August 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: s3c24xx platform: Fix s3c2410_dma_started called at wrong time
by Shine Liu 25 Aug '09

25 Aug '09

s3c24xx dma has the auto reload feature, when the the trnasfer is done, CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So the transmission is repeated. IRQ is issued while auto reload occurs. We change the DISRC and DCON[19:0] in the ISR, but at this time, the auto reload has been performed already. The first block is being re-transmitted by the DMA. So we need rewrite the DISRC and DCON[19:0] for the next block immediatly after the this block has been started to be transported. The function s3c2410_dma_started() is for this perpose, which is called in the form of "s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger(). But it is not correct. DMA transmission won't start until DMA REQ signal arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in s3c24xx_i2s_trigger(). In the current framework, s3c24xx_pcm_trigger() is always called before s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1) is called in this function. However, s3c2410_dma_started() is dma related, to call this function we should provide the channel number, which is given by substream->runtime->private_data->params->channel. The private_data points to a struct s3c24xx_runtime_data object, which is define in s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c To move the struct s3c24xx_runtime_data defination to s3c24xx-pcm.h is one solution. If not moved, the patch shold be like: Signed-off-by: Shine Liu <shinel(a)foxmail.com> ---------------------------------------------------------------- --- a/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.h 2009-08-20 16:06:14.000000000 +0800 @@ -28,4 +28,6 @@ extern struct snd_soc_platform s3c24xx_soc_platform; extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream); + #endif --- a/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-08-20 15:56:46.000000000 +0800 @@ -66,6 +66,28 @@ struct s3c24xx_pcm_dma_params *params; }; +/* s3c24xx_pcm_dma_started + * + * To load the second dma buffer immediately after the + * first dma buffer has been started transferring, so that + * in the dma autoreload situation, the first data block + * won't be re-transferred after the first irq issued. + * + * This function is called in the s3c24xx-i2s-trigger() of + * s3c24xx-i2s.c, just after the dma transfer is really started. + * + * It's ugly here. Need a elegant way to fix this bug. + * FIXME + * + */ + +void s3c24xx_pcm_dma_started(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); + +} + /* s3c24xx_pcm_enqueue * * place a dma buffer onto the queue for the dma system @@ -255,7 +277,6 @@ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: --- a/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-08-20 16:08:06.000000000 +0800 @@ -296,6 +296,9 @@ s3c24xx_snd_rxctrl(1); else s3c24xx_snd_txctrl(1); + + /* DMA transfer is really started, equeue the next buffer */ + s3c24xx_pcm_dma_started(substream); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND:

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 25 Aug '09

25 Aug '09

The following changes since commit c6ea2af76ab79b82a0bfed313e4c2b68b10c2bc1: Takashi Iwai (1): ASoC: Remove unneeded inclusion of linux/regulator/consumer.h are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Janusz Krzysztofik (1): ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 Jarkko Nikula (3): OMAP: McBSP: Use textual values in DMA operating mode sysfs files ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop Mark Brown (4): ASoC: Add input and output AIF widgets ASoC: WM8993 digital mixing support ASoC: Select core DMA when building for S3C64xx Merge branch 'topic/digital-mixing' into for-2.6.32 arch/arm/plat-omap/dma.c | 10 +++ arch/arm/plat-omap/include/mach/mcbsp.h | 2 - arch/arm/plat-omap/mcbsp.c | 132 ++++++++++++------------------ include/sound/soc-dapm.h | 8 ++ sound/soc/codecs/wm8993.c | 88 ++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 13 +--- sound/soc/s3c24xx/Kconfig | 1 + sound/soc/soc-dapm.c | 58 ++++++++++---- 8 files changed, 199 insertions(+), 113 deletions(-)

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[alsa-devel] [PATCH 0/3] Few patches to recent OMAP McBSP and ASoC changes
by Jarkko Nikula 25 Aug '09

25 Aug '09

Hi I would like to add these tree patches to recent set: [PATCHv5 00/20] OMAP ASoC changes in DMA utilization http://mailman.alsa-project.org/pipermail/alsa-devel/2009-August/020619.html These are actually independent patches but probably easier to handle as with the set above since some of the patches are touching both ASoC and OMAP subsystems. First patch is the most important since it changes the user space API for McBSP DMA operating mode selection. It would be more difficult to change it later if software starts to use current numeric value based operating mode selection in sysfs files. -- Jarkko

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[alsa-devel] multiple devices or one device w/ multiple subdevices?
by pl bossart 25 Aug '09

25 Aug '09

Hi, I have a piece of custom hardware that provides 7 independent PCM outputs. This is not intended for multichannel playback, the content on each output can be completely different; playback will not be synchronized and will start separately. The only restriction is that all outputs rely on the same baserate and share the same bitclock, i.e. all the outputs work at 48kHz. _From the mailling list archives, subdevices seem to be relevant when doing hw mixing or when lots of streams are handled. None of this applies to my case. Also it looks like PulseAudio knows nothing about subdevices in its detection loop, meaning the user would need to manually add some load-module commands in the PulseAudio configuration files. And last but not least, I didn't see very many lines of code (and no documentation) that showed how subdevices are registered/handled, etc. My conclusion is that multiple devices would seem like a much better option. But I wanted to check with the community that I am on the right track. Thanks for your feedback! - P

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[alsa-devel] [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
by Janusz Krzysztofik 25 Aug '09

25 Aug '09

Implement DMA channel self linking on OMAP1510 using AUTO_INIT and REPEAT flags of the DMA CCR register. Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Note: In its current form, enable_lnk() seems not very consistent (ie. result of a conditional assignment is then overwritten unconditionally). As rather different from OMAP1510 that I am familiar with, I am not quite sure how support for other OMAP models should really look like. So I decided to leave it as is and put my bit in a separate block, adding an additional FIXME comment. --- linux-2.6.31-rc5/arch/arm/plat-omap/dma.c.orig 2009-08-07 13:38:02.000000000 +0200 +++ linux-2.6.31-rc5/arch/arm/plat-omap/dma.c 2009-08-10 09:01:41.000000000 +0200 @@ -653,8 +653,21 @@ static inline void enable_lnk(int lch) { u32 l; + if (omap_dma_in_1510_mode()) { + /* only self linking is supported */ + if (dma_chan[lch].next_lch == lch) { + /* set AUTO_INIT | REPEAT bits in CCR */ + l = dma_read(CCR(lch)); + l |= 3 << 8; + dma_write(l, CCR(lch)); + } + return; + } + l = dma_read(CLNK_CTRL(lch)); + /* FIXME: + * this conditional assignment is then overwritten unconditionally */ if (cpu_class_is_omap1()) l &= ~(1 << 14); @@ -675,13 +688,21 @@ static inline void disable_lnk(int lch) { u32 l; - l = dma_read(CLNK_CTRL(lch)); + if (omap_dma_in_1510_mode()) + l = dma_read(CCR(lch)); + else + l = dma_read(CLNK_CTRL(lch)); /* Disable interrupts */ if (cpu_class_is_omap1()) { dma_write(0, CICR(lch)); - /* Set the STOP_LNK bit */ - l |= 1 << 14; + if (omap_dma_in_1510_mode()) { + /* Unset AUTO_INIT & REPEAT bits */ + l &= ~(3 << 8); + } else { + /* Set the STOP_LNK bit */ + l |= 1 << 14; + } } if (cpu_class_is_omap2()) { @@ -690,7 +711,10 @@ static inline void disable_lnk(int lch) l &= ~(1 << 15); } - dma_write(l, CLNK_CTRL(lch)); + if (omap_dma_in_1510_mode()) + dma_write(l, CCR(lch)); + else + dma_write(l, CLNK_CTRL(lch)); dma_chan[lch].flags &= ~OMAP_DMA_ACTIVE; } @@ -928,7 +952,7 @@ void omap_start_dma(int lch) { u32 l; - if (!omap_dma_in_1510_mode() && dma_chan[lch].next_lch != -1) { + if (dma_chan[lch].next_lch != -1) { int next_lch, cur_lch; char dma_chan_link_map[OMAP_DMA4_LOGICAL_DMA_CH_COUNT]; @@ -979,7 +1003,7 @@ void omap_stop_dma(int lch) { u32 l; - if (!omap_dma_in_1510_mode() && dma_chan[lch].next_lch != -1) { + if (dma_chan[lch].next_lch != -1) { int next_lch, cur_lch = lch; char dma_chan_link_map[OMAP_DMA4_LOGICAL_DMA_CH_COUNT]; @@ -1130,7 +1154,7 @@ int omap_dma_running(void) */ void omap_dma_link_lch(int lch_head, int lch_queue) { - if (omap_dma_in_1510_mode()) { + if (omap_dma_in_1510_mode() && (lch_head != lch_queue)) { printk(KERN_ERR "DMA linking is not supported in 1510 mode\n"); BUG(); return; @@ -1152,7 +1176,7 @@ EXPORT_SYMBOL(omap_dma_link_lch); */ void omap_dma_unlink_lch(int lch_head, int lch_queue) { - if (omap_dma_in_1510_mode()) { + if (omap_dma_in_1510_mode() && (lch_head != lch_queue)) { printk(KERN_ERR "DMA linking is not supported in 1510 mode\n"); BUG(); return; @@ -1822,7 +1846,7 @@ static int omap1_dma_handle_ch(int ch) if (unlikely(csr & OMAP_DMA_DROP_IRQ)) printk(KERN_WARNING "DMA synchronization event drop occurred " "with device %d\n", dma_chan[ch].dev_id); - if (likely(csr & OMAP_DMA_BLOCK_IRQ)) + if (likely(csr & OMAP_DMA_BLOCK_IRQ) && (dma_chan[ch].next_lch != ch)) dma_chan[ch].flags &= ~OMAP_DMA_ACTIVE; if (likely(dma_chan[ch].callback != NULL)) dma_chan[ch].callback(ch, csr, dma_chan[ch].data);

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Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Aug '09

24 Aug '09

2009/8/24 Takashi Iwai <tiwai(a)suse.de> > > At Mon, 24 Aug 2009 17:10:10 +0400, > Denis Kup wrote: > > > > Sorry, I was very busy last two months. We don't finish with this theme. > > What about my patch? > > Please check with the latest sound GIT tree (or alsa-driver snapshot) > whether it still work and repost the patch. There have been lots of > changes in HD-audio driver code, so an old patch might not work any > more. > > > thanks, > > Takashi I tested patch with alsa-driver-1.0.20.40.g13311.579.gcf341. All works as it should. It's diff between patched version( patch_realtek_new.c ) and unpatched: dener@linux-wpg9:~/Documents/1> diff ./patch_realtek_new.c ./patch_realtek_old.c 223d222 < ALC888_ACER_ASPIRE_7730G, 8341,8366d8339 < static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { < {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, < {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, < {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, < { } /* end */ < }; < < static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) < { < unsigned int present; < < present = snd_hda_codec_read(codec, 0x15, 0, < AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; < snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, < HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); < snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, < HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); < } < < static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, < unsigned int res) < { < if ((res >> 26) == ALC880_HP_EVENT) < alc888_acer_aspire_7730G_front_automute(codec); < } < 8558d8530 < [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", 8605d8576 < SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), 9094,9107d9064 < }, < [ALC888_ACER_ASPIRE_7730G] = { < .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, < .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_verbs }, < .num_dacs = ARRAY_SIZE(alc883_dac_nids), < .dac_nids = alc883_dac_nids, < .dig_out_nid = ALC883_DIGOUT_NID, < .dig_in_nid = ALC883_DIGIN_NID, < .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), < .channel_mode = alc883_3ST_6ch_modes, < .need_dac_fix = 1, < .input_mux = &alc883_capture_source, < .unsol_event = alc888_acer_aspire_7730G_unsol_event, < .init_hook = alc888_acer_aspire_7730G_front_automute,

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[alsa-devel] [PATCHv4 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 24 Aug '09

24 Aug '09

Hello again guys, Here is version 4 of these changes. No real changes from previous versions. The only thing is that now I've rebased on top of Takashi's tree: http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=summary as there are pending things there which Jarkko has pointed on previous series. BR, Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 377 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 78 ++++++- sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 511 insertions(+), 14 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Aug '09

24 Aug '09

2009/6/24 Denis Kup <dener.kup(a)gmail.com>: > 2009/6/24 Takashi Iwai <tiwai(a)suse.de>: >> At Wed, 24 Jun 2009 16:09:58 +0400, >> Denis Kup wrote: >>> >>> Hey, when i first installed alsa to my laptop, there no sound from >>> subwoofer, jacks were map very strange (2 surrounds for ex.). I try >>> all models from the list, but all don't support my subwoofer. After a >>> weak of hard work i made a patch that enables ACER_ASPIRE_7730G model >>> for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal >>> frontal speakers and internal subwoofer.You can see acer.com for more >>> info about hardware and jacks. >>> >>> It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: >> >> Can you try Emilio's patch recently posted (and pass >> model=acer-aspire-6530g) with the very latest alsa-driver-snapshot? >> This essentially includes your changes. It's in: >> From: Emilio López <buhitoescolar(a)gmail.com> >> Subject: Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G >> Date: Tue, 23 Jun 2009 22:32:45 -0300 >> Message-ID: <d54969c50906231832w48ce7315uc5cf8b3f81534c76(a)mail.gmail.com> >> >>> After changing file i install driver, and it automatically detect its >>> model (maybe there are old settings from previous install, so i dont >>> 100% sure that it was detected). With this settings subwoofer works >>> perfect, internal stereo and subwoofer mute when headphone are plugged >>> all works, except 1 thing: subwoofer uses center channel instead LFE. >>> To fix it i use such asound.conf file: >> ... >>> Is there any chance to swap Center and LFE in patch_realtek.c? >> >> Unlikely as ALC888* doesn't support the channel-swap feature (although >> some other ALC codecs do). >> >> >> thanks, >> >> Takashi >> > > You are talking about that > http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html > ?? Its right link to drivers http://www.alsa-project.org/snapshot/ ?? > And there is easier way to apply patches than manually adding strings > to files? > I manage with installing. I use alsa-driver-snapshot.tar.gz http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/ and this Amilo changes http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html . But this patch has smaller functionality against i have. There is advantages and disadvantages of both patches: My patch: + 1) when black jack plugged, internal speakers and subwoofer mute + 2) there is 3 maps for jacks: (black/pink/blue) 2ch: front/ext mic/ (something i dont know, it should be line in), 4ch: front/ ext mic/ side 6ch: front/ CLFE / side; (its changes in channel mode option in mixer) + 3) internal mic works + 4) there are separate boost mic channels for internal and external mics - 5) internal subwoofer controls with Center channel in mixer ? 6) i dont know how line in works (i will check it when reinstall to my patch) ? 7) i dont know how spdif works, because i have not any spdif speakers in house Amilo patch: - 1) when black jack pluged, only internal speaker mutes, subwoofer continue to work (it has no meaning when i listen to music in headphones) - 2) there is only 1 map for jacks: front/ext mic/ (something i dont know, it should be line in) - 3) there no side and center channels - 3) internal mic dont work - 4) there no line in in input source menu in mixer + 5) Subwoofer controls with LFE channel in mixer ? 7) i dont know how spdif works, because i have not any spdif speakers in house

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[alsa-devel] [RFC] [PATCH 3/3] ASoC: OMAP: Enhance OMAP1510 DMA progress software counter
by Janusz Krzysztofik 24 Aug '09

24 Aug '09

Enhance period index accuracy, particularly just before buffer rewind, by making use of DMA interrupt status flags in addition to simply counting interrupts This patch applies on top of patch 2 from this series: [RFC][PATCH 2/3] ASoC: OMAP: Make use of DMA channel self linking on OMAP1510 Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c.orig 2009-08-10 08:31:08.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c 2009-08-10 08:55:09.000000000 +0200 @@ -68,13 +68,28 @@ static void omap_pcm_dma_irq(int ch, u16 * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); - if (prtd->period_index >= 0) { - if (++prtd->period_index == runtime->periods) { + if (stat & OMAP_DMA_LAST_IRQ) { + /* last period of a buffer has been started */ + if (prtd->period_index == (runtime->periods - 1)) { + /* we are in sync, do nothing */ + spin_unlock_irqrestore(&prtd->lock, flags); + return; + } else if (prtd->period_index >= 0) { + /* possible IRQ loss, update the pointer */ + prtd->period_index = runtime->periods - 1; + } + } else if (prtd->period_index >= 0) { + /* playback in progress - increment the counter, + * check for end of buffer */ + if ((++prtd->period_index >= runtime->periods) || + (stat & OMAP_DMA_BLOCK_IRQ)) { + /* end of buffer reached, loop back */ prtd->period_index = 0; } } spin_unlock_irqrestore(&prtd->lock, flags); - } + } else if (stat & OMAP_DMA_LAST_IRQ) + return; snd_pcm_period_elapsed(substream); } @@ -175,7 +190,8 @@ static int omap_pcm_prepare(struct snd_p dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); - omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); + omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | + (cpu_is_omap1510() ? OMAP_DMA_LAST_IRQ : 0)); return 0; }

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[alsa-devel] [PATCH 1/2] ASoC: Add input and output AIF widgets
by Mark Brown 24 Aug '09

24 Aug '09

Currently DAPM interfaces with the audio streams to and from the processor at the DAC and ADC widgets. As the digital capabilities of parts increases this is becoming a less and less able to meet the needs of parts. To meet the needs of these devices create new widgets interfacing with the TDM bus but not integrated into any other functionality. Audio can then be routed to and from these widgets using existing routing widgets. A slot number is provided in the definition but this is currently not used yet. This is intended to support devices which can use more than one TDM slot on a single interface. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 8 ++++++ sound/soc/soc-dapm.c | 58 ++++++++++++++++++++++++++++++++------------- 2 files changed, 49 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 35814ce..3388405 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -137,6 +137,12 @@ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} /* stream domain */ +#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } +#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} @@ -312,6 +318,8 @@ enum snd_soc_dapm_type { snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_aif_in, /* audio interface input */ + snd_soc_dapm_aif_out, /* audio interface output */ }; /* diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8846131..a225e5a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,17 +55,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, [snd_soc_dapm_micbias] = 2, - [snd_soc_dapm_mic] = 3, - [snd_soc_dapm_mux] = 4, - [snd_soc_dapm_value_mux] = 4, - [snd_soc_dapm_dac] = 5, - [snd_soc_dapm_mixer] = 6, - [snd_soc_dapm_mixer_named_ctl] = 6, - [snd_soc_dapm_pga] = 7, - [snd_soc_dapm_adc] = 8, - [snd_soc_dapm_hp] = 9, - [snd_soc_dapm_spk] = 9, - [snd_soc_dapm_post] = 10, + [snd_soc_dapm_aif_in] = 3, + [snd_soc_dapm_aif_out] = 3, + [snd_soc_dapm_mic] = 4, + [snd_soc_dapm_mux] = 5, + [snd_soc_dapm_value_mux] = 5, + [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_mixer] = 7, + [snd_soc_dapm_mixer_named_ctl] = 7, + [snd_soc_dapm_pga] = 8, + [snd_soc_dapm_adc] = 9, + [snd_soc_dapm_hp] = 10, + [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_post] = 11, }; static int dapm_down_seq[] = { @@ -81,8 +83,10 @@ static int dapm_down_seq[] = { [snd_soc_dapm_micbias] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_value_mux] = 9, - [snd_soc_dapm_supply] = 10, - [snd_soc_dapm_post] = 11, + [snd_soc_dapm_aif_in] = 10, + [snd_soc_dapm_aif_out] = 10, + [snd_soc_dapm_supply] = 11, + [snd_soc_dapm_post] = 12, }; static void pop_wait(u32 pop_time) @@ -224,6 +228,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: p->connect = 1; break; /* does effect routing - dynamically connected */ @@ -497,8 +503,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) if (widget->id == snd_soc_dapm_supply) return 0; - if (widget->id == snd_soc_dapm_adc && widget->active) - return 1; + switch (widget->id) { + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + if (widget->active) + return 1; + default: + break; + } if (widget->connected) { /* connected pin ? */ @@ -537,8 +549,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return 0; /* active stream ? */ - if (widget->id == snd_soc_dapm_dac && widget->active) - return 1; + switch (widget->id) { + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: + if (widget->active) + return 1; + default: + break; + } if (widget->connected) { /* connected pin ? */ @@ -1052,6 +1070,8 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: if (w->name) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); @@ -1342,6 +1362,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, case snd_soc_dapm_pre: case snd_soc_dapm_post: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: list_add(&path->list, &codec->dapm_paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -1444,9 +1466,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) dapm_new_mux(codec, w); break; case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: -- 1.6.3.3

1 1

[alsa-devel] No Sound - X-Fi CA0110 IBG - Ubuntu 910 with Kernel 2.6.31-6
by David Favor 24 Aug '09

24 Aug '09

root@home1:~# uname -r 2.6.31-6-generic root@home1:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Generic [HD-Audio Generic], device 0: CA0110 Analog [CA0110 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Generic [HD-Audio Generic], device 1: CA0110 Digital [CA0110 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 root@home1:~# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.20. root@home1:~# cat /proc/asound/card0/codec#1 Codec: Creative CA0110-IBG Address: 1 Function Id: 0x1 Vendor Id: 0x1102000a Subsystem Id: 0x11021007 Revision Id: 0x100000 No Modem Function Group found Default PCM: N/A Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=255, o=255, i=255, unsolicited=1, wake=1 root@home1:~# lsmod | grep snd snd_hda_codec_ca0110 7868 1 snd_hda_intel 26600 6 snd_hda_codec 69276 2 snd_hda_codec_ca0110,snd_hda_intel snd_pcm 75360 4 snd_hda_intel,snd_hda_codec snd_timer 22244 1 snd_pcm snd 59204 15 snd_hda_codec_ca0110,snd_hda_intel,snd_hda_codec,snd_pcm,snd_timer soundcore 7264 1 snd snd_page_alloc 9156 2 snd_hda_intel,snd_pcm root@home1:~# lspci -vvnn 0d:00.0 Audio device [0403]: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG [1102:0009] Subsystem: Creative Labs Device [1102:0018] 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 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 26 Region 0: Memory at fbefc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel I find no mention anywhere of anyone getting this card to work. Someone let me know if you have a working card or if I should toss this one. Thanks.

