTIFACE_FILENAME "hdsploader/multiface_firmware.bin" +#define MULTIFACE_REV11_FILENAME "hdsploader/multiface_firmware_rev11.bin" +#define DIGIFACE_FILENAME "hdsploader/digiface_firmware.bin" +#define DIGIFACE_REV11_FILENAME "hdsploader/digiface_firmware_rev11.bin" + #ifdef HDSP_FW_LOADER - -MODULE_FIRMWARE("multiface_firmware.bin"); - -MODULE_FIRMWARE("multiface_firmware_rev11.bin"); - -MODULE_FIRMWARE("digiface_firmware.bin"); - -MODULE_FIRMWARE("digiface_firmware_rev11.bin"); +MODULE_FIRMWARE(MULTIFACE_FILENAME); +MODULE_FIRMWARE(MULTIFACE_REV11_FILENAME); +MODULE_FIRMWARE(DIGIFACE_FILENAME); +MODULE_FIRMWARE(DIGIFACE_REV11_FILENAME); #endif #define HDSP_MAX_CHANNELS 26 @@ -4928,15 +4934,15 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) switch (hdsp->io_type) { case Multiface: if (hdsp->firmware_rev == 0xa) - - fwfile = "multiface_firmware.bin"; + fwfile = MULTIFACE_FILENAME; else - - fwfile = "multiface_firmware_rev11.bin"; + fwfile = MULTIFACE_REV11_FILENAME; break; case Digiface: if (hdsp->firmware_rev == 0xa) - - fwfile = "digiface_firmware.bin"; + fwfile = DIGIFACE_FILENAME; else - - fwfile = "digiface_firmware_rev11.bin"; + fwfile = DIGIFACE_REV11_FILENAME; break; default: snd_printk(KERN_ERR "Hammerfall-DSP: invalid io_type %d\n", hdsp->io_type); - -- Raphaël Doursenaud -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkoWurIACgkQaZKmNAdXaVVYYACdEyheX1+7sUbWURScDRFV66XV s4gAniZHnEfwRG012ZQmaDAhDhNXQf9Z =Ex8j -----END PGP SIGNATURE-----

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[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 22 May '09

22 May '09

The following changes since commit e24805dd85283ac0912b9c400768a4d171b400ff: Atsushi Nemoto (1): ASoC: Add TXx9 AC link controller driver (v3) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Mark Brown (2): ASoC: Make snd_soc_read() and snd_soc_write() functions ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver Peter Ujfalusi (5): ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event ASoC: TWL4030: Add support for platform dependent configuration ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal ASoC: TWL4030: Differentiate the playback streams include/sound/soc.h | 17 +- include/sound/wm9081.h | 25 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 285 ++++++--- sound/soc/codecs/twl4030.h | 5 + sound/soc/codecs/wm9081.c | 1532 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9081.h | 787 +++++++++++++++++++++++ sound/soc/omap/sdp3430.c | 7 + 9 files changed, 2559 insertions(+), 105 deletions(-) create mode 100644 include/sound/wm9081.h create mode 100644 sound/soc/codecs/wm9081.c create mode 100644 sound/soc/codecs/wm9081.h

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[alsa-devel] bootup message "Control name ... truncated to ..." - what to fix?
by Andreas Mohr 22 May '09

22 May '09

Hi all, On my i8k, I'm getting this on bootup (2.6.30-rc5): ALSA sound/core/control.c:232: Control name 'Sigmatel Surround Phase Inversion Playback Switch' truncated to 'Sigmatel Surround Phase Inversion Playback ' I would quickly have submitted a patch for this (and similar items), problem is: - what to fix: control name or struct definition? - "Sigmatel Surround" might be a relatively important prefix, thus removing "Sigmatel " from relevant names possibly not a good idea andi@note:/usr/src/linux-2.6.30-rc5/sound$ find . -name "*.c" -o -name "*.h"|xargs grep "Sigmatel Surround" ./pci/ac97/ac97_patch.c:AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0); ./pci/ac97/ac97_patch.c: snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Sigmatel Surround Playback"); ./pci/ca0106/ca0106_mixer.c: "Sigmatel Surround Phase Inversion Playback ", ./pci/emu10k1/emu10k1x.c: * the PCM Playback Volume, Sigmatel Surround Playback Volume and sound/core/control.c/snd_ctl_new1(): if (ncontrol->name) { strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); if (strcmp(ncontrol->name, kctl.id.name) != 0) snd_printk(KERN_WARNING "Control name '%s' truncated to '%s'\n", ncontrol->name, kctl.id.name); } include/sound/asound.h: struct snd_ctl_elem_id { unsigned int numid; /* numeric identifier, zero = invalid */ snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ unsigned char name[44]; /* ASCII name of item */ unsigned int index; /* index of item */ }; I suppose (well, DAMN sure) that name is only 44 bytes due to exactly 64-byte cacheline-compatible struct alignment status. Note that status quo is pretty bad, since in another file already: ./pci/ca0106/ca0106_mixer.c: "Sigmatel Surround Phase Inversion Playback ", . IOW we already violate official mixer control name specs (by omitting "Switch" - see ControlNames.txt) simply in order to obey the 44-byte limit. Given that snd_ctl_elem_id 64 byte is a rather hard and arguably important limit, I'd suggest changing the names, but how? Unfortunately Sigmatel Surround PhaseInvert Playback Switch doesn't fit the bill either, since it's one char too long. Thus it will probably be Surround Phase Inversion Playback Switch (and thus changing "Sigmatel Surround" at all affected controls) Thanks, Andreas Mohr

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[alsa-devel] more on appl_ptr and DMA overrun at end of stream
by Jon Smirl 22 May '09

22 May '09

I've been reading Lennart's article on glitch free audio.... On my scatter/gather DMA hardware I generate an interrupt at the end of each period. On the last period this causes ALSA to issue a trigger(STOP). However, there is a latency from when that interrupt is generated and when ALSA does the trigger(STOP). During that latency my hardware starts playing the next buffer (which contains stale data). Playing stale data causes a noise burst. I fixed this by programming my DMA hardware to not overrun appl_ptr. There's another solution. ALSA already knows the stream ends mid-period and pads out the partial period with silence. If it simply padded out the next period with silence too the overrun problem would become non-audible and there would time to process trigger(STOP). If padding out two periods at end of stream is possible, it then becomes possible for me to turn off interrupts in my audio driver. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] [PATCH 0/1] ASoC: TWL4030: change the streams names for playbacks
by Peter Ujfalusi 22 May '09

22 May '09

Hello, rename the HiFi and Voice playback streams to have unique names to aid DAPM to correctly pick the active playback streams on the twl4030 codec. --- Peter Ujfalusi (1): ASoC: TWL4030: Differentiate the playback streams sound/soc/codecs/twl4030.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-)

2 2

[alsa-devel] [PATCH 0/2 take2] SoC: TWL4030 and SDP3430 HS ramp configuration
by Peter Ujfalusi 22 May '09

22 May '09

Hello, I have found few typos in the patches, resending... he following series adds support for platform dependent Headset pop removal configuration for the twl4030 codec. Platform drivers can fill the twl4030_setup_data structure and pass it with snd_soc_device->codec_data to the codec driver. As an example - based on the feedback for the ramp_delay value variations - the sdp3430.c board is updated to pass the correct ramp settings. If you know for the other platforms using the twl4030 codec the correct HS ramp configuration, please update those files... Mark: This series needs the DAPM restructuring and HS pop removal related patches for twl4030 codec. --- Peter Ujfalusi (2): ASoC: TWL4030: Add support for platform dependent configuration ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal sound/soc/codecs/twl4030.c | 22 +++++++++++++++++++--- sound/soc/codecs/twl4030.h | 5 +++++ sound/soc/omap/sdp3430.c | 7 +++++++ 3 files changed, 31 insertions(+), 3 deletions(-)

2 6

[alsa-devel] Disabling HFL/R outputs in twl4030
by Lopez Cruz, Misael 22 May '09

22 May '09

It seems that there is no way to disable handsfree L/R outputs for twl4030 codec. That means that: - handsfree amplifiers are turned-on even if we are not interested in routing through handsfree - if the selected input of the handsfree mux is an active DAC (streaming) sound will be produced on hf outputs and the only way to silence them is routing from a different DAC, which doesn't seem like the right solution Currently, handsfree amplifiers are turned-on when the HandsfreeL/R muxes are powered by the framework. I was thinking in adding a dapm switch to select whether or not handsfree amplifiers should be powered, but the switch probably is not the correct approach to handle this situation. Maybe a regular kcontrol could be the answer. -Misa

3 5

[alsa-devel] Releasing a handle without closing ?
by Werner Van Belle 22 May '09

22 May '09

Hello, I have an intruiging problem. I have two processes: a parent and a child. The parent forks off the child (as can be expected, the opposite would be somewhat unsettling), after it allocated and started to write to a pcm device. As can be expected, the child will also have a link to this dsp device. The problem now is that the child wants to release the pcm handle without closing it. Is there any particular function offering such possibility ? Wkr,- -- Dr. Werner Van Belle http://werner.yellowcouch.org/

1 0

[alsa-devel] [patch] ALSA: fix audio on HP TX25xx series notebooks
by Chuck Ebbert 22 May '09

22 May '09

ALSA: fix audio on HP TX25xx series notebooks Fixes https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4121 Taken from https://bugzilla.redhat.com/show_bug.cgi?id=498060 Signed-off-by: Adam Williamson <awilliam(a)redhat.com> --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12058,6 +12058,7 @@ SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),

2 1

[alsa-devel] [PATCH 0/2]ASoC: TWL4030 and SDP3430 HS ramp configuration
by Peter Ujfalusi 22 May '09

22 May '09

Hello, The following series adds support for platform dependent Headset pop removal configuration for the twl4030 codec. Platform drivers can fill the twl4030_setup_data structure and pass it with snd_soc_device->codec_data to the codec driver. As an example - based on the feedback for the ramp_delay value variations - the sdp3430.c board is updated to pass the correct ramp settings. If you know for the other platforms using the twl4030 codec the correct HS ramp configuration, please update those files... Mark: This series needs the DAPM restructuring and HS pop removal related patches for twl4030 codec. --- Peter Ujfalusi (2): ASoC: TWL4030: Add support for platform dependent configuration ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal sound/soc/codecs/twl4030.c | 22 +++++++++++++++++++--- sound/soc/codecs/twl4030.h | 5 +++++ sound/soc/omap/sdp3430.c | 7 +++++++ 3 files changed, 31 insertions(+), 3 deletions(-)

1 2

[alsa-devel] Program blocks with alsa pulse plugin
by Cristian Morales Vega 21 May '09

21 May '09

I'm not sure where exactly the problem is, probably in the alsa pulse plugin? And not sure where alsa pulse plugin should be reported, ALSA or PA? Using http://byuu.org/bsnes/ through the ALSA API with PulseAudio at some point provokes the game to block with 100% CPU usage. Stopping it I get this backtrace: Program received signal SIGTSTP, Stopped (user). 0x00007ffff2b8a386 in poll () from /lib64/libc.so.6 (gdb) bt #0 0x00007ffff2b8a386 in poll () from /lib64/libc.so.6 #1 0x00007ffff72ac94c in snd1_pcm_wait_nocheck () from /usr/lib64/libasound.so.2 #2 0x00007ffff72accb7 in snd1_pcm_write_areas () from /usr/lib64/libasound.so.2 #3 0x00007ffff72ebf3a in snd_pcm_ioplug_writei () from /usr/lib64/libasound.so.2 #4 0x000000000043e1b8 in ruby::AudioALSA::sample (this=<value optimized out>, left=<value optimized out>, right=<value optimized out>) at lib/ruby/audio/alsa.cpp:101 #5 0x000000000043ed8b in ruby::AudioInterface::sample (this=0x74c520, left=<value optimized out>, right=<value optimized out>) at lib/ruby/ruby_audio.cpp:110 #6 0x00000000004a5b82 in SNES::sDSP::enter (this=0x8528c0) at dsp/sdsp/sdsp.cpp:199 #7 0x00000000004a309a in SNES::sSMP::op_read (this=0x851860, addr=4532) at smp/ssmp/timing/timing.cpp:8 #8 0x00000000004a2652 in SNES::sSMP::enter (this=0x851860) at smp/core/memory.hpp:2 #9 0x000000000043bec0 in ?? () at lib/libco/x86-64.c:31 #10 0x0000000000000000 in ?? () There are lots of snd_pcm_writei() (lib/ruby/audio/alsa.cpp:101) calls before bsnes blocks, and the only errors are some underruns that are "fixed" with snd_pcm_recover(). Just that at some point (easier to trigger with 150% or 200% emu speed) blocks. It's using the "safe ALSA" everywhere, so... This is with openSUSE 11.1. But also with openSUSE 11.1 + PA from multimedia:libs (0.9.14) and a recent GIT snapshot of ALSA from multimedia:audio:snapshot and multimedia:audio:KMP. There is a package for Fedora in rpmfusion, and they have a readme that states the problem also happens there. To avoid this bsnes dev created a native PA backend. I inform just in case this is triggering a bug, perhaps affecting other software, you are interested in fix... PS. ...obviously bsnes never blocks when using ALSA directly.

