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[alsa-devel] NXP Semiconductors TFA9879 Amplifier Driver
by Peter Rosin 09 Nov '14

09 Nov '14

Hi! Sorry for not sending this from my axentia.se account, but I tend to get high spam-scores from there when I use git send-email. This is a new driver, and it's pretty minimalistic with support for only a few basic controls. However, it is usable and I'd be happy to see it included. I don't know if it's presumptious to add myself as a maintainer, but checkpatch complained... Are the alsa control names wisely chosen? Please review! Cheers, Peter

5 15

[alsa-devel] [PATCH 1/2] ASoC: Forward calls to snd_soc_cache_sync() to regcache_sync()
by Lars-Peter Clausen 09 Nov '14

09 Nov '14

For convenience for drivers that do not want to keep their own pointer to regmap struct around forward calls to snd_soc_cache_sync() to regcache_sync() if the driver is using regmap. This is similar to what we do for snd_soc_read()/snd_soc_write(). This patch also fixes drivers which already have been converted to regmap, but still use snd_soc_cache_sync() for trying to the sync the cache. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Unfortunately there are a few drivers that have already been converted to regmap, but still use snd_soc_cache_sync(). snd_soc_cache_sync() currently is a noop for those drivers. Some of these conversions were done as early as 3.4 and while this patch applies cleanly to earlier kernel versions, due to the restructuring of the ASoC core it will only work on 3.16+. Hence no Fixes tag. --- sound/soc/soc-cache.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index a9f82b5..6dab817 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -187,6 +187,9 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) const char *name = "flat"; int ret; + if (codec->component.regmap) + return regcache_sync(codec->component.regmap); + if (!codec->cache_sync) return 0; -- 1.8.0

2 2

[alsa-devel] [PATCH 0/2 v2] Digidesign Mbox 1 clock source and duplex mode
by damien.zammit＠gmail.com 09 Nov '14

09 Nov '14

From: Damien Zammit <damien(a)zamaudio.com> This is a cleaned up patchset version 2. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/card.h | 5 ++ sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 133 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 73 +++++++++++++++++--------- sound/usb/quirks.c | 30 +++++++++++ sound/usb/usbaudio.h | 1 + 6 files changed, 227 insertions(+), 24 deletions(-) -- 1.9.1

1 2

[alsa-devel] [PATCH 0/2] ASoC: More AC'97 cleanups
by Lars-Peter Clausen 08 Nov '14

08 Nov '14

More AC'97 cleanups in preparation for the conversion to regmap. * The mioa701_wm9713 directly accesses the read/write callback of the CODEC driver which will break when converting the driver to regmap. * Re-worked version of the wm9712/wm9713 virtual register removal, this time using custom put/get handlers to keep the control names. - Lars Lars-Peter Clausen (2): ASoC: mioa701_wm9713: Don't opencode CODEC register access ASoC: wm9712/wm9713: Replace virtual registers with custom put/get callbacks sound/soc/codecs/wm9712.c | 163 ++++++++++++++++++++++++++--------------- sound/soc/codecs/wm9713.c | 153 ++++++++++++++++++++++---------------- sound/soc/pxa/mioa701_wm9713.c | 7 +- 3 files changed, 196 insertions(+), 127 deletions(-) -- 1.8.0

3 6

[alsa-devel] newbie at helping alsa
by John Houston 08 Nov '14

08 Nov '14

I am a retired engineer with some years of experience. I would like to help with the ALC892 portion of the project. I have an ALC892 in my computer and would like to use it. I also have an application for it.

1 0

[alsa-devel] [PATCH] ALSA: pcm: Add snd_pcm_stop_xrun() helper
by Takashi Iwai 07 Nov '14

07 Nov '14

Add a new helper function snd_pcm_stop_xrun() to the standard sequnce lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the existing open codes with this helper. The function checks the PCM running state to prevent setting the wrong state, too, for more safety. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/media/pci/saa7134/saa7134-alsa.c | 4 +--- include/sound/pcm.h | 1 + sound/arm/pxa2xx-pcm-lib.c | 4 +--- sound/core/pcm_native.c | 23 +++++++++++++++++++++++ sound/firewire/amdtp.c | 8 ++------ sound/firewire/isight.c | 10 ++-------- sound/pci/asihpi/asihpi.c | 5 +---- sound/pci/atiixp.c | 4 +--- sound/pci/atiixp_modem.c | 4 +--- sound/soc/atmel/atmel-pcm-dma.c | 4 +--- sound/soc/fsl/fsl_dma.c | 9 +-------- sound/usb/6fire/pcm.c | 17 ++++------------- sound/usb/endpoint.c | 4 +--- sound/usb/misc/ua101.c | 18 ++++-------------- sound/usb/usx2y/usbusx2yaudio.c | 9 ++------- 15 files changed, 46 insertions(+), 78 deletions(-) diff --git a/drivers/media/pci/saa7134/saa7134-alsa.c b/drivers/media/pci/saa7134/saa7134-alsa.c index 40569894c1c9..ac3cd74e824e 100644 --- a/drivers/media/pci/saa7134/saa7134-alsa.c +++ b/drivers/media/pci/saa7134/saa7134-alsa.c @@ -173,9 +173,7 @@ static void saa7134_irq_alsa_done(struct saa7134_dev *dev, dprintk("irq: overrun [full=%d/%d] - Blocks in %d\n",dev->dmasound.read_count, dev->dmasound.bufsize, dev->dmasound.blocks); spin_unlock(&dev->slock); - snd_pcm_stream_lock(dev->dmasound.substream); - snd_pcm_stop(dev->dmasound.substream,SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(dev->dmasound.substream); + snd_pcm_stop_xrun(dev->dmasound.substream); return; } diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0b8daeed0a33..40289ec2451c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -506,6 +506,7 @@ int snd_pcm_status(struct snd_pcm_substream *substream, int snd_pcm_start(struct snd_pcm_substream *substream); int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status); int snd_pcm_drain_done(struct snd_pcm_substream *substream); +int snd_pcm_stop_xrun(struct snd_pcm_substream *substream); #ifdef CONFIG_PM int snd_pcm_suspend(struct snd_pcm_substream *substream); int snd_pcm_suspend_all(struct snd_pcm *pcm); diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index a61d7a9a995e..01f8fdc42b1b 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -200,9 +200,7 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) } else { printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n", dma_ch, dcsr); - snd_pcm_stream_lock(substream); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(substream); + snd_pcm_stop_xrun(substream); } } EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index dfb5031969f8..a3d122109704 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1098,6 +1098,29 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) SNDRV_PCM_STATE_SETUP); } +/** + * snd_pcm_stop_xrun - stop the running streams as XRUN + * @substream: the PCM substream instance + * @state: PCM state after stopping the stream + * + * This stops the given running substream (and all linked substreams) as XRUN. + * Unlike snd_pcm_stop(), this function takes the substream lock by itself. + * + * Return: Zero if successful, or a negative error code. + */ +int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) +{ + unsigned long flags; + int ret = 0; + + snd_pcm_stream_lock_irqsave(substream, flags); + if (snd_pcm_running(substream)) + ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(substream, flags); + return ret; +} +EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); + /* * pause callbacks */ diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 95fc2eaf11dc..3badc70124ab 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -1006,11 +1006,7 @@ void amdtp_stream_pcm_abort(struct amdtp_stream *s) struct snd_pcm_substream *pcm; pcm = ACCESS_ONCE(s->pcm); - if (pcm) { - snd_pcm_stream_lock_irq(pcm); - if (snd_pcm_running(pcm)) - snd_pcm_stop(pcm, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(pcm); - } + if (pcm) + snd_pcm_stop_xrun(pcm); } EXPORT_SYMBOL(amdtp_stream_pcm_abort); diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 7ac94439e758..48d6dca471c6 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -131,14 +131,8 @@ static void isight_samples(struct isight *isight, static void isight_pcm_abort(struct isight *isight) { - unsigned long flags; - - if (ACCESS_ONCE(isight->pcm_active)) { - snd_pcm_stream_lock_irqsave(isight->pcm, flags); - if (snd_pcm_running(isight->pcm)) - snd_pcm_stop(isight->pcm, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(isight->pcm, flags); - } + if (ACCESS_ONCE(isight->pcm_active)) + snd_pcm_stop_xrun(isight->pcm); } static void isight_dropped_samples(struct isight *isight, unsigned int total) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index ac66b3228a34..ff9f9f1c0391 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -769,10 +769,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) s->number); ds->drained_count++; if (ds->drained_count > 20) { - unsigned long flags; - snd_pcm_stream_lock_irqsave(s, flags); - snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(s, flags); + snd_pcm_stop_xrun(s); continue; } } else { diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 7895c5d300c7..9c1c4452a8ee 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -688,9 +688,7 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma) if (! dma->substream || ! dma->running) return; dev_dbg(chip->card->dev, "XRUN detected (DMA %d)\n", dma->ops->type); - snd_pcm_stream_lock(dma->substream); - snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(dma->substream); + snd_pcm_stop_xrun(dma->substream); } /* diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 3c3241309a30..b2f63e0727de 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -638,9 +638,7 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip, if (! dma->substream || ! dma->running) return; dev_dbg(chip->card->dev, "XRUN detected (DMA %d)\n", dma->ops->type); - snd_pcm_stream_lock(dma->substream); - snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(dma->substream); + snd_pcm_stop_xrun(dma->substream); } /* diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index b79a2a864513..33fb3bb133df 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -80,9 +80,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, /* stop RX and capture: will be enabled again at restart */ ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable); - snd_pcm_stream_lock(substream); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(substream); + snd_pcm_stop_xrun(substream); /* now drain RHR and read status to remove xrun condition */ ssc_readx(prtd->ssc->regs, SSC_RHR); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index a609aafc994d..b2b108805b24 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -151,14 +151,7 @@ static const struct snd_pcm_hardware fsl_dma_hardware = { */ static void fsl_dma_abort_stream(struct snd_pcm_substream *substream) { - unsigned long flags; - - snd_pcm_stream_lock_irqsave(substream, flags); - - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - - snd_pcm_stream_unlock_irqrestore(substream, flags); + snd_pcm_stop_xrun(substream); } /** diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index ba40489b2de4..36f4115eb1cd 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -679,25 +679,16 @@ int usb6fire_pcm_init(struct sfire_chip *chip) void usb6fire_pcm_abort(struct sfire_chip *chip) { struct pcm_runtime *rt = chip->pcm; - unsigned long flags; int i; if (rt) { rt->panic = true; - if (rt->playback.instance) { - snd_pcm_stream_lock_irqsave(rt->playback.instance, flags); - snd_pcm_stop(rt->playback.instance, - SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(rt->playback.instance, flags); - } + if (rt->playback.instance) + snd_pcm_stop_xrun(rt->playback.instance); - if (rt->capture.instance) { - snd_pcm_stream_lock_irqsave(rt->capture.instance, flags); - snd_pcm_stop(rt->capture.instance, - SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(rt->capture.instance, flags); - } + if (rt->capture.instance) + snd_pcm_stop_xrun(rt->capture.instance); for (i = 0; i < PCM_N_URBS; i++) { usb_poison_urb(&rt->in_urbs[i].instance); diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a4679913b0aa..03b074419964 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -391,9 +391,7 @@ static void snd_complete_urb(struct urb *urb) usb_audio_err(ep->chip, "cannot submit urb (err = %d)\n", err); if (ep->data_subs && ep->data_subs->pcm_substream) { substream = ep->data_subs->pcm_substream; - snd_pcm_stream_lock_irqsave(substream, flags); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(substream, flags); + snd_pcm_stop_xrun(substream); } exit_clear: diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index a1bab149df4d..9581089c28c5 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -613,24 +613,14 @@ static int start_usb_playback(struct ua101 *ua) static void abort_alsa_capture(struct ua101 *ua) { - unsigned long flags; - - if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) { - snd_pcm_stream_lock_irqsave(ua->capture.substream, flags); - snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(ua->capture.substream, flags); - } + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + snd_pcm_stop_xrun(ua->capture.substream); } static void abort_alsa_playback(struct ua101 *ua) { - unsigned long flags; - - if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) { - snd_pcm_stream_lock_irqsave(ua->playback.substream, flags); - snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(ua->playback.substream, flags); - } + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + snd_pcm_stop_xrun(ua->playback.substream); } static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index a63330dd1407..61d5dc2a3421 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -272,13 +272,8 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) for (s = 0; s < 4; s++) { struct snd_usX2Y_substream *subs = usX2Y->subs[s]; if (subs) { - if (atomic_read(&subs->state) >= state_PRERUNNING) { - unsigned long flags; - - snd_pcm_stream_lock_irqsave(subs->pcm_substream, flags); - snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(subs->pcm_substream, flags); - } + if (atomic_read(&subs->state) >= state_PRERUNNING) + snd_pcm_stop_xrun(subs->pcm_substream); for (u = 0; u < NRURBS; u++) { struct urb *urb = subs->urb[u]; if (NULL != urb) -- 2.1.3