3 4

[alsa-devel] [PATCH] hda: move open coded tricks into get_wcaps_channels()
by Wu Fengguang 24 Aug '09

24 Aug '09

Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_local.h | 10 ++++++++++ sound/pci/hda/hda_proc.c | 4 +--- sound/pci/hda/patch_atihdmi.c | 3 +-- 3 files changed, 12 insertions(+), 5 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_local.h 2009-08-24 09:48:57.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_local.h 2009-08-24 09:49:44.000000000 +0800 @@ -410,6 +410,16 @@ static inline u32 get_wcaps(struct hda_c /* get the widget type from widget capability bits */ #define get_wcaps_type(wcaps) (((wcaps) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT) +static inline unsigned int get_wcaps_channels(u32 wcaps) +{ + unsigned int chans; + + chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13; + chans = ((chans << 1) | 1) + 1; + + return chans; +} + u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); --- sound-2.6.orig/sound/pci/hda/hda_proc.c 2009-08-24 09:48:57.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_proc.c 2009-08-24 09:49:44.000000000 +0800 @@ -515,9 +515,7 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, get_wid_type_name(wid_type), wid_caps); if (wid_caps & AC_WCAP_STEREO) { - unsigned int chans; - chans = (wid_caps & AC_WCAP_CHAN_CNT_EXT) >> 13; - chans = ((chans << 1) | 1) + 1; + unsigned int chans = get_wcaps_channels(wid_caps); if (chans == 2) snd_iprintf(buffer, " Stereo"); else --- sound-2.6.orig/sound/pci/hda/patch_atihdmi.c 2009-08-24 09:48:57.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_atihdmi.c 2009-08-24 09:49:44.000000000 +0800 @@ -141,8 +141,7 @@ static int atihdmi_build_pcms(struct hda /* FIXME: we must check ELD and change the PCM parameters dynamically */ chans = get_wcaps(codec, CVT_NID); - chans = (chans & AC_WCAP_CHAN_CNT_EXT) >> 13; - chans = ((chans << 1) | 1) + 1; + chans = get_wcaps_channels(chans); info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; return 0;

2 1

[PATCH] ASoC: add missing inclusion of debugfs.h
by Takashi Iwai 24 Aug '09

24 Aug '09

To fix compile errors. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2f5295d..7b74547 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -37,6 +37,7 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/jiffies.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -- 1.6.3.3

1 0

[alsa-devel] [PATCH] ali5451: remove dead code
by Bartlomiej Zolnierkiewicz 23 Aug '09

23 Aug '09

From: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> Subject: [PATCH] ali5451: remove dead code Remove code covered by #if/endif 0 and #ifdef/endif CODEC_RESET (CODEC_RESET is never defined). Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> --- sound/pci/ali5451/ali5451.c | 65 -------------------------------------------- 1 file changed, 65 deletions(-) Index: b/sound/pci/ali5451/ali5451.c =================================================================== --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -477,45 +477,6 @@ static int snd_ali_reset_5451(struct snd return 0; } -#ifdef CODEC_RESET - -static int snd_ali_reset_codec(struct snd_ali *codec) -{ - struct pci_dev *pci_dev; - unsigned char bVal; - unsigned int dwVal; - unsigned short wCount, wReg; - - pci_dev = codec->pci_m1533; - - pci_read_config_dword(pci_dev, 0x7c, &dwVal); - pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000); - udelay(5000); - pci_read_config_dword(pci_dev, 0x7c, &dwVal); - pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff); - udelay(5000); - - bVal = inb(ALI_REG(codec,ALI_SCTRL)); - bVal |= 0x02; - outb(ALI_REG(codec,ALI_SCTRL),bVal); - udelay(5000); - bVal = inb(ALI_REG(codec,ALI_SCTRL)); - bVal &= 0xfd; - outb(ALI_REG(codec,ALI_SCTRL),bVal); - udelay(15000); - - wCount = 200; - while (wCount--) { - wReg = snd_ali_codec_read(codec->ac97, AC97_POWERDOWN); - if ((wReg & 0x000f) == 0x000f) - return 0; - udelay(5000); - } - return -1; -} - -#endif - /* * ALI 5451 Controller */ @@ -560,22 +521,6 @@ static void snd_ali_disable_address_inte outl(gc, ALI_REG(codec, ALI_GC_CIR)); } -#if 0 /* not used */ -static void snd_ali_enable_voice_irq(struct snd_ali *codec, - unsigned int channel) -{ - unsigned int mask; - struct snd_ali_channel_control *pchregs = &(codec->chregs); - - snd_ali_printk("enable_voice_irq channel=%d\n",channel); - - mask = 1 << (channel & 0x1f); - pchregs->data.ainten = inl(ALI_REG(codec, pchregs->regs.ainten)); - pchregs->data.ainten |= mask; - outl(pchregs->data.ainten, ALI_REG(codec, pchregs->regs.ainten)); -} -#endif - static void snd_ali_disable_voice_irq(struct snd_ali *codec, unsigned int channel) { @@ -676,16 +621,6 @@ static void snd_ali_free_channel_pcm(str } } -#if 0 /* not used */ -static void snd_ali_start_voice(struct snd_ali *codec, unsigned int channel) -{ - unsigned int mask = 1 << (channel & 0x1f); - - snd_ali_printk("start_voice: channel=%d\n",channel); - outl(mask, ALI_REG(codec,codec->chregs.regs.start)); -} -#endif - static void snd_ali_stop_voice(struct snd_ali *codec, unsigned int channel) { unsigned int mask = 1 << (channel & 0x1f);

2 1

[alsa-devel] [PATCH] ali5451: fix timeout handling in snd_ali_{codecs, timer}_ready()
by Bartlomiej Zolnierkiewicz 23 Aug '09

23 Aug '09

From: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> Subject: [PATCH] ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready() Modify loops in such way that the register value is checked also after the timeout condition, just in case the heavy interrupt load etc. caused the thread to sleep for the time period exceeding the timeout value. While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready(). Reported-by: Jack Byer <ojbyer(a)usa.net> Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier(a)gmail.com> --- sound/pci/ali5451/ali5451.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) Index: b/sound/pci/ali5451/ali5451.c =================================================================== --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -314,8 +314,11 @@ static int snd_ali_codec_ready(struct sn res = snd_ali_5451_peek(codec,port); if (!(res & 0x8000)) return 0; + if (!time_after_eq(end_time, jiffies)) + break; schedule_timeout_uninterruptible(1); - } while (time_after_eq(end_time, jiffies)); + } while (1); + snd_ali_5451_poke(codec, port, res & ~0x8000); snd_printdd("ali_codec_ready: codec is not ready.\n "); return -EIO; @@ -327,15 +330,17 @@ static int snd_ali_stimer_ready(struct s unsigned long dwChk1,dwChk2; dwChk1 = snd_ali_5451_peek(codec, ALI_STIMER); - dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); end_time = jiffies + msecs_to_jiffies(250); do { dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); if (dwChk2 != dwChk1) return 0; + if (!time_after_eq(end_time, jiffies)) + break; schedule_timeout_uninterruptible(1); - } while (time_after_eq(end_time, jiffies)); + } while (1); + snd_printk(KERN_ERR "ali_stimer_read: stimer is not ready.\n"); return -EIO; }

2 3

[alsa-devel] [PATCH] ASoC: free socdev if init_card() fails in wm9705_soc_probe()
by Roel Kluin 23 Aug '09

23 Aug '09

Free socdev if snd_soc_init_card() fails. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- It doesn't seem to happen in snd_soc_init_card(), but please review. diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index fa88b46..e7d2840 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -406,7 +406,7 @@ static int wm9705_soc_probe(struct platform_device *pdev) ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register card\n"); - goto pcm_err; + goto reset_err; } return 0;

2 1

Re: [alsa-devel] ALSA synchronization
by arif setiawan 23 Aug '09

23 Aug '09

Hi, We are working on audio processing (encryption-decryption) and audio will be sent using radio. Our configuration is : Audio source - ALSA audio in - encryption - ALSA audio out - transmission over radio (now we still use audio cable) - ALSA audio in - decryption - ALSA audio out - speaker. So data will be sent through radio channel and its converted to analog form by ALSA PCM during playback. We want to solve synchronization issue by using data header so that we can decrypt correct data block. So far we still unable to verify that data we received in the decryption side is similar to data that we sent from encryption box. Supposed that we filled buffer in encryption box with some values, when we check buffer in decryption box, we can't get same data. Our guess that data somehow changed in digital-analog/analog-digital conversion or we might make mistakes in filling the buffer so data is not compatible with ALSA PCM. Any suggestion to solve this problem? Thank you, Best Regards, Arif

3 3

[alsa-devel] [PATCH] Convert WM97xx-battery to dev_pm_ops
by Marek Vasut 22 Aug '09

22 Aug '09

Hi, Anton, are you fine with this one? Mark, in case Anton is ok, would you mind applying? Thanks! Cheers! Marek Vasut

1 0

[alsa-devel] ALSA synchronization
by arif setiawan 21 Aug '09

21 Aug '09

Hi all, We are developing a AES based speech encryption module using beagleboard. We have two beagleboard: one for encryption and other for decryption. We use audio-in/audio-out for sending speech. We have problem with synchronization of audio stream from encryption board to decryption board. Since AES is block cipher, the buffer content received by decryption board must be the same as sent from encryption board. Right now, we want to use some bytes as data header, but we don't now how to make this work. Any idea how to solve synchronization in ALSA? Best Regards, Arif