1 0

[alsa-devel] [PATCH] beyond ARRAY_SIZE of s3c64xx_i2s
by Roel Kluin 21 May '09

21 May '09

Do not go beyond ARRAY_SIZE of s3c64xx_i2s Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- In this case you are Maintainer, correct? diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 33c5de7..f2d3021 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -126,7 +126,7 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev, dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id); - if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) { + if (pdev->id < 0 || pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) { dev_err(dev, "id %d out of range\n", pdev->id); return -EINVAL; }

2 2

[alsa-devel] About my issue with the sound on a Dell Dimension 4550.
by Kristoffer Grundström 21 May '09

21 May '09

What is the problem? I really want to know if it's about to be fixed. I'm getting tired of having no sound. https://qa.mandriva.com/show_bug.cgi?id=49048 /Kristoffer

2 1

[alsa-devel] [PATCH 0/2] ASoC: TWL4030: DAPM restructuring and Headset pop-attenuation fix
by Peter Ujfalusi 21 May '09

21 May '09

Hello, The following series aims to fix the Headset power on/off pop noise problems. Since the twl4030 codec internal implementation have this order in HW: DAC -> Analog PGA -> Mixer While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA I needed to move the APGA control from SND_SOC_DAPM_PGA to be handled with _MIXER. Than adding a PGA to the headset outputs results correct power sequences. This patch also separates the analog and digital loopback paths, so when the analog loopback is enabled none of the DACs will be powered on, while in case of digital loopback the corresponding DACs will be powered. These are internal changes, not visible for the user. I have tested it with my custom board with the headset soldered to HeadsetL, the pop noise is really small if any. The user must also set the correct ramp delay for the given board. On my board: MCLK = 38.4MHz RAMP_DELAY = 2 (109/81/55 ms) works best (meaning 55ms ramp delay), if it is shorter, than it clicks, if it is longer, than it does not click, but the beep tone can be observed. I hope that I have not missed anything with the patches, since these are forward ported from my working tree to sound-2.6:topic/asoc. I have applied the patch from Misael on top of the current topic/asoc, so the head was: ASoC: TWL4030: Enable/disable voice digital filters I believe that this should be able to minimize the pop noise on the headset outputs. --- Peter Ujfalusi (2): ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event sound/soc/codecs/twl4030.c | 257 ++++++++++++++++++++++++++++---------------- 1 files changed, 162 insertions(+), 95 deletions(-)

5 18

[alsa-devel] [GIT branch] Fix envy24control on glibc-2.10
by Diego E. “Flameeyes” Pettenò 21 May '09

21 May '09

I've pushed to git://git.flameeyes.eu/others/alsa/alsa-tools.git a single commit to fix building of envy24control on glibc-2.10. HTH, -- Diego Elio Pettenò — “Flameeyes” http://blog.flameeyes.eu/ If you found a .asc file in this mail and know not what it is, it's a GnuPG digital signature: http://www.gnupg.org/

1 1

[alsa-devel] Testers wanted: New SB X-Fi driver
by Noel Bourke 21 May '09

21 May '09

Hi Takashi, I set this up with an Auzentech X-Fi Prelude. Driver works OK but sound seemed a bit off. According to speaker-test I seem to only have one audio channel! Any thoughts? Regards, Noel

3 5

[alsa-devel] Testers wanted: New SB X-Fi driver
by Marcel Mulders 20 May '09

20 May '09

Hi Takashi, I tried the snd-ctxfi driver on Ubuntu Jaunty x64. Seems to work just fine for me. Audacious and Totem have no problems as far as I can see. Great work, your efforts are very much appreciated! I was wondering, is there any chance to get the headphones-output on the frontpanel working? I don't really need all the other jacks to work, just plain old analog stereo.. ;-) In one of the closed-source drivers from Creative from a year ago or so, version 1.18, that worked fine. In the opensource driver from last February, the headphones were broken again, just like with the latest snapshot alsa drivers. Can you comment on this maybe? Thanks in advance. Marcel

1 0

Re: [alsa-devel] Acer Aspire One codec: Realtek ALC272 not found in ALSA-Configuration.txt Mic does not capture in alsamixer
by Takashi Iwai 20 May '09

20 May '09

[Please don't drop Cc from alsa-devel ML so that other people can follow.] At Wed, 20 May 2009 12:51:17 -0400, wirechief wrote: > > Takashi > I got brave. i did compile this: > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… > I used this: > sudo apt-get install build-essential linux-headers-$(uname -r) > ./configure --with-cards=all > make > sudo make install Note that you may have already the driver modules in the update directory in /lib/modules/*, depending on distro. As default, alsa-driver external build tries to replace the existing /lib/module/*/kernel/ module files, but the update directory always wins. Takashi > then i rebooted and re-ran the alsa script : > http://www.alsa-project.org/db/?f=249eea933a054742dc27c86822edecc0c731e1b9 > > Sorry if but i am new to this and wonder if i should of compiled more > than just this > "snapshot" the report does show the new driver however, i am including a .png of > alsamixer if that helps. > > > > On Wed, May 20, 2009 at 10:34 AM, wirechief <silvermachineman(a)gmail.com> wrote: > > thank you very much, I am trying to contact dtchen perhaps he can advise > > perhaps there is a backport for jaunty 9.04 remix that i can get a .deb > > > > On Wed, May 20, 2009 at 10:26 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> At Wed, 20 May 2009 10:23:30 -0400, > >> wirechief wrote: > >>> > >>> My skills at doing compiling is limited and not sure this netbook can > >>> accomplish > >>> this. I will look for 1.0.20 in a deb as my first option, perhaps that > >>> will work.. > >> > >> There are lots of changes since 1.0.20, so you should try anyway the > >> later version. > >> > >> Cc'ed ML, so that anyone who knows well Debian could help... > >> > >> > >> thanks, > >> > >> Takashi > >> > > > > > > > > -- > > Reach out and share life, care for others, > > > > > > -- > Reach out and share life, care for others, > [2 alsamixer.png <image/png (base64)>] >

1 0

Re: [alsa-devel] Acer Aspire One codec: Realtek ALC272 not found in ALSA-Configuration.txt Mic does not capture in alsamixer
by Takashi Iwai 20 May '09

20 May '09

At Wed, 20 May 2009 10:23:30 -0400, wirechief wrote: > > My skills at doing compiling is limited and not sure this netbook can > accomplish > this. I will look for 1.0.20 in a deb as my first option, perhaps that > will work.. There are lots of changes since 1.0.20, so you should try anyway the later version. Cc'ed ML, so that anyone who knows well Debian could help... thanks, Takashi

1 0

[alsa-devel] Acer Aspire One codec: Realtek ALC272 not found in ALSA-Configuration.txt Mic does not capture in alsamixer
by wirechief 20 May '09

20 May '09

Host/Kernel/OS "wirechief-notebook" running Linux 2.6.30-020630rc5-generic i686 [ Jaunty 9.04 remix ] CPU Info (1) Intel Atom N270 @ 512 KB cache flags( sse3 ) clocked at [ 1333.000 MHz ] (2) Intel Atom N270 @ 512 KB cache flags( sse3 ) clocked at [ 800.000 MHz ] Videocard Intel Mobile 945GME Express Integrated Graphics Controller X.Org 1.6.0 [ 1024x600(a)60.0hz ] Network cards Attansic Corp. L1e Gigabit Ethernet Adapter, at port: 1000 Atheros AR242x 802.11abg Wireless PCI Express Adapter Processes 154 | Uptime 2:44 | Memory 357.3/993.4MB | HDD Verbatim STORE N GO,ATA TOSHIBA MK1652GS,Generic- Multi-Card Size 166GB (1%used) | Client Shell | Infobash v3.11 my alsa info report: http://www.alsa-project.org/db/?f=83ac843507b5450f5c4d3252a0082ef6f9c6f124 I am not able to see a sound indication with : arecord -f cd -vv /dev/null -- Reach out and share life, care for others,