1 0

[alsa-devel] [PATCH 1/4] ASoC:rt5670:Add runtime PM support
by bardliao＠realtek.com 07 Nov '14

07 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds runtime PM support on rt5670 codec. And will switch sysclk to internal clock during runtime suspend. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index da4b689..97359b3 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -14,6 +14,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/firmware.h> @@ -2666,6 +2667,13 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, /*Give sysclk a default value*/ rt5670->sysclk = 24576000; + /* The set initial power status to active, to be consist with ACPI power + * state D0. The codec will be suspended after probe and runtime status + * change will trigger ACPI D3 method for power saving. + */ + pm_runtime_set_active(&i2c->dev); + pm_runtime_enable(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, rt5670_dai, ARRAY_SIZE(rt5670_dai)); if (ret < 0) @@ -2673,20 +2681,61 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, return 0; err: + pm_runtime_disable(&i2c->dev); + return ret; } static int rt5670_i2c_remove(struct i2c_client *i2c) { + pm_runtime_disable(&i2c->dev); snd_soc_unregister_codec(&i2c->dev); return 0; } +#ifdef CONFIG_PM_RUNTIME +static void rt5670_use_internal_clk(struct rt5670_priv *rt5670) +{ + struct snd_soc_codec *codec = rt5670->codec; + + if (!codec || + (0 == rt5670->sysclk && RT5670_SCLK_S_RCCLK == rt5670->sysclk_src)) + return; + + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK); + + rt5670->sysclk = 0; + rt5670->sysclk_src = RT5670_SCLK_S_RCCLK; +} + +static int rt5670_runtime_suspend(struct device *dev) +{ + struct rt5670_priv *rt5670 = dev_get_drvdata(dev); + + rt5670_use_internal_clk(rt5670); + return 0; +} + +static int rt5670_runtime_resume(struct device *dev) +{ + return 0; +} + +static const struct dev_pm_ops rt5670_pm = { + SET_RUNTIME_PM_OPS(rt5670_runtime_suspend, rt5670_runtime_resume, NULL) +}; +#define RT5670_PM_OPS rt5670_pm +#else +#define RT5670_PM_OPS NULL +#endif /* CONFIG_PM_RUNTIME */ + static struct i2c_driver rt5670_i2c_driver = { .driver = { .name = "rt5670", .owner = THIS_MODULE, + .pm = &RT5670_PM_OPS, }, .probe = rt5670_i2c_probe, .remove = rt5670_i2c_remove, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ALSA: snd_ctl_activate_id(): Fix index look-up
by Lars-Peter Clausen 07 Nov '14

07 Nov '14

We want to know the offset for the id that was passed to the function, not the offset of the first id of the control (which is always 0). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- The function seems currently unused, so no need to add this for stable --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 99aa3aa..bb96a46 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -572,7 +572,7 @@ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, ret = -ENOENT; goto unlock; } - index_offset = snd_ctl_get_ioff(kctl, &kctl->id); + index_offset = snd_ctl_get_ioff(kctl, id); vd = &kctl->vd[index_offset]; ret = 0; if (active) { -- 1.8.0

2 1

[alsa-devel] [PATCH 0/2] add support for exynos7 I2S controller
by Padmavathi Venna 07 Nov '14

07 Nov '14

I2S on exynos7 includes the below changes - has got support for lower root clock sampling frequencies like 64, 128, 96, 192fs. I2S on previous SoCs supports only 256,512,384 and 768. - supports 7.1CH TDM mode for recording. exynos5 has only 7.1CH TDM mode for playback. - secondary dai works with only external DMA. on previous SoCs secondary dai can work with either both or only internal DMA. - I2S1 controller upgraded to 5.1CH with slighlty modified bit offsets for supporting lower RFS and more no.of BFS values. As internal dma is not mandatory on all SoCs this patchset introduced a quirk for the same and added a structure to hold all the modified bit offsets across all SoCs. This patchset is made based on linux-next master branch and tested with exynos7 base support from the below link. http://www.spinics.net/lists/linux-samsung-soc/msg37047.html Padmavathi Venna (2): ASoC: Samsung: Add quirk for internal DMA ASoC: samsung: add support for exynos7 I2S controller .../devicetree/bindings/sound/samsung-i2s.txt | 15 +- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/i2s-regs.h | 10 +- sound/soc/samsung/i2s.c | 228 ++++++++++++++------ 5 files changed, 180 insertions(+), 76 deletions(-) -- 1.7.4.4

