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

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