2 2

[alsa-devel] [PATCH] TXx9: Add ACLC support (v2)
by Atsushi Nemoto 20 May '09

20 May '09

Add platform support for ACLC of TXx9 SoCs. Signed-off-by: Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> --- Changes since v1: * use one platform device to provide irq/mem/dma resources and another platform device for machine driver. arch/mips/include/asm/txx9/generic.h | 5 ++++ arch/mips/include/asm/txx9/tx4927.h | 2 + arch/mips/include/asm/txx9/tx4938.h | 1 + arch/mips/include/asm/txx9/tx4939.h | 1 + arch/mips/txx9/Kconfig | 3 ++ arch/mips/txx9/generic/setup.c | 36 +++++++++++++++++++++++++++ arch/mips/txx9/generic/setup_tx4927.c | 43 +++++++++++++++++++++++++++++++++ arch/mips/txx9/generic/setup_tx4938.c | 11 ++++++++ arch/mips/txx9/generic/setup_tx4939.c | 9 +++++++ arch/mips/txx9/rbtx4927/setup.c | 8 ++++- arch/mips/txx9/rbtx4938/setup.c | 2 + arch/mips/txx9/rbtx4939/setup.c | 2 + 12 files changed, 121 insertions(+), 2 deletions(-) diff --git a/arch/mips/include/asm/txx9/generic.h b/arch/mips/include/asm/txx9/generic.h index 9cde009..8169477 100644 --- a/arch/mips/include/asm/txx9/generic.h +++ b/arch/mips/include/asm/txx9/generic.h @@ -91,4 +91,9 @@ void txx9_7segled_init(unsigned int num, void (*putc)(unsigned int pos, unsigned char val)); int txx9_7segled_putc(unsigned int pos, char c); +void __init txx9_aclc_init(unsigned long baseaddr, int irq, + unsigned int dmac_id, + unsigned int dma_chan_out, + unsigned int dma_chan_in); + #endif /* __ASM_TXX9_GENERIC_H */ diff --git a/arch/mips/include/asm/txx9/tx4927.h b/arch/mips/include/asm/txx9/tx4927.h index d92ae07..18c98c5 100644 --- a/arch/mips/include/asm/txx9/tx4927.h +++ b/arch/mips/include/asm/txx9/tx4927.h @@ -50,6 +50,7 @@ #define TX4927_NR_SIO 2 #define TX4927_SIO_REG(ch) (TX4927_REG_BASE + 0xf300 + (ch) * 0x100) #define TX4927_PIO_REG (TX4927_REG_BASE + 0xf500) +#define TX4927_ACLC_REG (TX4927_REG_BASE + 0xf700) #define TX4927_IR_ECCERR 0 #define TX4927_IR_WTOERR 1 @@ -267,5 +268,6 @@ void tx4927_setup_pcierr_irq(void); void tx4927_irq_init(void); void tx4927_mtd_init(int ch); void tx4927_dmac_init(int memcpy_chan); +void tx4927_aclc_init(unsigned int dma_chan_out, unsigned int dma_chan_in); #endif /* __ASM_TXX9_TX4927_H */ diff --git a/arch/mips/include/asm/txx9/tx4938.h b/arch/mips/include/asm/txx9/tx4938.h index 0758a0c..54e4674 100644 --- a/arch/mips/include/asm/txx9/tx4938.h +++ b/arch/mips/include/asm/txx9/tx4938.h @@ -306,5 +306,6 @@ struct tx4938ide_platform_info { void tx4938_ata_init(unsigned int irq, unsigned int shift, int tune); void tx4938_dmac_init(int memcpy_chan0, int memcpy_chan1); +void tx4938_aclc_init(void); #endif diff --git a/arch/mips/include/asm/txx9/tx4939.h b/arch/mips/include/asm/txx9/tx4939.h index 1be9798..f13b708 100644 --- a/arch/mips/include/asm/txx9/tx4939.h +++ b/arch/mips/include/asm/txx9/tx4939.h @@ -545,5 +545,6 @@ void tx4939_rtc_init(void); void tx4939_ndfmc_init(unsigned int hold, unsigned int spw, unsigned char ch_mask, unsigned char wide_mask); void tx4939_dmac_init(int memcpy_chan0, int memcpy_chan1); +void tx4939_aclc_init(void); #endif /* __ASM_TXX9_TX4939_H */ diff --git a/arch/mips/txx9/Kconfig b/arch/mips/txx9/Kconfig index 0db7cf3..852ae4b 100644 --- a/arch/mips/txx9/Kconfig +++ b/arch/mips/txx9/Kconfig @@ -69,6 +69,7 @@ config SOC_TX4927 select IRQ_TXX9 select PCI_TX4927 select GPIO_TXX9 + select HAS_TXX9_ACLC config SOC_TX4938 bool @@ -78,6 +79,7 @@ config SOC_TX4938 select IRQ_TXX9 select PCI_TX4927 select GPIO_TXX9 + select HAS_TXX9_ACLC config SOC_TX4939 bool @@ -85,6 +87,7 @@ config SOC_TX4939 select HAS_TXX9_SERIAL select HW_HAS_PCI select PCI_TX4927 + select HAS_TXX9_ACLC config TXX9_7SEGLED bool diff --git a/arch/mips/txx9/generic/setup.c b/arch/mips/txx9/generic/setup.c index 369d863..7f91012 100644 --- a/arch/mips/txx9/generic/setup.c +++ b/arch/mips/txx9/generic/setup.c @@ -876,3 +876,39 @@ void __init txx9_dmac_init(int id, unsigned long baseaddr, int irq, } #endif } + +void __init txx9_aclc_init(unsigned long baseaddr, int irq, + unsigned int dmac_id, + unsigned int dma_chan_out, + unsigned int dma_chan_in) +{ +#if defined(CONFIG_SND_SOC_TXX9ACLC) || \ + defined(CONFIG_SND_SOC_TXX9ACLC_MODULE) + unsigned int dma_base = dmac_id * TXX9_DMA_MAX_NR_CHANNELS; + struct resource res[] = { + { + .start = baseaddr, + .end = baseaddr + 0x100 - 1, + .flags = IORESOURCE_MEM, + }, { + .start = irq, + .flags = IORESOURCE_IRQ, + }, { + .name = "txx9dmac-chan", + .start = dma_base + dma_chan_out, + .flags = IORESOURCE_DMA, + }, { + .name = "txx9dmac-chan", + .start = dma_base + dma_chan_in, + .flags = IORESOURCE_DMA, + } + }; + struct platform_device *pdev = + platform_device_alloc("txx9aclc-ac97", -1); + + if (!pdev || + platform_device_add_resources(pdev, res, ARRAY_SIZE(res)) || + platform_device_add(pdev)) + platform_device_put(pdev); +#endif +} diff --git a/arch/mips/txx9/generic/setup_tx4927.c b/arch/mips/txx9/generic/setup_tx4927.c index 6b681cd..3418b2a 100644 --- a/arch/mips/txx9/generic/setup_tx4927.c +++ b/arch/mips/txx9/generic/setup_tx4927.c @@ -265,6 +265,49 @@ void __init tx4927_dmac_init(int memcpy_chan) TXX9_IRQ_BASE + TX4927_IR_DMA(0), &plat_data); } +void __init tx4927_aclc_init(unsigned int dma_chan_out, + unsigned int dma_chan_in) +{ + u64 pcfg = __raw_readq(&tx4927_ccfgptr->pcfg); + __u64 dmasel_mask = 0, dmasel = 0; + unsigned long flags; + + if (!(pcfg & TX4927_PCFG_SEL2)) + return; + /* setup DMASEL (playback:ACLC ch0, capture:ACLC ch1) */ + switch (dma_chan_out) { + case 0: + dmasel_mask |= TX4927_PCFG_DMASEL0_MASK; + dmasel |= TX4927_PCFG_DMASEL0_ACL0; + break; + case 2: + dmasel_mask |= TX4927_PCFG_DMASEL2_MASK; + dmasel |= TX4927_PCFG_DMASEL2_ACL0; + break; + default: + return; + } + switch (dma_chan_in) { + case 1: + dmasel_mask |= TX4927_PCFG_DMASEL1_MASK; + dmasel |= TX4927_PCFG_DMASEL1_ACL1; + break; + case 3: + dmasel_mask |= TX4927_PCFG_DMASEL3_MASK; + dmasel |= TX4927_PCFG_DMASEL3_ACL1; + break; + default: + return; + } + local_irq_save(flags); + txx9_clear64(&tx4927_ccfgptr->pcfg, dmasel_mask); + txx9_set64(&tx4927_ccfgptr->pcfg, dmasel); + local_irq_restore(flags); + txx9_aclc_init(TX4927_ACLC_REG & 0xfffffffffULL, + TXX9_IRQ_BASE + TX4927_IR_ACLC, + 0, dma_chan_out, dma_chan_in); +} + static void __init tx4927_stop_unused_modules(void) { __u64 pcfg, rst = 0, ckd = 0; diff --git a/arch/mips/txx9/generic/setup_tx4938.c b/arch/mips/txx9/generic/setup_tx4938.c index b2b8529..4dfdb52 100644 --- a/arch/mips/txx9/generic/setup_tx4938.c +++ b/arch/mips/txx9/generic/setup_tx4938.c @@ -414,6 +414,17 @@ void __init tx4938_dmac_init(int memcpy_chan0, int memcpy_chan1) } } +void __init tx4938_aclc_init(void) +{ + u64 pcfg = __raw_readq(&tx4938_ccfgptr->pcfg); + + if ((pcfg & TX4938_PCFG_SEL2) && + !(pcfg & TX4938_PCFG_ETH0_SEL)) + txx9_aclc_init(TX4938_ACLC_REG & 0xfffffffffULL, + TXX9_IRQ_BASE + TX4938_IR_ACLC, + 1, 0, 1); +} + static void __init tx4938_stop_unused_modules(void) { __u64 pcfg, rst = 0, ckd = 0; diff --git a/arch/mips/txx9/generic/setup_tx4939.c b/arch/mips/txx9/generic/setup_tx4939.c index 98effef..7139686 100644 --- a/arch/mips/txx9/generic/setup_tx4939.c +++ b/arch/mips/txx9/generic/setup_tx4939.c @@ -485,6 +485,15 @@ void __init tx4939_dmac_init(int memcpy_chan0, int memcpy_chan1) } } +void __init tx4939_aclc_init(void) +{ + u64 pcfg = __raw_readq(&tx4939_ccfgptr->pcfg); + + if ((pcfg & TX4939_PCFG_I2SMODE_MASK) == TX4939_PCFG_I2SMODE_ACLC) + txx9_aclc_init(TX4939_ACLC_REG & 0xfffffffffULL, + TXX9_IRQ_BASE + TX4939_IR_ACLC, 1, 0, 1); +} + static void __init tx4939_stop_unused_modules(void) { __u64 pcfg, rst = 0, ckd = 0; diff --git a/arch/mips/txx9/rbtx4927/setup.c b/arch/mips/txx9/rbtx4927/setup.c index 332cdbc..ee468ea 100644 --- a/arch/mips/txx9/rbtx4927/setup.c +++ b/arch/mips/txx9/rbtx4927/setup.c @@ -337,10 +337,14 @@ static void __init rbtx4927_device_init(void) rbtx4927_ne_init(); tx4927_wdt_init(); rbtx4927_mtd_init(); - if (TX4927_REV_PCODE() == 0x4927) + if (TX4927_REV_PCODE() == 0x4927) { tx4927_dmac_init(2); - else + tx4927_aclc_init(0, 1); + } else { tx4938_dmac_init(0, 2); + tx4938_aclc_init(); + } + platform_device_register_simple("txx9aclc-generic", -1, NULL, 0); txx9_iocled_init(RBTX4927_LED_ADDR - IO_BASE, -1, 3, 1, "green", NULL); rbtx4927_gpioled_init(); } diff --git a/arch/mips/txx9/rbtx4938/setup.c b/arch/mips/txx9/rbtx4938/setup.c index 37c5e3d..8da66e9 100644 --- a/arch/mips/txx9/rbtx4938/setup.c +++ b/arch/mips/txx9/rbtx4938/setup.c @@ -356,6 +356,8 @@ static void __init rbtx4938_device_init(void) tx4938_ndfmc_init(10, 35); tx4938_ata_init(RBTX4938_IRQ_IOC_ATA, 0, 1); tx4938_dmac_init(0, 2); + tx4938_aclc_init(); + platform_device_register_simple("txx9aclc-generic", -1, NULL, 0); txx9_iocled_init(RBTX4938_LED_ADDR - IO_BASE, -1, 8, 1, "green", NULL); } diff --git a/arch/mips/txx9/rbtx4939/setup.c b/arch/mips/txx9/rbtx4939/setup.c index 91f2ec8..d5ad5ab 100644 --- a/arch/mips/txx9/rbtx4939/setup.c +++ b/arch/mips/txx9/rbtx4939/setup.c @@ -499,6 +499,8 @@ static void __init rbtx4939_device_init(void) tx4939_ata_init(); tx4939_rtc_init(); tx4939_dmac_init(0, 2); + tx4939_aclc_init(); + platform_device_register_simple("txx9aclc-generic", -1, NULL, 0); } static void __init rbtx4939_setup(void) -- 1.5.6.5