2 1

[alsa-devel] [PATCH] various acer-mic-internal bugs solution
by epic 21 Aug '09

21 Aug '09

Several mic related bugs maybe solved with this patch or a derived one 1 -My internal-mic was not working and front jack mic sound was terrible (useless for headset mic). 2 -Detected offset level from inputs {line, mic}, this offset level increases with specific [boost] and/or [capture(NID 0x23)] gain. * I just played around several days compiling and testing alsa with my acer-alc268, checked all data communication with alc268, pin config etc. and tested different values for offset in pins. * Found internal-mic comes alive with a 50% offset v_reference and mic sound is much better. *This patch applies/solves only for alc268 with option model=acer in alsa-base.conf, but i think the others acer-alcxxx bugs maybe solved in the same way. changing PIN_VREF80 to PIN_VREF50 at NID 0x18 mic NID 0x19 internal-mic NID 0x1a line-in (not tested) *Don't know what can happen to other acer-alcxxx, so i only patched acer-alc268. But this patch solved my mic-internal issue and most important for me now i can use a headset mic. *It remains a level offset, lower than before, but mic is usable, the level is showed in the pictures. *Pictures: input with two channels and a positive offset in left channel (red), negative offset in rigth channel (blue) http://imagebin.ca/view/Y8TyujP.html working internal microphone ok http://imagebin.ca/view/nzTzeX.html Realtek datasheet 1.3 is available for alc268 2.6.26-2-686 #1 SMP i686 GNU/Linux debian/squeeze description: Computer product: Aspire 7720 vendor: Acer version: V1.17 TAGS: alc268 offset level internal mic microphone alsa hda intel realtek Signed-off-by: EliasTG <epicadocan a.t gmail.com> upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Tue Aug 18 21:27:39 UTC 2009 !!Linux Distribution !!------------------ Debian GNU/Linux squeeze/sid \n \l !!Kernel Information !!------------------ Kernel release: 2.6.26-2-686 Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.20 Utilities version: 1.0.20 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 11 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 1025:011e !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 snd-hda-intel: index=0 model=test !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : 0,-1,-1,-1,-1,-1,-1,-1 model : test,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC268 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0268 Subsystem Id: 0x1025011e Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=04, enabled=1 Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19840: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99a30941: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0281304e: [Jack] Line In at Ext Front Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xe Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4017952d: [N/A] Speaker at Ext N/A Conn = Analog, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000010: OUT Pin Default 0x02451130: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=10 Processing Coefficient: 0x03 Coefficient Index: 0x01 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x15 0x15] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x13 Codec: Conexant ID 2c06 Address: 1 Function Id: 0x2 Vendor Id: 0x14f12c06 Subsystem Id: 0x1025011e Revision Id: 0x100000 Modem Function Group: 0x2 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Aug 18 22:02 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Aug 18 22:02 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 Aug 18 22:02 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 24 Aug 18 22:02 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Aug 18 22:02 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Aug 18 05:24 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Aug 18 05:24 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0x9b300000 irq 11' Mixer name : 'Realtek ALC268' Components : 'HDA:10ec0268,1025011e,00100003 HDA:14f12c06,1025011e,00100000' Controls : 12 Simple ctrls : 9 Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 44 [69%] [-20.00dB] [on] Front Right: Playback 44 [69%] [-20.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 185 [73%] [-14.00dB] Front Right: Playback 185 [73%] [-14.00dB] Simple mixer control 'Line In Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 2 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 2 Front Left: 2 [100%] Front Right: 2 [100%] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 21 [68%] [15.00dB] [on] Front Right: Capture 21 [68%] [15.00dB] [on] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 12 Mono: Front Left: Playback 0 [0%] [-24.00dB] [off] Front Right: Playback 0 [0%] [-24.00dB] [off] Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 0 [0%] [-30.00dB] Front Right: Capture 0 [0%] [-30.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Internal Mic' 'Line' Item0: 'Mic' Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 2 Front Left: 2 [100%] Front Right: 2 [100%] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 44 value.1 44 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Master Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 2' comment.dbmin 0 comment.dbmax 4000 iface MIXER name 'Mic Boost' value.0 2 value.1 2 } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 2' comment.dbmin 0 comment.dbmax 4000 iface MIXER name 'Internal Mic Boost' value.0 2 value.1 2 } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 2' comment.dbmin 0 comment.dbmax 4000 iface MIXER name 'Line In Boost' value.0 0 value.1 0 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -1650 comment.dbmax 3000 iface MIXER name 'Capture Volume' value.0 21 value.1 21 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.8 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Internal Mic' comment.item.2 Line iface MIXER name 'Input Source' value Mic } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 12' comment.dbmin -2400 comment.dbmax 0 iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 } control.10 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Beep Playback Switch' value.0 false value.1 false } control.11 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 185 value.1 185 } control.12 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 120' comment.tlv '0000000100000008fffff44800000032' comment.dbmin -3000 comment.dbmax 3000 iface MIXER name 'Digital Capture Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_hwdep usb_storage nvidia ppdev parport_pc lp parport ipv6 sbp2 loop snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_dummy snd_seq_oss snd_seq_midi snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device mxl5005s zl10353 snd arc4 ecb crypto_blkcipher soundcore joydev dvb_usb_ce6230 dvb_usb iwl3945 snd_page_alloc dvb_core uvcvideo serio_raw mac80211 i2c_i801 acer_wmi firmware_class videodev v4l1_compat cfg80211 i2c_core psmouse pcspkr led_class evdev video output battery button ac wmi ext2 mbcache usbhid hid ff_memless ide_cd_mod cdrom ide_pci_generic sd_mod ata_generic uhci_hcd ahci sdhci libata ricoh_mmc mmc_core ohci1394 scsi_mod dock ieee1394 piix ide_core tg3 ehci_hcd usbcore intel_agp agpgart thermal processor fan thermal_sys dm_mirror dm_log dm_mod !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x12 0x411111f0 0x13 0x411111f0 0x14 0x02211020 0x15 0x99130110 0x16 0x411111f0 0x18 0x02a19840 0x19 0x99a30941 0x1a 0x0281304e 0x1c 0x411111f0 0x1d 0x4017952d 0x1e 0x02451130 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D1/init_pin_configs: 0x73 0x016a10f0 /sys/class/sound/hwC0D1/driver_pin_configs: /sys/class/sound/hwC0D1/user_pin_configs: /sys/class/sound/hwC0D1/init_verbs: Codec: Realtek ALC268 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0268 Subsystem Id: 0x1025011e Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=04, enabled=1 Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19840: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99a30941: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0281304e: [Jack] Line In at Ext Front Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xe Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4017952d: [N/A] Speaker at Ext N/A Conn = Analog, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000010: OUT Pin Default 0x02451130: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=10 Processing Coefficient: 0xa948 Coefficient Index: 0x02 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x15 0x15] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x13 [617100.073588] hda_codec:patch_alc268 here we go! v2 [617100.079773] hda_codec:snd_hda_codec_write: 0x170e01 [617100.079856] hda_codec:snd_hda_codec_read: 0x1f000d --> 0x0 [617100.079954] hda_codec:snd_hda_codec_read: 0x100f0000 --> 0x14f12c06 [617100.080028] hda_codec:snd_hda_codec_read: 0x100f0001 --> 0x0 [617100.080105] hda_codec:snd_hda_codec_read: 0x100f0002 --> 0x100000 [617100.080170] hda_codec:snd_hda_codec_read: 0x100f0004 --> 0x20001 [617100.080257] hda_codec:snd_hda_codec_read: 0x102f0005 --> 0x102 [617100.080333] hda_codec:snd_hda_codec_read: 0x102f0004 --> 0x700004 [617100.080398] hda_codec:snd_hda_codec_read: 0x170f0009 --> 0xf00000 [617100.080464] hda_codec:snd_hda_codec_read: 0x171f0009 --> 0x100100 [617100.080548] hda_codec:snd_hda_codec_read: 0x172f0009 --> 0x0 [617100.080625] hda_codec:snd_hda_codec_read: 0x173f0009 --> 0x400100 [617100.080689] hda_codec:snd_hda_codec_read: 0x173f1c00 --> 0x16a10f0 [617100.080756] hda_codec:snd_hda_codec_read: 0x102f2000 --> 0x1025011e [617100.080762] hda_codec:snd_hda_codec_write: 0x10270500 [617100.110915] hda_codec:snd_hda_codec_read: 0x102f0500 --> 0x0 [617100.114318] hda_codec:snd_hda_codec_read: 0x2f000a --> 0xe0560 [617100.114318] hda_codec:snd_hda_codec_read: 0x2f000b --> 0x1 [617100.114318] hda_codec:snd_hda_codec_read: 0x8f000a --> 0x60160 [617100.114318] hda_codec:snd_hda_codec_read: 0x8f000b --> 0x1 [617100.114318] hda_codec:snd_hda_codec_write: 0x17ff00 [617100.114318] hda_codec:snd_hda_codec_write: 0x23b000 [617100.114318] hda_codec:snd_hda_codec_write: 0x33b000 [617100.114318] hda_codec:snd_hda_codec_write: 0xe37000 [617100.114318] hda_codec:snd_hda_codec_write: 0xe70100 [617100.114318] hda_codec:snd_hda_codec_write: 0xf37000 [617100.114318] hda_codec:snd_hda_codec_write: 0x1037000 [617100.114318] hda_codec:snd_hda_codec_write: 0x1470740 [617100.114318] hda_codec:snd_hda_codec_write: 0x15707c0 [617100.114318] hda_codec:snd_hda_codec_write: 0x1670740 [617100.114318] hda_codec:snd_hda_codec_write: 0x1870722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1970722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1a70720 [617100.114318] hda_codec:snd_hda_codec_write: 0x1c70720 [617100.114318] hda_codec:snd_hda_codec_write: 0x1d70720 [617100.114318] hda_codec:snd_hda_codec_write: 0x143b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x153b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x163b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x183b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x1a3b080 [617100.114318] hda_codec:snd_hda_codec_write: 0x1d37000 [617100.114318] hda_codec:snd_hda_codec_write: 0xf37180 [617100.114318] hda_codec:snd_hda_codec_write: 0x1037180 [617100.114318] hda_codec:snd_hda_codec_write: 0x2370100 [617100.114318] hda_codec:snd_hda_codec_write: 0x233b000 [617100.114318] hda_codec:snd_hda_codec_write: 0x2470100 [617100.114318] hda_codec:snd_hda_codec_write: 0x243b000 [617100.114318] hda_codec:snd_hda_codec_write: 0x1470c02 [617100.114318] hda_codec:snd_hda_codec_write: 0x1570c02 [617100.114318] hda_codec:snd_hda_codec_write: 0x1270720 [617100.114318] hda_codec:snd_hda_codec_write: 0x1370720 [617100.114318] hda_codec:snd_hda_codec_write: 0x14707c0 [617100.114318] hda_codec:snd_hda_codec_write: 0x1570740 [617100.114318] hda_codec:snd_hda_codec_write: 0x1870722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1a70722 [617100.114318] hda_codec:snd_hda_codec_write: 0x1470884 [617100.114318] hda_codec:snd_hda_codec_read: 0x14f0900 --> 0x0 [617100.114318] hda_codec:snd_hda_codec_read: 0x14ba000 --> 0x80 [617100.114318] hda_codec:snd_hda_codec_read: 0x15ba000 --> 0x80 [617100.114318] hda_codec:snd_hda_codec_read: 0x15b8000 --> 0x80 [617100.114318] hda_codec:snd_hda_codec_read: 0x2f0012 --> 0x34040 [617100.114318] hda_codec:snd_hda_codec_read: 0x23f0012 --> 0x80051f0b [617100.135224] hda_codec:snd_hda_codec_read: 0x14f0900 --> 0x80000000 [617100.162363] hda_codec:snd_hda_codec_read: 0x18f000d --> 0x4f0200 [617100.162363] hda_codec:snd_hda_codec_read: 0x19f000d --> 0x4f0200 [617100.162363] hda_codec:snd_hda_codec_read: 0x1af000d --> 0x4f0200 [617100.162363] hda_codec:snd_hda_codec_read: 0x2ba000 --> 0x0 [617100.162363] hda_codec:snd_hda_codec_write: 0x23a036 [617100.162363] hda_codec:snd_hda_codec_read: 0x2b8000 --> 0x0 [617100.162363] hda_codec:snd_hda_codec_write: 0x239036 [617100.162363] hda_codec:snd_hda_codec_read: 0x3ba000 --> 0x0 [617100.162363] hda_codec:snd_hda_codec_write: 0x33a036 [617100.162363] hda_codec:snd_hda_codec_read: 0x3b8000 --> 0x0 [617100.162377] hda_codec:snd_hda_codec_write: 0x339036 [617100.162387] hda_codec:snd_hda_codec_write: 0x143a000 [617100.162491] hda_codec:snd_hda_codec_read: 0x14b8000 --> 0x80 [617100.162498] hda_codec:snd_hda_codec_write: 0x1439000 [617100.162557] hda_codec:snd_hda_codec_read: 0x18b2000 --> 0x0 [617100.162621] hda_codec:snd_hda_codec_read: 0x18b0000 --> 0x0 [617100.162702] hda_codec:snd_hda_codec_read: 0x19b2000 --> 0x0 [617100.162789] hda_codec:snd_hda_codec_read: 0x19b0000 --> 0x0 [617100.162869] hda_codec:snd_hda_codec_read: 0x1ab2000 --> 0x0 [617100.162955] hda_codec:snd_hda_codec_read: 0x1ab0000 --> 0x0 [617100.163035] hda_codec:snd_hda_codec_read: 0x23ba000 --> 0x0 [617100.163122] hda_codec:snd_hda_codec_read: 0x23b8000 --> 0x0 [617100.163128] hda_codec:snd_hda_codec_write: 0x2339006 [617100.163136] hda_codec:snd_hda_codec_write: 0x233a080 [617100.163239] hda_codec:snd_hda_codec_read: 0x1db2000 --> 0x0 [617100.163306] hda_codec:snd_hda_codec_read: 0x1db0000 --> 0x0 [617100.163385] hda_codec:snd_hda_codec_read: 0xfb2001 --> 0x80 [617100.163452] hda_codec:snd_hda_codec_read: 0xfb0001 --> 0x80 [617100.163538] hda_codec:snd_hda_codec_read: 0x10b2001 --> 0x80 [617100.163615] hda_codec:snd_hda_codec_read: 0x10b0001 --> 0x80 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f000a --> 0xe0560 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f000b --> 0x1 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f000d --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f0012 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f0004 --> 0x20023 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f0011 --> 0x40000004 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1600 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1700 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1800 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1900 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1a00 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x1f1500 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f0009 --> 0x1d [617165.253398] hda_codec:snd_hda_codec_read: 0x2f0012 --> 0x34040 [617165.253398] hda_codec:snd_hda_codec_read: 0x2ba000 --> 0x36 [617165.253398] hda_codec:snd_hda_codec_read: 0x2b8000 --> 0x36 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f0600 --> 0x0 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f000a --> 0xe0560 [617165.253398] hda_codec:snd_hda_codec_read: 0x2f000b --> 0x1 [617165.253398] hda_codec:snd_hda_codec_read: 0x3f0009 --> 0x1d [617165.253398] hda_codec:snd_hda_codec_read: 0x3f0012 --> 0x34040 [617165.253398] hda_codec:snd_hda_codec_read: 0x3ba000 --> 0x36 [617165.255878] hda_codec:snd_hda_codec_read: 0x3b8000 --> 0x36 [617165.255942] hda_codec:snd_hda_codec_read: 0x3f0600 --> 0x0 [617165.256009] hda_codec:snd_hda_codec_read: 0x3f000a --> 0xe0560 [617165.256095] hda_codec:snd_hda_codec_read: 0x3f000b --> 0x1 [617165.256174] hda_codec:snd_hda_codec_read: 0x4f0009 --> 0xf00000 [617165.256239] hda_codec:snd_hda_codec_read: 0x5f0009 --> 0xf00000 [617165.256316] hda_codec:snd_hda_codec_read: 0x6f0009 --> 0x211 [617165.256383] hda_codec:snd_hda_codec_read: 0x6f0600 --> 0x0 [617165.256447] hda_codec:snd_hda_codec_read: 0x6f0d00 --> 0x0 [617165.256524] hda_codec:snd_hda_codec_read: 0x6f000a --> 0x1e05e0 [617165.256590] hda_codec:snd_hda_codec_read: 0x6f000b --> 0x1 [617165.256676] hda_codec:snd_hda_codec_read: 0x7f0009 --> 0x100111 [617165.256754] hda_codec:snd_hda_codec_read: 0x7f000e --> 0x1 [617165.256818] hda_codec:snd_hda_codec_read: 0x7f0200 --> 0x24 [617165.256884] hda_codec:snd_hda_codec_read: 0x7f0600 --> 0x0 [617165.256961] hda_codec:snd_hda_codec_read: 0x7f0400 --> 0x0 [617165.257025] hda_codec:snd_hda_codec_read: 0x7f000a --> 0x60160 [617165.257092] hda_codec:snd_hda_codec_read: 0x7f000b --> 0x1 [617165.257171] hda_codec:snd_hda_codec_read: 0x8f0009 --> 0x100111 [617165.257235] hda_codec:snd_hda_codec_read: 0x8f000e --> 0x1 [617165.257321] hda_codec:snd_hda_codec_read: 0x8f0200 --> 0x23 [617165.257398] hda_codec:snd_hda_codec_read: 0x8f0600 --> 0x0 [617165.257464] hda_codec:snd_hda_codec_read: 0x8f0400 --> 0x0 [617165.257529] hda_codec:snd_hda_codec_read: 0x8f000a --> 0x60160 [617165.257605] hda_codec:snd_hda_codec_read: 0x8f000b --> 0x1 [617165.257674] hda_codec:snd_hda_codec_read: 0x9f0009 --> 0xf00000 [617165.257759] hda_codec:snd_hda_codec_read: 0xaf0009 --> 0xf00000 [617165.257836] hda_codec:snd_hda_codec_read: 0xbf0009 --> 0xf00000 [617165.257903] hda_codec:snd_hda_codec_read: 0xcf0009 --> 0xf00000 [617165.257980] hda_codec:snd_hda_codec_read: 0xdf0009 --> 0xf00000 [617165.258047] hda_codec:snd_hda_codec_read: 0xef0009 --> 0x20010a [617165.258112] hda_codec:snd_hda_codec_read: 0xef000e --> 0x1 [617165.258188] hda_codec:snd_hda_codec_read: 0xef0200 --> 0x2 [617165.258255] hda_codec:snd_hda_codec_read: 0xef000d --> 0x80000000 [617165.258320] hda_codec:snd_hda_codec_read: 0xeb0000 --> 0x0 [617165.258397] hda_codec:snd_hda_codec_read: 0xff0009 --> 0x20010b [617165.258464] hda_codec:snd_hda_codec_read: 0xff000e --> 0x2 [617165.258528] hda_codec:snd_hda_codec_read: 0xff0200 --> 0x1d02 [617165.258605] hda_codec:snd_hda_codec_read: 0xff000d --> 0x80000000 [617165.258672] hda_codec:snd_hda_codec_read: 0xfb2000 --> 0x0 [617165.258756] hda_codec:snd_hda_codec_read: 0xfb0000 --> 0x0 [617165.259833] hda_codec:snd_hda_codec_read: 0xfb2001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0xfb0001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10f0009 --> 0x20010b [617165.259833] hda_codec:snd_hda_codec_read: 0x10f000e --> 0x3 [617165.259833] hda_codec:snd_hda_codec_read: 0x10f0200 --> 0x21d03 [617165.259833] hda_codec:snd_hda_codec_read: 0x10f000d --> 0x80000000 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b2000 --> 0x0 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b0000 --> 0x0 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b2001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b0001 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b2002 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x10b0002 --> 0x80 [617165.259833] hda_codec:snd_hda_codec_read: 0x11f0009 --> 0xf00000 [617165.259833] hda_codec:snd_hda_codec_read: 0x12f0009 --> 0x400001 [617165.259868] hda_codec:snd_hda_codec_read: 0x12f000c --> 0x20 [617165.259945] hda_codec:snd_hda_codec_read: 0x12f1c00 --> 0x411111f0 [617165.260012] hda_codec:snd_hda_codec_read: 0x12f0700 --> 0x20 [617165.260089] hda_codec:snd_hda_codec_read: 0x13f0009 --> 0x400001 [617165.260155] hda_codec:snd_hda_codec_read: 0x13f000c --> 0x20 [617165.260220] hda_codec:snd_hda_codec_read: 0x13f1c00 --> 0x411111f0 [617165.260298] hda_codec:snd_hda_codec_read: 0x13f0700 --> 0x20 [617165.260364] hda_codec:snd_hda_codec_read: 0x14f0009 --> 0x40018d [617165.260449] hda_codec:snd_hda_codec_read: 0x14f000e --> 0x1 [617165.260526] hda_codec:snd_hda_codec_read: 0x14f0200 --> 0xf [617165.260590] hda_codec:snd_hda_codec_read: 0x14f0012 --> 0x80000000 [617165.260657] hda_codec:snd_hda_codec_read: 0x14ba000 --> 0x0 [617165.260734] hda_codec:snd_hda_codec_read: 0x14b8000 --> 0x0 [617165.260798] hda_codec:snd_hda_codec_read: 0x14f000c --> 0x1003c [617165.260865] hda_codec:snd_hda_codec_read: 0x14f0c00 --> 0x2 [617165.260949] hda_codec:snd_hda_codec_read: 0x14f1c00 --> 0x2211020 [617165.261027] hda_codec:snd_hda_codec_read: 0x14f0700 --> 0xc0 [617165.261094] hda_codec:snd_hda_codec_read: 0x14f0800 --> 0x84 [617165.261179] hda_codec:snd_hda_codec_read: 0x15f0009 --> 0x40018d [617165.261256] hda_codec:snd_hda_codec_read: 0x15f000e --> 0x1 [617165.261322] hda_codec:snd_hda_codec_read: 0x15f0200 --> 0x10 [617165.261399] hda_codec:snd_hda_codec_read: 0x15f0012 --> 0x80000000 [617165.261464] hda_codec:snd_hda_codec_read: 0x15ba000 --> 0x80 [617165.261530] hda_codec:snd_hda_codec_read: 0x15b8000 --> 0x80 [617165.261615] hda_codec:snd_hda_codec_read: 0x15f000c --> 0x1003c [617165.261692] hda_codec:snd_hda_codec_read: 0x15f0c00 --> 0x2 [617165.261756] hda_codec:snd_hda_codec_read: 0x15f1c00 --> 0x99130110 [617165.261845] hda_codec:snd_hda_codec_read: 0x15f0700 --> 0x40 [617165.261921] hda_codec:snd_hda_codec_read: 0x15f0800 --> 0x0 [617165.262004] hda_codec:snd_hda_codec_read: 0x16f0009 --> 0x40010c [617165.262071] hda_codec:snd_hda_codec_read: 0x16f000e --> 0x1 [617165.262156] hda_codec:snd_hda_codec_read: 0x16f0200 --> 0xe [617165.262232] hda_codec:snd_hda_codec_read: 0x16f0012 --> 0x80000000 [617165.262299] hda_codec:snd_hda_codec_read: 0x16b8000 --> 0x80 [617165.262386] hda_codec:snd_hda_codec_read: 0x16f000c --> 0x10 [617165.262462] hda_codec:snd_hda_codec_read: 0x16f1c00 --> 0x411111f0 [617165.262551] hda_codec:snd_hda_codec_read: 0x16f0700 --> 0x40 [617165.262628] hda_codec:snd_hda_codec_read: 0x17f0009 --> 0xf00000 [617165.262693] hda_codec:snd_hda_codec_read: 0x18f0009 --> 0x40018f [617165.262780] hda_codec:snd_hda_codec_read: 0x18f000e --> 0x1 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0200 --> 0x2 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f000d --> 0x4f0200 [617165.263834] hda_codec:snd_hda_codec_read: 0x18b2000 --> 0x0 [617165.263834] hda_codec:snd_hda_codec_read: 0x18b0000 --> 0x0 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0012 --> 0x80000000 [617165.263834] hda_codec:snd_hda_codec_read: 0x18ba000 --> 0x80 [617165.263834] hda_codec:snd_hda_codec_read: 0x18b8000 --> 0x80 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f000c --> 0x3734 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f1c00 --> 0x2a19840 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0700 --> 0x22 [617165.263834] hda_codec:snd_hda_codec_read: 0x18f0800 --> 0x0 [617165.263834] hda_codec:snd_hda_codec_read: 0x19f0009 --> 0x40008b [617165.263834] hda_codec:snd_hda_codec_read: 0x19f000d --> 0x4f0200 [617165.263838] hda_codec:snd_hda_codec_read: 0x19b2000 --> 0x0 [617165.263904] hda_codec:snd_hda_codec_read: 0x19b0000 --> 0x0 [617165.263969] hda_codec:snd_hda_codec_read: 0x19f000c --> 0x3724 [617165.264046] hda_codec:snd_hda_codec_read: 0x19f1c00 --> 0x99a30941 [617165.264114] hda_codec:snd_hda_codec_read: 0x19f0700 --> 0x22 [617165.264199] hda_codec:snd_hda_codec_read: 0x19f0800 --> 0x0 [617165.264276] hda_codec:snd_hda_codec_read: 0x1af0009 --> 0x40018f [617165.264343] hda_codec:snd_hda_codec_read: 0x1af000e --> 0x1 [617165.264429] hda_codec:snd_hda_codec_read: 0x1af0200 --> 0x2 [617165.264505] hda_codec:snd_hda_codec_read: 0x1af000d --> 0x4f0200 [617165.264570] hda_codec:snd_hda_codec_read: 0x1ab2000 --> 0x0 [617165.264657] hda_codec:snd_hda_codec_read: 0x1ab0000 --> 0x0 [617165.264733] hda_codec:snd_hda_codec_read: 0x1af0012 --> 0x80000000 [617165.264800] hda_codec:snd_hda_codec_read: 0x1aba000 --> 0x80 [617165.264884] hda_codec:snd_hda_codec_read: 0x1ab8000 --> 0x80 [617165.264961] hda_codec:snd_hda_codec_read: 0x1af000c --> 0x3734 [617165.265026] hda_codec:snd_hda_codec_read: 0x1af1c00 --> 0x281304e [617165.265114] hda_codec:snd_hda_codec_read: 0x1af0700 --> 0x22 [617165.265191] hda_codec:snd_hda_codec_read: 0x1af0800 --> 0x0 [617165.265258] hda_codec:snd_hda_codec_read: 0x1bf0009 --> 0xf00000 [617165.265335] hda_codec:snd_hda_codec_read: 0x1cf0009 --> 0x400001 [617165.265400] hda_codec:snd_hda_codec_read: 0x1cf000c --> 0x20 [617165.265467] hda_codec:snd_hda_codec_read: 0x1cf1c00 --> 0x411111f0 [617165.265545] hda_codec:snd_hda_codec_read: 0x1cf0700 --> 0x20 [617165.265610] hda_codec:snd_hda_codec_read: 0x1df0009 --> 0x400000 [617165.265697] hda_codec:snd_hda_codec_read: 0x1df000c --> 0x20 [617165.265773] hda_codec:snd_hda_codec_read: 0x1df1c00 --> 0x4017952d [617165.265842] hda_codec:snd_hda_codec_read: 0x1df0700 --> 0x20 [617165.265927] hda_codec:snd_hda_codec_read: 0x1ef0009 --> 0x400380 [617165.266003] hda_codec:snd_hda_codec_read: 0x1ef000e --> 0x1 [617165.266070] hda_codec:snd_hda_codec_read: 0x1ef0200 --> 0x6 [617165.266146] hda_codec:snd_hda_codec_read: 0x1ef000c --> 0x10 [617165.266210] hda_codec:snd_hda_codec_read: 0x1ef1c00 --> 0x2451130 [617165.266278] hda_codec:snd_hda_codec_read: 0x1ef0700 --> 0x40 [617165.266355] hda_codec:snd_hda_codec_read: 0x1ef0800 --> 0x0 [617165.266420] hda_codec:snd_hda_codec_read: 0x1ff0009 --> 0xf00000 [617165.266507] hda_codec:snd_hda_codec_read: 0x20f0009 --> 0xf00040 [617165.266584] hda_codec:snd_hda_codec_read: 0x20f0010 --> 0xa00 [617165.266651] hda_codec:snd_hda_codec_read: 0x20c0000 --> 0x3 [617165.266738] hda_codec:snd_hda_codec_read: 0x20d0000 --> 0x1 [617165.266815] hda_codec:snd_hda_codec_read: 0x21f0009 --> 0xf00000 [617165.267835] hda_codec:snd_hda_codec_read: 0x22f0009 --> 0xf00000 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0009 --> 0x30010d [617165.267835] hda_codec:snd_hda_codec_read: 0x23f000e --> 0x7 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0200 --> 0x1c1a1918 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0204 --> 0x121514 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0012 --> 0x80051f0b [617165.267835] hda_codec:snd_hda_codec_read: 0x23ba000 --> 0x80 [617165.267835] hda_codec:snd_hda_codec_read: 0x23b8000 --> 0x6 [617165.267835] hda_codec:snd_hda_codec_read: 0x23f0100 --> 0x0 [617165.267835] hda_codec:snd_hda_codec_read: 0x24f0009 --> 0x30010d [617165.267835] hda_codec:snd_hda_codec_read: 0x24f000e --> 0x7 [617165.267835] hda_codec:snd_hda_codec_read: 0x24f0200 --> 0x1c1a1918 [617165.267835] hda_codec:snd_hda_codec_read: 0x24f0204 --> 0x131514 [617165.267870] hda_codec:snd_hda_codec_read: 0x24f0012 --> 0x80051f0b [617165.267947] hda_codec:snd_hda_codec_read: 0x24ba000 --> 0x0 [617165.268012] hda_codec:snd_hda_codec_read: 0x24b8000 --> 0x0 [617165.268098] hda_codec:snd_hda_codec_read: 0x24f0100 --> 0x0 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) Subsystem: Acer Incorporated [ALI] Device [1025:011e] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 11 Region 0: Memory at 9b300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

2 1

[alsa-devel] [PATCH v3 4/4] sh: add FSI driver support for ms7724se
by Kuninori Morimoto 21 Aug '09

21 Aug '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o remove un-used dma-sh.h I used Paul's git for base and merge from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 arch/sh/boards/mach-se/7724/setup.c | 89 +++++++++++++++++++++++++++++++++++ 1 files changed, 89 insertions(+), 0 deletions(-) diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 9162081..eb144f6 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -22,6 +22,7 @@ #include <linux/usb/r8a66597.h> #include <video/sh_mobile_lcdc.h> #include <media/sh_mobile_ceu.h> +#include <sound/sh_fsi.h> #include <asm/io.h> #include <asm/heartbeat.h> #include <asm/sh_eth.h> @@ -246,6 +247,65 @@ static struct platform_device ceu1_device = { }, }; +/* FSI */ +/* + * FSI-A use external clock which came from ak464x. + * So, we should change parent of fsi + */ +#define FCLKACR 0xa4150008 +static void fsimck_init(struct clk *clk) +{ + u32 status = ctrl_inl(clk->enable_reg); + + /* use external clock */ + status &= ~0x000000ff; + status |= 0x00000080; + ctrl_outl(status, clk->enable_reg); +} + +static struct clk_ops fsimck_clk_ops = { + .init = fsimck_init, +}; + +static struct clk fsimcka_clk = { + .name = "fsimcka_clk", + .id = -1, + .ops = &fsimck_clk_ops, + .enable_reg = (void __iomem *)FCLKACR, + .rate = 0, /* unknown */ +}; + +struct sh_fsi_platform_info fsi_info = { + .porta_flags = SH_FSI_BRS_INV | + SH_FSI_OUT_SLAVE_MODE | + SH_FSI_IN_SLAVE_MODE | + SH_FSI_OFMT(PCM) | + SH_FSI_IFMT(PCM), +}; + +static struct resource fsi_resources[] = { + [0] = { + .name = "FSI", + .start = 0xFE3C0000, + .end = 0xFE3C021d, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = 108, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device fsi_device = { + .name = "sh_fsi", + .id = 0, + .num_resources = ARRAY_SIZE(fsi_resources), + .resource = fsi_resources, + .dev = { + .platform_data = &fsi_info, + }, +}; + /* KEYSC in SoC (Needs SW33-2 set to ON) */ static struct sh_keysc_info keysc_info = { .mode = SH_KEYSC_MODE_1, @@ -351,6 +411,7 @@ static struct platform_device *ms7724se_devices[] __initdata = { &keysc_device, &sh_eth_device, &sh7724_usb0_host_device, + &fsi_device, }; #define EEPROM_OP 0xBA206000 @@ -418,11 +479,13 @@ static void __init sh_eth_init(void) static int __init devices_setup(void) { u16 sw = ctrl_inw(SW4140); /* select camera, monitor */ + struct clk *fsia_clk; /* Reset Release */ ctrl_outw(ctrl_inw(FPGA_OUT) & ~((1 << 1) | /* LAN */ (1 << 6) | /* VIDEO DAC */ + (1 << 7) | /* AK4643 */ (1 << 12) | /* USB0 */ (1 << 14)), /* RMII */ FPGA_OUT); @@ -558,6 +621,32 @@ static int __init devices_setup(void) gpio_request(GPIO_FN_KEYOUT1, NULL); gpio_request(GPIO_FN_KEYOUT0, NULL); + /* enable FSI */ + gpio_request(GPIO_FN_FSIMCKB, NULL); + gpio_request(GPIO_FN_FSIMCKA, NULL); + gpio_request(GPIO_FN_FSIOASD, NULL); + gpio_request(GPIO_FN_FSIIABCK, NULL); + gpio_request(GPIO_FN_FSIIALRCK, NULL); + gpio_request(GPIO_FN_FSIOABCK, NULL); + gpio_request(GPIO_FN_FSIOALRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOAO, NULL); + gpio_request(GPIO_FN_FSIIBSD, NULL); + gpio_request(GPIO_FN_FSIOBSD, NULL); + gpio_request(GPIO_FN_FSIIBBCK, NULL); + gpio_request(GPIO_FN_FSIIBLRCK, NULL); + gpio_request(GPIO_FN_FSIOBBCK, NULL); + gpio_request(GPIO_FN_FSIOBLRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOBO, NULL); + gpio_request(GPIO_FN_FSIIASD, NULL); + + /* change parent of FSI A */ + fsia_clk = clk_get(NULL, "fsia_clk"); + clk_register(&fsimcka_clk); + clk_set_parent(fsia_clk, &fsimcka_clk); + clk_set_rate(fsia_clk, 11000); + clk_set_rate(&fsimcka_clk, 11000); + clk_put(fsia_clk); + /* * enable SH-Eth * -- 1.6.0.4

2 1

[alsa-devel] [PATCH v3 3/4] ASoC: Add FSI-AK4642 sound support for SuperH
by Kuninori Morimoto 21 Aug '09

21 Aug '09

This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o remove un-used ak4642_setup_data I used Paul's git for base and merge from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 sound/soc/sh/Kconfig | 8 +++ sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-ak4642.c | 107 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 117 insertions(+), 0 deletions(-) create mode 100644 sound/soc/sh/fsi-ak4642.c diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 01943a1..9154b43 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -40,4 +40,12 @@ config SND_SH7760_AC97 This option enables generic sound support for the first AC97 unit of the SH7760. +config SND_FSI_AK4642 + bool "FSI-AK4642 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_AK4642 + help + This option enables generic sound support for the + FSI - AK4642 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 9fbcc4a..a699787 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -12,5 +12,7 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o +snd-soc-fsi-ak4642-objs := fsi-ak4642.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o +obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c new file mode 100644 index 0000000..c7af097 --- /dev/null +++ b/sound/soc/sh/fsi-ak4642.c @@ -0,0 +1,107 @@ +/* + * FSI-AK464x sound support for ms7724se + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <sound/sh_fsi.h> +#include <../sound/soc/codecs/ak4642.h> + +static struct snd_soc_dai_link fsi_dai_link = { + .name = "AK4642", + .stream_name = "AK4642", + .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .codec_dai = &ak4642_dai, + .ops = NULL, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_dai_link, + .num_links = 1, +}; + +static struct snd_soc_device fsi_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_ak4642, +}; + +#define AK4642_BUS 0 +#define AK4642_ADR 0x12 +static int ak4642_add_i2c_device(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = AK4642_ADR; + strlcpy(info.type, "ak4642", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(AK4642_BUS); + if (!adapter) { + printk(KERN_DEBUG "can't get i2c adapter\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_DEBUG "can't add i2c device\n"); + return -ENODEV; + } + + return 0; +} + +static struct platform_device *fsi_snd_device; + +static int __init fsi_ak4642_init(void) +{ + int ret = -ENOMEM; + + ak4642_add_i2c_device(); + + fsi_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_snd_device) + goto out; + + platform_set_drvdata(fsi_snd_device, + &fsi_snd_devdata); + fsi_snd_devdata.dev = &fsi_snd_device->dev; + ret = platform_device_add(fsi_snd_device); + + if (ret) + platform_device_put(fsi_snd_device); + +out: + return ret; +} + +static void __exit fsi_ak4642_exit(void) +{ + platform_device_unregister(fsi_snd_device); +} + +module_init(fsi_ak4642_init); +module_exit(fsi_ak4642_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.6.0.4