2 4

[alsa-devel] Serial MIDI driver for PL011 - as found on BCM2835 hardware
by Kim Taylor 07 Nov '14

07 Nov '14

Hi, I've written a driver for raw MIDI using the PL011. It is obviously based on the snd-serial-u16550 module, with a couple of improvements: - I've implemented a drain() callback. - I've also implemented a half duplex mode based on a timer callback which was necessary for reliable communication with my eMagic Unitor 8 hardware. (Testing with a serial link to other hardware shows that the full duplex mode is also working.) It can be found here: https://github.com/kmtaylor/rpi_patches/blob/master/serial-pl011.c If you think it should be included in the ALSA repository, please let me know what to do next. Regards, Kim

4 6

[alsa-devel] [PATCH 0/6] ASoC/ARM: move Marvell Armada 370 DB to simple-audio-card
by Thomas Petazzoni 07 Nov '14

07 Nov '14

Hello, This set of patches moves the Marvell Armada 370 DB audio complex description from a custom audio machine driver to the simple-audio-card DT binding. The first three patches are meant to go through the ASoC tree, the last three patches are meant to go through the mvebu maintainers tree. Quick description of the patches: - First patch fixes a small issue in the cs42l51 driver following its split in two parts. - Second patch makes the cs42l51 user-selectable by adding a prompt to its Kconfig option. - Third patch gets rid of the machine driver. - Fourth patch removes some now incorrect pin-muxing selection in the Armada 370 DB platform. - Fifth patch actually switches to use the simple-audio-card DT binding on Armada 370 DB. - Sixth patch updates mvebu_v7_defconfig to select the relevant options to get audio to work. Note: obviously, after PATCH 3 is applied, the audio will no longer work until PATCH 5 is applied. But since those patches will anyway go through different trees, it is a bit inevitable. And it is not dramatically important, since audio support on Armada 370 DB is already broken since the split of the CS42L51 driver. Thanks, Thomas Thomas Petazzoni (6): ASoC: cs42l51: re-add of_match_table pointer ASoC: cs42l51: make driver user-selectable ASoC: kirkwood: get rid of armada-370-db driver ARM: mvebu: remove conflicting muxing on Armada 370 DB ARM: mvebu: use simple-card DT binding for audio on Armada 370 DB ARM: mvebu: update mvebu_v7_defconfig for Armada 370 audio arch/arm/boot/dts/armada-370-db.dts | 80 +++++++++++-------- arch/arm/boot/dts/armada-370.dtsi | 1 + arch/arm/configs/mvebu_v7_defconfig | 4 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/cs42l51-i2c.c | 7 ++ sound/soc/codecs/cs42l51.c | 5 -- sound/soc/kirkwood/Kconfig | 13 +--- sound/soc/kirkwood/Makefile | 4 - sound/soc/kirkwood/armada-370-db.c | 148 ------------------------------------ 9 files changed, 63 insertions(+), 201 deletions(-) delete mode 100644 sound/soc/kirkwood/armada-370-db.c -- 2.0.0

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[alsa-devel] [PATCH 0/8] ASoC: rsnd: add Volume Ramp support
by Kuninori Morimoto 06 Nov '14

06 Nov '14

Hi Mark These patches add Volume Ramp support on Renesas R-Car sound driver. Kuninori Morimoto (8): ASoC: rsnd: tidyup debug information when read/write ASoC: rsnd: tidyup RSND_DVC_VOLUME_NUM to RSND_DVC_CHANNELS ASoC: rsnd: add struct rsnd_dvc_cfg and control DVC settings ASoC: rsnd: control DVC_DVUCR under rsnd_dvc_volume_update() ASoC: rsnd: move DVC_DVUER settings under rsnd_dvc_volume_update() ASoC: rsnd: enable multiple DVC valume settings ASoC: rsnd: enable single DVC valume settings ASoC: rsnd: Add Volume Ramp support sound/soc/sh/rcar/dvc.c | 155 ++++++++++++++++++++++++++++++++++------------ sound/soc/sh/rcar/gen.c | 30 ++++++--- sound/soc/sh/rcar/rsnd.h | 6 ++ 3 files changed, 145 insertions(+), 46 deletions(-) Best regards --- Kuninori Morimoto

2 24

[alsa-devel] [PATCH] ASoC: Intel: mrfld - remove non static definition
by Vinod Koul 06 Nov '14

06 Nov '14

sst_save_shim64() is defined as static in code but header is non static. Since this is not used other than file where defined remove non static definition Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/sst/sst.h | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 3ee555e..2dcbf47 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -508,8 +508,6 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, size_t mbox_data_len, const void *mbox_data, void **data, bool large, bool fill_dsp, bool sync, bool response); -void sst_save_shim64(struct intel_sst_drv *ctx, void __iomem *shim, - struct sst_shim_regs64 *shim_regs); void sst_process_pending_msg(struct work_struct *work); int sst_assign_pvt_id(struct intel_sst_drv *sst_drv_ctx); void sst_init_stream(struct stream_info *stream, -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: ssm4567: Add DAC high-pass-filter control
by Lars-Peter Clausen 06 Nov '14

06 Nov '14

Add a switch which can be used to enable/disable the DAC high-pass-filter. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm4567.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 4b5c17f..e1e33d8 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -145,6 +145,8 @@ static const struct snd_kcontrol_new ssm4567_snd_controls[] = { SOC_SINGLE_TLV("Master Playback Volume", SSM4567_REG_DAC_VOLUME, 0, 0xff, 1, ssm4567_vol_tlv), SOC_SINGLE("DAC Low Power Mode Switch", SSM4567_REG_DAC_CTRL, 4, 1, 0), + SOC_SINGLE("DAC High Pass Filter Switch", SSM4567_REG_DAC_CTRL, + 5, 1, 0), }; static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { -- 1.8.0

2 3

[alsa-devel] [PATCH] ALSA: usb-audio: Trigger PCM XRUN at XRUN
by Takashi Iwai 06 Nov '14

06 Nov '14

The usb-audio driver detects XRUN at its complete callback, but the actual code to trigger PCM XRUN is commented out because it caused deadlock in the past. This patch revives the PCM trigger properly. It resulted in more than just enabling snd_pcm_stop(), but it had to deduce the PCM substream with proper NULL checks and holds the stream lock around the call. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 167d0c1643e1..a4679913b0aa 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -348,6 +348,8 @@ static void snd_complete_urb(struct urb *urb) { struct snd_urb_ctx *ctx = urb->context; struct snd_usb_endpoint *ep = ctx->ep; + struct snd_pcm_substream *substream; + unsigned long flags; int err; if (unlikely(urb->status == -ENOENT || /* unlinked */ @@ -364,8 +366,6 @@ static void snd_complete_urb(struct urb *urb) goto exit_clear; if (snd_usb_endpoint_implicit_feedback_sink(ep)) { - unsigned long flags; - spin_lock_irqsave(&ep->lock, flags); list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); spin_unlock_irqrestore(&ep->lock, flags); @@ -389,7 +389,12 @@ static void snd_complete_urb(struct urb *urb) return; usb_audio_err(ep->chip, "cannot submit urb (err = %d)\n", err); - //snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + if (ep->data_subs && ep->data_subs->pcm_substream) { + substream = ep->data_subs->pcm_substream; + snd_pcm_stream_lock_irqsave(substream, flags); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(substream, flags); + } exit_clear: clear_bit(ctx->index, &ep->active_mask); -- 2.1.3

1 0

[alsa-devel] [PATCH] ASoC:rt5645:Add JD function support
by bardliao＠realtek.com 06 Nov '14

06 Nov '14

From: Bard Liao <bardliao(a)realtek.com> rt5645 codec support jack detection function. The patch will set related registers if JD function is used. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5645.h | 3 +++ sound/soc/codecs/rt5645.c | 20 ++++++++++++++++++++ sound/soc/codecs/rt5645.h | 5 +++++ 3 files changed, 28 insertions(+) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a535271..937f421 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,9 @@ struct rt5645_platform_data { unsigned int hp_det_gpio; bool gpio_hp_det_active_high; + + /* true if codec's jd function is used */ + bool en_jd_func; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 0e92e63..1c099ee 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2205,6 +2205,13 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); + /* for JD function */ + if (rt5645->pdata.en_jd_func) { + snd_soc_dapm_force_enable_pin(&codec->dapm, "JD Power"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_sync(&codec->dapm); + } + return 0; } @@ -2438,6 +2445,19 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } + if (rt5645->pdata.en_jd_func) { + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU, + RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); + regmap_update_bits(rt5645->regmap, RT5645_JD_CTRL3, + RT5645_JD_CBJ_EN | RT5645_JD_CBJ_POL, + RT5645_JD_CBJ_EN | RT5645_JD_CBJ_POL); + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_IRQ_CLK_INT, RT5645_IRQ_CLK_INT); + } + if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 5ec2520..82f681b 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1348,6 +1348,8 @@ #define RT5645_PWR_CLK25M_SFT 4 #define RT5645_PWR_CLK25M_PD (0x0 << 4) #define RT5645_PWR_CLK25M_PU (0x1 << 4) +#define RT5645_IRQ_CLK_MCLK (0x0 << 3) +#define RT5645_IRQ_CLK_INT (0x1 << 3) /* VAD Control 4 (0x9d) */ #define RT5645_VAD_SEL_MASK (0x3 << 8) @@ -2116,6 +2118,9 @@ enum { #define RT5645_RXDP2_SEL_ADC (0x1 << 3) #define RT5645_RXDP2_SEL_SFT (3) +/* General Control3 (0xfc) */ +#define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) +#define RT5645_MICINDET_MANU (0x1 << 7) /* Vendor ID (0xfd) */ #define RT5645_VER_C 0x2 -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] [PATCH] ALSA: pcm: Update the state properly before notification
by Takashi Iwai 06 Nov '14