3 2

[alsa-devel] [PATCH 1/1] Add AIC3106 support.
by Ambrose, Martin 20 May '09

20 May '09

Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Martin Ambrose <martin(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 185 ++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic3x.h | 11 ++- 2 files changed, 190 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ab099f4..5720cad 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -57,13 +57,17 @@ struct aic3x_priv { int master; }; +static u8 pg0_end = AIC3X_GEN_PG0_END; +static u8 pg1_end = AIC3X_GEN_PG1_END; +static enum aic3x_codec_variant codec_variant; + /* * AIC3X register cache * We can't read the AIC3X register space when we are * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { +static const u8 aic3x_reg[] = { 0x00, 0x00, 0x00, 0x10, /* 0 */ 0x04, 0x00, 0x00, 0x00, /* 4 */ 0x00, 0x00, 0x00, 0x01, /* 8 */ @@ -90,6 +94,35 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x00, 0x00, /* 92 */ 0x00, 0x00, 0x00, 0x00, /* 96 */ 0x00, 0x00, 0x02, /* 100 */ + + 0x00, 0x00, 0x00, 0x00, /* 103-127 unused */ + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, + + 0x00, 0x6b, 0xe3, 0x96, /* 0 */ + 0x66, 0x67, 0x5d, 0x6b, /* 4 */ + 0xe3, 0x96, 0x66, 0x67, /* 8 */ + 0x5d, 0x7d, 0x83, 0x84, /* 12 */ + 0xee, 0x7d, 0x83, 0x84, /* 16 */ + 0xee, 0x39, 0x55, 0xf3, /* 20 */ + 0x2d, 0x53, 0x7e, 0x6b, /* 24 */ + 0xe3, 0x96, 0x66, 0x67, /* 28 */ + 0x5d, 0x6b, 0xe3, 0x96, /* 32 */ + 0x66, 0x67, 0x5d, 0x7d, /* 36 */ + 0x83, 0x84, 0xee, 0x7d, /* 40 */ + 0x83, 0x84, 0xee, 0x39, /* 44 */ + 0x55, 0xf3, 0x2d, 0x53, /* 48 */ + 0x7e, 0x7f, 0xff, 0x00, /* 52 */ + 0x00, 0x00, 0x00, 0x00, /* 56 */ + 0x00, 0x00, 0x00, 0x00, /* 60 */ + 0x00, 0x39, 0x55, 0xf3, /* 64 */ + 0x2d, 0x53, 0x7e, 0x39, /* 68 */ + 0x55, 0xf3, 0x2d, 0x53, /* 72 */ + 0x7e, /* 76 */ }; /* @@ -99,7 +132,8 @@ static inline unsigned int aic3x_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u8 *cache = codec->reg_cache; - if (reg >= AIC3X_CACHEREGNUM) + if ((reg >= pg1_end) || + ((reg >= pg0_end) && (reg < AIC3X_GEN_PG1_BEG))) return -1; return cache[reg]; } @@ -111,7 +145,8 @@ static inline void aic3x_write_reg_cache(struct snd_soc_codec *codec, u8 reg, u8 value) { u8 *cache = codec->reg_cache; - if (reg >= AIC3X_CACHEREGNUM) + if ((reg >= pg1_end) || + ((reg >= pg0_end) && (reg < AIC3X_GEN_PG1_BEG))) return; cache[reg] = value; } @@ -119,8 +154,8 @@ static inline void aic3x_write_reg_cache(struct snd_soc_codec *codec, /* * write to the aic3x register space */ -static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) +static int _aic3x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) { u8 data[2]; @@ -132,12 +167,41 @@ static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0xff; aic3x_write_reg_cache(codec, data[0], data[1]); + + /* adjust for page 1 before updating hardware if necessary */ + if (data[0] >= AIC3X_GEN_PG1_BEG) + data[0] -= AIC3X_GEN_PG1_BEG; + if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else return -EIO; } +static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 cur_pg; + u8 reg_pg; + int ret = 0; + + cur_pg = aic3x_read_reg_cache(codec, 0); + if (reg < pg0_end) + reg_pg = 0; + else if ((reg >= AIC3X_GEN_PG1_BEG) && (reg < pg1_end)) + reg_pg = 1; + else + return -EIO; + + if (cur_pg != reg_pg) + ret = _aic3x_write(codec, 0, reg_pg); + + if (ret == 0) + ret = _aic3x_write(codec, reg, value); + + return ret; +} + /* * read from the aic3x register space */ @@ -218,6 +282,65 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, return ret; } +static int tlv320aic3x_dual_reg_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xffff; + return 0; +} + +static int tlv320aic3x_dual_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg_msb = kcontrol->private_value & 0xff; + int reg_lsb = (kcontrol->private_value >> 8) & 0xff; + int val = aic3x_read_reg_cache(codec, reg_msb) << 8; + + val |= aic3x_read_reg_cache(codec, reg_lsb); + + /* convert 2's complement to unsigned int */ + val ^= 0x8000; + + ucontrol->value.integer.value[0] = val; + + return 0; +} + +static int tlv320aic3x_dual_reg_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int err; + int reg_msb = kcontrol->private_value & 0xff; + int reg_lsb = (kcontrol->private_value >> 8) & 0xff; + int val_msb, val_lsb; + + val_msb = (ucontrol->value.integer.value[0] >> 8) & 0xff; + val_lsb = ucontrol->value.integer.value[0] & 0xff; + + /* convert unsigned int to 2's complement */ + val_msb ^= 0x80; + + err = snd_soc_update_bits(codec, reg_msb, 0xff, val_msb); + if (err < 0) + return err; + err = snd_soc_update_bits(codec, reg_lsb, 0xff, val_lsb); + return err; +} + +#define TLV320AIC3X_DUAL_R(xname, page, reg_msb, reg_lsb) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = tlv320aic3x_dual_reg_info, \ + .get = tlv320aic3x_dual_reg_get, .put = tlv320aic3x_dual_reg_put, \ + .private_value = ((reg_msb) + page) | (((reg_lsb) + page) << 8) } + +#define TLV320AIC3X_PG1_DUAL_R(xname, reg_msb, reg_lsb) \ + TLV320AIC3X_DUAL_R(xname, AIC3X_GEN_PG1_BEG, reg_msb, reg_lsb) + static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; static const char *aic3x_left_hpcom_mux[] = @@ -344,6 +467,36 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N0", 1, 2), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N1", 3, 4), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N2", 5, 6), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N3", 7, 8), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N4", 9, 10), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N5", 11, 12), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient D1", 13, 14), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient D2", 15, 16), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient D4", 17, 18), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient D5", 19, 20), + TLV320AIC3X_PG1_DUAL_R("Left De-Emphasis Coefficient N0", 21, 22), + TLV320AIC3X_PG1_DUAL_R("Left De-Emphasis Coefficient N1", 23, 24), + TLV320AIC3X_PG1_DUAL_R("Left De-Emphasis Coefficient D1", 25, 26), + + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N0", 27, 28), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N1", 29, 30), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N2", 31, 32), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N3", 33, 34), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N4", 35, 36), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N5", 37, 38), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient D1", 39, 40), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient D2", 41, 42), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient D4", 43, 44), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient D5", 45, 46), + TLV320AIC3X_PG1_DUAL_R("Right De-Emphasis Coefficient N0", 47, 48), + TLV320AIC3X_PG1_DUAL_R("Right De-Emphasis Coefficient N1", 49, 50), + TLV320AIC3X_PG1_DUAL_R("Right De-Emphasis Coefficient D1", 51, 52), + + TLV320AIC3X_PG1_DUAL_R("3-D Attenuation Coefficient", 53, 54), }; /* Left DAC Mux */ @@ -454,6 +607,18 @@ static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = { SOC_DAPM_SINGLE("HPRCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), }; +static const struct snd_kcontrol_new aic3106_snd_controls[] = { + TLV320AIC3X_PG1_DUAL_R("Left Capture High Pass Coefficient N0", 65, 66), + TLV320AIC3X_PG1_DUAL_R("Left Capture High Pass Coefficient N1", 67, 68), + TLV320AIC3X_PG1_DUAL_R("Left Capture High Pass Coefficient D1", 69, 70), + TLV320AIC3X_PG1_DUAL_R("Right Capture High Pass Coefficient N0", + 71, 72), + TLV320AIC3X_PG1_DUAL_R("Right Capture High Pass Coefficient N1", + 73, 74), + TLV320AIC3X_PG1_DUAL_R("Right Capture High Pass Coefficient D1", + 75, 76), +}; + static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Left DAC to Left Outputs */ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0), @@ -1165,6 +1330,10 @@ static int aic3x_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + /* setup register cache sizes */ + if (codec_variant == AIC3106_CODEC) + pg1_end = AIC3106_PG1_END; + aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); aic3x_write(codec, AIC3X_RESET, SOFT_RESET); @@ -1247,6 +1416,11 @@ static int aic3x_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, aic3x_snd_controls, ARRAY_SIZE(aic3x_snd_controls)); + + if (codec_variant == AIC3106_CODEC) + snd_soc_add_controls(codec, aic3106_snd_controls, + ARRAY_SIZE(aic3106_snd_controls)); + aic3x_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -1374,6 +1548,7 @@ static int aic3x_probe(struct platform_device *pdev) printk(KERN_INFO "AIC3X Audio Codec %s\n", AIC3X_VERSION); setup = socdev->codec_data; + codec_variant = setup->variant; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index ac827e5..ee40e9c 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -13,7 +13,10 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_GEN_PG0_END 103 +#define AIC3X_GEN_PG1_BEG 128 +#define AIC3X_GEN_PG1_END 183 +#define AIC3106_PG1_END 205 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -199,6 +202,11 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +enum aic3x_codec_variant { + AIC3X_GENERIC_CODEC, + AIC3106_CODEC, +}; + /* GPIO API */ enum { AIC3X_GPIO1_FUNC_DISABLED = 0, @@ -284,6 +292,7 @@ int aic3x_button_pressed(struct snd_soc_codec *codec); struct aic3x_setup_data { int i2c_bus; unsigned short i2c_address; + enum aic3x_codec_variant variant; unsigned int gpio_func[2]; }; -- 1.6.0.6

2 3

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 19 May '09

19 May '09

The following changes since commit f83fba8baab9e95fff0fe2be0e1e32a1650bdd7f: Mark Brown (1): ASoC: Add debug trace for bias level transitions are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Atsushi Nemoto (1): ASoC: Add TXx9 AC link controller driver (v3) Lopez Cruz, Misael (2): ASoC: TWL4030: Add control for selecting codec operation mode ASoC: SDP3430: Connect twl4030 voice DAI to McBSP3 Peter Ujfalusi (1): ASoC: TWL4030: Fix Analog capture path for AUXR sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/twl4030.c | 49 ++++- sound/soc/omap/sdp3430.c | 87 +++++++- sound/soc/txx9/Kconfig | 29 +++ sound/soc/txx9/Makefile | 11 + sound/soc/txx9/txx9aclc-ac97.c | 255 ++++++++++++++++++++++ sound/soc/txx9/txx9aclc-generic.c | 98 +++++++++ sound/soc/txx9/txx9aclc.c | 430 +++++++++++++++++++++++++++++++++++++ sound/soc/txx9/txx9aclc.h | 83 +++++++ 10 files changed, 1033 insertions(+), 11 deletions(-) create mode 100644 sound/soc/txx9/Kconfig create mode 100644 sound/soc/txx9/Makefile create mode 100644 sound/soc/txx9/txx9aclc-ac97.c create mode 100644 sound/soc/txx9/txx9aclc-generic.c create mode 100644 sound/soc/txx9/txx9aclc.c create mode 100644 sound/soc/txx9/txx9aclc.h