2 1

[alsa-devel] [PATCH v3 2/4] ASoC: Add ak4642/ak4643 codec support
by Kuninori Morimoto 21 Aug '09

21 Aug '09

This is very simple driver for ALSA It supprt headphone output and stereo input only This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v2 -> v3 o add ak4643 in i2c table o remove un-used hw_read o remove un-used ak4642_setup_data I used Paul's git for base and merge from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 502 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4642.h | 20 ++ 4 files changed, 528 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4642.c create mode 100644 sound/soc/codecs/ak4642.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 167a5ce..0edca93 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4642 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -92,6 +93,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4642 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fbab43b..fb4af28 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4642-objs := ak4642.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -54,6 +55,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c new file mode 100644 index 0000000..e057c7b --- /dev/null +++ b/sound/soc/codecs/ak4642.c @@ -0,0 +1,502 @@ +/* + * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * 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. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It can use headphone output / stereo input only + * + * AK4642 is not tested. + * AK4643 is tested. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ak4642.h" + +#define AK4642_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK4642_CACHEREGNUM 0x25 + +struct snd_soc_codec_device soc_codec_dev_ak4642; + +/* codec private data */ +struct ak4642_priv { + struct snd_soc_codec codec; + unsigned int sysclk; +}; + +static struct snd_soc_codec *ak4642_codec; + +/* + * ak4642 register cache + */ +static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak4642 register cache + */ +static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write ak4642 register cache + */ +static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK4642 register space + */ +static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4642 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak4642_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak4642_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + r |= ak4642_write(codec, i, cache[i]); + + return r; +}; + +static int ak4642_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* + * start headphone output + * + * PLL, Master Mode + * Audio I/F Format :MSB justified (ADC & DAC) + * Sampling Frequency: 44.1kHz + * Digital Volume: −8dB + * Bass Boost Level : Middle + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p97. + * + * Example code use 0x39, 0x79 value for 0x01 address, + * But we need MCKO (0x02) bit now + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x0f, 0x09); + ak4642_write(codec, 0x0e, 0x19); + ak4642_write(codec, 0x09, 0x91); + ak4642_write(codec, 0x0c, 0x91); + ak4642_write(codec, 0x0a, 0x28); + ak4642_write(codec, 0x0d, 0x28); + ak4642_write(codec, 0x00, 0x64); + ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ + ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + } else { + /* + * start stereo input + * + * PLL Master Mode + * Audio I/F Format:MSB justified (ADC & DAC) + * Sampling Frequency:44.1kHz + * Pre MIC AMP:+20dB + * MIC Power On + * ALC setting:Refer to Table 35 + * ALC bit=“1” + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p94. + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x02, 0x05); + ak4642_write(codec, 0x06, 0x3c); + ak4642_write(codec, 0x08, 0xe1); + ak4642_write(codec, 0x0b, 0x00); + ak4642_write(codec, 0x07, 0x21); + ak4642_write(codec, 0x00, 0x41); + ak4642_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak4642_write(codec, 0x01, 0x3b); + ak4642_write(codec, 0x01, 0x0b); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x0e, 0x11); + ak4642_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x10, 0x00); + ak4642_write(codec, 0x07, 0x01); + } +} + +static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4642_priv *ak4642 = codec->private_data; + + ak4642->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak4642_dai_ops = { + .startup = ak4642_dai_startup, + .shutdown = ak4642_dai_shutdown, + .set_sysclk = ak4642_dai_set_sysclk, +}; + +struct snd_soc_dai ak4642_dai = { + .name = "AK4642", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .ops = &ak4642_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4642_dai); + +static int ak4642_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak4642_sync(codec); + return 0; +} + +/* + * initialise the AK4642 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4642_init(struct ak4642_priv *ak4642) +{ + struct snd_soc_codec *codec = &ak4642->codec; + int ret = 0; + + if (ak4642_codec) { + dev_err(codec->dev, "Another ak4642 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4642; + codec->name = "AK4642"; + codec->owner = THIS_MODULE; + codec->read = ak4642_read_reg_cache; + codec->write = ak4642_write; + codec->dai = &ak4642_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); + codec->reg_cache = kmemdup(ak4642_reg, + sizeof(ak4642_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + ak4642_dai.dev = codec->dev; + ak4642_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto reg_cache_err; + } + + ret = snd_soc_register_dai(&ak4642_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto reg_cache_err; + } + + /* + * clock setting + * + * Audio I/F Format: MSB justified (ADC & DAC) + * BICK frequency at Master Mode: 64fs + * Input Master Clock Select at PLL Mode: 11.2896MHz + * MCKO: Enable + * Sampling Frequency: 44.1kHz + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p89. + * + * please fix-me + */ + ak4642_write(codec, 0x01, 0x08); + ak4642_write(codec, 0x04, 0x4a); + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x01, 0x0b); + + return ret; + +reg_cache_err: + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + return ret; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ak4642_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4642_priv *ak4642; + struct snd_soc_codec *codec; + int ret; + + ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); + if (!ak4642) + return -ENOMEM; + + codec = &ak4642->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, ak4642); + codec->control_data = i2c; + + ret = ak4642_init(ak4642); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK4642\n"); + + return ret; +} + +static int ak4642_i2c_remove(struct i2c_client *client) +{ + struct ak4642_priv *ak4642 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&ak4642_dai); + snd_soc_unregister_codec(&ak4642->codec); + kfree(ak4642->codec.reg_cache); + kfree(ak4642); + ak4642_codec = NULL; + + return 0; +} + +static const struct i2c_device_id ak4642_i2c_id[] = { + { "ak4642", 0 }, + { "ak4643", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); + +static struct i2c_driver ak4642_i2c_driver = { + .driver = { + .name = "AK4642 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak4642_i2c_probe, + .remove = ak4642_i2c_remove, + .id_table = ak4642_i2c_id, +}; + +#endif + +static int ak4642_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret; + + if (!ak4642_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4642_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to register card\n"); + goto card_err; + } + + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; + +} + +/* power down chip */ +static int ak4642_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4642 = { + .probe = ak4642_probe, + .remove = ak4642_remove, + .resume = ak4642_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); + +static int __init ak4642_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&ak4642_i2c_driver); +#endif + return ret; + +} +module_init(ak4642_modinit); + +static void __exit ak4642_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4642_i2c_driver); +#endif + +} +module_exit(ak4642_exit); + +MODULE_DESCRIPTION("Soc AK4642 driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h new file mode 100644 index 0000000..e476833 --- /dev/null +++ b/sound/soc/codecs/ak4642.h @@ -0,0 +1,20 @@ +/* + * ak4642.h -- AK4642 Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on ak4535.c + * + * 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 _AK4642_H +#define _AK4642_H + +extern struct snd_soc_dai ak4642_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4642; + +#endif -- 1.6.0.4

2 1

[alsa-devel] [PATCH] pyalsa: fix integer overflow in alsaseq.c
by Chris Coleman 21 Aug '09

21 Aug '09

Hello alsa-devel, I've been using the alsaseq python module and I found a bug. Sometimes the SEQ_* constants have extremely large and incorrect values. For example, 25769803811 instead of 35. The lower 32-bits are always correct. Obviously, I'm running a 64-bit operating system. The problem is that the `value` member of the `ConstantObject` structure is an `unsigned int` whereas it should be a `long`. I've attached a patch. It's against the latest released version, 1.0.20. Regards, Chris Coleman Signed-off-by: Chris Coleman <chris.coleman83 at gmail.com>

2 3

[alsa-devel] [patch 0/4] ASoC S3C24XX/Simtec support
by Ben Dooks 21 Aug '09

21 Aug '09

re-worked against ASoC git branch Will be away from until 26th, any comments will be sorted when back. --

2 6

[alsa-devel] Need help compiling alsa-kernel from git
by Damien Zammit 21 Aug '09

21 Aug '09

Hi, I downloaded a 2.6.30.4 vanilla kernel source tree, merged it with alsa 1.0.20 by copying the files into my kernel source tree, and used that as a base for modifying a sound driver module. I want to submit a patch because I got a new device working. I cloned alsa-kernel git and merged my changes into the tree and committed it. Now, how do I compile it to test it? Do I need to clone linux-2.6 git? I would prefer not to clone that whole tree if possible... I tried copying the sound directory from my working alsa-kernel git tree into a 2.6.30.4 vanilla kernel tree, but it doesnt work. Please help.

1 0

[alsa-devel] [PATCH v2 4/4] sh: add FSI driver support for ms7724se
by Kuninori Morimoto 21 Aug '09

21 Aug '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o become independent patch arch/sh/boards/mach-se/7724/setup.c | 90 +++++++++++++++++++++++++++++++++++ 1 files changed, 90 insertions(+), 0 deletions(-) diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 9162081..bffc90e 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -22,11 +22,13 @@ #include <linux/usb/r8a66597.h> #include <video/sh_mobile_lcdc.h> #include <media/sh_mobile_ceu.h> +#include <sound/sh_fsi.h> #include <asm/io.h> #include <asm/heartbeat.h> #include <asm/sh_eth.h> #include <asm/clock.h> #include <asm/sh_keysc.h> +#include <asm/dma-sh.h> #include <cpu/sh7724.h> #include <mach-se/mach/se7724.h> @@ -246,6 +248,65 @@ static struct platform_device ceu1_device = { }, }; +/* FSI */ +/* + * FSI-A use external clock which came from ak464x. + * So, we should change parent of fsi + */ +#define FCLKACR 0xa4150008 +static void fsimck_init(struct clk *clk) +{ + u32 status = ctrl_inl(clk->enable_reg); + + /* use external clock */ + status &= ~0x000000ff; + status |= 0x00000080; + ctrl_outl(status, clk->enable_reg); +} + +static struct clk_ops fsimck_clk_ops = { + .init = fsimck_init, +}; + +static struct clk fsimcka_clk = { + .name = "fsimcka_clk", + .id = -1, + .ops = &fsimck_clk_ops, + .enable_reg = (void __iomem *)FCLKACR, + .rate = 0, /* unknown */ +}; + +struct sh_fsi_platform_info fsi_info = { + .porta_flags = SH_FSI_BRS_INV | + SH_FSI_OUT_SLAVE_MODE | + SH_FSI_IN_SLAVE_MODE | + SH_FSI_OFMT(PCM) | + SH_FSI_IFMT(PCM), +}; + +static struct resource fsi_resources[] = { + [0] = { + .name = "FSI", + .start = 0xFE3C0000, + .end = 0xFE3C021d, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = 108, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device fsi_device = { + .name = "sh_fsi", + .id = 0, + .num_resources = ARRAY_SIZE(fsi_resources), + .resource = fsi_resources, + .dev = { + .platform_data = &fsi_info, + }, +}; + /* KEYSC in SoC (Needs SW33-2 set to ON) */ static struct sh_keysc_info keysc_info = { .mode = SH_KEYSC_MODE_1, @@ -351,6 +412,7 @@ static struct platform_device *ms7724se_devices[] __initdata = { &keysc_device, &sh_eth_device, &sh7724_usb0_host_device, + &fsi_device, }; #define EEPROM_OP 0xBA206000 @@ -418,11 +480,13 @@ static void __init sh_eth_init(void) static int __init devices_setup(void) { u16 sw = ctrl_inw(SW4140); /* select camera, monitor */ + struct clk *fsia_clk; /* Reset Release */ ctrl_outw(ctrl_inw(FPGA_OUT) & ~((1 << 1) | /* LAN */ (1 << 6) | /* VIDEO DAC */ + (1 << 7) | /* AK4643 */ (1 << 12) | /* USB0 */ (1 << 14)), /* RMII */ FPGA_OUT); @@ -558,6 +622,32 @@ static int __init devices_setup(void) gpio_request(GPIO_FN_KEYOUT1, NULL); gpio_request(GPIO_FN_KEYOUT0, NULL); + /* enable FSI */ + gpio_request(GPIO_FN_FSIMCKB, NULL); + gpio_request(GPIO_FN_FSIMCKA, NULL); + gpio_request(GPIO_FN_FSIOASD, NULL); + gpio_request(GPIO_FN_FSIIABCK, NULL); + gpio_request(GPIO_FN_FSIIALRCK, NULL); + gpio_request(GPIO_FN_FSIOABCK, NULL); + gpio_request(GPIO_FN_FSIOALRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOAO, NULL); + gpio_request(GPIO_FN_FSIIBSD, NULL); + gpio_request(GPIO_FN_FSIOBSD, NULL); + gpio_request(GPIO_FN_FSIIBBCK, NULL); + gpio_request(GPIO_FN_FSIIBLRCK, NULL); + gpio_request(GPIO_FN_FSIOBBCK, NULL); + gpio_request(GPIO_FN_FSIOBLRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOBO, NULL); + gpio_request(GPIO_FN_FSIIASD, NULL); + + /* change parent of FSI A */ + fsia_clk = clk_get(NULL, "fsia_clk"); + clk_register(&fsimcka_clk); + clk_set_parent(fsia_clk, &fsimcka_clk); + clk_set_rate(fsia_clk, 11000); + clk_set_rate(&fsimcka_clk, 11000); + clk_put(fsia_clk); + /* * enable SH-Eth * -- 1.6.0.4

3 3

[alsa-devel] Making only some targets on cross-compilation
by Philip A. Prindeville 21 Aug '09

21 Aug '09

Hi. I'm working on the astlinux project (www.astlinux.org) and had a question about doing a build for only a subset of the $SUBDIRS targets. I tried doing something like: make -C $(ALSA-LIB_DIR) \ SUBDIRS='src include modules aserver' but this fails. The problem is that if we look into the environment in the top-level $(RECURSIVE_TARGETS): we get: SUBDIRS=include src modules aserver ... MAKEFLAGS=w --jobserver-fds=3,4 -j -- SUBDIRS=include\ src\ modules\ aserver CC=/home/philipp/asterisk/build_i586/staging_dir/bin/i586-linux-uclibc-gcc so those targets eventually get picked up by the make running in include/, which then overrides his own Makefile value: SUBDIRS = sound and tries to build the same top-level targets now in include/. Clearly this is wrong. Anyone have a work-around to this? We're on an embedded system, and don't want to build (nor install, obviously) "doc", "test", "alsalisp", "utils", etc. Having something like: SUBDIRS=$(filter-out $(SUBDIRS_EXCLUDE),$(SUBDIRS)) would be really sweet. Then I could just build as: make -C $(ALSA-LIB_DIR) \ all SUBDIRS_EXCLUDE='doc test utils' and only get the targets I want... Thanks, -Philip

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 21 Aug '09

21 Aug '09

The following changes since commit 474e09ca017bd0f6e9bf41af643364a21bc27f43: Mark Brown (1): ASoC: Provide default set_bias_level() implementation are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Kuninori Morimoto (1): ASoC: Add SuperH FSI driver support for ALSA Mark Brown (1): ALSA: Restore support for DMAless DAIs on PXA Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1, 3, 4, 5 OMAP: McBSP: Add link DMA mode selection Shine Liu (1): ASoC: S3C24XX : Align the peroid size to the buffer size arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++ arch/arm/plat-omap/mcbsp.c | 377 ++++++++++++- include/sound/sh_fsi.h | 83 +++ sound/arm/pxa2xx-pcm-lib.c | 3 + sound/soc/omap/omap-mcbsp.c | 77 +++- sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-pcm.h | 2 + sound/soc/s3c24xx/s3c24xx-pcm.c | 1 + sound/soc/sh/Kconfig | 7 +- sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi.c | 1004 +++++++++++++++++++++++++++++++ 12 files changed, 1610 insertions(+), 14 deletions(-) create mode 100644 include/sound/sh_fsi.h create mode 100644 sound/soc/sh/fsi.c

2 1

[alsa-devel] [PATCH v2 3/4] Add FSI-AK4642 sound support for SuperH
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o move to sound/soc/sh sound/soc/sh/Kconfig | 8 +++ sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-ak4642.c | 111 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 121 insertions(+), 0 deletions(-) create mode 100644 sound/soc/sh/fsi-ak4642.c diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 01943a1..9154b43 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -40,4 +40,12 @@ config SND_SH7760_AC97 This option enables generic sound support for the first AC97 unit of the SH7760. +config SND_FSI_AK4642 + bool "FSI-AK4642 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_AK4642 + help + This option enables generic sound support for the + FSI - AK4642 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 9fbcc4a..a699787 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -12,5 +12,7 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o +snd-soc-fsi-ak4642-objs := fsi-ak4642.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o +obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c new file mode 100644 index 0000000..3e5498e --- /dev/null +++ b/sound/soc/sh/fsi-ak4642.c @@ -0,0 +1,111 @@ +/* + * FSI-AK464x sound support for ms7724se + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <sound/sh_fsi.h> +#include <../sound/soc/codecs/ak4642.h> + +static struct snd_soc_dai_link fsi_dai_link = { + .name = "AK4642", + .stream_name = "AK4642", + .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .codec_dai = &ak4642_dai, + .ops = NULL, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_dai_link, + .num_links = 1, +}; + +struct ak4642_setup_data ak4642_setup = { + .i2c_bus = 0, + .i2c_address = 0x12, /* 0x13 */ +}; + +static struct snd_soc_device fsi_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_ak4642, + .codec_data = &ak4642_setup, +}; + +static int ak4642_add_i2c_device(const struct ak4642_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "ak4642", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + printk(KERN_DEBUG "can't get i2c adapter\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_DEBUG "can't add i2c device\n"); + return -ENODEV; + } + + return 0; +} + +static struct platform_device *fsi_snd_device; + +static int __init fsi_ak4642_init(void) +{ + int ret = -ENOMEM; + + ak4642_add_i2c_device(&ak4642_setup); + + fsi_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_snd_device) + goto out; + + platform_set_drvdata(fsi_snd_device, + &fsi_snd_devdata); + fsi_snd_devdata.dev = &fsi_snd_device->dev; + ret = platform_device_add(fsi_snd_device); + + if (ret) + platform_device_put(fsi_snd_device); + +out: + return ret; +} + +static void __exit fsi_ak4642_exit(void) +{ + platform_device_unregister(fsi_snd_device); +} + +module_init(fsi_ak4642_init); +module_exit(fsi_ak4642_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.6.0.4

2 1

[alsa-devel] [PATCH v2 2/4] Add ak464x codec support
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This is very simple driver for ALSA It supprt headphone output and stereo input only This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o based on wm8731.c o add explain for register setting sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 516 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4642.h | 25 +++ 4 files changed, 547 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4642.c create mode 100644 sound/soc/codecs/ak4642.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..ee794f1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4642 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF @@ -74,6 +75,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4642 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..c8c567e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4642-objs := ak4642.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(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 diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c new file mode 100644 index 0000000..1b439b4 --- /dev/null +++ b/sound/soc/codecs/ak4642.c @@ -0,0 +1,516 @@ +/* + * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * 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. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It can use headphone output / stereo input only + * + * AK4642 is not tested. + * AK4643 is tested. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ak4642.h" + +#define AK4642_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK4642_CACHEREGNUM 0x25 + +struct snd_soc_codec_device soc_codec_dev_ak4642; + +/* codec private data */ +struct ak4642_priv { + struct snd_soc_codec codec; + unsigned int sysclk; +}; + +static struct snd_soc_codec *ak4642_codec; + +/* + * ak4642 register cache + */ +static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak4642 register cache + */ +static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak4642_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak4642 register cache + */ +static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK4642 register space + */ +static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4642 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak4642_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak4642_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + r |= ak4642_write(codec, i, cache[i]); + + return r; +}; + +static int ak4642_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* + * start headphone output + * + * PLL, Master Mode + * Audio I/F Format :MSB justified (ADC & DAC) + * Sampling Frequency: 44.1kHz + * Digital Volume: −8dB + * Bass Boost Level : Middle + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p97. + * + * Example code use 0x39, 0x79 value for 0x01 address, + * But we need MCKO (0x02) bit now + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x0f, 0x09); + ak4642_write(codec, 0x0e, 0x19); + ak4642_write(codec, 0x09, 0x91); + ak4642_write(codec, 0x0c, 0x91); + ak4642_write(codec, 0x0a, 0x28); + ak4642_write(codec, 0x0d, 0x28); + ak4642_write(codec, 0x00, 0x64); + ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ + ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + } else { + /* + * start stereo input + * + * PLL Master Mode + * Audio I/F Format:MSB justified (ADC & DAC) + * Sampling Frequency:44.1kHz + * Pre MIC AMP:+20dB + * MIC Power On + * ALC setting:Refer to Table 35 + * ALC bit=“1” + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p94. + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x02, 0x05); + ak4642_write(codec, 0x06, 0x3c); + ak4642_write(codec, 0x08, 0xe1); + ak4642_write(codec, 0x0b, 0x00); + ak4642_write(codec, 0x07, 0x21); + ak4642_write(codec, 0x00, 0x41); + ak4642_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak4642_write(codec, 0x01, 0x3b); + ak4642_write(codec, 0x01, 0x0b); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x0e, 0x11); + ak4642_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x10, 0x00); + ak4642_write(codec, 0x07, 0x01); + } +} + +static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4642_priv *ak4642 = codec->private_data; + + ak4642->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak4642_dai_ops = { + .startup = ak4642_dai_startup, + .shutdown = ak4642_dai_shutdown, + .set_sysclk = ak4642_dai_set_sysclk, +}; + +struct snd_soc_dai ak4642_dai = { + .name = "AK4642", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .ops = &ak4642_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4642_dai); + +static int ak4642_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak4642_sync(codec); + return 0; +} + +/* + * initialise the AK4642 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4642_init(struct ak4642_priv *ak4642) +{ + struct snd_soc_codec *codec = &ak4642->codec; + int ret = 0; + + if (ak4642_codec) { + dev_err(codec->dev, "Another ak4642 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4642; + codec->name = "AK4642"; + codec->owner = THIS_MODULE; + codec->read = ak4642_read_reg_cache; + codec->write = ak4642_write; + codec->dai = &ak4642_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); + codec->reg_cache = kmemdup(ak4642_reg, + sizeof(ak4642_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + ak4642_dai.dev = codec->dev; + ak4642_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto reg_cache_err; + } + + ret = snd_soc_register_dai(&ak4642_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto reg_cache_err; + } + + /* + * clock setting + * + * Audio I/F Format: MSB justified (ADC & DAC) + * BICK frequency at Master Mode: 64fs + * Input Master Clock Select at PLL Mode: 11.2896MHz + * MCKO: Enable + * Sampling Frequency: 44.1kHz + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p89. + * + * please fix-me + */ + ak4642_write(codec, 0x01, 0x08); + ak4642_write(codec, 0x04, 0x4a); + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x01, 0x0b); + + return ret; + +reg_cache_err: + kfree(codec->reg_cache); + + return ret; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ak4642_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4642_priv *ak4642; + struct snd_soc_codec *codec; + int ret; + + ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); + if (!ak4642) + return -ENOMEM; + + codec = &ak4642->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, ak4642); + codec->control_data = i2c; + + ret = ak4642_init(ak4642); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK4642\n"); + + return ret; +} + +static int ak4642_i2c_remove(struct i2c_client *client) +{ + struct ak4642_priv *ak4642 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&ak4642_dai); + snd_soc_unregister_codec(&ak4642->codec); + kfree(ak4642->codec.reg_cache); + kfree(ak4642); + ak4642_codec = NULL; + + return 0; +} + +static const struct i2c_device_id ak4642_i2c_id[] = { + { "ak4642", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); + +static struct i2c_driver ak4642_i2c_driver = { + .driver = { + .name = "AK4642 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak4642_i2c_probe, + .remove = ak4642_i2c_remove, + .id_table = ak4642_i2c_id, +}; + +#endif + +static int ak4642_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret; + + if (!ak4642_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4642_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to register card\n"); + goto card_err; + } + + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; + +} + +/* power down chip */ +static int ak4642_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4642 = { + .probe = ak4642_probe, + .remove = ak4642_remove, + .resume = ak4642_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); + +static int __init ak4642_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&ak4642_i2c_driver); +#endif + return ret; + +} +module_init(ak4642_modinit); + +static void __exit ak4642_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4642_i2c_driver); +#endif + +} +module_exit(ak4642_exit); + +MODULE_DESCRIPTION("Soc AK4642 driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h new file mode 100644 index 0000000..aa58c14 --- /dev/null +++ b/sound/soc/codecs/ak4642.h @@ -0,0 +1,25 @@ +/* + * ak464x.h -- AK464x Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based of ak4535.c + * + * 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 _AK4642_H +#define _AK4642_H + +struct ak4642_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ak4642_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4642; + +#endif -- 1.6.0.4