06 Nov '14

Some state changes (e.g. snd_pcm_stop()) sets the runtime state after calling snd_timer_notify(). This is basically racy, since the notification may wakes up the user even before the state change. Although the possibility is low, we should set the state before the notifications. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ca224fa2a33a..dfb5031969f8 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1052,10 +1052,10 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; if (runtime->status->state != state) { snd_pcm_trigger_tstamp(substream); + runtime->status->state = state; if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP, &runtime->trigger_tstamp); - runtime->status->state = state; } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); @@ -1204,11 +1204,11 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); + runtime->status->suspended_state = runtime->status->state; + runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND, &runtime->trigger_tstamp); - runtime->status->suspended_state = runtime->status->state; - runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } @@ -1311,10 +1311,10 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); + runtime->status->state = runtime->status->suspended_state; if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME, &runtime->trigger_tstamp); - runtime->status->state = runtime->status->suspended_state; } static struct action_ops snd_pcm_action_resume = { -- 2.1.3

1 0

[alsa-devel] [PATCH alsa-lib] pcm: dmix: Don't ignore SND_TIMER_EVENT_MSTOP
by Takashi Iwai 06 Nov '14

06 Nov '14

When a slave PCM gets an error like XRUN, it stops and notifies with SND_TIMER_EVENT_MSTOP event. But the current code filters out this type and eventually hang due to the empty timer queue. The fix is to just add this event type to the filter bit mask. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 8e37bcba5e19..195fddf06cda 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1148,6 +1148,7 @@ int snd_pcm_direct_initialize_poll_fd(snd_pcm_direct_t *dmix) dmix->timer_events = (1<<SND_TIMER_EVENT_MSUSPEND) | (1<<SND_TIMER_EVENT_MRESUME) | + (1<<SND_TIMER_EVENT_MSTOP) | (1<<SND_TIMER_EVENT_STOP); /* -- 2.1.3

1 0

[alsa-devel] [PATCH 1/2] ASoC:rt286: set combo jack by dmi
by bardliao＠realtek.com 06 Nov '14

06 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch enables combo jack configuration according to dmi. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * rename intel_platform to force_combo_jack_table --- --- sound/soc/codecs/rt286.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 97daa80..d4acb64 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -17,6 +17,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1205,6 +1206,16 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); +static struct dmi_system_id force_combo_jack_table[] __initdata = { + { + .ident = "Intel Wilson Beach", + .matches = { + DMI_MATCH(DMI_BOARD_NAME, "Wilson Beach SDS") + } + }, + { } +}; + static int rt286_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1240,6 +1251,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + if (dmi_check_system(force_combo_jack_table)) + rt286->pdata.cbj_en = true; + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); for (i = 0; i < RT286_POWER_REG_LEN; i++) -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Remove '.owner' field of driver
by Manish Badarkhe 06 Nov '14

06 Nov '14

Remove '.owner' field of driver structure as it is going to be filled via "module_i2c_driver" call. Signed-off-by: Manish Badarkhe <manishvb(a)ti.com> --- :100644 100644 0307909... 191b423... M sound/soc/codecs/tlv320aic31xx.c sound/soc/codecs/tlv320aic31xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 0307909..191b423 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1303,7 +1303,6 @@ MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, -- 1.7.9.5

1 0

Re: [alsa-devel] Restore hidden register for default value
by Kailang 06 Nov '14

06 Nov '14

Hi Takashi, I mail it again. BR, Kailang > -----Original Message----- > From: Kailang > Sent: Friday, October 31, 2014 3:12 PM > To: 'Takashi Iwai' > Cc: (alsa-devel(a)alsa-project.org) > Subject: RE: Restore hidden register for default value > > Hi Takashi, > > It's just the default value refill in initial state. > It's only for ALC668(ALC3661). > The patch was tested pass with issue machine in ODM site. > > Many Thanks. > > BR, > Kailang > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Wednesday, October 22, 2014 4:00 PM > > To: Kailang > > Subject: Re: Restore hidden register for default value > > > > At Wed, 22 Oct 2014 07:56:43 +0000, > > Kailang wrote: > > > > > > Hi Takashi, > > > > > > We want to post ALC668 for hidden register issue. > > > > > > Could I post like below array for codec initial state? > > > > > > WORD coefIndexDefault[][2] = {{0x1, > > 0xbebe},{0x2,0xaaaa},{0x3,0x0},{0x4,0x180},{0x6,0x0}, > > > > > {0x7,0xf80},{0x8,0x31},{0xa,0x60},{0xb,0x0},{0xc,0x7cf7},{0xd, > > 0x1080},{0xe,0x7f7f}, > > > > > {0xf,0xcccc},{0x10,0xddcc},{0x11,0x1},{0x13,0x0},{0x14,0x2aa0} > > ,{0x17,0xA940}, > > > > > {0x19,0},{0x1A,0},{0x1B,0},{0x1c,0x0},{0x1d,0x0},{0x1e,0x7418} > > ,{0x1f,0x804}, > > > > > {0x20,0x4200},{0x21,0x468},{0x22,0x8ccc},{0x23,0x250},{0x24,0x7418}, > > > > > > > > > {0x27,0x0000},{0x28,0x8ccc},{0x2a,0xff00},{0x2b,0x8000},{0xa7,0xff00}} > > > ; Because codec sometimes has wrong value for hidden > register after > > > resume back or reboot. > > > It needed to refill the default value with upper array in > > initial state. > > > But the hidden registers was more for 59 records. > > > Could I post it with no remark? Because it just the default > > value. Thanks. > > > > Some comments would be nice, of course, but not mandatory. > > > > One concern is exactly which codecs should this be applied. > > There are a bunch of codecs compatible with alc668. It's only for > > ALC668, and not for others (ALC668, 670, whatever). > > > > > > Takashi > > > > ------Please consider the environment before printing this e-mail. > >

2 3

[alsa-devel] [PATCH] ASoC:rt286: set combo jack by dmi
by bardliao＠realtek.com 06 Nov '14

06 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch enables combo jack configuration according to dmi. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 97daa80..aa572b6 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -17,6 +17,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1205,6 +1206,16 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); +static struct dmi_system_id intel_platform[] __initdata = { + { + .ident = "Intel Wilson Beach", + .matches = { + DMI_MATCH(DMI_BOARD_NAME, "Wilson Beach SDS") + } + }, + { } +}; + static int rt286_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1240,6 +1251,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + if (dmi_check_system(intel_platform)) + rt286->pdata.cbj_en = true; + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); for (i = 0; i < RT286_POWER_REG_LEN; i++) -- 1.8.1.1.439.g50a6b54

3 2

[alsa-devel] [PATCH] ASoC: samsung: Add MODULE_DEVICE_TABLE for Snow
by Andreas Färber 05 Nov '14

05 Nov '14

This enables the snd_soc_snow module to be auto-loaded. Fixes: da5993df2374 ("ASoC: samsung: Add sound card driver for Snow board") Signed-off-by: Andreas Färber <afaerber(a)suse.de> --- Tested using patches for Spring, but should apply to Snow/Peach-pi equally. Rebased onto sound.git for-next branch. sound/soc/samsung/snow.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 0acf5d0eed53..72118a77dd5b 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -110,6 +110,7 @@ static const struct of_device_id snow_of_match[] = { { .compatible = "google,snow-audio-max98095", }, {}, }; +MODULE_DEVICE_TABLE(of, snow_of_match); static struct platform_driver snow_driver = { .driver = { -- 2.1.2

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[alsa-devel] [PATCH] ASoC: ad193x: Keep DAC output stage active in idle
by Lars-Peter Clausen 05 Nov '14