2 1

[alsa-devel] [PATCH] ASoC: Add TXx9 AC link controller driver (v3)
by Atsushi Nemoto 19 May '09

19 May '09

This patch adds support for the integrated ACLC of the TXx9 family. Signed-off-by: Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> --- Changes since v2: * register DAI driver as a platform driver * use platform device to obtain irq/mem/dma resources. sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/txx9/Kconfig | 29 +++ sound/soc/txx9/Makefile | 11 + sound/soc/txx9/txx9aclc-ac97.c | 255 ++++++++++++++++++++++ sound/soc/txx9/txx9aclc-generic.c | 98 +++++++++ sound/soc/txx9/txx9aclc.c | 430 +++++++++++++++++++++++++++++++++++++ sound/soc/txx9/txx9aclc.h | 83 +++++++ 8 files changed, 908 insertions(+), 0 deletions(-) create mode 100644 sound/soc/txx9/Kconfig create mode 100644 sound/soc/txx9/Makefile create mode 100644 sound/soc/txx9/txx9aclc-ac97.c create mode 100644 sound/soc/txx9/txx9aclc-generic.c create mode 100644 sound/soc/txx9/txx9aclc.c create mode 100644 sound/soc/txx9/txx9aclc.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 3d2bb6f..85e985a 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,6 +33,7 @@ source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/txx9/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0237879..6039b0d 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -11,3 +11,4 @@ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/txx9/Kconfig b/sound/soc/txx9/Kconfig new file mode 100644 index 0000000..ebc9327 --- /dev/null +++ b/sound/soc/txx9/Kconfig @@ -0,0 +1,29 @@ +## +## TXx9 ACLC +## +config SND_SOC_TXX9ACLC + tristate "SoC Audio for TXx9" + depends on HAS_TXX9_ACLC && TXX9_DMAC + help + This option enables support for the AC Link Controllers in TXx9 SoC. + +config HAS_TXX9_ACLC + bool + +config SND_SOC_TXX9ACLC_AC97 + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + + +## +## Boards +## +config SND_SOC_TXX9ACLC_GENERIC + tristate "Generic TXx9 ACLC sound machine" + depends on SND_SOC_TXX9ACLC + select SND_SOC_TXX9ACLC_AC97 + select SND_SOC_AC97_CODEC + help + This is a generic AC97 sound machine for use in TXx9 based systems. diff --git a/sound/soc/txx9/Makefile b/sound/soc/txx9/Makefile new file mode 100644 index 0000000..551f16c --- /dev/null +++ b/sound/soc/txx9/Makefile @@ -0,0 +1,11 @@ +# Platform +snd-soc-txx9aclc-objs := txx9aclc.o +snd-soc-txx9aclc-ac97-objs := txx9aclc-ac97.o + +obj-$(CONFIG_SND_SOC_TXX9ACLC) += snd-soc-txx9aclc.o +obj-$(CONFIG_SND_SOC_TXX9ACLC_AC97) += snd-soc-txx9aclc-ac97.o + +# Machine +snd-soc-txx9aclc-generic-objs := txx9aclc-generic.o + +obj-$(CONFIG_SND_SOC_TXX9ACLC_GENERIC) += snd-soc-txx9aclc-generic.o diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c new file mode 100644 index 0000000..0f83bdb --- /dev/null +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -0,0 +1,255 @@ +/* + * TXx9 ACLC AC97 driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include "txx9aclc.h" + +#define AC97_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_48000 + +#ifdef __BIG_ENDIAN +#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_BE +#else +#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_LE +#endif + +static DECLARE_WAIT_QUEUE_HEAD(ac97_waitq); + +/* REVISIT: How to find txx9aclc_soc_device from snd_ac97? */ +static struct txx9aclc_soc_device *txx9aclc_soc_dev; + +static int txx9aclc_regready(struct txx9aclc_soc_device *dev) +{ + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + + return __raw_readl(drvdata->base + ACINTSTS) & ACINT_REGACCRDY; +} + +/* AC97 controller reads codec register */ +static unsigned short txx9aclc_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + u32 dat; + + if (!(__raw_readl(base + ACINTSTS) & ACINT_CODECRDY(ac97->num))) + return 0xffff; + reg |= ac97->num << 7; + dat = (reg << ACREGACC_REG_SHIFT) | ACREGACC_READ; + __raw_writel(dat, base + ACREGACC); + __raw_writel(ACINT_REGACCRDY, base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) { + __raw_writel(ACINT_REGACCRDY, base + ACINTDIS); + dev_err(dev->soc_dev.dev, "ac97 read timeout (reg %#x)\n", reg); + dat = 0xffff; + goto done; + } + dat = __raw_readl(base + ACREGACC); + if (((dat >> ACREGACC_REG_SHIFT) & 0xff) != reg) { + dev_err(dev->soc_dev.dev, "reg mismatch %x with %x\n", + dat, reg); + dat = 0xffff; + goto done; + } + dat = (dat >> ACREGACC_DAT_SHIFT) & 0xffff; +done: + __raw_writel(ACINT_REGACCRDY, base + ACINTDIS); + return dat; +} + +/* AC97 controller writes to codec register */ +static void txx9aclc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + + __raw_writel(((reg | (ac97->num << 7)) << ACREGACC_REG_SHIFT) | + (val << ACREGACC_DAT_SHIFT), + base + ACREGACC); + __raw_writel(ACINT_REGACCRDY, base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) { + dev_err(dev->soc_dev.dev, + "ac97 write timeout (reg %#x)\n", reg); + } + __raw_writel(ACINT_REGACCRDY, base + ACINTDIS); +} + +static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + u32 ready = ACINT_CODECRDY(ac97->num) | ACINT_REGACCRDY; + + __raw_writel(ACCTL_ENLINK, base + ACCTLDIS); + mmiowb(); + udelay(1); + __raw_writel(ACCTL_ENLINK, base + ACCTLEN); + /* wait for primary codec ready status */ + __raw_writel(ready, base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, + (__raw_readl(base + ACINTSTS) & ready) == ready, + HZ)) { + dev_err(&ac97->dev, "primary codec is not ready " + "(status %#x)\n", + __raw_readl(base + ACINTSTS)); + } + __raw_writel(ACINT_REGACCRDY, base + ACINTSTS); + __raw_writel(ready, base + ACINTDIS); +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = txx9aclc_ac97_read, + .write = txx9aclc_ac97_write, + .reset = txx9aclc_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id) +{ + struct txx9aclc_plat_drvdata *drvdata = dev_id; + void __iomem *base = drvdata->base; + + __raw_writel(__raw_readl(base + ACINTMSTS), base + ACINTDIS); + wake_up(&ac97_waitq); + return IRQ_HANDLED; +} + +static int txx9aclc_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + + dev->aclc_pdev = to_platform_device(dai->dev); + txx9aclc_soc_dev = dev; + return 0; +} + +static void txx9aclc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct platform_device *aclc_pdev = to_platform_device(dai->dev); + struct txx9aclc_plat_drvdata *drvdata = platform_get_drvdata(aclc_pdev); + + /* disable AC-link */ + __raw_writel(ACCTL_ENLINK, drvdata->base + ACCTLDIS); + txx9aclc_soc_dev = NULL; +} + +struct snd_soc_dai txx9aclc_ac97_dai = { + .name = "txx9aclc_ac97", + .ac97_control = 1, + .probe = txx9aclc_ac97_probe, + .remove = txx9aclc_ac97_remove, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, +}; +EXPORT_SYMBOL_GPL(txx9aclc_ac97_dai); + +static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev) +{ + struct txx9aclc_plat_drvdata *drvdata; + struct resource *r; + int err; + int irq; + + irq = platform_get_irq(pdev, 0); + if (irq < 0) + return irq; + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) + return -EBUSY; + + if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r), + dev_name(&pdev->dev))) + return -EBUSY; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + platform_set_drvdata(pdev, drvdata); + drvdata->physbase = r->start; + if (sizeof(drvdata->physbase) > sizeof(r->start) && + r->start >= TXX9_DIRECTMAP_BASE && + r->start < TXX9_DIRECTMAP_BASE + 0x400000) + drvdata->physbase |= 0xf00000000ull; + drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r)); + if (!drvdata->base) + return -EBUSY; + err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, + IRQF_DISABLED, dev_name(&pdev->dev), drvdata); + if (err < 0) + return err; + + txx9aclc_ac97_dai.dev = &pdev->dev; + return snd_soc_register_dai(&txx9aclc_ac97_dai); +} + +static int __devexit txx9aclc_ac97_dev_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&txx9aclc_ac97_dai); + return 0; +} + +static struct platform_driver txx9aclc_ac97_driver = { + .probe = txx9aclc_ac97_dev_probe, + .remove = __devexit_p(txx9aclc_ac97_dev_remove), + .driver = { + .name = "txx9aclc-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init txx9aclc_ac97_init(void) +{ + return platform_driver_register(&txx9aclc_ac97_driver); +} + +static void __exit txx9aclc_ac97_exit(void) +{ + platform_driver_unregister(&txx9aclc_ac97_driver); +} + +module_init(txx9aclc_ac97_init); +module_exit(txx9aclc_ac97_exit); + +MODULE_AUTHOR("Atsushi Nemoto <anemo(a)mba.ocn.ne.jp>"); +MODULE_DESCRIPTION("TXx9 ACLC AC97 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c new file mode 100644 index 0000000..3175de9 --- /dev/null +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -0,0 +1,98 @@ +/* + * Generic TXx9 ACLC machine driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This is a very generic AC97 sound machine driver for boards which + * have (AC97) audio at ACLC (e.g. RBTX49XX boards). + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include "../codecs/ac97.h" +#include "txx9aclc.h" + +static struct snd_soc_dai_link txx9aclc_generic_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &txx9aclc_ac97_dai, + .codec_dai = &ac97_dai, +}; + +static struct snd_soc_card txx9aclc_generic_card = { + .name = "Generic TXx9 ACLC Audio", + .platform = &txx9aclc_soc_platform, + .dai_link = &txx9aclc_generic_dai, + .num_links = 1, +}; + +static struct txx9aclc_soc_device txx9aclc_generic_soc_device = { + .soc_dev = { + .card = &txx9aclc_generic_card, + .codec_dev = &soc_codec_dev_ac97, + }, +}; + +static int __init txx9aclc_generic_probe(struct platform_device *pdev) +{ + struct txx9aclc_soc_device *dev = &txx9aclc_generic_soc_device; + struct platform_device *soc_pdev; + int ret; + + soc_pdev = platform_device_alloc("soc-audio", -1); + if (!soc_pdev) + return -ENOMEM; + platform_set_drvdata(soc_pdev, &dev->soc_dev); + dev->soc_dev.dev = &soc_pdev->dev; + ret = platform_device_add(soc_pdev); + if (ret) { + platform_device_put(soc_pdev); + return ret; + } + platform_set_drvdata(pdev, soc_pdev); + return 0; +} + +static int __exit txx9aclc_generic_remove(struct platform_device *pdev) +{ + struct platform_device *soc_pdev = platform_get_drvdata(pdev); + + platform_device_unregister(soc_pdev); + return 0; +} + +static struct platform_driver txx9aclc_generic_driver = { + .remove = txx9aclc_generic_remove, + .driver = { + .name = "txx9aclc-generic", + .owner = THIS_MODULE, + }, +}; + +static int __init txx9aclc_generic_init(void) +{ + return platform_driver_probe(&txx9aclc_generic_driver, + txx9aclc_generic_probe); +} + +static void __exit txx9aclc_generic_exit(void) +{ + platform_driver_unregister(&txx9aclc_generic_driver); +} + +module_init(txx9aclc_generic_init); +module_exit(txx9aclc_generic_exit); + +MODULE_AUTHOR("Atsushi Nemoto <anemo(a)mba.ocn.ne.jp>"); +MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c new file mode 100644 index 0000000..fa33661 --- /dev/null +++ b/sound/soc/txx9/txx9aclc.c @@ -0,0 +1,430 @@ +/* + * Generic TXx9 ACLC platform driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * 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/init.h> +#include <linux/platform_device.h> +#include <linux/scatterlist.