2 1

[alsa-devel] [PATCH] pcm_lib.c: Fixed inaccurate calculation of hw_ptr_interrupt in snd_pcm_update_hw_ptr_interrupt function
by Shine Liu 20 Aug '09

20 Aug '09

There's no restriction on that the runtime->period_size should be the multiplicator of the runtime->buffer_size in current kernel code. So "hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size;" is not always the trueth. I've encounter the problem of this bug, the kernel told me "PCM: Unexpected hw_pointer value (stream=0, pos=8, intr_ptr=16384)". The period_size and buffer_size I used was 0x800 and 0x3e80 when the bug occured. Following patch will fix the bug. Signed-off-by: Shine Liu <shinel(a)foxmail.com> ----------------------------------------------------------------------- --- a/sound/core/pcm_lib.c 2009-08-14 06:43:34.000000000 +0800 +++ b/sound/core/pcm_lib.c 2009-08-19 17:59:00.000000000 +0800 @@ -248,6 +248,12 @@ hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; + /* if the period_size is not the multiplicator of the + * buffer_size, the hw_ptr_interrupt calculated above + * may not be correct, should be fixed. + */ + if(hw_ptr_interrupt > hw_base + runtime->buffer_size) + hw_ptr_interrupt = hw_base + runtime->buffer_size; delta = new_hw_ptr - hw_ptr_interrupt; if (hw_ptr_interrupt >= runtime->boundary) { hw_ptr_interrupt -= runtime->boundary;

4 10

[alsa-devel] [PATCHv5 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 20 Aug '09

20 Aug '09

Hello guys, I've updated this series after some previous confusion from my side. Now that all dependencies are merged (Janusz's patches and Jarkko's patch), I've based this series on Takashi's tree (master branch). I've also talked with Tony and he agrees to have this changes merged by alsa tree. And as pointed by Jarkko, the mcbsp code is currently used only by Audio inside OMAP code. So, that should be safe. It is the same series. It now includes patch description for every patch. And also, as I already said, it is based on master branch of sound-2.6. Tested with beagle and rx51. Compiled tested for n770. BR, Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 377 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 77 ++++++- sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 511 insertions(+), 13 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

5 4

[PATCH] ALSA: pcm - Fix drain behavior in non-blocking mode
by Takashi Iwai 20 Aug '09

20 Aug '09

The current PCM core has the following problems regarding PCM draining in non-blocking mode: - the current f_flags isn't checked in snd_pcm_drain(), thus changing the mode dynamically via snd_pcm_nonblock() after open doesn't work. - calling drain in non-blocking mode just return -EAGAIN error, but doesn't provide any way to sync with draining. This patch fixes these issues. - check file->f_flags in snd_pcm_drain() properly - when O_NONBLOCK is set, PCM core sets the stream(s) to DRAIN state but quits ioctl immediately without waiting the whole drain; the caller can sync the drain manually via poll() Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 12 +++++++--- sound/core/pcm_native.c | 52 ++++++++++++++++++++++++++-------------------- 2 files changed, 37 insertions(+), 27 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 72cfd47..e3e78c7 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -197,12 +197,16 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, avail = snd_pcm_capture_avail(runtime); if (avail > runtime->avail_max) runtime->avail_max = avail; - if (avail >= runtime->stop_threshold) { - if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING) + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { + if (avail >= runtime->buffer_size) { snd_pcm_drain_done(substream); - else + return -EPIPE; + } + } else { + if (avail >= runtime->stop_threshold) { xrun(substream); - return -EPIPE; + return -EPIPE; + } } if (avail >= runtime->control->avail_min) wake_up(&runtime->sleep); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ac2150e..b08898c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1343,8 +1343,6 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state) { - if (substream->f_flags & O_NONBLOCK) - return -EAGAIN; substream->runtime->trigger_master = substream; return 0; } @@ -1392,7 +1390,6 @@ static struct action_ops snd_pcm_action_drain_init = { struct drain_rec { struct snd_pcm_substream *substream; wait_queue_t wait; - snd_pcm_uframes_t stop_threshold; }; static int snd_pcm_drop(struct snd_pcm_substream *substream); @@ -1404,13 +1401,15 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream); * After this call, all streams are supposed to be either SETUP or DRAINING * (capture only) state. */ -static int snd_pcm_drain(struct snd_pcm_substream *substream) +static int snd_pcm_drain(struct snd_pcm_substream *substream, + struct file *file) { struct snd_card *card; struct snd_pcm_runtime *runtime; struct snd_pcm_substream *s; int result = 0; int i, num_drecs; + int nonblock = 0; struct drain_rec *drec, drec_tmp, *d; card = substream->pcm->card; @@ -1428,6 +1427,15 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) } } + if (file) { + if (file->f_flags & O_NONBLOCK) + nonblock = 1; + } else if (substream->f_flags & O_NONBLOCK) + nonblock = 1; + + if (nonblock) + goto lock; /* no need to allocate waitqueues */ + /* allocate temporary record for drain sync */ down_read(&snd_pcm_link_rwsem); if (snd_pcm_stream_linked(substream)) { @@ -1449,16 +1457,11 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) d->substream = s; init_waitqueue_entry(&d->wait, current); add_wait_queue(&runtime->sleep, &d->wait); - /* stop_threshold fixup to avoid endless loop when - * stop_threshold > buffer_size - */ - d->stop_threshold = runtime->stop_threshold; - if (runtime->stop_threshold > runtime->buffer_size) - runtime->stop_threshold = runtime->buffer_size; } } up_read(&snd_pcm_link_rwsem); + lock: snd_pcm_stream_lock_irq(substream); /* resume pause */ if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED) @@ -1466,9 +1469,12 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) /* pre-start/stop - all running streams are changed to DRAINING state */ result = snd_pcm_action(&snd_pcm_action_drain_init, substream, 0); - if (result < 0) { - snd_pcm_stream_unlock_irq(substream); - goto _error; + if (result < 0) + goto unlock; + /* in non-blocking, we don't wait in ioctl but let caller poll */ + if (nonblock) { + result = -EAGAIN; + goto unlock; } for (;;) { @@ -1504,18 +1510,18 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) } } + unlock: snd_pcm_stream_unlock_irq(substream); - _error: - for (i = 0; i < num_drecs; i++) { - d = &drec[i]; - runtime = d->substream->runtime; - remove_wait_queue(&runtime->sleep, &d->wait); - runtime->stop_threshold = d->stop_threshold; + if (!nonblock) { + for (i = 0; i < num_drecs; i++) { + d = &drec[i]; + runtime = d->substream->runtime; + remove_wait_queue(&runtime->sleep, &d->wait); + } + if (drec != &drec_tmp) + kfree(drec); } - - if (drec != &drec_tmp) - kfree(drec); snd_power_unlock(card); return result; @@ -2544,7 +2550,7 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_hw_params_old_user(substream, arg); #endif case SNDRV_PCM_IOCTL_DRAIN: - return snd_pcm_drain(substream); + return snd_pcm_drain(substream, file); case SNDRV_PCM_IOCTL_DROP: return snd_pcm_drop(substream); case SNDRV_PCM_IOCTL_PAUSE: -- 1.6.3.3

1 0

[alsa-devel] [PATCHv5 01/20] OMAP: McBSP: Provide functions for ASoC frame syncronization
by Eduardo Valentin 20 Aug '09

20 Aug '09

From: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> ASoC has an annoying bug letting either L or R channel to be played on L channel. In other words, L and R channels can switch at random. This provides McBSP funtionality that may be used to fix this feature. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> --- arch/arm/plat-omap/include/mach/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 52 +++++++++++++++++++++++++++++++ 2 files changed, 54 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-omap/include/mach/mcbsp.h b/arch/arm/plat-omap/include/mach/mcbsp.h index 57249bb..b85e6e6 100644 --- a/arch/arm/plat-omap/include/mach/mcbsp.h +++ b/arch/arm/plat-omap/include/mach/mcbsp.h @@ -389,6 +389,8 @@ int omap_mcbsp_request(unsigned int id); void omap_mcbsp_free(unsigned int id); void omap_mcbsp_start(unsigned int id, int tx, int rx); void omap_mcbsp_stop(unsigned int id, int tx, int rx); +void omap_mcbsp_xmit_enable(unsigned int id, u8 enable); +void omap_mcbsp_recv_enable(unsigned int id, u8 enable); void omap_mcbsp_xmit_word(unsigned int id, u32 word); u32 omap_mcbsp_recv_word(unsigned int id); diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c index a3d2313..0aa2524 100644 --- a/arch/arm/plat-omap/mcbsp.c +++ b/arch/arm/plat-omap/mcbsp.c @@ -412,6 +412,58 @@ void omap_mcbsp_stop(unsigned int id, int tx, int rx) } EXPORT_SYMBOL(omap_mcbsp_stop); +void omap_mcbsp_xmit_enable(unsigned int id, u8 enable) +{ + struct omap_mcbsp *mcbsp; + void __iomem *io_base; + u16 w; + + if (!(cpu_is_omap2430() || cpu_is_omap34xx())) + return; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + + mcbsp = id_to_mcbsp_ptr(id); + io_base = mcbsp->io_base; + + w = OMAP_MCBSP_READ(io_base, XCCR); + + if (enable) + OMAP_MCBSP_WRITE(io_base, XCCR, w & ~(XDISABLE)); + else + OMAP_MCBSP_WRITE(io_base, XCCR, w | XDISABLE); +} +EXPORT_SYMBOL(omap_mcbsp_xmit_enable); + +void omap_mcbsp_recv_enable(unsigned int id, u8 enable) +{ + struct omap_mcbsp *mcbsp; + void __iomem *io_base; + u16 w; + + if (!(cpu_is_omap2430() || cpu_is_omap34xx())) + return; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + + mcbsp = id_to_mcbsp_ptr(id); + io_base = mcbsp->io_base; + + w = OMAP_MCBSP_READ(io_base, RCCR); + + if (enable) + OMAP_MCBSP_WRITE(io_base, RCCR, w & ~(RDISABLE)); + else + OMAP_MCBSP_WRITE(io_base, RCCR, w | RDISABLE); +} +EXPORT_SYMBOL(omap_mcbsp_recv_enable); + /* polled mcbsp i/o operations */ int omap_mcbsp_pollwrite(unsigned int id, u16 buf) { -- 1.6.2.GIT -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 19

[alsa-devel] [PATCH v2 1/4] Add SuperH FSI driver support for ALSA
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This driver is very simple. It support playback only now. This patch is tested by ms7724se board. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o it direct use clk_enable/disable o rename fsi.h -> sh_fsi.h include/sound/sh_fsi.h | 83 ++++ sound/soc/sh/Kconfig | 7 +- sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi.c | 1004 ++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1095 insertions(+), 1 deletions(-) create mode 100644 include/sound/sh_fsi.h create mode 100644 sound/soc/sh/fsi.c diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h new file mode 100644 index 0000000..c022736 --- /dev/null +++ b/include/sound/sh_fsi.h @@ -0,0 +1,83 @@ +#ifndef __SOUND_FSI_H +#define __SOUND_FSI_H + +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.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. + */ + +/* flags format + + * 0xABCDEEFF + * + * A: channel size for TDM (input) + * B: channel size for TDM (ooutput) + * C: inversion + * D: mode + * E: input format + * F: output format + */ + +#include <linux/clk.h> +#include <sound/soc.h> + +/* TDM channel */ +#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28) +#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24) + +#define SH_FSI_CH_IMASK 0xF0000000 +#define SH_FSI_CH_OMASK 0x0F000000 +#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28) +#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24) + +/* clock inversion */ +#define SH_FSI_INVERSION_MASK 0x00F00000 +#define SH_FSI_LRM_INV (1 << 20) +#define SH_FSI_BRM_INV (1 << 21) +#define SH_FSI_LRS_INV (1 << 22) +#define SH_FSI_BRS_INV (1 << 23) + +/* mode */ +#define SH_FSI_MODE_MASK 0x000F0000 +#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */ +#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */ + +/* DI format */ +#define SH_FSI_FMT_MASK 0x000000FF +#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8) +#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0) +#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) +#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) + +#define SH_FSI_FMT_MONO (1 << 0) +#define SH_FSI_FMT_MONO_DELAY (1 << 1) +#define SH_FSI_FMT_PCM (1 << 2) +#define SH_FSI_FMT_I2S (1 << 3) +#define SH_FSI_FMT_TDM (1 << 4) +#define SH_FSI_FMT_TDM_DELAY (1 << 5) + +#define SH_FSI_IFMT_TDM_CH(x) \ + (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) +#define SH_FSI_IFMT_TDM_DELAY_CH(x) \ + (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x)) + +#define SH_FSI_OFMT_TDM_CH(x) \ + (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x)) +#define SH_FSI_OFMT_TDM_DELAY_CH(x) \ + (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x)) + +struct sh_fsi_platform_info { + unsigned long porta_flags; + unsigned long portb_flags; +}; + +extern struct snd_soc_dai fsi_soc_dai[2]; +extern struct snd_soc_platform fsi_soc_platform; + +#endif /* __SOUND_FSI_H */ diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 54bd604..01943a1 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -20,7 +20,12 @@ config SND_SOC_SH4_HAC config SND_SOC_SH4_SSI tristate - +config SND_SOC_SH4_FSI + tristate "SH4 FSI support" + depends on CPU_SUBTYPE_SH7724 + select SH_DMA + help + This option enables FSI sound support ## ## Boards diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a8e8ab8..9fbcc4a 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -5,8 +5,10 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o ## audio units found on some SH-4 snd-soc-hac-objs := hac.o snd-soc-ssi-objs := ssi.o +snd-soc-fsi-objs := fsi.o obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o +obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c new file mode 100644 index 0000000..4412324 --- /dev/null +++ b/sound/soc/sh/fsi.c @@ -0,0 +1,1004 @@ +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on ssi.c + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * 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/platform_device.h> +#include <linux/delay.h> +#include <linux/list.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/sh_fsi.h> +#include <asm/atomic.h> +#include <asm/dma.h> +#include <asm/dma-sh.h> + +#define DO_FMT 0x0000 +#define DOFF_CTL 0x0004 +#define DOFF_ST 0x0008 +#define DI_FMT 0x000C +#define DIFF_CTL 0x0010 +#define DIFF_ST 0x0014 +#define CKG1 0x0018 +#define CKG2 0x001C +#define DIDT 0x0020 +#define DODT 0x0024 +#define MUTE_ST 0x0028 +#define REG_END MUTE_ST + +#define INT_ST 0x0200 +#define IEMSK 0x0204 +#define IMSK 0x0208 +#define MUTE 0x020C +#define CLK_RST 0x0210 +#define SOFT_RST 0x0214 +#define MREG_START INT_ST +#define MREG_END SOFT_RST + +/* DO_FMT */ +/* DI_FMT */ +#define CR_FMT(param) ((param) << 4) +# define CR_MONO 0x0 +# define CR_MONO_D 0x1 +# define CR_PCM 0x2 +# define CR_I2S 0x3 +# define CR_TDM 0x4 +# define CR_TDM_D 0x5 + +/* DOFF_CTL */ +/* DIFF_CTL */ +#define IRQ_HALF 0x00100000 +#define FIFO_CLR 0x00000001 + +/* DOFF_ST */ +#define ERR_OVER 0x00000010 +#define ERR_UNDER 0x00000001 + +/* CLK_RST */ +#define B_CLK 0x00000010 +#define A_CLK 0x00000001 + +/* INT_ST */ +#define INT_B_IN (1 << 12) +#define INT_B_OUT (1 << 8) +#define INT_A_IN (1 << 4) +#define INT_A_OUT (1 << 0) + +#define FSI_RATES SNDRV_PCM_RATE_8000_96000 + +#define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/************************************************************************ + + + struct + + +************************************************************************/ +struct fsi_priv { + void __iomem *base; + struct snd_pcm_substream *substream; + + int fifo_max; + int chan; + int dma_chan; + + int byte_offset; + int period_len; + int buffer_len; + int periods; +}; + +struct fsi_master { + void __iomem *base; + int irq; + struct clk *clk; + struct fsi_priv fsia; + struct fsi_priv fsib; + struct sh_fsi_platform_info *info; +}; + +static struct fsi_master *master; + +/************************************************************************ + + + basic read write function + + +************************************************************************/ +static int __fsi_reg_write(u32 reg, u32 data) +{ + /* valid data area is 24bit */ + data &= 0x00ffffff; + + return ctrl_outl(data, reg); +} + +static u32 __fsi_reg_read(u32 reg) +{ + return ctrl_inl(reg); +} + +static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +{ + u32 val = __fsi_reg_read(reg); + + val &= ~mask; + val |= data & mask; + + return __fsi_reg_write(reg, val); +} + +static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_write((u32)(fsi->base + reg), data); +} + +static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) +{ + if (reg > REG_END) + return 0; + + return __fsi_reg_read((u32)(fsi->base + reg)); +} + +static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); +} + +static int fsi_master_write(u32 reg, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_write((u32)(master->base + reg), data); +} + +static u32 fsi_master_read(u32 reg) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return 0; + + return __fsi_reg_read((u32)(master->base + reg)); +} + +static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_mask_set((u32)(master->base + reg), mask, data); +} + +/************************************************************************ + + + basic function + + +************************************************************************/ +static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd; + struct fsi_priv *fsi = NULL; + + if (!substream || !master) + return NULL; + + rtd = substream->private_data; + switch (rtd->dai->cpu_dai->id) { + case 0: + fsi = &master->fsia; + break; + case 1: + fsi = &master->fsib; + break; + } + + return fsi; +} + +static int fsi_is_port_a(struct fsi_priv *fsi) +{ + /* return + * 1 : port a + * 0 : port b + */ + + if (fsi == &master->fsia) + return 1; + + return 0; +} + +static u32 fsi_get_info_flags(struct fsi_priv *fsi) +{ + int is_porta = fsi_is_port_a(fsi); + + return is_porta ? master->info->porta_flags : + master->info->portb_flags; +} + +static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play) +{ + u32 mode; + u32 flags = fsi_get_info_flags(fsi); + + mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE; + + /* return + * 1 : master mode + * 0 : slave mode + */ + + return (mode & flags) != mode; +} + +static u32 fsi_port_ab_io_bit(struct fsi_priv *fsi, int is_play) +{ + int is_porta = fsi_is_port_a(fsi); + u32 data; + + if (is_porta) + data = is_play ? (1 << 0) : (1 << 4); + else + data = is_play ? (1 << 8) : (1 << 12); + + return data; +} + +static void fsi_stream_push(struct fsi_priv *fsi, + struct snd_pcm_substream *substream, + u32 buffer_len, + u32 period_len) +{ + fsi->substream = substream; + fsi->buffer_len = buffer_len; + fsi->period_len = period_len; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static void fsi_stream_pop(struct fsi_priv *fsi) +{ + fsi->substream = NULL; + fsi->buffer_len = 0; + fsi->period_len = 0; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) +{ + u32 status; + u32 reg = is_play ? DOFF_ST : DIFF_ST; + int residue; + + status = fsi_reg_read(fsi, reg); + residue = 0x1ff & (status >> 8); + residue *= fsi->chan; + + return residue; +} + +static int fsi_get_residue(struct fsi_priv *fsi, int is_play) +{ + int residue; + int width; + struct snd_pcm_runtime *runtime; + + runtime = fsi->substream->runtime; + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + if (2 == width) + residue = fsi_get_fifo_residue(fsi, is_play); + else + residue = get_dma_residue(fsi->dma_chan); + + return residue; +} + +/************************************************************************ + + + basic dma function + + +************************************************************************/ +#define PORTA_DMA 0 +#define PORTB_DMA 1 + +static int fsi_get_dma_chan(void) +{ + if (0 != request_dma(PORTA_DMA, "fsia")) + return -EIO; + + if (0 != request_dma(PORTB_DMA, "fsib")) { + free_dma(PORTA_DMA); + return -EIO; + } + + master->fsia.dma_chan = PORTA_DMA; + master->fsib.dma_chan = PORTB_DMA; + + return 0; +} + +static void fsi_free_dma_chan(void) +{ + dma_wait_for_completion(PORTA_DMA); + dma_wait_for_completion(PORTB_DMA); + free_dma(PORTA_DMA); + free_dma(PORTB_DMA); + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; +} + +/************************************************************************ + + + ctrl function + + +************************************************************************/ +static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, data); + fsi_master_mask_set(IEMSK, data, data); +} + +static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, 0); + fsi_master_mask_set(IEMSK, data, 0); +} + +static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) +{ + u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4); + + if (enable) + fsi_master_mask_set(CLK_RST, val, val); + else + fsi_master_mask_set(CLK_RST, val, 0); +} + +static void fsi_irq_init(struct fsi_priv *fsi, int is_play) +{ + u32 data; + u32 ctrl; + + data = fsi_port_ab_io_bit(fsi, is_play); + ctrl = is_play ? DOFF_CTL : DIFF_CTL; + + /* set IMSK */ + fsi_irq_disable(fsi, is_play); + + /* set interrupt generation factor */ + fsi_reg_write(fsi, ctrl, IRQ_HALF); + + /* clear FIFO */ + fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR); + + /* clear interrupt factor */ + fsi_master_mask_set(INT_ST, data, 0); +} + +static void fsi_soft_all_reset(void) +{ + u32 status = fsi_master_read(SOFT_RST); + + /* port AB reset */ + status &= 0x000000ff; + fsi_master_write(SOFT_RST, status); + mdelay(10); + + /* soft reset */ + status &= 0x000000f0; + fsi_master_write(SOFT_RST, status); + status |= 0x00000001; + fsi_master_write(SOFT_RST, status); + mdelay(10); +} + +static void fsi_16data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u16 *dma_start; + u32 snd; + int i; + + /* get dma start position for FSI */ + dma_start = (u16 *)runtime->dma_area; + dma_start += fsi->byte_offset / 2; + + /* + * soft dma + * FSI can not use DMA when 16bpp + */ + for (i = 0; i < send; i++) { + snd = (u32)dma_start[i]; + fsi_reg_write(fsi, DODT, snd << 8); + } +} + +static void fsi_32data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u32 *dma_start; + + /* get dma start position for FSI */ + dma_start = (u32 *)runtime->dma_area; + dma_start += fsi->byte_offset / 4; + + dma_wait_for_completion(fsi->dma_chan); + dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR)); + dma_write(fsi->dma_chan, (u32)dma_start, + (u32)(fsi->base + DODT), send * 4); +} + +/* playback interrupt */ +static int fsi_data_push(struct fsi_priv *fsi) +{ + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream = NULL; + int send; + int fifo_free; + int width; + + if (!fsi || + !fsi->substream || + !fsi->substream->runtime) + return -EINVAL; + + runtime = fsi->substream->runtime; + + /* FSI FIFO has limit. + * So, this driver can not send periods data at a time + */ + if (fsi->byte_offset >= + fsi->period_len * (fsi->periods + 1)) { + + substream = fsi->substream; + fsi->periods = (fsi->periods + 1) % runtime->periods; + + if (0 == fsi->periods) + fsi->byte_offset = 0; + } + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + /* get send size for alsa */ + send = (fsi->buffer_len - fsi->byte_offset) / width; + + /* get FIFO free size */ + fifo_free = (fsi->fifo_max * fsi->chan) - fsi_get_fifo_residue(fsi, 1); + + /* size check */ + if (fifo_free < send) + send = fifo_free; + + if (2 == width) + fsi_16data_push(fsi, runtime, send); + else if (4 == width) + fsi_32data_push(fsi, runtime, send); + else + return -EINVAL; + + fsi->byte_offset += send * width; + + fsi_irq_enable(fsi, 1); + + if (substream) + snd_pcm_period_elapsed(substream); + + return 0; +} + +static irqreturn_t fsi_interrupt(int irq, void *data) +{ + u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; + u32 int_st = fsi_master_read(INT_ST); + + /* clear irq status */ + fsi_master_write(SOFT_RST, status); + fsi_master_write(SOFT_RST, status | 0x00000010); + + if (int_st & INT_A_OUT) + fsi_data_push(&master->fsia); + if (int_st & INT_B_OUT) + fsi_data_push(&master->fsib); + + fsi_master_write(INT_ST, 0x0000000); + + return IRQ_HANDLED; +} + +/************************************************************************ + + + dai ops + + +************************************************************************/ +static int fsi_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + const char *msg; + u32 flags = fsi_get_info_flags(fsi); + u32 fmt; + u32 reg; + u32 data; + int is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int is_master; + int ret = 0; + + clk_enable(master->clk); + + /* CKG1 */ + data = is_play ? (1 << 0) : (1 << 4); + is_master = fsi_is_master_mode(fsi, is_play); + if (is_master) + fsi_reg_mask_set(fsi, CKG1, data, data); + else + fsi_reg_mask_set(fsi, CKG1, data, 0); + + /* clock inversion (CKG2) */ + data = 0; + switch (SH_FSI_INVERSION_MASK & flags) { + case SH_FSI_LRM_INV: + data = 1 << 12; + break; + case SH_FSI_BRM_INV: + data = 1 << 8; + break; + case SH_FSI_LRS_INV: + data = 1 << 4; + break; + case SH_FSI_BRS_INV: + data = 1 << 0; + break; + } + fsi_reg_write(fsi, CKG2, data); + + /* do fmt, di fmt */ + data = 0; + reg = is_play ? DO_FMT : DI_FMT; + fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags); + switch (fmt) { + case SH_FSI_FMT_MONO: + msg = "MONO"; + data = CR_FMT(CR_MONO); + fsi->chan = 1; + break; + case SH_FSI_FMT_MONO_DELAY: + msg = "MONO Delay"; + data = CR_FMT(CR_MONO_D); + fsi->chan = 1; + break; + case SH_FSI_FMT_PCM: + msg = "PCM"; + data = CR_FMT(CR_PCM); + fsi->chan = 2; + break; + case SH_FSI_FMT_I2S: + msg = "I2S"; + data = CR_FMT(CR_I2S); + fsi->chan = 2; + break; + case SH_FSI_FMT_TDM: + msg = "TDM"; + data = CR_FMT(CR_TDM) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + case SH_FSI_FMT_TDM_DELAY: + msg = "TDM Delay"; + data = CR_FMT(CR_TDM_D) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + default: + dev_err(dai->dev, "unknown format.\n"); + return -EINVAL; + } + + switch (fsi->chan) { + case 1: + fsi->fifo_max = 256; + break; + case 2: + fsi->fifo_max = 128; + break; + case 3: + case 4: + fsi->fifo_max = 64; + break; + case 5: + case 6: + case 7: + case 8: + fsi->fifo_max = 32; + break; + default: + dev_err(dai->dev, "channel size error.\n"); + return -EINVAL; + } + + fsi_reg_write(fsi, reg, data); + dev_dbg(dai->dev, "use %s format (%d channel) use %d DMAC\n", + msg, fsi->chan, fsi->dma_chan); + + /* + * clear clk reset if master mode + */ + if (is_master) + fsi_clk_ctrl(fsi, 1); + + /* irq setting */ + fsi_irq_init(fsi, is_play); + + return ret; +} + +static void fsi_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + fsi_irq_disable(fsi, is_play); + fsi_clk_ctrl(fsi, 0); + + clk_disable(master->clk); +} + +static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int ret = 0; + + /* capture not supported */ + if (!is_play) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + fsi_stream_push(fsi, substream, + frames_to_bytes(runtime, runtime->buffer_size), + frames_to_bytes(runtime, runtime->period_size)); + ret = fsi_data_push(fsi); + break; + case SNDRV_PCM_TRIGGER_STOP: + fsi_irq_disable(fsi, is_play); + fsi_stream_pop(fsi); + break; + } + + return ret; +} + +static struct snd_soc_dai_ops fsi_dai_ops = { + .startup = fsi_dai_startup, + .shutdown = fsi_dai_shutdown, + .trigger = fsi_dai_trigger, +}; + +/************************************************************************ + + + pcm ops + + +************************************************************************/ +static struct snd_pcm_hardware fsi_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE, + .formats = FSI_FMTS, + .rates = FSI_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int fsi_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &fsi_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + + return ret; +} + +static int fsi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int fsi_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + long location; + + location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play); + if (location < 0) + location = 0; + + return bytes_to_frames(runtime, location); +} + +static struct snd_pcm_ops fsi_pcm_ops = { + .open = fsi_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsi_hw_params, + .hw_free = fsi_hw_free, + .pointer = fsi_pointer, +}; + +/************************************************************************ + + + snd_soc_platform + + +************************************************************************/ +#define PREALLOC_BUFFER (32 * 1024) +#define PREALLOC_BUFFER_MAX (32 * 1024) + +static void fsi_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int fsi_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + /* + * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel + * in MMAP mode (i.e. aplay -M) + */ + return snd_pcm_lib_preallocate_pages_for_all( + pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); +} + +/************************************************************************ + + + alsa struct + + +************************************************************************/ +struct snd_soc_dai fsi_soc_dai[] = { + { + .name = "FSIA", + .id = 0, + .playback = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, + /* capture not supported */ + .ops = &fsi_dai_ops, + }, + { + .name = "FSIB", + .id = 1, + .playback = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, + /* capture not supported */ + .ops = &fsi_dai_ops, + }, +}; +EXPORT_SYMBOL_GPL(fsi_soc_dai); + +struct snd_soc_platform fsi_soc_platform = { + .name = "fsi-pcm", + .pcm_ops = &fsi_pcm_ops, + .pcm_new = fsi_pcm_new, + .pcm_free = fsi_pcm_free, +}; +EXPORT_SYMBOL_GPL(fsi_soc_platform); + +/************************************************************************ + + + platform function + + +************************************************************************/ +static int fsi_probe(struct platform_device *pdev) +{ + struct resource *res; + char clk_name[8]; + unsigned int irq; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + irq = platform_get_irq(pdev, 0); + if (!res || !irq) { + dev_err(&pdev->dev, "Not enough FSI platform resources.\n"); + ret = -ENODEV; + goto exit; + } + + master = kzalloc(sizeof(*master), GFP_KERNEL); + if (!master) { + dev_err(&pdev->dev, "Could not allocate master\n"); + ret = -ENOMEM; + goto exit; + } + + master->base = ioremap_nocache(res->start, resource_size(res)); + if (!master->base) { + ret = -ENXIO; + dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n"); + goto exit_kfree; + } + + master->irq = irq; + master->info = pdev->dev.platform_data; + master->fsia.base = master->base; + master->fsib.base = master->base + 0x40; + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; + + ret = fsi_get_dma_chan(); + if (ret < 0) { + dev_err(&pdev->dev, "cannot get dma api\n"); + goto exit_iounmap; + } + + /* FSI is based on SPU mstp */ + snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); + master->clk = clk_get(NULL, clk_name); + if (IS_ERR(master->clk)) { + dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); + ret = -EIO; + goto exit_free_dma; + } + + fsi_soc_dai[0].dev = &pdev->dev; + fsi_soc_dai[1].dev = &pdev->dev; + + fsi_soft_all_reset(); + + ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); + if (ret) { + dev_err(&pdev->dev, "irq request err\n"); + goto exit_free_dma; + } + + ret = snd_soc_register_platform(&fsi_soc_platform); + if (ret < 0) { + dev_err(&pdev->dev, "cannot snd soc register\n"); + goto exit_free_irq; + } + + return snd_soc_register_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + +exit_free_irq: + free_irq(irq, master); +exit_free_dma: + fsi_free_dma_chan(); +exit_iounmap: + iounmap(master->base); +exit_kfree: + kfree(master); + master = NULL; +exit: + return ret; +} + +static int fsi_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + snd_soc_unregister_platform(&fsi_soc_platform); + + clk_put(master->clk); + + fsi_free_dma_chan(); + + free_irq(master->irq, master); + + iounmap(master->base); + kfree(master); + master = NULL; + return 0; +} + +static struct platform_driver fsi_driver = { + .driver = { + .name = "sh_fsi", + }, + .probe = fsi_probe, + .remove = fsi_remove, +}; + +static int __init fsi_mobile_init(void) +{ + return platform_driver_register(&fsi_driver); +} + +static void __exit fsi_mobile_exit(void) +{ + platform_driver_unregister(&fsi_driver); +} +module_init(fsi_mobile_init); +module_exit(fsi_mobile_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.6.0.4