05 Nov '14

Setting the DAC power-down bit for the ad193x will also disable the DAC output amplifier. This will cause audible clicks and pops when starting or stopping playback. To prevent this a new widget is introduced that controls the DAC power-down bit. This widget is connected to both the DAC and a newly introduced VMID widget. This makes sure that the DAC power-down bit is not set as long as a audio sink is connected to the DAC output. At the same time the PLL and SYSCLK will still be disabled when no playback or capture stream is active. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad193x.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 6844d0b..387530b 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -72,11 +72,13 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", AD193X_DAC_CTRL0, 0, 1), + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), + SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), @@ -87,13 +89,15 @@ static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, + { "DAC Output", NULL, "DAC" }, + { "DAC Output", NULL, "VMID" }, { "ADC", NULL, "SYSCLK" }, { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, - { "DAC1OUT", NULL, "DAC" }, - { "DAC2OUT", NULL, "DAC" }, - { "DAC3OUT", NULL, "DAC" }, - { "DAC4OUT", NULL, "DAC" }, + { "DAC1OUT", NULL, "DAC Output" }, + { "DAC2OUT", NULL, "DAC Output" }, + { "DAC3OUT", NULL, "DAC Output" }, + { "DAC4OUT", NULL, "DAC Output" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, { "SYSCLK", NULL, "PLL_PWR" }, -- 1.8.0

2 1

[alsa-devel] [PATCH v2] ALSA: echoaudio: cleanup of unnecessary messages
by Sudip Mukherjee 05 Nov '14