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "txx9aclc.h" + +static const struct snd_pcm_hardware txx9aclc_pcm_hardware = { + /* + * REVISIT: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID + * needs more works for noncoherent MIPS. + */ + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE, +#ifdef __BIG_ENDIAN + .formats = SNDRV_PCM_FMTBIT_S16_BE, +#else + .formats = SNDRV_PCM_FMTBIT_S16_LE, +#endif + .period_bytes_min = 1024, + .period_bytes_max = 8 * 1024, + .periods_min = 2, + .periods_max = 4096, + .buffer_bytes_max = 32 * 1024, +}; + +static int txx9aclc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_device *socdev = rtd->socdev; + struct snd_pcm_runtime *runtime = substream->runtime; + struct txx9aclc_dmadata *dmadata = runtime->private_data; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + dev_dbg(socdev->dev, + "runtime->dma_area = %#lx dma_addr = %#lx dma_bytes = %zd " + "runtime->min_align %ld\n", + (unsigned long)runtime->dma_area, + (unsigned long)runtime->dma_addr, runtime->dma_bytes, + runtime->min_align); + dev_dbg(socdev->dev, + "periods %d period_bytes %d stream %d\n", + params_periods(params), params_period_bytes(params), + substream->stream); + + dmadata->substream = substream; + dmadata->pos = 0; + return 0; +} + +static int txx9aclc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int txx9aclc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct txx9aclc_dmadata *dmadata = runtime->private_data; + + dmadata->dma_addr = runtime->dma_addr; + dmadata->buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + dmadata->period_bytes = snd_pcm_lib_period_bytes(substream); + + if (dmadata->buffer_bytes == dmadata->period_bytes) { + dmadata->frag_bytes = dmadata->period_bytes >> 1; + dmadata->frags = 2; + } else { + dmadata->frag_bytes = dmadata->period_bytes; + dmadata->frags = dmadata->buffer_bytes / dmadata->period_bytes; + } + dmadata->frag_count = 0; + dmadata->pos = 0; + return 0; +} + +static void txx9aclc_dma_complete(void *arg) +{ + struct txx9aclc_dmadata *dmadata = arg; + unsigned long flags; + + /* dma completion handler cannot submit new operations */ + spin_lock_irqsave(&dmadata->dma_lock, flags); + if (dmadata->frag_count >= 0) { + dmadata->dmacount--; + BUG_ON(dmadata->dmacount < 0); + tasklet_schedule(&dmadata->tasklet); + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); +} + +static struct dma_async_tx_descriptor * +txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) +{ + struct dma_chan *chan = dmadata->dma_chan; + struct dma_async_tx_descriptor *desc; + struct scatterlist sg; + + sg_init_table(&sg, 1); + sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf_dma_addr)), + dmadata->frag_bytes, buf_dma_addr & (PAGE_SIZE - 1)); + sg_dma_address(&sg) = buf_dma_addr; + desc = chan->device->device_prep_slave_sg(chan, &sg, 1, + dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_TO_DEVICE : DMA_FROM_DEVICE, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(&chan->dev->device, "cannot prepare slave dma\n"); + return NULL; + } + desc->callback = txx9aclc_dma_complete; + desc->callback_param = dmadata; + desc->tx_submit(desc); + return desc; +} + +#define NR_DMA_CHAIN 2 + +static void txx9aclc_dma_tasklet(unsigned long data) +{ + struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data; + struct dma_chan *chan = dmadata->dma_chan; + struct dma_async_tx_descriptor *desc; + struct snd_pcm_substream *substream = dmadata->substream; + u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + ACCTL_AUDODMA : ACCTL_AUDIDMA; + int i; + unsigned long flags; + + spin_lock_irqsave(&dmadata->dma_lock, flags); + if (dmadata->frag_count < 0) { + struct txx9aclc_soc_device *dev = + container_of(dmadata, struct txx9aclc_soc_device, + dmadata[substream->stream]); + struct txx9aclc_plat_drvdata *drvdata = + txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + chan->device->device_terminate_all(chan); + /* first time */ + for (i = 0; i < NR_DMA_CHAIN; i++) { + desc = txx9aclc_dma_submit(dmadata, + dmadata->dma_addr + i * dmadata->frag_bytes); + if (!desc) + return; + } + dmadata->dmacount = NR_DMA_CHAIN; + chan->device->device_issue_pending(chan); + spin_lock_irqsave(&dmadata->dma_lock, flags); + __raw_writel(ctlbit, base + ACCTLEN); + dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags; + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + return; + } + BUG_ON(dmadata->dmacount >= NR_DMA_CHAIN); + while (dmadata->dmacount < NR_DMA_CHAIN) { + dmadata->dmacount++; + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + desc = txx9aclc_dma_submit(dmadata, + dmadata->dma_addr + + dmadata->frag_count * dmadata->frag_bytes); + if (!desc) + return; + chan->device->device_issue_pending(chan); + + spin_lock_irqsave(&dmadata->dma_lock, flags); + dmadata->frag_count++; + dmadata->frag_count %= dmadata->frags; + dmadata->pos += dmadata->frag_bytes; + dmadata->pos %= dmadata->buffer_bytes; + if ((dmadata->frag_count * dmadata->frag_bytes) % + dmadata->period_bytes == 0) + snd_pcm_period_elapsed(substream); + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); +} + +static int txx9aclc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct txx9aclc_soc_device *dev = + container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + unsigned long flags; + int ret = 0; + u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + ACCTL_AUDODMA : ACCTL_AUDIDMA; + + spin_lock_irqsave(&dmadata->dma_lock, flags); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dmadata->frag_count = -1; + tasklet_schedule(&dmadata->tasklet); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + __raw_writel(ctlbit, base + ACCTLDIS); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + __raw_writel(ctlbit, base + ACCTLEN); + break; + default: + ret = -EINVAL; + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + return ret; +} + +static snd_pcm_uframes_t +txx9aclc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + + return bytes_to_frames(substream->runtime, dmadata->pos); +} + +static int txx9aclc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct txx9aclc_soc_device *dev = + container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_dmadata *dmadata = &dev->dmadata[substream->stream]; + int ret; + + ret = snd_soc_set_runtime_hwparams(substream, &txx9aclc_pcm_hardware); + if (ret) + return ret; + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + substream->runtime->private_data = dmadata; + return 0; +} + +static int txx9aclc_pcm_close(struct snd_pcm_substream *substream) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + struct dma_chan *chan = dmadata->dma_chan; + + dmadata->frag_count = -1; + chan->device->device_terminate_all(chan); + return 0; +} + +static struct snd_pcm_ops txx9aclc_pcm_ops = { + .open = txx9aclc_pcm_open, + .close = txx9aclc_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = txx9aclc_pcm_hw_params, + .hw_free = txx9aclc_pcm_hw_free, + .prepare = txx9aclc_pcm_prepare, + .trigger = txx9aclc_pcm_trigger, + .pointer = txx9aclc_pcm_pointer, +}; + +static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, 64 * 1024, 4 * 1024 * 1024); +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct txx9aclc_dmadata *dmadata = param; + char devname[BUS_ID_SIZE + 2]; + + sprintf(devname, "%s.%d", dmadata->dma_res->name, + (int)dmadata->dma_res->start); + if (strcmp(dev_name(chan->device->dev), devname) == 0) { + chan->private = &dmadata->dma_slave; + return true; + } + return false; +} + +static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, + struct txx9aclc_dmadata *dmadata) +{ + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + struct txx9dmac_slave *ds = &dmadata->dma_slave; + dma_cap_mask_t mask; + + spin_lock_init(&dmadata->dma_lock); + + ds->reg_width = sizeof(u32); + if (dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ds->tx_reg = drvdata->physbase + ACAUDODAT; + ds->rx_reg = 0; + } else { + ds->tx_reg = 0; + ds->rx_reg = drvdata->physbase + ACAUDIDAT; + } + + /* Try to grab a DMA channel */ + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dmadata->dma_chan = dma_request_channel(mask, filter, dmadata); + if (!dmadata->dma_chan) { + dev_err(dev->soc_dev.dev, + "DMA channel for %s is not available\n", + dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"); + return -EBUSY; + } + tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet, + (unsigned long)dmadata); + return 0; +} + +static int txx9aclc_pcm_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + struct resource *r; + int i; + int ret; + + dev->dmadata[0].stream = SNDRV_PCM_STREAM_PLAYBACK; + dev->dmadata[1].stream = SNDRV_PCM_STREAM_CAPTURE; + for (i = 0; i < 2; i++) { + r = platform_get_resource(dev->aclc_pdev, IORESOURCE_DMA, i); + if (!r) { + ret = -EBUSY; + goto exit; + } + dev->dmadata[i].dma_res = r; + ret = txx9aclc_dma_init(dev, &dev->dmadata[i]); + if (ret) + goto exit; + } + return 0; + +exit: + for (i = 0; i < 2; i++) { + if (dev->dmadata[i].dma_chan) + dma_release_channel(dev->dmadata[i].dma_chan); + dev->dmadata[i].dma_chan = NULL; + } + return ret; +} + +static int txx9aclc_pcm_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + int i; + + /* disable all FIFO DMAs */ + __raw_writel(ACCTL_AUDODMA | ACCTL_AUDIDMA, base + ACCTLDIS); + /* dummy R/W to clear pending DMAREQ if any */ + __raw_writel(__raw_readl(base + ACAUDIDAT), base + ACAUDODAT); + + for (i = 0; i < 2; i++) { + struct txx9aclc_dmadata *dmadata = &dev->dmadata[i]; + struct dma_chan *chan = dmadata->dma_chan; + if (chan) { + dmadata->frag_count = -1; + chan->device->device_terminate_all(chan); + dma_release_channel(chan); + } + dev->dmadata[i].dma_chan = NULL; + } + return 0; +} + +struct snd_soc_platform txx9aclc_soc_platform = { + .name = "txx9aclc-audio", + .probe = txx9aclc_pcm_probe, + .remove = txx9aclc_pcm_remove, + .pcm_ops = &txx9aclc_pcm_ops, + .pcm_new = txx9aclc_pcm_new, + .pcm_free = txx9aclc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(txx9aclc_soc_platform); + +static int __init txx9aclc_soc_platform_init(void) +{ + return snd_soc_register_platform(&txx9aclc_soc_platform); +} + +static void __exit txx9aclc_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&txx9aclc_soc_platform); +} + +module_init(txx9aclc_soc_platform_init); +module_exit(txx9aclc_soc_platform_exit); + +MODULE_AUTHOR("Atsushi Nemoto <anemo(a)mba.ocn.ne.jp>"); +MODULE_DESCRIPTION("TXx9 ACLC Audio DMA driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc.h b/sound/soc/txx9/txx9aclc.h new file mode 100644 index 0000000..6769aab --- /dev/null +++ b/sound/soc/txx9/txx9aclc.h @@ -0,0 +1,83 @@ +/* + * TXx9 SoC AC Link Controller + * + * 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 __TXX9ACLC_H +#define __TXX9ACLC_H + +#include <linux/interrupt.h> +#include <asm/txx9/dmac.h> + +#define ACCTLEN 0x00 /* control enable */ +#define ACCTLDIS 0x04 /* control disable */ +#define ACCTL_ENLINK 0x00000001 /* enable/disable AC-link */ +#define ACCTL_AUDODMA 0x00000100 /* AUDODMA enable/disable */ +#define ACCTL_AUDIDMA 0x00001000 /* AUDIDMA enable/disable */ +#define ACCTL_AUDOEHLT 0x00010000 /* AUDO error halt + enable/disable */ +#define ACCTL_AUDIEHLT 0x00100000 /* AUDI error halt + enable/disable */ +#define ACREGACC 0x08 /* codec register access */ +#define ACREGACC_DAT_SHIFT 0 /* data field */ +#define ACREGACC_REG_SHIFT 16 /* address field */ +#define ACREGACC_CODECID_SHIFT 24 /* CODEC ID field */ +#define ACREGACC_READ 0x80000000 /* CODEC read */ +#define ACREGACC_WRITE 0x00000000 /* CODEC write */ +#define ACINTSTS 0x10 /* interrupt status */ +#define ACINTMSTS 0x14 /* interrupt masked status */ +#define ACINTEN 0x18 /* interrupt enable */ +#define ACINTDIS 0x1c /* interrupt disable */ +#define ACINT_CODECRDY(n) (0x00000001 << (n)) /* CODECn ready */ +#define ACINT_REGACCRDY 0x00000010 /* ACREGACC ready */ +#define ACINT_AUDOERR 0x00000100 /* AUDO underrun error */ +#define ACINT_AUDIERR 0x00001000 /* AUDI overrun error */ +#define ACDMASTS 0x80 /* DMA request status */ +#define ACDMA_AUDO 0x00000001 /* AUDODMA pending */ +#define ACDMA_AUDI 0x00000010 /* AUDIDMA pending */ +#define ACAUDODAT 0xa0 /* audio out data */ +#define ACAUDIDAT 0xb0 /* audio in data */ +#define ACREVID 0xfc /* revision ID */ + +struct txx9aclc_dmadata { + struct resource *dma_res; + struct txx9dmac_slave dma_slave; + struct dma_chan *dma_chan; + struct tasklet_struct tasklet; + spinlock_t dma_lock; + int stream; /* SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE */ + struct snd_pcm_substream *substream; + unsigned long pos; + dma_addr_t dma_addr; + unsigned long buffer_bytes; + unsigned long period_bytes; + unsigned long frag_bytes; + int frags; + int frag_count; + int dmacount; +}; + +struct txx9aclc_plat_drvdata { + void __iomem *base; + u64 physbase; +}; + +struct txx9aclc_soc_device { + struct snd_soc_device soc_dev; + struct platform_device *aclc_pdev; /* for ioresources, drvdata */ + struct txx9aclc_dmadata dmadata[2]; +}; + +static inline struct txx9aclc_plat_drvdata *txx9aclc_get_plat_drvdata( + struct txx9aclc_soc_device *sdev) +{ + return platform_get_drvdata(sdev->aclc_pdev); +} + +extern struct snd_soc_platform txx9aclc_soc_platform; +extern struct snd_soc_dai txx9aclc_ac97_dai; + +#endif /* __TXX9ACLC_H */ -- 1.5.6.5