4 5

[alsa-devel] [PATCH 1/3] Add SuperH FSI driver support for ALSA
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This driver is very simple. It support playback only now. It is tested by ms7724se board. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- include/sound/fsi.h | 83 ++++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/Makefile | 1 + sound/soc/sh/fsi.c | 1001 +++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1090 insertions(+), 0 deletions(-) create mode 100644 include/sound/fsi.h create mode 100644 sound/soc/sh/fsi.c diff --git a/include/sound/fsi.h b/include/sound/fsi.h new file mode 100644 index 0000000..c022736 --- /dev/null +++ b/include/sound/fsi.h @@ -0,0 +1,83 @@ +#ifndef __SOUND_FSI_H +#define __SOUND_FSI_H + +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.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. + */ + +/* flags format + + * 0xABCDEEFF + * + * A: channel size for TDM (input) + * B: channel size for TDM (ooutput) + * C: inversion + * D: mode + * E: input format + * F: output format + */ + +#include <linux/clk.h> +#include <sound/soc.h> + +/* TDM channel */ +#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28) +#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24) + +#define SH_FSI_CH_IMASK 0xF0000000 +#define SH_FSI_CH_OMASK 0x0F000000 +#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28) +#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24) + +/* clock inversion */ +#define SH_FSI_INVERSION_MASK 0x00F00000 +#define SH_FSI_LRM_INV (1 << 20) +#define SH_FSI_BRM_INV (1 << 21) +#define SH_FSI_LRS_INV (1 << 22) +#define SH_FSI_BRS_INV (1 << 23) + +/* mode */ +#define SH_FSI_MODE_MASK 0x000F0000 +#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */ +#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */ + +/* DI format */ +#define SH_FSI_FMT_MASK 0x000000FF +#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8) +#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0) +#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) +#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) + +#define SH_FSI_FMT_MONO (1 << 0) +#define SH_FSI_FMT_MONO_DELAY (1 << 1) +#define SH_FSI_FMT_PCM (1 << 2) +#define SH_FSI_FMT_I2S (1 << 3) +#define SH_FSI_FMT_TDM (1 << 4) +#define SH_FSI_FMT_TDM_DELAY (1 << 5) + +#define SH_FSI_IFMT_TDM_CH(x) \ + (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) +#define SH_FSI_IFMT_TDM_DELAY_CH(x) \ + (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x)) + +#define SH_FSI_OFMT_TDM_CH(x) \ + (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x)) +#define SH_FSI_OFMT_TDM_DELAY_CH(x) \ + (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x)) + +struct sh_fsi_platform_info { + unsigned long porta_flags; + unsigned long portb_flags; +}; + +extern struct snd_soc_dai fsi_soc_dai[2]; +extern struct snd_soc_platform fsi_soc_platform; + +#endif /* __SOUND_FSI_H */ diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 54bd604..afb0e60 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -20,6 +20,11 @@ config SND_SOC_SH4_HAC config SND_SOC_SH4_SSI tristate +config SND_SOC_SH4_FSI + tristate "SH4 FSI support" + depends on CPU_SUBTYPE_SH7724 + help + This option enables FSI sound support ## diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a8e8ab8..dfa4949 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -7,6 +7,7 @@ snd-soc-hac-objs := hac.o snd-soc-ssi-objs := ssi.o obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o +obj-$(CONFIG_SND_SOC_SH4_FSI) += fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c new file mode 100644 index 0000000..017f170 --- /dev/null +++ b/sound/soc/sh/fsi.c @@ -0,0 +1,1001 @@ +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on ssi.c + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * 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/platform_device.h> +#include <linux/delay.h> +#include <linux/list.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/fsi.h> +#include <asm/atomic.h> +#include <asm/dma.h> +#include <asm/dma-sh.h> + +#define DO_FMT 0x0000 +#define DOFF_CTL 0x0004 +#define DOFF_ST 0x0008 +#define DI_FMT 0x000C +#define DIFF_CTL 0x0010 +#define DIFF_ST 0x0014 +#define CKG1 0x0018 +#define CKG2 0x001C +#define DIDT 0x0020 +#define DODT 0x0024 +#define MUTE_ST 0x0028 +#define REG_END MUTE_ST + +#define INT_ST 0x0200 +#define IEMSK 0x0204 +#define IMSK 0x0208 +#define MUTE 0x020C +#define CLK_RST 0x0210 +#define SOFT_RST 0x0214 +#define MREG_START INT_ST +#define MREG_END SOFT_RST + +/* DO_FMT */ +/* DI_FMT */ +#define CR_FMT(param) ((param) << 4) +# define CR_MONO 0x0 +# define CR_MONO_D 0x1 +# define CR_PCM 0x2 +# define CR_I2S 0x3 +# define CR_TDM 0x4 +# define CR_TDM_D 0x5 + +/* DOFF_CTL */ +/* DIFF_CTL */ +#define IRQ_HALF 0x00100000 +#define FIFO_CLR 0x00000001 + +/* DOFF_ST */ +#define ERR_OVER 0x00000010 +#define ERR_UNDER 0x00000001 + +/* CLK_RST */ +#define B_CLK 0x00000010 +#define A_CLK 0x00000001 + +/* INT_ST */ +#define INT_B_IN (1 << 12) +#define INT_B_OUT (1 << 8) +#define INT_A_IN (1 << 4) +#define INT_A_OUT (1 << 0) + +#define FSI_RATES SNDRV_PCM_RATE_8000_96000 + +#define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/************************************************************************ + + + struct + + +************************************************************************/ +struct fsi_priv { + void __iomem *base; + struct snd_pcm_substream *substream; + + int fifo_max; + int chan; + int dma_chan; + + int byte_offset; + int period_len; + int buffer_len; + int periods; +}; + +struct fsi_master { + void __iomem *base; + atomic_t user; + int irq; + struct clk *clk; + struct fsi_priv fsia; + struct fsi_priv fsib; + struct sh_fsi_platform_info *info; +}; + +struct fsi_master *master; + +/************************************************************************ + + + basic read write function + + +************************************************************************/ +static int __fsi_reg_write(u32 reg, u32 data) +{ + /* valid data area is 24bit */ + data &= 0x00ffffff; + + return ctrl_outl(data, reg); +} + +static u32 __fsi_reg_read(u32 reg) +{ + return ctrl_inl(reg); +} + +static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +{ + u32 val = __fsi_reg_read(reg); + + val &= ~mask; + val |= data & mask; + + return __fsi_reg_write(reg, val); +} + +static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_write((u32)(fsi->base + reg), data); +} + +static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) +{ + if (reg > REG_END) + return 0; + + return __fsi_reg_read((u32)(fsi->base + reg)); +} + +static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); +} + +static int fsi_master_write(u32 reg, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_write((u32)(master->base + reg), data); +} + +static u32 fsi_master_read(u32 reg) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return 0; + + return __fsi_reg_read((u32)(master->base + reg)); +} + +static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_mask_set((u32)(master->base + reg), mask, data); +} + +/************************************************************************ + + + basic function + + +************************************************************************/ +static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd; + struct fsi_priv *fsi = NULL; + + if (!substream || !master) + return NULL; + + rtd = substream->private_data; + switch (rtd->dai->cpu_dai->id) { + case 0: + fsi = &master->fsia; + break; + case 1: + fsi = &master->fsib; + break; + } + + return fsi; +} + +static int fsi_is_port_a(struct fsi_priv *fsi) +{ + /* return + * 1 : port a + * 0 : port b + */ + + if (fsi == &master->fsia) + return 1; + + return 0; +} + +static u32 fsi_get_info_flags(struct fsi_priv *fsi) +{ + int is_porta = fsi_is_port_a(fsi); + + return is_porta ? master->info->porta_flags : + master->info->portb_flags; +} + +static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play) +{ + u32 mode; + u32 flags = fsi_get_info_flags(fsi); + + mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE; + + /* return + * 1 : master mode + * 0 : slave mode + */ + + return (mode & flags) != mode; +} + +static u32 fsi_port_ab_io_bit(struct fsi_priv *fsi, int is_play) +{ + int is_porta = fsi_is_port_a(fsi); + u32 data; + + if (is_porta) + data = is_play ? (1 << 0) : (1 << 4); + else + data = is_play ? (1 << 8) : (1 << 12); + + return data; +} + +static void fsi_stream_push(struct fsi_priv *fsi, + struct snd_pcm_substream *substream, + u32 buffer_len, + u32 period_len) +{ + fsi->substream = substream; + fsi->buffer_len = buffer_len; + fsi->period_len = period_len; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static void fsi_stream_pop(struct fsi_priv *fsi) +{ + fsi->substream = NULL; + fsi->buffer_len = 0; + fsi->period_len = 0; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) +{ + u32 status; + u32 reg = is_play ? DOFF_ST : DIFF_ST; + int residue; + + status = fsi_reg_read(fsi, reg); + residue = 0x1ff & (status >> 8); + residue *= fsi->chan; + + return residue; +} + +static int fsi_get_residue(struct fsi_priv *fsi, int is_play) +{ + int residue; + int width; + struct snd_pcm_runtime *runtime; + + runtime = fsi->substream->runtime; + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + if (2 == width) + residue = fsi_get_fifo_residue(fsi, is_play); + else + residue = get_dma_residue(fsi->dma_chan); + + return residue; +} + +/************************************************************************ + + + basic dma function + + +************************************************************************/ +#define PORTA_DMA 0 +#define PORTB_DMA 1 + +static int fsi_get_dma_chan(void) +{ + if (0 != request_dma(PORTA_DMA, "fsia")) + return -EIO; + + if (0 != request_dma(PORTB_DMA, "fsib")) { + free_dma(PORTA_DMA); + return -EIO; + } + + master->fsia.dma_chan = PORTA_DMA; + master->fsib.dma_chan = PORTB_DMA; + + return 0; +} + +static void fsi_free_dma_chan(void) +{ + dma_wait_for_completion(PORTA_DMA); + dma_wait_for_completion(PORTB_DMA); + free_dma(PORTA_DMA); + free_dma(PORTB_DMA); + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; +} + +/************************************************************************ + + + ctrl function + + +************************************************************************/ +static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, data); + fsi_master_mask_set(IEMSK, data, data); +} + +static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, 0); + fsi_master_mask_set(IEMSK, data, 0); +} + +static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) +{ + u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4); + + if (enable) + fsi_master_mask_set(CLK_RST, val, val); + else + fsi_master_mask_set(CLK_RST, val, 0); +} + +static void fsi_irq_init(struct fsi_priv *fsi, int is_play) +{ + u32 data; + u32 ctrl; + + data = fsi_port_ab_io_bit(fsi, is_play); + ctrl = is_play ? DOFF_CTL : DIFF_CTL; + + /* set IMSK */ + fsi_irq_disable(fsi, is_play); + + /* set interrupt generation factor */ + fsi_reg_write(fsi, ctrl, IRQ_HALF); + + /* clear FIFO */ + fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR); + + /* clear interrupt factor */ + fsi_master_mask_set(INT_ST, data, 0); +} + +static void fsi_soft_all_reset(void) +{ + u32 status = fsi_master_read(SOFT_RST); + + /* port AB reset */ + status &= 0x000000ff; + fsi_master_write(SOFT_RST, status); + mdelay(10); + + /* soft reset */ + status &= 0x000000f0; + fsi_master_write(SOFT_RST, status); + status |= 0x00000001; + fsi_master_write(SOFT_RST, status); + mdelay(10); +} + +static void fsi_16data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u16 *dma_start; + u32 snd; + int i; + + /* get dma start position for FSI */ + dma_start = (u16 *)runtime->dma_area; + dma_start += fsi->byte_offset / 2; + + /* + * soft dma + * FSI can not use DMA when 16bpp + */ + for (i = 0; i < send; i++) { + snd = (u32)dma_start[i]; + fsi_reg_write(fsi, DODT, snd << 8); + } +} + +static void fsi_32data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u32 *dma_start; + + /* get dma start position for FSI */ + dma_start = (u32 *)runtime->dma_area; + dma_start += fsi->byte_offset / 4; + + dma_wait_for_completion(fsi->dma_chan); + dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR)); + dma_write(fsi->dma_chan, (u32)dma_start, + (u32)(fsi->base + DODT), send * 4); +} + +/* playback interrupt */ +static int fsi_data_push(struct fsi_priv *fsi) +{ + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream = NULL; + int send; + int fifo_free; + int width; + + if (!fsi || + !fsi->substream || + !fsi->substream->runtime) + return -EINVAL; + + runtime = fsi->substream->runtime; + + /* FSI FIFO has limit. + * So, this driver can not send periods data at a time + */ + if (fsi->byte_offset >= + fsi->period_len * (fsi->periods + 1)) { + + substream = fsi->substream; + fsi->periods = (fsi->periods + 1) % runtime->periods; + + if (0 == fsi->periods) + fsi->byte_offset = 0; + } + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + /* get send size for alsa */ + send = (fsi->buffer_len - fsi->byte_offset) / width; + + /* get FIFO free size */ + fifo_free = (fsi->fifo_max * fsi->chan) - fsi_get_fifo_residue(fsi, 1); + + /* size check */ + if (fifo_free < send) + send = fifo_free; + + if (2 == width) + fsi_16data_push(fsi, runtime, send); + else if (4 == width) + fsi_32data_push(fsi, runtime, send); + else + return -EINVAL; + + fsi->byte_offset += send * width; + + fsi_irq_enable(fsi, 1); + + if (substream) + snd_pcm_period_elapsed(substream); + + return 0; +} + +static irqreturn_t fsi_interrupt(int irq, void *data) +{ + u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; + u32 int_st = fsi_master_read(INT_ST); + + /* clear irq status */ + fsi_master_write(SOFT_RST, status); + fsi_master_write(SOFT_RST, status | 0x00000010); + + if (int_st & INT_A_OUT) + fsi_data_push(&master->fsia); + if (int_st & INT_B_OUT) + fsi_data_push(&master->fsib); + + fsi_master_write(INT_ST, 0x0000000); + + return IRQ_HANDLED; +} + +/************************************************************************ + + + dai ops + + +************************************************************************/ +static int fsi_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + const char *msg; + u32 flags = fsi_get_info_flags(fsi); + u32 fmt; + u32 reg; + u32 data; + int is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int is_master; + int ret = 0; + + if (1 == atomic_inc_return(&master->user)) + clk_enable(master->clk); + + /* CKG1 */ + data = is_play ? (1 << 0) : (1 << 4); + is_master = fsi_is_master_mode(fsi, is_play); + if (is_master) + fsi_reg_mask_set(fsi, CKG1, data, data); + else + fsi_reg_mask_set(fsi, CKG1, data, 0); + + /* clock inversion (CKG2) */ + data = 0; + switch (SH_FSI_INVERSION_MASK & flags) { + case SH_FSI_LRM_INV: + data = 1 << 12; + break; + case SH_FSI_BRM_INV: + data = 1 << 8; + break; + case SH_FSI_LRS_INV: + data = 1 << 4; + break; + case SH_FSI_BRS_INV: + data = 1 << 0; + break; + } + fsi_reg_write(fsi, CKG2, data); + + /* do fmt, di fmt */ + data = 0; + reg = is_play ? DO_FMT : DI_FMT; + fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags); + switch (fmt) { + case SH_FSI_FMT_MONO: + msg = "MONO"; + data = CR_FMT(CR_MONO); + fsi->chan = 1; + break; + case SH_FSI_FMT_MONO_DELAY: + msg = "MONO Delay"; + data = CR_FMT(CR_MONO_D); + fsi->chan = 1; + break; + case SH_FSI_FMT_PCM: + msg = "PCM"; + data = CR_FMT(CR_PCM); + fsi->chan = 2; + break; + case SH_FSI_FMT_I2S: + msg = "I2S"; + data = CR_FMT(CR_I2S); + fsi->chan = 2; + break; + case SH_FSI_FMT_TDM: + msg = "TDM"; + data = CR_FMT(CR_TDM) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + case SH_FSI_FMT_TDM_DELAY: + msg = "TDM Delay"; + data = CR_FMT(CR_TDM_D) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + default: + dev_err(dai->dev, "unknown format.\n"); + return -EINVAL; + } + + switch (fsi->chan) { + case 1: + fsi->fifo_max = 256; + break; + case 2: + fsi->fifo_max = 128; + break; + case 3: + case 4: + fsi->fifo_max = 64; + break; + case 5: + case 6: + case 7: + case 8: + fsi->fifo_max = 32; + break; + default: + dev_err(dai->dev, "channel size error.\n"); + return -EINVAL; + } + + fsi_reg_write(fsi, reg, data); + dev_info(dai->dev, "use %s format (%d channel) use %d DMAC\n", + msg, fsi->chan, fsi->dma_chan); + + /* + * clear clk reset if master mode + */ + if (is_master) + fsi_clk_ctrl(fsi, 1); + + /* irq setting */ + fsi_irq_init(fsi, is_play); + + return ret; +} + +static void fsi_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + fsi_irq_disable(fsi, is_play); + fsi_clk_ctrl(fsi, 0); + + if (atomic_dec_and_test(&master->user)) + clk_disable(master->clk); +} + +static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int ret = 0; + + /* capture not supported */ + if (!is_play) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + fsi_stream_push(fsi, substream, + frames_to_bytes(runtime, runtime->buffer_size), + frames_to_bytes(runtime, runtime->period_size)); + ret = fsi_data_push(fsi); + break; + case SNDRV_PCM_TRIGGER_STOP: + fsi_irq_disable(fsi, is_play); + fsi_stream_pop(fsi); + break; + } + + return ret; +} + +static struct snd_soc_dai_ops fsi_dai_ops = { + .startup = fsi_dai_startup, + .shutdown = fsi_dai_shutdown, + .trigger = fsi_dai_trigger, +}; + +/************************************************************************ + + + pcm ops + + +************************************************************************/ +static struct snd_pcm_hardware fsi_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE, + .formats = FSI_FMTS, + .rates = FSI_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int fsi_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &fsi_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + + return ret; +} + +static int fsi_pcm_close(struct snd_pcm_substream *substream) +{ + return 0; +} + +static int fsi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int fsi_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + long location; + + location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play); + if (location < 0) + location = 0; + + return bytes_to_frames(runtime, location); +} + +static struct snd_pcm_ops fsi_pcm_ops = { + .open = fsi_pcm_open, + .close = fsi_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsi_hw_params, + .hw_free = fsi_hw_free, + .pointer = fsi_pointer, +}; + +/************************************************************************ + + + snd_soc_platform + + +************************************************************************/ +#define PREALLOC_BUFFER (32 * 1024) +#define PREALLOC_BUFFER_MAX (32 * 1024) + +static void fsi_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int fsi_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + /* + * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel + * in MMAP mode (i.e. aplay -M) + */ + return snd_pcm_lib_preallocate_pages_for_all( + pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); +} + +/************************************************************************ + + + alsa struct + + +************************************************************************/ +struct snd_soc_dai fsi_soc_dai[] = { + { + .name = "FSI0", + .id = 0, + .playback = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, + /* capture not supported */ + .ops = &fsi_dai_ops, + } +}; +EXPORT_SYMBOL_GPL(fsi_soc_dai); + +struct snd_soc_platform fsi_soc_platform = { + .name = "fsi-pcm", + .pcm_ops = &fsi_pcm_ops, + .pcm_new = fsi_pcm_new, + .pcm_free = fsi_pcm_free, +}; +EXPORT_SYMBOL_GPL(fsi_soc_platform); + +/************************************************************************ + + + platform function + + +************************************************************************/ +static int fsi_probe(struct platform_device *pdev) +{ + struct resource *res; + char clk_name[8]; + unsigned int irq; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + irq = platform_get_irq(pdev, 0); + if (!res || !irq) { + dev_err(&pdev->dev, "Not enough FSI platform resources.\n"); + ret = -ENODEV; + goto exit; + } + + master = kzalloc(sizeof(*master), GFP_KERNEL); + if (!master) { + dev_err(&pdev->dev, "Could not allocate master\n"); + ret = -ENOMEM; + goto exit; + } + + master->base = ioremap_nocache(res->start, resource_size(res)); + if (!master->base) { + ret = -ENXIO; + dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n"); + goto exit_kfree; + } + + master->irq = irq; + master->info = pdev->dev.platform_data; + master->fsia.base = master->base; + master->fsib.base = master->base + 0x40; + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; + + ret = fsi_get_dma_chan(); + if (ret < 0) + goto exit_iounmap; + + /* FSI is based on SPU mstp */ + snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); + master->clk = clk_get(&pdev->dev, clk_name); + if (IS_ERR(master->clk)) { + dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); + ret = -EIO; + goto exit_free_dma; + } + + /* FSIA */ + fsi_soc_dai[0].dev = &pdev->dev; + + master->user = ATOMIC_INIT(0); + + fsi_soft_all_reset(); + + ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); + if (ret) { + dev_err(&pdev->dev, "irq request err\n"); + goto exit_free_dma; + } + + ret = snd_soc_register_platform(&fsi_soc_platform); + if (ret < 0) { + dev_err(&pdev->dev, "cannot snd soc register\n"); + goto exit_free_irq; + } + + return snd_soc_register_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + +exit_free_irq: + free_irq(irq, master); +exit_free_dma: + fsi_free_dma_chan(); +exit_iounmap: + iounmap(master->base); +exit_kfree: + kfree(master); + master = NULL; +exit: + return ret; +} + +static int fsi_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + snd_soc_unregister_platform(&fsi_soc_platform); + + clk_put(master->clk); + + fsi_free_dma_chan(); + + free_irq(master->irq, master); + + iounmap(master->base); + kfree(master); + master = NULL; + return 0; +} + +static struct platform_driver fsi_driver = { + .driver = { + .name = "sh_fsi", + }, + .probe = fsi_probe, + .remove = fsi_remove, +}; + +static int __init fsi_mobile_init(void) +{ + return platform_driver_register(&fsi_driver); +} + +static void __exit fsi_mobile_exit(void) +{ + platform_driver_unregister(&fsi_driver); +} +module_init(fsi_mobile_init); +module_exit(fsi_mobile_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.6.0.4