05 Nov '14

commit "b5b4a41b392960010fccf1f9ccf8334d612bd450" was dereferencing chip after it has been freed. This patch fixes that and at the same time removes some debugging messages, which are unnecessary, as they are just printing information about entry and exit from a function, and which switch-case it is executing. we can easily get from ftrace the information about the entry and exit from a function. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change from v1: removing some more debug messages sound/pci/echoaudio/darla20_dsp.c | 2 -- sound/pci/echoaudio/darla24_dsp.c | 2 -- sound/pci/echoaudio/echo3g_dsp.c | 2 -- sound/pci/echoaudio/echoaudio.c | 31 ------------------------------- sound/pci/echoaudio/echoaudio_3g.c | 5 ----- sound/pci/echoaudio/echoaudio_dsp.c | 12 ------------ sound/pci/echoaudio/gina20_dsp.c | 6 ------ sound/pci/echoaudio/gina24_dsp.c | 9 --------- sound/pci/echoaudio/indigo_dsp.c | 2 -- sound/pci/echoaudio/indigodj_dsp.c | 2 -- sound/pci/echoaudio/indigodjx_dsp.c | 2 -- sound/pci/echoaudio/indigoio_dsp.c | 2 -- sound/pci/echoaudio/indigoiox_dsp.c | 2 -- sound/pci/echoaudio/layla20_dsp.c | 9 --------- sound/pci/echoaudio/layla24_dsp.c | 8 -------- sound/pci/echoaudio/mia_dsp.c | 2 -- sound/pci/echoaudio/midi.c | 6 ------ sound/pci/echoaudio/mona_dsp.c | 6 ------ 18 files changed, 110 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index c94e92e..febee5b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20)) return -ENODEV; @@ -58,7 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index b1272f88..7b4f6fd 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24)) return -ENODEV; @@ -57,7 +56,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index bc37168..ae11ce1 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -46,7 +46,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) int err; local_irq_enable(); - dev_dbg(chip->card->dev, "init_hw() - Echo3G\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G)) return -ENODEV; @@ -99,7 +98,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1ef29e5..60e4003 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -79,7 +79,6 @@ static void free_firmware(const struct firmware *fw_entry, dev_dbg(chip->card->dev, "firmware not released (kept in cache)\n"); #else release_firmware(fw_entry); - dev_dbg(chip->card->dev, "firmware released\n"); #endif } @@ -96,7 +95,6 @@ static void free_firmware_cache(struct echoaudio *chip) dev_dbg(chip->card->dev, "release_firmware(%d)\n", i); } - dev_dbg(chip->card->dev, "firmware_cache released\n"); #endif } @@ -354,7 +352,6 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err; - dev_dbg(chip->card->dev, "pcm_analog_in_open\n"); if ((err = pcm_open(substream, num_analog_busses_in(chip) - substream->number)) < 0) return err; @@ -389,7 +386,6 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) #else max_channels = num_analog_busses_out(chip); #endif - dev_dbg(chip->card->dev, "pcm_analog_out_open\n"); if ((err = pcm_open(substream, max_channels - substream->number)) < 0) return err; if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, @@ -422,7 +418,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_in_open\n"); max_channels = num_digital_busses_in(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -464,7 +459,6 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_out_open\n"); max_channels = num_digital_busses_out(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -511,7 +505,6 @@ static int pcm_close(struct snd_pcm_substream *substream) /* Nothing to do here. Audio is already off and pipe will be * freed by its callback */ - dev_dbg(chip->card->dev, "pcm_close\n"); atomic_dec(&chip->opencount); oc = atomic_read(&chip->opencount); @@ -620,7 +613,6 @@ static int init_engine(struct snd_pcm_substream *substream, spin_lock_irq(&chip->lock); set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den); spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_params ok\n"); return 0; } @@ -691,7 +683,6 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) } spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_freed\n"); snd_pcm_lib_free_pages(substream); return 0; } @@ -763,10 +754,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - dev_dbg(chip->card->dev, "pcm_trigger resume\n"); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dev_dbg(chip->card->dev, "pcm_trigger start\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -788,9 +777,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) chip->pipe_cyclic_mask); break; case SNDRV_PCM_TRIGGER_SUSPEND: - dev_dbg(chip->card->dev, "pcm_trigger suspend\n"); case SNDRV_PCM_TRIGGER_STOP: - dev_dbg(chip->card->dev, "pcm_trigger stop\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -800,7 +787,6 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = stop_transport(chip, channelmask); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev_dbg(chip->card->dev, "pcm_trigger pause\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -937,7 +923,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital inputs, no outputs */ @@ -950,7 +935,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #else /* ECHOCARD_HAS_VMIXER */ @@ -972,7 +956,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital i/o */ @@ -987,7 +970,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #endif /* ECHOCARD_HAS_VMIXER */ @@ -1881,10 +1863,8 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) static int snd_echo_free(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "Stop DSP...\n"); if (chip->comm_page) rest_in_peace(chip); - dev_dbg(chip->card->dev, "Stopped.\n"); if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1898,14 +1878,12 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->iores) release_and_free_resource(chip->iores); - dev_dbg(chip->card->dev, "MMIO freed.\n"); pci_disable_device(chip->pci); /* release chip data */ free_firmware_cache(chip); kfree(chip); - dev_dbg(chip->card->dev, "Chip freed.\n"); return 0; } @@ -1915,7 +1893,6 @@ static int snd_echo_dev_free(struct snd_device *device) { struct echoaudio *chip = device->device_data; - dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n"); return snd_echo_free(chip); } @@ -2008,7 +1985,6 @@ static int snd_echo_create(struct snd_card *card, snd_echo_free(chip); return err; } - dev_dbg(card->dev, "Card init OK\n"); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_echo_free(chip); @@ -2038,7 +2014,6 @@ static int snd_echo_probe(struct pci_dev *pci, return -ENOENT; } - dev_dbg(&pci->dev, "Echoaudio driver starting...\n"); i = 0; err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -2191,7 +2166,6 @@ static int snd_echo_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct echoaudio *chip = dev_get_drvdata(dev); - dev_dbg(dev, "suspend start\n"); snd_pcm_suspend_all(chip->analog_pcm); snd_pcm_suspend_all(chip->digital_pcm); @@ -2218,7 +2192,6 @@ static int snd_echo_suspend(struct device *dev) pci_save_state(pci); pci_disable_device(pci); - dev_dbg(dev, "suspend done\n"); return 0; } @@ -2232,7 +2205,6 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - dev_dbg(dev, "resume start\n"); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); if (commpage_bak == NULL) @@ -2247,7 +2219,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_free(chip); return err; } - dev_dbg(dev, "resume init OK\n"); /* Temporarily set chip->pipe_alloc_mask=0 otherwise * restore_dsp_settings() fails. @@ -2260,7 +2231,6 @@ static int snd_echo_resume(struct device *dev) kfree(commpage_bak); return err; } - dev_dbg(dev, "resume restore OK\n"); memcpy(&commpage->audio_format, &commpage_bak->audio_format, sizeof(commpage->audio_format)); @@ -2286,7 +2256,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif - dev_dbg(dev, "resume done\n"); return 0; } diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 50a2169..2fa66dc 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -328,7 +328,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -337,13 +336,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Echo3G clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Echo3G clock to SPDIF\n"); control_reg |= E3G_SPDIF_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96) control_reg |= E3G_DOUBLE_SPEED_MODE; @@ -353,12 +350,10 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Echo3G clock to ADAT\n"); control_reg |= E3G_ADAT_CLOCK; control_reg &= ~E3G_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Echo3G clock to WORD\n"); control_reg |= E3G_WORD_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96) control_reg |= E3G_DOUBLE_SPEED_MODE; diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index ba9d4f1..1a9427a 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,7 +206,6 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) goto la_error; } - dev_dbg(chip->card->dev, "ASIC loaded\n"); free_firmware(fw, chip); return 0; @@ -473,7 +472,6 @@ static int load_dsp(struct echoaudio *chip, u16 *code) chip->dsp_code = code; /* Show which DSP code loaded */ chip->bad_board = FALSE; /* DSP OK */ - dev_dbg(chip->card->dev, "load_dsp: OK!\n"); return 0; } udelay(100); @@ -674,7 +672,6 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { int i, o, err; - dev_dbg(chip->card->dev, "restore_dsp_settings\n"); if ((err = check_asic_status(chip)) < 0) return err; @@ -771,7 +768,6 @@ static int restore_dsp_rettings(struct echoaudio *chip) if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) return -EIO; - dev_dbg(chip->card->dev, "restore_dsp_rettings done\n"); return 0; } @@ -865,7 +861,6 @@ Same thing for pause_ and stop_ -trasport below. */ static int start_transport(struct echoaudio *chip, u32 channel_mask, u32 cyclic_mask) { - dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -891,7 +886,6 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask, static int pause_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -918,7 +912,6 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask) static int stop_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -954,8 +947,6 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index) stopped and unallocated. */ static int rest_in_peace(struct echoaudio *chip) { - dev_dbg(chip->card->dev, - "rest_in_peace() open=%x\n", chip->pipe_alloc_mask); /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); @@ -1018,7 +1009,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) */ static int init_line_levels(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "init_line_levels\n"); memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); @@ -1099,7 +1089,6 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, it moves data. The DMA counter is in units of bytes, not samples. */ pipe->dma_counter = &chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; - dev_dbg(chip->card->dev, "allocate_pipes: ok\n"); return pipe_index; } @@ -1110,7 +1099,6 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe) u32 channel_mask; int i; - dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index))) return -EINVAL; if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED)) diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index a959eae..5dafe92 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -37,7 +37,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20)) return -ENODEV; @@ -63,7 +62,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -150,7 +148,6 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) static int set_input_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_input_clock:\n"); switch (clock) { case ECHO_CLOCK_INTERNAL: @@ -159,7 +156,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) chip->spdif_status = GD_SPDIF_STATUS_UNDEF; set_sample_rate(chip, chip->sample_rate); chip->input_clock = clock; - dev_dbg(chip->card->dev, "Set Gina clock to INTERNAL\n"); break; case ECHO_CLOCK_SPDIF: chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN; @@ -167,7 +163,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) clear_handshake(chip); send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE); chip->clock_state = GD_CLOCK_SPDIFIN; - dev_dbg(chip->card->dev, "Set Gina20 clock to SPDIF\n"); chip->input_clock = clock; break; default: @@ -209,7 +204,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index c8ea576..6971766 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24)) return -ENODEV; @@ -79,7 +78,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -156,7 +154,6 @@ static int load_asic(struct echoaudio *chip) control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; err = write_control_reg(chip, control_reg, TRUE); } - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -238,7 +235,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -247,13 +243,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Gina24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Gina24 clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -263,17 +257,14 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Gina24 clock to ADAT\n"); control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC: - dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC\n"); control_reg |= GML_ESYNC_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC96: - dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC96\n"); control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE; break; default: diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index cdeb073..54edd67 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 133915c..2cf5cc0 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 26cdfcf..5252863 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 5e6df7c..4e81787 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 90cdd27..6de3f9b 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 7f0f6ea..f2024a3 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20)) return -ENODEV; @@ -65,7 +64,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -190,24 +188,19 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) u16 clock; u32 rate; - dev_dbg(chip->card->dev, "set_input_clock:\n"); rate = 0; switch (clock_source) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Layla20 clock to INTERNAL\n"); rate = chip->sample_rate; clock = LAYLA20_CLOCK_INTERNAL; break; case ECHO_CLOCK_SPDIF: - dev_dbg(chip->card->dev, "Set Layla20 clock to SPDIF\n"); clock = LAYLA20_CLOCK_SPDIF; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Layla20 clock to WORD\n"); clock = LAYLA20_CLOCK_WORD; break; case ECHO_CLOCK_SUPER: - dev_dbg(chip->card->dev, "Set Layla20 clock to SUPER\n"); clock = LAYLA20_CLOCK_SUPER; break; default: @@ -232,7 +225,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) static int set_output_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock); switch (clock) { case ECHO_CLOCK_SUPER: clock = LAYLA20_OUTPUT_CLOCK_SUPER; @@ -286,7 +278,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index eb8f218..4f11e81 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24)) return -ENODEV; @@ -70,7 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -118,7 +116,6 @@ static int load_asic(struct echoaudio *chip) if (chip->asic_loaded) return 1; - dev_dbg(chip->card->dev, "load_asic\n"); /* Give the DSP a few milliseconds to settle down */ mdelay(10); @@ -152,7 +149,6 @@ static int load_asic(struct echoaudio *chip) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, TRUE); - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -262,7 +258,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) /* Pick the new clock */ switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Layla24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -271,7 +266,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_SPDIF_CLOCK; /* Layla24 doesn't support 96KHz S/PDIF */ control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to SPDIF\n"); break; case ECHO_CLOCK_WORD: control_reg |= GML_WORD_CLOCK; @@ -279,14 +273,12 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_DOUBLE_SPEED_MODE; else control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to WORD\n"); break; case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to ADAT\n"); break; default: dev_err(chip->card->dev, diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index ed2f21d..fdad079 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mia\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MIA)) return -ENODEV; @@ -67,7 +66,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 8d43c5a..d913749 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -157,7 +157,6 @@ static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = substream; - dev_dbg(chip->card->dev, "rawmidi_iopen\n"); return 0; } @@ -183,7 +182,6 @@ static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = NULL; - dev_dbg(chip->card->dev, "rawmidi_iclose\n"); return 0; } @@ -196,7 +194,6 @@ static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream) chip->tinuse = 0; chip->midi_full = 0; chip->midi_out = substream; - dev_dbg(chip->card->dev, "rawmidi_open\n"); return 0; } @@ -209,7 +206,6 @@ static void snd_echo_midi_output_write(unsigned long data) int bytes, sent, time; unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1]; - dev_dbg(chip->card->dev, "snd_echo_midi_output_write\n"); /* No interrupts are involved: we have to check at regular intervals if the card's output buffer has room for new data. */ sent = bytes = 0; @@ -288,7 +284,6 @@ static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_out = NULL; - dev_dbg(chip->card->dev, "rawmidi_oclose\n"); return 0; } @@ -328,6 +323,5 @@ static int snd_echo_midi_create(struct snd_card *card, chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; - dev_dbg(chip->card->dev, "MIDI ok\n"); return 0; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index cc46a8c..843c7a5 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mona\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA)) return -ENODEV; @@ -72,7 +71,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -302,7 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) u32 control_reg, clocks_from_dsp; int err; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Prevent two simultaneous calls to switch_asic() */ if (atomic_read(&chip->opencount)) @@ -315,7 +312,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Mona clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -327,7 +323,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) spin_lock_irq(&chip->lock); if (err < 0) return err; - dev_dbg(chip->card->dev, "Set Mona clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -335,7 +330,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Mona clock to WORD\n"); spin_unlock_irq(&chip->lock); err = switch_asic(chip, clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96); -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/6] ASoC: sn95031: Use table based control setup
by Lars-Peter Clausen 05 Nov '14

05 Nov '14

Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sn95031.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index cf8fa40..6167c59 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -867,9 +867,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); - snd_soc_add_codec_controls(codec, sn95031_snd_controls, - ARRAY_SIZE(sn95031_snd_controls)); - return 0; } @@ -886,6 +883,9 @@ static struct snd_soc_codec_driver sn95031_codec = { .remove = sn95031_codec_remove, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, + + .controls = sn95031_snd_controls, + .num_controls = ARRAY_SIZE(sn95031_snd_controls), .dapm_widgets = sn95031_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), .dapm_routes = sn95031_audio_map, -- 1.8.0

4 10

[alsa-devel] [PATCH] ALSA: usb-audio: Fix device_del() sysfs warnings at disconnect
by Takashi Iwai 05 Nov '14