2 1

Re: [alsa-devel] mainline tlv320aic26 driver
by Grant Likely 19 May '09

19 May '09

BTW, Please cc: the the alsa-devel(a)alsa-project.org mailing list when asking questions. I don't know everything and others can chime in that way. On Mon, May 18, 2009 at 12:44 PM, Joey Oravec <joravec(a)drewtech.com> wrote: > Grant - > > I'm trying to create an ASoC driver for my at91 platform with a tlv320aic26, > and I have a question about your driver. I'm looking at mainline 2.6.29. > > Inside aic26_probe() it's looking for struct aic26 to from > socdev->codec_data. I see how aic26_spi_probe() kzalloc's the structure, > sets some values in aic26->codec, but I have no idea when you expected > codec_data would get set. Could you briefly explain which function is > responsible to set that pointer and when it should happen? Since the aic26 is an SPI device, it is first probed (as you found) in the aic26_spi_probe() function. On my platform, the connection between the codec and the bus driver is described in an OF Flattened Device Tree and it uses the snd-of-simple driver to match the codec and platform drivers. So, the codec_data value is pass via the of_snd_soc_register_codec() function. I don't know the best way for your fabric driver to find the aic26 spi device. It probably should use bus_find_device() and dev_get_drvdata() to resolve to the value that codec_data should be set to. g. -- Grant Likely, B.Sc., P.Eng. Secret Lab Technologies Ltd.

2 1

[alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 19 May '09

19 May '09

Hi, after some communications with Creative, I finally got a mergeable version of their X-Fi driver. It's not my crappy hack but by genuine Creative :) The patch is now merged into sound-unstable GIT tree topic/ctxfi branch: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git The corresponding alsa-driver snapshot tarball is: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… The driver module is named as snd-ctxfi. I tested only the driver builds with recent kernels, but unfortunately I have no test hardware now, so it'd be helpful if anyone can test it with a real hardware. Note that the ctxfi driver might conflict with the HD-audio driver. So it'd be nice if you save the output of lspci before loading the drivers and let me know if any problem happens. Also, at testing, please check the kernel message. If this conflict really happens, try to build without hd-audio, that is, run configure with --with-cards=ctxfi option in addition to others, and test whether it works with this workaround (better after reboot to reset the board info by BIOS). Once after I get positive feedbacks (or fix after negative ones), I'm going to merge it to the main tree so that it'll be in the next kernel. So, a quick test report would be greatly appreciated. thanks, Takashi

8 30

[alsa-devel] Testers wanted: New SB X-Fi driver
by Daniel Bankmann 19 May '09

19 May '09

Hi! I just tested the newest driver version with Open Suse 11.1 and pae-kernel. I seems not to work probably, although the latest "official" driver by Creative worked more or less good. The problem is, that always, when hearing some music or any other sound, the signal gets interrupted every few seconds. At the other side, the startup jingle for example is accompanied by a knock every few seconds. Any idea, or what information do you need to fix this?

1 0

[alsa-devel] [PATCH] snd-hda-intel: add controls to toggle DC bias on mic ports
by Nickolas Lloyd 19 May '09

19 May '09

This patch adds a mixer control for the STAC92XX boards to control the DC bias of mic ports, allowing recording from both powered and non-powered sources. It replaces the "Mic Output Switch" with "Mic Jack Mode" to switch between Mic, Line In, and Line Out. Signed-off by: Nickolas Lloyd <ultrageek.lloyd(a)gmail.com> Note: diffed against alsa-driver-20090513 ---- diff -urpN alsa-driver/sound/pci/hda/patch_sigmatel.c alsa-driver.a/sound/pci/hda/patch_sigmatel.c --- alsa-driver/sound/pci/hda/patch_sigmatel.c 2009-05-12 22:05:31.000000000 +0000 +++ alsa-driver.a/sound/pci/hda/patch_sigmatel.c 2009-05-13 12:24:17.000000000 +0000 @@ -634,6 +634,96 @@ static int stac92xx_smux_enum_put(struct return 0; } +static unsigned int stac92xx_vref_set(struct hda_codec *codec, + hda_nid_t nid, unsigned int new_vref) +{ + unsigned int error; + unsigned int pincfg; + pincfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + + pincfg &= 0xff; + pincfg &= ~(AC_PINCTL_VREFEN | AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + pincfg |= new_vref; + + if (new_vref == AC_PINCTL_VREF_HIZ) + pincfg |= AC_PINCTL_OUT_EN; + else + pincfg |= AC_PINCTL_IN_EN; + + error = snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg); + if (error < 0) + return error; + else + return 1; +} + +static unsigned int stac92xx_vref_get(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int vref; + vref = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + vref &= AC_PINCTL_VREFEN; + return vref; +} + +static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int new_vref; + unsigned int error; + + if (ucontrol->value.enumerated.item[0] == 0) + new_vref = AC_PINCTL_VREF_80; + else if (ucontrol->value.enumerated.item[0] == 1) + new_vref = AC_PINCTL_VREF_GRD; + else + new_vref = AC_PINCTL_VREF_HIZ; + + if (new_vref != stac92xx_vref_get(codec, kcontrol->private_value)) { + error = stac92xx_vref_set(codec, + kcontrol->private_value, new_vref); + return error; + } + + return 0; +} + +static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vref = stac92xx_vref_get(codec, kcontrol->private_value); + if (vref == AC_PINCTL_VREF_80) + ucontrol->value.enumerated.item[0] = 0; + else if (vref == AC_PINCTL_VREF_GRD) + ucontrol->value.enumerated.item[0] = 1; + else if (vref == AC_PINCTL_VREF_HIZ) + ucontrol->value.enumerated.item[0] = 2; + + return 0; +} + +static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { + "Mic In", "Line In", "Line Out" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = 3; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 2; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -995,6 +1085,17 @@ static struct hda_verb stac9205_core_ini .private_value = verb_read | (verb_write << 16), \ } +#define DC_BIAS(xname, idx, nid) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = idx, \ + .info = stac92xx_dc_bias_info, \ + .get = stac92xx_dc_bias_get, \ + .put = stac92xx_dc_bias_put, \ + .private_value = nid, \ + } + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -2702,7 +2803,8 @@ enum { STAC_CTL_WIDGET_AMP_VOL, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, - STAC_CTL_WIDGET_CLFE_SWITCH + STAC_CTL_WIDGET_CLFE_SWITCH, + STAC_CTL_WIDGET_DC_BIAS }; static struct snd_kcontrol_new stac92xx_control_templates[] = { @@ -2714,6 +2816,7 @@ static struct snd_kcontrol_new stac92xx_ STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), + DC_BIAS(NULL, 0, 0), }; /* add dynamic controls */ @@ -3165,9 +3268,9 @@ static int stac92xx_auto_create_multi_ou } if (spec->mic_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, - "Mic as Output Switch", - (spec->mic_switch << 8) | 1); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_DC_BIAS, + "Mic Jack Mode", + spec->mic_switch); if (err < 0) return err; }

2 4

[alsa-devel] POLLOUT warnings dropped by Pulseaudio
by Ozan Çağlayan 19 May '09

19 May '09

Hi, Pulseaudio keeps telling May 19 12:22:59 mecbuk pulseaudio[1630]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! May 19 12:22:59 mecbuk pulseaudio[1630]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. May 19 12:22:59 mecbuk pulseaudio[1630]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. on quite every system (say 8 out of 10). I remember the long discussion thread about a few problems found and fixed in alsa-lib. Is there anything left to do, any way of helping by sending some debugging output or else, to fix those issues totally? Or should I silently ignore those messages? Thanks, Ozan Caglayan

1 0

Re: [alsa-devel] [RFC] ALSA vs. dedicated char device for a USB Audio Class gadget driver
by Laurent Pinchart 19 May '09