4 6

[alsa-devel] [PATCH v3 2/4] Add ak4642 codec support
by Kuninori Morimoto 20 Aug '09

20 Aug '09

This is very simple driver for ALSA It supprt headphone output and stereo input only This patch is tested by ms7724se Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v3 I'm so sorry. I took miss. v2 patch still use ak464x name. This patch modify name o ak464x -> ak4642 sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4642.c | 516 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4642.h | 25 +++ 4 files changed, 547 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4642.c create mode 100644 sound/soc/codecs/ak4642.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..ee794f1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4642 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF @@ -74,6 +75,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4642 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..c8c567e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4642-objs := ak4642.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(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 diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c new file mode 100644 index 0000000..1b439b4 --- /dev/null +++ b/sound/soc/codecs/ak4642.c @@ -0,0 +1,516 @@ +/* + * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * 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. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It can use headphone output / stereo input only + * + * AK4642 is not tested. + * AK4643 is tested. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ak4642.h" + +#define AK4642_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK4642_CACHEREGNUM 0x25 + +struct snd_soc_codec_device soc_codec_dev_ak4642; + +/* codec private data */ +struct ak4642_priv { + struct snd_soc_codec codec; + unsigned int sysclk; +}; + +static struct snd_soc_codec *ak4642_codec; + +/* + * ak4642 register cache + */ +static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak4642 register cache + */ +static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak4642_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak4642 register cache + */ +static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK4642 register space + */ +static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4642 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak4642_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak4642_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + r |= ak4642_write(codec, i, cache[i]); + + return r; +}; + +static int ak4642_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* + * start headphone output + * + * PLL, Master Mode + * Audio I/F Format :MSB justified (ADC & DAC) + * Sampling Frequency: 44.1kHz + * Digital Volume: −8dB + * Bass Boost Level : Middle + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p97. + * + * Example code use 0x39, 0x79 value for 0x01 address, + * But we need MCKO (0x02) bit now + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x0f, 0x09); + ak4642_write(codec, 0x0e, 0x19); + ak4642_write(codec, 0x09, 0x91); + ak4642_write(codec, 0x0c, 0x91); + ak4642_write(codec, 0x0a, 0x28); + ak4642_write(codec, 0x0d, 0x28); + ak4642_write(codec, 0x00, 0x64); + ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ + ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + } else { + /* + * start stereo input + * + * PLL Master Mode + * Audio I/F Format:MSB justified (ADC & DAC) + * Sampling Frequency:44.1kHz + * Pre MIC AMP:+20dB + * MIC Power On + * ALC setting:Refer to Table 35 + * ALC bit=“1” + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p94. + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x02, 0x05); + ak4642_write(codec, 0x06, 0x3c); + ak4642_write(codec, 0x08, 0xe1); + ak4642_write(codec, 0x0b, 0x00); + ak4642_write(codec, 0x07, 0x21); + ak4642_write(codec, 0x00, 0x41); + ak4642_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak4642_write(codec, 0x01, 0x3b); + ak4642_write(codec, 0x01, 0x0b); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x0e, 0x11); + ak4642_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x10, 0x00); + ak4642_write(codec, 0x07, 0x01); + } +} + +static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4642_priv *ak4642 = codec->private_data; + + ak4642->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak4642_dai_ops = { + .startup = ak4642_dai_startup, + .shutdown = ak4642_dai_shutdown, + .set_sysclk = ak4642_dai_set_sysclk, +}; + +struct snd_soc_dai ak4642_dai = { + .name = "AK4642", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .ops = &ak4642_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4642_dai); + +static int ak4642_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak4642_sync(codec); + return 0; +} + +/* + * initialise the AK4642 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4642_init(struct ak4642_priv *ak4642) +{ + struct snd_soc_codec *codec = &ak4642->codec; + int ret = 0; + + if (ak4642_codec) { + dev_err(codec->dev, "Another ak4642 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4642; + codec->name = "AK4642"; + codec->owner = THIS_MODULE; + codec->read = ak4642_read_reg_cache; + codec->write = ak4642_write; + codec->dai = &ak4642_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); + codec->reg_cache = kmemdup(ak4642_reg, + sizeof(ak4642_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + ak4642_dai.dev = codec->dev; + ak4642_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto reg_cache_err; + } + + ret = snd_soc_register_dai(&ak4642_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto reg_cache_err; + } + + /* + * clock setting + * + * Audio I/F Format: MSB justified (ADC & DAC) + * BICK frequency at Master Mode: 64fs + * Input Master Clock Select at PLL Mode: 11.2896MHz + * MCKO: Enable + * Sampling Frequency: 44.1kHz + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p89. + * + * please fix-me + */ + ak4642_write(codec, 0x01, 0x08); + ak4642_write(codec, 0x04, 0x4a); + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x01, 0x0b); + + return ret; + +reg_cache_err: + kfree(codec->reg_cache); + + return ret; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ak4642_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4642_priv *ak4642; + struct snd_soc_codec *codec; + int ret; + + ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); + if (!ak4642) + return -ENOMEM; + + codec = &ak4642->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, ak4642); + codec->control_data = i2c; + + ret = ak4642_init(ak4642); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK4642\n"); + + return ret; +} + +static int ak4642_i2c_remove(struct i2c_client *client) +{ + struct ak4642_priv *ak4642 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&ak4642_dai); + snd_soc_unregister_codec(&ak4642->codec); + kfree(ak4642->codec.reg_cache); + kfree(ak4642); + ak4642_codec = NULL; + + return 0; +} + +static const struct i2c_device_id ak4642_i2c_id[] = { + { "ak4642", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); + +static struct i2c_driver ak4642_i2c_driver = { + .driver = { + .name = "AK4642 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak4642_i2c_probe, + .remove = ak4642_i2c_remove, + .id_table = ak4642_i2c_id, +}; + +#endif + +static int ak4642_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret; + + if (!ak4642_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4642_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to register card\n"); + goto card_err; + } + + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; + +} + +/* power down chip */ +static int ak4642_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4642 = { + .probe = ak4642_probe, + .remove = ak4642_remove, + .resume = ak4642_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); + +static int __init ak4642_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&ak4642_i2c_driver); +#endif + return ret; + +} +module_init(ak4642_modinit); + +static void __exit ak4642_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4642_i2c_driver); +#endif + +} +module_exit(ak4642_exit); + +MODULE_DESCRIPTION("Soc AK4642 driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h new file mode 100644 index 0000000..28c267a --- /dev/null +++ b/sound/soc/codecs/ak4642.h @@ -0,0 +1,25 @@ +/* + * ak4642.h -- AK4642 Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based of ak4535.c + * + * 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 _AK4642_H +#define _AK4642_H + +struct ak4642_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ak4642_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4642; + +#endif -- 1.6.0.4

1 0

[alsa-devel] [PATCH v2 0/3] Add SuperH FSI driver support for ALSA
by Kuninori Morimoto 20 Aug '09

20 Aug '09

These patches are FSI driver for ALSA v2 1) Add SuperH FSI driver support for ALSA 2) Add ak464x codec support 3) Add FSI-AK4642 sound support for SuperH 4) sh: add FSI driver support for ms7724se Best regards -- Kuninori Morimoto

1 0

[alsa-devel] [PATCH 3/3] Add FSI-AK464x sound support for ms7724se
by Kuninori Morimoto 20 Aug '09

20 Aug '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- arch/sh/boards/Kconfig | 9 +++ arch/sh/boards/mach-se/7724/Makefile | 3 +- arch/sh/boards/mach-se/7724/fsi-ak464x.c | 90 ++++++++++++++++++++++++++++++ arch/sh/boards/mach-se/7724/setup.c | 90 ++++++++++++++++++++++++++++++ 4 files changed, 191 insertions(+), 1 deletions(-) create mode 100644 arch/sh/boards/mach-se/7724/fsi-ak464x.c diff --git a/arch/sh/boards/Kconfig b/arch/sh/boards/Kconfig index db04c85..44f4c8b 100644 --- a/arch/sh/boards/Kconfig +++ b/arch/sh/boards/Kconfig @@ -55,6 +55,15 @@ config SH_7724_SOLUTION_ENGINE Select 7724 SolutionEngine if configuring for a Hitachi SH7724 evaluation board. +config SND_SH4_FSI_AK464X + bool "FSI-AK464X sound support" + depends on SND_SOC_SH4_FSI + depends on SH_7724_SOLUTION_ENGINE + select SND_SOC_AK464X + help + This option enables generic sound support for the + FSI - AK464x unit of the SH4. + config SH_7751_SOLUTION_ENGINE bool "SolutionEngine7751" select SOLUTION_ENGINE diff --git a/arch/sh/boards/mach-se/7724/Makefile b/arch/sh/boards/mach-se/7724/Makefile index 349cbd6..e0f5366 100644 --- a/arch/sh/boards/mach-se/7724/Makefile +++ b/arch/sh/boards/mach-se/7724/Makefile @@ -7,4 +7,5 @@ # # -obj-y := setup.o irq.o \ No newline at end of file +obj-y := setup.o irq.o +obj-$(CONFIG_SND_SH4_FSI_AK464X) += fsi-ak464x.o diff --git a/arch/sh/boards/mach-se/7724/fsi-ak464x.c b/arch/sh/boards/mach-se/7724/fsi-ak464x.c new file mode 100644 index 0000000..bab118e --- /dev/null +++ b/arch/sh/boards/mach-se/7724/fsi-ak464x.c @@ -0,0 +1,90 @@ +/* + * FSI-AK464x sound support for ms7724se + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.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/io.h> + +#include <sound/fsi.h> +#include <../sound/soc/codecs/ak464x.h> + +static int machine_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link fsi_dai_link = { + .name = "AK464x", + .stream_name = "AK464x", + .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .codec_dai = &ak464x_dai, + .init = machine_init, + .ops = NULL, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "SH4 I2S (FSI)", + .platform = &fsi_soc_platform, + .dai_link = &fsi_dai_link, + .num_links = 1, +}; + +struct ak464x_setup_data ak464x_setup = { + .i2c_bus = 0, + .i2c_address = 0x12, /* 0x13 */ +}; + +static struct snd_soc_device fsi_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_ak464x, + .codec_data = &ak464x_setup, +}; + +static struct platform_device *fsi_snd_device; + +static int __init fsi_ak464x_init(void) +{ + int ret; + + ret = -ENOMEM; + fsi_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_snd_device) + goto out; + + platform_set_drvdata(fsi_snd_device, + &fsi_snd_devdata); + fsi_snd_devdata.dev = &fsi_snd_device->dev; + ret = platform_device_add(fsi_snd_device); + + if (ret) + platform_device_put(fsi_snd_device); + +out: + return ret; +} + +static void __exit fsi_ak464x_exit(void) +{ + platform_device_unregister(fsi_snd_device); +} + +module_init(fsi_ak464x_init); +module_exit(fsi_ak464x_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH4 FSI-AK464x sound card"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 9162081..0099a79 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -22,11 +22,13 @@ #include <linux/usb/r8a66597.h> #include <video/sh_mobile_lcdc.h> #include <media/sh_mobile_ceu.h> +#include <sound/fsi.h> #include <asm/io.h> #include <asm/heartbeat.h> #include <asm/sh_eth.h> #include <asm/clock.h> #include <asm/sh_keysc.h> +#include <asm/dma-sh.h> #include <cpu/sh7724.h> #include <mach-se/mach/se7724.h> @@ -246,6 +248,65 @@ static struct platform_device ceu1_device = { }, }; +/* FSI */ +/* + * FSI-A use external clock which came from ak464x. + * So, we should change parent of fsi + */ +#define FCLKACR 0xa4150008 +static void fsimck_init(struct clk *clk) +{ + u32 status = ctrl_inl(clk->enable_reg); + + /* use external clock */ + status &= ~0x000000ff; + status |= 0x00000080; + ctrl_outl(status, clk->enable_reg); +} + +static struct clk_ops fsimck_clk_ops = { + .init = fsimck_init, +}; + +static struct clk fsimcka_clk = { + .name = "fsimcka_clk", + .id = -1, + .ops = &fsimck_clk_ops, + .enable_reg = (void __iomem *)FCLKACR, + .rate = 0, /* unknown */ +}; + +struct sh_fsi_platform_info fsi_info = { + .porta_flags = SH_FSI_BRS_INV | + SH_FSI_OUT_SLAVE_MODE | + SH_FSI_IN_SLAVE_MODE | + SH_FSI_OFMT(PCM) | + SH_FSI_IFMT(PCM), +}; + +static struct resource fsi_resources[] = { + [0] = { + .name = "FSI", + .start = 0xFE3C0000, + .end = 0xFE3C021d, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = 108, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device fsi_device = { + .name = "sh_fsi", + .id = 0, + .num_resources = ARRAY_SIZE(fsi_resources), + .resource = fsi_resources, + .dev = { + .platform_data = &fsi_info, + }, +}; + /* KEYSC in SoC (Needs SW33-2 set to ON) */ static struct sh_keysc_info keysc_info = { .mode = SH_KEYSC_MODE_1, @@ -351,6 +412,7 @@ static struct platform_device *ms7724se_devices[] __initdata = { &keysc_device, &sh_eth_device, &sh7724_usb0_host_device, + &fsi_device, }; #define EEPROM_OP 0xBA206000 @@ -418,11 +480,13 @@ static void __init sh_eth_init(void) static int __init devices_setup(void) { u16 sw = ctrl_inw(SW4140); /* select camera, monitor */ + struct clk *fsia_clk; /* Reset Release */ ctrl_outw(ctrl_inw(FPGA_OUT) & ~((1 << 1) | /* LAN */ (1 << 6) | /* VIDEO DAC */ + (1 << 7) | /* AK4643 */ (1 << 12) | /* USB0 */ (1 << 14)), /* RMII */ FPGA_OUT); @@ -558,6 +622,32 @@ static int __init devices_setup(void) gpio_request(GPIO_FN_KEYOUT1, NULL); gpio_request(GPIO_FN_KEYOUT0, NULL); + /* enable FSI */ + gpio_request(GPIO_FN_FSIMCKB, NULL); + gpio_request(GPIO_FN_FSIMCKA, NULL); + gpio_request(GPIO_FN_FSIOASD, NULL); + gpio_request(GPIO_FN_FSIIABCK, NULL); + gpio_request(GPIO_FN_FSIIALRCK, NULL); + gpio_request(GPIO_FN_FSIOABCK, NULL); + gpio_request(GPIO_FN_FSIOALRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOAO, NULL); + gpio_request(GPIO_FN_FSIIBSD, NULL); + gpio_request(GPIO_FN_FSIOBSD, NULL); + gpio_request(GPIO_FN_FSIIBBCK, NULL); + gpio_request(GPIO_FN_FSIIBLRCK, NULL); + gpio_request(GPIO_FN_FSIOBBCK, NULL); + gpio_request(GPIO_FN_FSIOBLRCK, NULL); + gpio_request(GPIO_FN_CLKAUDIOBO, NULL); + gpio_request(GPIO_FN_FSIIASD, NULL); + + /* change parent of FSI A */ + fsia_clk = clk_get(NULL, "fsia_clk"); + clk_register(&fsimcka_clk); + clk_set_parent(fsia_clk, &fsimcka_clk); + clk_set_rate(fsia_clk, 11000); + clk_set_rate(&fsimcka_clk, 11000); + clk_put(fsia_clk); + /* * enable SH-Eth * -- 1.6.0.4