05 Nov '14

Some USB-audio devices show weird sysfs warnings at disconnecting the devices, e.g. usb 1-3: USB disconnect, device number 3 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 973 at fs/sysfs/group.c:216 device_del+0x39/0x180() sysfs group ffffffff8183df40 not found for kobject 'midiC1D0' Call Trace: [<ffffffff814a3e38>] ? dump_stack+0x49/0x71 [<ffffffff8103cb72>] ? warn_slowpath_common+0x82/0xb0 [<ffffffff8103cc55>] ? warn_slowpath_fmt+0x45/0x50 [<ffffffff813521e9>] ? device_del+0x39/0x180 [<ffffffff81352339>] ? device_unregister+0x9/0x20 [<ffffffff81352384>] ? device_destroy+0x34/0x40 [<ffffffffa00ba29f>] ? snd_unregister_device+0x7f/0xd0 [snd] [<ffffffffa025124e>] ? snd_rawmidi_dev_disconnect+0xce/0x100 [snd_rawmidi] [<ffffffffa00c0192>] ? snd_device_disconnect+0x62/0x90 [snd] [<ffffffffa00c025c>] ? snd_device_disconnect_all+0x3c/0x60 [snd] [<ffffffffa00bb574>] ? snd_card_disconnect+0x124/0x1a0 [snd] [<ffffffffa02e54e8>] ? usb_audio_disconnect+0x88/0x1c0 [snd_usb_audio] [<ffffffffa015260e>] ? usb_unbind_interface+0x5e/0x1b0 [usbcore] [<ffffffff813553e9>] ? __device_release_driver+0x79/0xf0 [<ffffffff81355485>] ? device_release_driver+0x25/0x40 [<ffffffff81354e11>] ? bus_remove_device+0xf1/0x130 [<ffffffff813522b9>] ? device_del+0x109/0x180 [<ffffffffa01501d5>] ? usb_disable_device+0x95/0x1f0 [usbcore] [<ffffffffa014634f>] ? usb_disconnect+0x8f/0x190 [usbcore] [<ffffffffa0149179>] ? hub_thread+0x539/0x13a0 [usbcore] [<ffffffff810669f5>] ? sched_clock_local+0x15/0x80 [<ffffffff81066c98>] ? sched_clock_cpu+0xb8/0xd0 [<ffffffff81070730>] ? bit_waitqueue+0xb0/0xb0 [<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore] [<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore] [<ffffffff8105973e>] ? kthread+0xce/0xf0 [<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0 [<ffffffff814a8b7c>] ? ret_from_fork+0x7c/0xb0 [<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0 ---[ end trace 40b1928d1136b91e ]--- This comes from the fact that usb-audio driver may receive the disconnect callback multiple times, per each usb interface. When a device has both audio and midi interfaces, it gets called twice, and currently the driver tries to release resources at the last call. At this point, the first parent interface has been already deleted, thus deleting a child of the first parent hits such a warning. For fixing this problem, we need to call snd_card_disconnect() and cancel pending operations at the very first disconnect while the release of the whole objects waits until the last disconnect call. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=80931 Reported-and-tested-by: Tomas Gayoso <tgayoso(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 7ecd0e8a5c51..f61ebb17cc64 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -591,18 +591,19 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, { struct snd_card *card; struct list_head *p; + bool was_shutdown; if (chip == (void *)-1L) return; card = chip->card; down_write(&chip->shutdown_rwsem); + was_shutdown = chip->shutdown; chip->shutdown = 1; up_write(&chip->shutdown_rwsem); mutex_lock(&register_mutex); - chip->num_interfaces--; - if (chip->num_interfaces <= 0) { + if (!was_shutdown) { struct snd_usb_endpoint *ep; snd_card_disconnect(card); @@ -622,6 +623,10 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, list_for_each(p, &chip->mixer_list) { snd_usb_mixer_disconnect(p); } + } + + chip->num_interfaces--; + if (chip->num_interfaces <= 0) { usb_chip[chip->index] = NULL; mutex_unlock(&register_mutex); snd_card_free_when_closed(card); -- 2.1.3

1 0

[alsa-devel] [RFC] AXD Audio Processing IP ALSA support - Questions
by Qais Yousef 05 Nov '14

05 Nov '14

Hi, I have several questions on the best way to add AXD support in ALSA. The discussion of the previous patch can be found here: https://lkml.org/lkml/2014/10/28/465 Questions: 1- What is the best example to follow to add a simple mp3 support for AXD? The only one I find is in sst-mfld-platform-compress.c in sound/soc/intel directory but it's a bit confusing. I think because it's sharing code with several other sst drivers/platforms. I find the documentation for compress_offload generally lacking. Is there a plan to improve on that? Being a new comer to ALSA framework api, I'm confused what is the correct way to do things :-/ So far I know I need to call snd_soc_register_card(). I thought snd_compress_register() (from compress_driver.h) is how you add compressed nodes to the card but apparently not. It looks like I need to define a compress_dai? Hmmm. 2- Is tinycompress the only userland support for compress_offload? Is there anyone working on gstreamer and omx plugins to support that? Would tinycompress be part of alsa-utils and alsa-libs in the future? I know it needs more work at the moment but it'd be nice if compress_offload support is part of the standard alsa-utils and alsa-libs. 3- Can we get an example of how transcoding (back to disk) is supposed to be working? 4- How can we reconfigure complex audio effect components (like shelving filters) which need filter co-effecient changes to be applied all at once atomically to avoid instability? My plan is to add mp3 driver first then build on that to add the rest of the features. Hopefully I can minimise the amount of work that needs doing (due to me doing it wrongly) and get features upstream sooner. Thanks, Qais

5 9

[alsa-devel] [PATCH] ALSA: echoaudio: cleanup of unnecessary messages
by Sudip Mukherjee 05 Nov '14