19 May '09

Hi Hal, On Friday 15 May 2009 22:15:18 Hal Murray wrote: > > If I use a synchronous endpoint, isn't the number of samples per frame > > determined by the nominal sampling rate and the nominal SOF frequency > > ? With the SOF clock running at 1kHz, I expect a synchronous endpoint > > for a 16 bits mono 48kHz stream to deliver exactly 48 frames (96 > > bytes) per USB frame. > > Beware of the "exact" in there. In real life, crystals have a tolerance. > The USB clock will not match the audio clock perfectly. What should be 1 > kHz might be 1.00003 kHz or it might be 0.99997 kHz. > > If the USB clock in this example is slightly fast, you will get occasional > times when there are only 47 samples ready. If it's slightly slow, you > will occasionally have 49 samples. > > That's slow/fast relative to the audio clock. If the USB clock is 0.001% > slow but the audio clock is 0.002% slow, then the USB clock will be fast > for this discussion. Except that the SOF clock *is* the reference clock. It will be slower or faster than the audio clock, and the userspace application will have to perform sample rate matching so that packets will be exactly 48 bytes long. > This is a common problem in communications. There are several techniques > to cope with it. > > The simplest is to make the transport mechanism have a bit of extra > bandwidth so you can always keep up. It's just a matter of how often you > use that extra bandwidth. In this case, you would allocate enough > bandwidth for 49 samples and only use 48 most of the time. An occasional > USB frame would have 47 or 49. That's what would happen with an asynchronous endpoint. The hard bit is to find out when to send that occasional packet, as the driver doesn't have access to the audio clock. > Another approach is to use big enough FIFO so you never get in trouble. > For this to work, you need a limit on the length of the data stream. For > example, suppose the longest song you want to send is 1000 seconds and both > clocks are accurate to 50 ppm (parts per million). The max difference in > clocks is 100 ppm. 48 k samples/second * 1000 seconds is 48 million > samples total. 100 ppm means that the worst difference would be 4800 > samples. So if the USB side waits until it gets 4800 samples before it > starts forwarding data, it won't run out if the audio clock is worst-case > slow and the USB clock is worst-case fast. You also need buffering for > another 4800 samples in case the clock differences are reversed. This adds > a delay to get started while you collect the 4800 samples. That won't work here, the stream can last forever (well, not quite, but still a long time). > You can just drop or duplicate a sample whenever the clocks slip. > You can derive both clocks from a common source. > > You can lock one clock to the other with a PLL (Phase Locked Loop). The SOF clock is driven by the USB host, and the audio clock is driven by the audio codec. I can't lock one to another or derive both of them from a common source. There could even be no audio clock at all if I stream audio data from a file. > There are probably other approaches. The applicable techniques require knowledge of both the audio clock and the SOF clock in a common place. My driver has no access to the audio clock. All it knows about is the SOF clock. There are only two options I can think of. The first one is to use an asynchronous endpoint and sent the occasional smaller or bigger packet (or duplicate/drop one sample). As the driver can't access the audio clock it needs to derive the information from the amount of data present in the ALSA ring buffer. To be honest I'm not sure if that will be possible at all, as the application will write data at a non-constant rate. The second one, which sounds easier, at least on the driver side, is to use a synchronous endpoint with a fixed packet size. The application will perform rate matching (duplicating/dropping a sample or resampling the audio stream) using the audio clock and the SOF clock. What I'm still unsure about is how the application can access the audio clock and the SOF clock through ALSA, but I suppose that's possible. Best regards, Laurent Pinchart

3 5

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: fix analog right microphone path
by Peter Ujfalusi 19 May '09

19 May '09

Hello, Here is the fix for AUXR recording that Anuj reported. Bit 2 is selecting the AUXR input in the ANAMICR register and not bit 1. --- Peter Ujfalusi (1): ASoC: TWL4030: Fix Analog capture path for AUXR sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

3 4

[alsa-devel] [PATCH 2/3 - RESEND] ASoC: TWL4030: Add control for selecting codec operation mode
by Lopez Cruz, Misael 19 May '09

19 May '09

Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 47 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 47 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d683d..8eca0f4 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -814,6 +814,48 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +/* Codec operation modes */ +static const char *twl4030_op_modes_texts[] = { + "Option 2 (voice/audio)", "Option 1 (audio)" +}; + +static const struct soc_enum twl4030_op_modes_enum = + SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0, + ARRAY_SIZE(twl4030_op_modes_texts), + twl4030_op_modes_texts); + +int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl4030_priv *twl4030 = codec->private_data; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned short val; + unsigned short mask, bitmask; + + if (twl4030->configured) { + printk(KERN_ERR "twl4030 operation mode cannot be " + "changed on-the-fly\n"); + return -EBUSY; + } + + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) + ; + if (ucontrol->value.enumerated.item[0] > e->max - 1) + return -EINVAL; + + val = ucontrol->value.enumerated.item[0] << e->shift_l; + mask = (bitmask - 1) << e->shift_l; + if (e->shift_l != e->shift_r) { + if (ucontrol->value.enumerated.item[1] > e->max - 1) + return -EINVAL; + val |= ucontrol->value.enumerated.item[1] << e->shift_r; + mask |= (bitmask - 1) << e->shift_r; + } + + return snd_soc_update_bits(codec, e->reg, mask, val); +} + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -895,6 +937,11 @@ static const struct soc_enum twl4030_vibradir_enum = twl4030_vibradir_texts); static const struct snd_kcontrol_new twl4030_snd_controls[] = { + /* Codec operation mode control */ + SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum, + snd_soc_get_enum_double, + snd_soc_put_twl4030_opmode_enum_double), + /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, -- 1.5.4.3

3 2

[alsa-devel] [PATCH] soc: Correct print specifiers for unsigneds
by Roel Kluin 19 May '09

19 May '09

Unsigned variables should use `%u' rather than `%d'. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- This also comtains some changes that I already sent here http://lkml.org/lkml/2009/5/11/497 Sorry for that, Is there some way to split these changes? sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/codecs/tlv320aic23.c | 4 ++-- sound/soc/codecs/uda134x.c | 4 ++-- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 8 ++++---- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 4 ++-- sound/soc/codecs/wm8753.c | 6 +++--- sound/soc/codecs/wm8900.c | 6 +++--- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 ++-- sound/soc/s3c24xx/s3c-i2s-v2.c | 4 ++-- sound/soc/sh/ssi.c | 2 +- 14 files changed, 26 insertions(+), 26 deletions(-) diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 7065753..9eb610c 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -117,7 +117,7 @@ static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( * Find actual rate, compare to requested rate */ actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n", rate, actual_rate); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3f4afb..a06b6bb 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -86,7 +86,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, */ if ((reg < 0 || reg > 9) && (reg != 15)) { - printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; } @@ -98,7 +98,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; - printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, value, reg); return -EIO; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index ddefb8f..269b108 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); if (reg >= UDA134X_REGS_NUM) { - printk(KERN_ERR "%s unkown register: reg: %d", + printk(KERN_ERR "%s unkown register: reg: %u", __func__, reg); return -EINVAL; } @@ -296,7 +296,7 @@ static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct uda134x_priv *uda134x = codec->private_data; - pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__, + pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__, clk_id, freq, dir); /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 0275321..e7348d3 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1108,7 +1108,7 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, if (ret < 0) return ret; dev_dbg(wm8350->dev, - "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d", + "FLL in %u FLL out %u N 0x%x K 0x%x div %d ratio %d", freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div, fll_div.ratio); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 510efa6..502eefa 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -954,7 +954,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, factors->outdiv *= 2; if (factors->outdiv > 32) { dev_err(wm8400->wm8400->dev, - "Unsupported FLL output frequency %dHz\n", + "Unsupported FLL output frequency %uHz\n", Fout); return -EINVAL; } @@ -1003,7 +1003,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, factors->k = K / 10; dev_dbg(wm8400->wm8400->dev, - "FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n", + "FLL: Fref=%u Fout=%u N=%x K=%x, FRATIO=%x OUTDIV=%x\n", Fref, Fout, factors->n, factors->k, factors->fratio, factors->outdiv); @@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev) codec = &priv->codec; codec->private_data = priv; - codec->control_data = dev->dev.driver_data; - priv->wm8400 = dev->dev.driver_data; + codec->control_data = dev_get_drvdata(&dev->dev); + priv->wm8400 = dev_get_drvdata(&dev->dev); ret = regulator_bulk_get(priv->wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6a4cea0..c8b8dba 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -298,7 +298,7 @@ static void pll_factors(unsigned int target, unsigned int source) if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8510 N value %d outwith recommended range!d\n", + "WM8510 N value %u outwith recommended range!d\n", Ndiv); pll_div.n = Ndiv; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 9f6be3d..86c4b24 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -415,7 +415,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, unsigned int K, Ndiv, Nmod; int i; - pr_debug("wm8580: PLL %dHz->%dHz\n", source, target); + pr_debug("wm8580: PLL %uHz->%uHz\n", source, target); /* Scale the output frequency up; the PLL should run in the * region of 90-100MHz. @@ -447,7 +447,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 5) || (Ndiv > 13)) { printk(KERN_ERR - "WM8580 N=%d outside supported range\n", Ndiv); + "WM8580 N=%u outside supported range\n", Ndiv); return -EINVAL; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a6e8f3f..d28eeac 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -703,7 +703,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "wm8753: unsupported N = %d\n", Ndiv); + "wm8753: unsupported N = %u\n", Ndiv); pll_div->n = Ndiv; Nmod = target % source; @@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8753_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8753; + dev_set_drvdata(&spi->dev, wm8753); return wm8753_register(wm8753); } static int __devexit wm8753_spi_remove(struct spi_device *spi) { - struct wm8753_priv *wm8753 = spi->dev.driver_data; + struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev); wm8753_unregister(wm8753); return 0; } diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 46c5ea1..3c78945 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -778,11 +778,11 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, } if (target > 100000000) - printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d" - " Fout=%d\n", target, Fref, Fout); + printk(KERN_WARNING "wm8900: FLL rate %u out of range, Fref=%u" + " Fout=%u\n", target, Fref, Fout); if (div > 32) { printk(KERN_ERR "wm8900: Invalid FLL division rate %u, " - "Fref=%d, Fout=%d, target=%d\n", + "Fref=%u, Fout=%u, target=%u\n", div, Fref, Fout, target); return -EINVAL; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 40cd274..d029818 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -998,7 +998,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8990 N value outwith recommended range! N = %d\n", Ndiv); + "WM8990 N value outwith recommended range! N = %u\n", Ndiv); pll_div->n = Ndiv; Nmod = target % source; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 523bad0..c79e666 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -689,7 +689,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source) Ndiv = target / source; if ((Ndiv < 5) || (Ndiv > 12)) printk(KERN_WARNING - "WM9713 PLL N value %d out of recommended range!\n", + "WM9713 PLL N value %u out of recommended range!\n", Ndiv); pll_div->n = Ndiv; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 286be31..d656a90 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -323,7 +323,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, - "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n", + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); switch (clk_id) { @@ -472,7 +472,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, ssacd |= (0x6 << 4); dev_dbg(&ssp->pdev->dev, - "Using SSACDD %x to supply %dHz\n", + "Using SSACDD %x to supply %uHz\n", val, freq_out); break; } diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index ab680aa..3cd94c5 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -507,7 +507,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, actual = clkrate / (fsdiv * div); deviation = actual - rate; - printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", + printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", fsdiv, div, actual, deviation); deviation = abs(deviation); @@ -523,7 +523,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, break; } - printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", + printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", best_fs, best_div, best_rate); info->fs_div = best_fs; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 56fa087..b378096 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -145,7 +145,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream, recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); - pr_debug("bits: %d channels: %d\n", bits, channels); + pr_debug("bits: %u channels: %u\n", bits, channels); ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA | CR_SWL_MASK);

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