3 9

[alsa-devel] unable to do simultaneous playback and recording
by prashant chopade 20 Aug '09

20 Aug '09

Hello, We using linux-2.6.27 on target board-AT91SAM9263EK. We are using ALSA library utilities (aplay & arecord). But by using these utilities we can’t do simultaneous playback and recording. We performed following activity to come to this conclusion: //At the background we were recording #arecord –f cd record_test_file & //At the forefront we were playing pre-recorded audio file #aplay –f cd pre_recorded_file Our observation is that, recording stops when start playing file. Kindly need your guidance for simultaneous playback and recording of an audio file. Appreciates your help if you provide sample test programs (using ALSA API) for recording. Thanks & Regards, Prashant C. Love Cricket? Check out live scores, photos, video highlights and more. Click here http://cricket.yahoo.com

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Re: [alsa-devel] unable to do simultaneous playback and recording
by prashant chopade 20 Aug '09

20 Aug '09

Hello, My understandings: >Define your meaning of "simultaneous playback and recording" more >exactly. I mean: Playing audio file stored on disk on speaker and at the same time recording audio from MIC and storing it on disk. I don,t want to do multiplayback, I mean i don,t want to play two or more audio files at a time. So you are saying that my driver(ATMEL-AT91SAM9263-AC97c) is not supporting the full duplex mode? Thanks & Regards, Prashant C. >prashant chopade wrote: > > Hello, > > As per discussion, I know that ALSA driver does not have support for > simultaneous playback and recording. Huh? The full-duplex is of course supported. Your example shown in the thread doesn't indicate the multi playbacks or so. > We are using AD1981B codec, so our hardware has a support for > simultaneous playback and recording of an audio. I think you are confused. > So what exactly I have to do to get a support for simultaneous > playback and recording? > Shall I have to use sound server demons? > > Is it really possible to do simultaneous playback and recording in > linux? Define your meaning of "simultaneous playback and recording" more exactly. If it's about full-duplex, the answer is definitely yes. If it's multi-streams, the answer is yes, but there are choices, e.g. dmix/dsnoop plugin or a sound server. Takashi > prashant chopade wrote: > > > > > > > > > > > > Hello, > > > > We using linux-2.6.27 on target board-AT91SAM9263EK. > > We are using ALSA library utilities (aplay & > > arecord). But by using these utilities we can’t do simultaneous playback and > > recording. > > > > > > > > We performed following activity to come to this conclusion: > > > > > > > > //At the background we were recording > > > > #arecord –f cd record_test_file & > > > > > > > > //At the forefront we were playing pre-recorded audio file > > > > #aplay –f cd pre_recorded_file > > > > > > > > Our observation is that, recording stops when start playing > > file. > > In the API side, the above should work fine. It must be the driver > that doesn't work with full-duplex. > > > Takashi Yahoo! recommends that you upgrade to the new and safer Internet Explorer 8. http://downloads.yahoo.com/in/internetexplorer/

2 1

Re: [alsa-devel] Nvidia MSI Geforce 9500
by Clemens Ladisch 20 Aug '09

20 Aug '09

Shane W wrote: > think this might belong on the devel list. After some > searching, looks like my Geforce 9500 card is not > recognized by the hda_intel driver. I tried adding: > { PCI_DEVICE(0x10de, 0x0640), .driver_data = AZX_DRIVER_NVIDIA }, > >> lspci -vvv: >> 01:00.0 VGA compatible controller: nVidia Corporation G96 [GeForce 9500 GT] >> (rev a1) (prog-if 00 [VGA controller]) This PCI ID is for the video part; the HDMI interface should have a second entry in the lspci output. HTH Clemens

1 0

[alsa-devel] Using 8bit mono at 8000Hz with arecord and aplay
by Shilpa Kedar Walvekar 20 Aug '09

20 Aug '09

Hello, We tried to record and play the recorded file using arecord and aplay as follows: arecord -f cd /home/test.wav aplay -f cd /home/test.wav The file gets recorded and played correctly. But if I try to record file of format S8 or U8, I get error: # arecord -f U8 /home/test_alsa_record_2 Recording WAVE '/home/test_alsa_record_2' : Unsigned 8 bit, Rate 8000 Hz, Mono arecord: set_params:979: Sample format non available # arecord -f S8 /home/alsa_record_2 Recording WAVE '/home/alsa_record_2' : Signed 8 bit, Rate 8000 Hz, Mono arecord: set_params:979: Sample format non available Is there any other way to set the format as S8 or U8? Or only cd/cdr/dat formats can be played using alsa? Regards, --Shilpa.

3 4

[alsa-devel] Support for ALC662 on FUJITSU SIEMENS AMILO Li3710
by Mads Kiilerich 20 Aug '09

20 Aug '09

On a friends FUJITSU SIEMENS AMILO Li3710 10601011427 sound didn't work with kernel-PAE-2.6.29.6-213.fc11.i686 Out of the box we get no sound and HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 hda_codec: Unknown model for ALC662, trying auto-probe from BIOS... ALSA sound/pci/hda/hda_codec.c:3507: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3511: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3515: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3516: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3524: inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x0, aux=0x0 But with /etc/modprobe.d/local.conf (from http://ubuntuforums.org/archive/index.php/t-940689.html) options snd-hda-intel model=3stack-dig options snd-hda-intel single_cmd=1 options snd-hda-intel enable_msi=1 we get sound and HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: irq 32 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 ALSA sound/pci/hda/hda_codec.c:2698: hda_codec: model '3stack-dig' is selected I would like to have this fixed so Linux sound works on my friends machine without manual configuration. After having read some documentation I assume that single_cmd and enable_msi perhaps shouldn't be needed. I will test that next time I get access to that machine. HAL reports the sound card as /org/freedesktop/Hal/devices/pci_8086_293e, so should this patch be sufficient? diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e1563f4..0385353 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16699,6 +16699,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x8086, 0x293e, "FUJITSU SIEMENS AMILO Li3710", + ALC662_3ST_2ch_DIG), {} }; Untested and hard to test because I don't have direct access to the machine. This issue was also reported on https://bugzilla.redhat.com/show_bug.cgi?id=487889 Please CC me. /Mads

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Re: [alsa-devel] [PATCH] ALSA: hda: partially revert ALC268 quirk collapse
by Takashi Iwai 19 Aug '09

19 Aug '09

At Sat, 15 Aug 2009 20:53:38 +0400, Alexey Dobriyan wrote: > > On Sat, Aug 15, 2009 at 05:41:04PM +0200, Takashi Iwai wrote: > > At Sat, 15 Aug 2009 16:38:46 +0400, > > Alexey Dobriyan wrote: > > > > > > Fix alsamixer inability to start: > > > > > > ALSA lib simple_none.c:1491:(simple_add1) helem (MIXER,'Master Playback Switch',0,3,0) appears twice or more > > > > > > ALC268 codec shipped with Toshiba Satellite X200 has ids 1179:ff00 and > > > began to match quirk table. > > > > > > If it doesn't match and auto-probe from BIOS is used, alsamixer works: > > > > > > hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... > > > > > > 8871e5b91518a47284b6bc2603b44dbc79c85446 is first bad commit > > > commit 8871e5b91518a47284b6bc2603b44dbc79c85446 > > > Author: Takashi Iwai <tiwai(a)suse.de> > > > Date: Tue Jun 2 01:02:50 2009 +0200 > > > > > > ALSA: hda - Reorder and clean-up ALC268 quirk table > > > > > > Rearrange alc268_cfg_tbl[] in the order of vendor id, and group some > > > entries using SND_PCI_QUIRK_MASK(). > > > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > > > > > :040000 040000 2497bb6fc5305acf3191b8e9b82956ec0ea1aee9 6f1d9ff9fed20550a1b2b398c7d2d5fec705fcd9 M sound > > > > > > Signed-off-by: Alexey Dobriyan <adobriyan(a)gmail.com> > > > > Well, it's no right fix. The symptom implies that something wrong > > with the current model=toshiba, so fixing the quirk is the way to go > > at this moment. > > > > Could you give alsa-info.sh output (run with --no-upload option) > > as vanilla kernel? Run with both model=auto and without model option > > and attach both results to compare. The former means actually the > > condition as your patch does. > > Here is diff for quick look: Thanks. The problem is that this laptop has two ALC268 codecs, and these conflict. Could you try the patch below? thanks, Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fea9767..6f683e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12521,8 +12521,6 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), - SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", - ALC268_TOSHIBA), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), @@ -12530,6 +12528,15 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { {} }; +/* Toshiba laptops have no unique PCI SSID but only codec SSID */ +static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { + SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO), + SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO), + SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", + ALC268_TOSHIBA), + {} +}; + static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, @@ -12696,6 +12703,10 @@ static int patch_alc268(struct hda_codec *codec) alc268_models, alc268_cfg_tbl); + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) + board_config = snd_hda_check_board_codec_sid_config(codec, + ALC882_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl); + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { printk(KERN_INFO "hda_codec: Unknown model for %s, " "trying auto-probe from BIOS...\n", codec->chip_name);

2 2

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 19 Aug '09

19 Aug '09

The following changes since commit d3c9e9a1390f8a34da8b69e09fa1afa90f5067f4: Mark Brown (1): ASoC: Implement TDM configuration for WM8993 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Joonyoung Shim (1): ASoC: Fix data format configuration for S3C64XX IISv2 Mark Brown (3): ASoC: Reenable S3C64xx I2S support ASoC: Remove duplicate ADC/DAC widgets from wm_hubs.c ASoC: Provide default set_bias_level() implementation arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 5 +++++ sound/soc/codecs/wm_hubs.c | 6 ------ sound/soc/s3c24xx/Kconfig | 2 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 6 +++--- sound/soc/soc-dapm.c | 8 ++++++-- 5 files changed, 15 insertions(+), 12 deletions(-)

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[alsa-devel] [PATCH 2/3] Add ak464x codec support
by Kuninori Morimoto 19 Aug '09

19 Aug '09

This is very simple driver for ALSA It supprt headphone output and stereo input only Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak464x.c | 525 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak464x.h | 25 +++ 4 files changed, 556 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak464x.c create mode 100644 sound/soc/codecs/ak464x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..f66689c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK464X if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF @@ -74,6 +75,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK464X + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..4400ad4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak464x-objs := ak464x.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK464X) += snd-soc-ak464x.o obj-$(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 diff --git a/sound/soc/codecs/ak464x.c b/sound/soc/codecs/ak464x.c new file mode 100644 index 0000000..d2ef80d --- /dev/null +++ b/sound/soc/codecs/ak464x.c @@ -0,0 +1,525 @@ +/* + * ak464x.c -- AK464x ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based of ak4535.c + * + * Copyright 2005 Openedhand Ltd. + * Author: Richard Purdie <richard(a)openedhand.com> + * + * Based on wm8753.c by Liam Girdwood + * + * 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. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It is able to use headphone output + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ak464x.h" + +#define AK464X_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK464X_CACHEREGNUM 0x25 + +#define DPRINTK(param...) printk(param) + +struct snd_soc_codec_device soc_codec_dev_ak464x; + +/* codec private data */ +struct ak464x_priv { + unsigned int sysclk; +}; + +/* + * ak464x register cache + */ +static const u16 ak464x_reg[AK464X_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak464x register cache + */ +static inline unsigned int ak464x_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK464X_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak464x_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak464x register cache + */ +static inline void ak464x_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK464X_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK464X register space + */ +static int ak464x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK464X register + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak464x_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak464x_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK464X_CACHEREGNUM; i++) + r |= ak464x_write(codec, i, cache[i]); + + return r; +}; + +static int ak464x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + codec->bias_level = level; + return 0; +} + +static int ak464x_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* start headphone output */ + ak464x_write(codec, 0x0f, 0x09); + ak464x_write(codec, 0x0e, 0x19); + ak464x_write(codec, 0x09, 0x91); + ak464x_write(codec, 0x0c, 0x91); + ak464x_write(codec, 0x0a, 0x28); + ak464x_write(codec, 0x0d, 0x28); + ak464x_write(codec, 0x00, 0x64); + ak464x_write(codec, 0x01, 0x3b); + ak464x_write(codec, 0x01, 0x7b); + } else { + /* start stereo input */ + ak464x_write(codec, 0x02, 0x05); + ak464x_write(codec, 0x06, 0x3c); + ak464x_write(codec, 0x08, 0xe1); + ak464x_write(codec, 0x0b, 0x00); + ak464x_write(codec, 0x07, 0x21); + ak464x_write(codec, 0x00, 0x41); + ak464x_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak464x_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak464x_write(codec, 0x01, 0x3b); + ak464x_write(codec, 0x01, 0x0b); + ak464x_write(codec, 0x00, 0x40); + ak464x_write(codec, 0x0e, 0x11); + ak464x_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak464x_write(codec, 0x00, 0x40); + ak464x_write(codec, 0x10, 0x00); + ak464x_write(codec, 0x07, 0x01); + } +} + +static int ak464x_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak464x_priv *ak464x = codec->private_data; + + ak464x->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak464x_dai_ops = { + .startup = ak464x_dai_startup, + .shutdown = ak464x_dai_shutdown, + .set_sysclk = ak464x_dai_set_sysclk, +}; + +struct snd_soc_dai ak464x_dai = { + .name = "AK464X", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,}, + .ops = &ak464x_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak464x_dai); + +static int ak464x_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; + + ak464x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ak464x_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak464x_sync(codec); + ak464x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ak464x_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the AK464X driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak464x_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + int ret = 0; + + codec->name = "AK464X"; + codec->owner = THIS_MODULE; + codec->read = ak464x_read_reg_cache; + codec->write = ak464x_write; + codec->set_bias_level = ak464x_set_bias_level; + codec->dai = &ak464x_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(ak464x_reg); + codec->reg_cache = kmemdup(ak464x_reg, + sizeof(ak464x_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak464x: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + ak464x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak464x: failed to register card\n"); + goto card_err; + } + + /* clock setting */ + ak464x_write(codec, 0x01, 0x08); + ak464x_write(codec, 0x04, 0x4a); + ak464x_write(codec, 0x05, 0x27); + ak464x_write(codec, 0x00, 0x40); + ak464x_write(codec, 0x01, 0x0b); + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + + return ret; +} + +static struct snd_soc_device *ak464x_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static int ak464x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = ak464x_socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = ak464x_init(socdev); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK464X\n"); + + return ret; +} + +static int ak464x_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id ak464x_i2c_id[] = { + { "ak464x", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak464x_i2c_id); + +static struct i2c_driver ak464x_i2c_driver = { + .driver = { + .name = "AK464X I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak464x_i2c_probe, + .remove = ak464x_i2c_remove, + .id_table = ak464x_i2c_id, +}; + +static int ak464x_add_i2c_device(struct platform_device *pdev, + const struct ak464x_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&ak464x_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "ak464x", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&ak464x_i2c_driver); + return -ENODEV; +} +#endif + +static int ak464x_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ak464x_setup_data *setup; + struct snd_soc_codec *codec; + struct ak464x_priv *ak464x; + int ret; + + printk(KERN_INFO "AK464X Audio Codec %s", AK464X_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ak464x = kzalloc(sizeof(struct ak464x_priv), GFP_KERNEL); + if (ak464x == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ak464x; + socdev->card->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ak464x_socdev = socdev; + ret = -ENODEV; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + ret = ak464x_add_i2c_device(pdev, setup); + } +#endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } + + return ret; +} + +/* power down chip */ +static int ak464x_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec->control_data) + ak464x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); + i2c_del_driver(&ak464x_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak464x = { + .probe = ak464x_probe, + .remove = ak464x_remove, + .suspend = ak464x_suspend, + .resume = ak464x_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak464x); + +static int __init ak464x_modinit(void) +{ + return snd_soc_register_dai(&ak464x_dai); +} +module_init(ak464x_modinit); + +static void __exit ak464x_exit(void) +{ + snd_soc_unregister_dai(&ak464x_dai); +} +module_exit(ak464x_exit); + +MODULE_DESCRIPTION("Soc AK464x driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak464x.h b/sound/soc/codecs/ak464x.h new file mode 100644 index 0000000..07cd1bf --- /dev/null +++ b/sound/soc/codecs/ak464x.h @@ -0,0 +1,25 @@ +/* + * ak464x.h -- AK464x Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * Based of ak4535.c + * + * 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 _AK464X_H +#define _AK464X_H + +struct ak464x_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ak464x_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak464x; + +#endif -- 1.6.0.4

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[alsa-devel] [PATCH] snd-usb-audio: bootup sequence provided for new device - Digidesign Mbox 2
by Damien Zammit 19 Aug '09

19 Aug '09

Digidesign Mbox 2 support for Linux - Damien Zammit 19/08/2009 dczammit@flop:~$ dmesg [132064.931715] ALSA sound/usb/usbaudio.c:3279: Sending Digidesign Mbox 2 boot sequence... [132064.934474] ALSA sound/usb/usbaudio.c:3306: device initialised! [132064.937473] ALSA sound/usb/usbaudio.c:3315: mbox2_boot: new boot length = 646 [132064.945769] ALSA sound/usb/usbaudio.c:2814: 36:2:2: add audio endpoint 0x3 [132064.950765] ALSA sound/usb/usbaudio.c:2814: 36:2:3: add audio endpoint 0x3 [132064.957314] ALSA sound/usb/usbaudio.c:2814: 36:3:2: add audio endpoint 0x4 [132064.961622] ALSA sound/usb/usbaudio.c:2814: 36:3:3: add audio endpoint 0x4 [132064.964575] ALSA sound/usb/usbaudio.c:2814: 36:3:6: add audio endpoint 0x4 [132064.971046] ALSA sound/usb/usbaudio.c:2814: 36:4:2: add audio endpoint 0x85 [132064.973545] ALSA sound/usb/usbaudio.c:2814: 36:4:3: add audio endpoint 0x85 [132064.979576] ALSA sound/usb/usbaudio.c:2814: 36:5:2: add audio endpoint 0x86 [132064.982522] ALSA sound/usb/usbaudio.c:2814: 36:5:3: add audio endpoint 0x86 [132064.986584] ALSA sound/usb/usbmidi.c:1357: created 1 output and 1 input ports [132064.986955] usbcore: registered new interface driver snd-usb-audio dczammit@flop:~$ dczammit@flop:~$ ls /proc/asound/M2/ id midi0 pcm0c pcm0p pcm1c pcm1p pcm2p stream0 stream1 stream2 usbbus usbid dczammit@flop:~$ dczammit@flop:~$ cat /proc/asound/M2/stream0 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed : USB Audio Playback: Status: Stop Interface 2 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Interface 2 Altset 3 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Capture: Status: Stop Interface 5 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 6 IN (ASYNC) Rates: 8000 - 48000 (continuous) Interface 5 Altset 3 Format: 0x20 (24 bits) Channels: 2 Endpoint: 6 IN (SYNC) Rates: 8000 - 48000 (continuous) dczammit@flop:~$ dczammit@flop:~$ cat /proc/asound/M2/stream1 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed : USB Audio #1 Playback: Status: Stop Interface 3 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 4 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Interface 3 Altset 3 Format: 0x20 (24 bits) Channels: 2 Endpoint: 4 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Capture: Status: Stop Interface 4 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 8000 - 48000 (continuous) Interface 4 Altset 3 Format: 0x20 (24 bits) Channels: 2 Endpoint: 5 IN (SYNC) Rates: 8000 - 48000 (continuous) dczammit@flop:~$ dczammit@flop:~$ cat /proc/asound/M2/stream2 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed : USB Audio #2 Playback: Status: Stop Interface 3 Altset 6 Format: 0x2 (16 bits) Channels: 2 Endpoint: 4 OUT (SYNC) Rates: 8000 - 48000 (continuous) dczammit@flop:~$ I fixed the broken pipes!! I can get speaker-test working, but I dont get any sound... 1) Is format 0x20 (S24_3LE) supported yet? I can't get it to work with jackd version 0.116.2 2) Which interfaces/altsettings should I disable? I can't quite get the sound to come out. Damien

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[alsa-devel] [PATCH 0/3] Add SuperH FSI driver support for ALSA
by Kuninori Morimoto 19 Aug '09

19 Aug '09

These patches are ALSA driver support for SuperH SH7724 1: Add SuperH FSI driver support for ALSA 2: Add ak464x codec support for ALSA 3: Add FSI-AK464x sound support for ms7724se 1 and 2 are very simple driver and these for ALSA. 3 is FSI-AK464x combination patch for ms7724se board. Best regards -- Kuninori Morimoto

1 0