05 Nov '14

commit "b5b4a41b392960010fccf1f9ccf8334d612bd450" was dereferencing chip after it has been freed. This patch fixes that and at the same time removes some debugging messages, which are unnecessary, as they are just printing information about entry and exit from a function. This information we can easily get from ftrace. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/echoaudio/darla20_dsp.c | 2 -- sound/pci/echoaudio/darla24_dsp.c | 2 -- sound/pci/echoaudio/echo3g_dsp.c | 2 -- sound/pci/echoaudio/echoaudio.c | 25 ------------------------- sound/pci/echoaudio/echoaudio_3g.c | 1 - sound/pci/echoaudio/echoaudio_dsp.c | 12 ------------ sound/pci/echoaudio/gina20_dsp.c | 4 ---- sound/pci/echoaudio/gina24_dsp.c | 4 ---- sound/pci/echoaudio/indigo_dsp.c | 2 -- sound/pci/echoaudio/indigodj_dsp.c | 2 -- sound/pci/echoaudio/indigodjx_dsp.c | 2 -- sound/pci/echoaudio/indigoio_dsp.c | 2 -- sound/pci/echoaudio/indigoiox_dsp.c | 2 -- sound/pci/echoaudio/layla20_dsp.c | 5 ----- sound/pci/echoaudio/layla24_dsp.c | 4 ---- sound/pci/echoaudio/mona_dsp.c | 3 --- 16 files changed, 74 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index c94e92e..febee5b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20)) return -ENODEV; @@ -58,7 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index b1272f88..7b4f6fd 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24)) return -ENODEV; @@ -57,7 +56,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index bc37168..ae11ce1 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -46,7 +46,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) int err; local_irq_enable(); - dev_dbg(chip->card->dev, "init_hw() - Echo3G\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G)) return -ENODEV; @@ -99,7 +98,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1ef29e5..146ffc5 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -96,7 +96,6 @@ static void free_firmware_cache(struct echoaudio *chip) dev_dbg(chip->card->dev, "release_firmware(%d)\n", i); } - dev_dbg(chip->card->dev, "firmware_cache released\n"); #endif } @@ -354,7 +353,6 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err; - dev_dbg(chip->card->dev, "pcm_analog_in_open\n"); if ((err = pcm_open(substream, num_analog_busses_in(chip) - substream->number)) < 0) return err; @@ -389,7 +387,6 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) #else max_channels = num_analog_busses_out(chip); #endif - dev_dbg(chip->card->dev, "pcm_analog_out_open\n"); if ((err = pcm_open(substream, max_channels - substream->number)) < 0) return err; if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, @@ -422,7 +419,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_in_open\n"); max_channels = num_digital_busses_in(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -464,7 +460,6 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_out_open\n"); max_channels = num_digital_busses_out(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -511,7 +506,6 @@ static int pcm_close(struct snd_pcm_substream *substream) /* Nothing to do here. Audio is already off and pipe will be * freed by its callback */ - dev_dbg(chip->card->dev, "pcm_close\n"); atomic_dec(&chip->opencount); oc = atomic_read(&chip->opencount); @@ -620,7 +614,6 @@ static int init_engine(struct snd_pcm_substream *substream, spin_lock_irq(&chip->lock); set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den); spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_params ok\n"); return 0; } @@ -691,7 +684,6 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) } spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_freed\n"); snd_pcm_lib_free_pages(substream); return 0; } @@ -763,10 +755,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - dev_dbg(chip->card->dev, "pcm_trigger resume\n"); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dev_dbg(chip->card->dev, "pcm_trigger start\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -788,9 +778,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) chip->pipe_cyclic_mask); break; case SNDRV_PCM_TRIGGER_SUSPEND: - dev_dbg(chip->card->dev, "pcm_trigger suspend\n"); case SNDRV_PCM_TRIGGER_STOP: - dev_dbg(chip->card->dev, "pcm_trigger stop\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -800,7 +788,6 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = stop_transport(chip, channelmask); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev_dbg(chip->card->dev, "pcm_trigger pause\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -1881,10 +1868,8 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) static int snd_echo_free(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "Stop DSP...\n"); if (chip->comm_page) rest_in_peace(chip); - dev_dbg(chip->card->dev, "Stopped.\n"); if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1898,14 +1883,12 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->iores) release_and_free_resource(chip->iores); - dev_dbg(chip->card->dev, "MMIO freed.\n"); pci_disable_device(chip->pci); /* release chip data */ free_firmware_cache(chip); kfree(chip); - dev_dbg(chip->card->dev, "Chip freed.\n"); return 0; } @@ -1915,7 +1898,6 @@ static int snd_echo_dev_free(struct snd_device *device) { struct echoaudio *chip = device->device_data; - dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n"); return snd_echo_free(chip); } @@ -2038,7 +2020,6 @@ static int snd_echo_probe(struct pci_dev *pci, return -ENOENT; } - dev_dbg(&pci->dev, "Echoaudio driver starting...\n"); i = 0; err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -2191,7 +2172,6 @@ static int snd_echo_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct echoaudio *chip = dev_get_drvdata(dev); - dev_dbg(dev, "suspend start\n"); snd_pcm_suspend_all(chip->analog_pcm); snd_pcm_suspend_all(chip->digital_pcm); @@ -2218,7 +2198,6 @@ static int snd_echo_suspend(struct device *dev) pci_save_state(pci); pci_disable_device(pci); - dev_dbg(dev, "suspend done\n"); return 0; } @@ -2232,7 +2211,6 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - dev_dbg(dev, "resume start\n"); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); if (commpage_bak == NULL) @@ -2247,7 +2225,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_free(chip); return err; } - dev_dbg(dev, "resume init OK\n"); /* Temporarily set chip->pipe_alloc_mask=0 otherwise * restore_dsp_settings() fails. @@ -2260,7 +2237,6 @@ static int snd_echo_resume(struct device *dev) kfree(commpage_bak); return err; } - dev_dbg(dev, "resume restore OK\n"); memcpy(&commpage->audio_format, &commpage_bak->audio_format, sizeof(commpage->audio_format)); @@ -2286,7 +2262,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif - dev_dbg(dev, "resume done\n"); return 0; } diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 50a2169..7258328 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -328,7 +328,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index ba9d4f1..1a9427a 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,7 +206,6 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) goto la_error; } - dev_dbg(chip->card->dev, "ASIC loaded\n"); free_firmware(fw, chip); return 0; @@ -473,7 +472,6 @@ static int load_dsp(struct echoaudio *chip, u16 *code) chip->dsp_code = code; /* Show which DSP code loaded */ chip->bad_board = FALSE; /* DSP OK */ - dev_dbg(chip->card->dev, "load_dsp: OK!\n"); return 0; } udelay(100); @@ -674,7 +672,6 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { int i, o, err; - dev_dbg(chip->card->dev, "restore_dsp_settings\n"); if ((err = check_asic_status(chip)) < 0) return err; @@ -771,7 +768,6 @@ static int restore_dsp_rettings(struct echoaudio *chip) if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) return -EIO; - dev_dbg(chip->card->dev, "restore_dsp_rettings done\n"); return 0; } @@ -865,7 +861,6 @@ Same thing for pause_ and stop_ -trasport below. */ static int start_transport(struct echoaudio *chip, u32 channel_mask, u32 cyclic_mask) { - dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -891,7 +886,6 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask, static int pause_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -918,7 +912,6 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask) static int stop_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -954,8 +947,6 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index) stopped and unallocated. */ static int rest_in_peace(struct echoaudio *chip) { - dev_dbg(chip->card->dev, - "rest_in_peace() open=%x\n", chip->pipe_alloc_mask); /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); @@ -1018,7 +1009,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) */ static int init_line_levels(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "init_line_levels\n"); memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); @@ -1099,7 +1089,6 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, it moves data. The DMA counter is in units of bytes, not samples. */ pipe->dma_counter = &chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; - dev_dbg(chip->card->dev, "allocate_pipes: ok\n"); return pipe_index; } @@ -1110,7 +1099,6 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe) u32 channel_mask; int i; - dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index))) return -EINVAL; if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED)) diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index a959eae..a8dcf1e 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -37,7 +37,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20)) return -ENODEV; @@ -63,7 +62,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -150,7 +148,6 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) static int set_input_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_input_clock:\n"); switch (clock) { case ECHO_CLOCK_INTERNAL: @@ -209,7 +206,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index c8ea576..165fe3d 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24)) return -ENODEV; @@ -79,7 +78,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -156,7 +154,6 @@ static int load_asic(struct echoaudio *chip) control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; err = write_control_reg(chip, control_reg, TRUE); } - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -238,7 +235,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index cdeb073..54edd67 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 133915c..2cf5cc0 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 26cdfcf..5252863 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 5e6df7c..4e81787 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 90cdd27..6de3f9b 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 7f0f6ea..637e72f 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20)) return -ENODEV; @@ -65,7 +64,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -190,7 +188,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) u16 clock; u32 rate; - dev_dbg(chip->card->dev, "set_input_clock:\n"); rate = 0; switch (clock_source) { case ECHO_CLOCK_INTERNAL: @@ -232,7 +229,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) static int set_output_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock); switch (clock) { case ECHO_CLOCK_SUPER: clock = LAYLA20_OUTPUT_CLOCK_SUPER; @@ -286,7 +282,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index eb8f218..93813a8 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24)) return -ENODEV; @@ -70,7 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -118,7 +116,6 @@ static int load_asic(struct echoaudio *chip) if (chip->asic_loaded) return 1; - dev_dbg(chip->card->dev, "load_asic\n"); /* Give the DSP a few milliseconds to settle down */ mdelay(10); @@ -152,7 +149,6 @@ static int load_asic(struct echoaudio *chip) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, TRUE); - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index cc46a8c..63a6a50 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mona\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA)) return -ENODEV; @@ -72,7 +71,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -302,7 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) u32 control_reg, clocks_from_dsp; int err; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Prevent two simultaneous calls to switch_asic() */ if (atomic_read(&chip->opencount)) -- 1.8.1.2

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[alsa-devel] [PATCH v2 0/5] ASoC: Intel: ACPI support and machines
by Vinod Koul 05 Nov '14

05 Nov '14

This is second rev of patch series which fixes the modules split as discussed in ML. The core part is now module along with PCI and ACPI one. Added fix reported by Dan as well. Also added the BYTCR machine driver Subhransu S. Prusty (1): ASoC: Intel: add BYTCR machine driver with RT5640 Vinod Koul (4): ASoC: Intel: mrfld - remove unnecessary check for pointer ASoC: Intel: mrfld - create separate module for pci part ASoC: Intel: mrfld - add shim save restore ASoC: Intel: mrfld- add ACPI module sound/soc/intel/Kconfig | 22 ++- sound/soc/intel/Makefile | 2 + sound/soc/intel/bytcr_dpcm_rt5640.c | 258 +++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.c | 2 +- sound/soc/intel/sst/Makefile | 8 +- sound/soc/intel/sst/sst.c | 243 +++++----------------- sound/soc/intel/sst/sst.h | 9 +- sound/soc/intel/sst/sst_acpi.c | 386 +++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pci.c | 225 ++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 2 + 10 files changed, 958 insertions(+), 199 deletions(-) create mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_pci.c

4 18

[alsa-devel] [PATCH] ASoC: Intel: Fix the driver data not set issue
by Jie Yang 05 Nov '14

05 Nov '14

The priv_data is allocated again here wrongly, and it is not set to the driver data after assignment. This make the pdata->dev is NULL and oops occurs on the first call to hsw_volume_put. The resource has been allocated in driver probe callback hsw_pcm_dev_probe, so here just remove this sencond allocation is OK. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 21f1316..bc02a6e 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -866,7 +866,6 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) dev = platform->dev; dma_dev = pdata->dma_dev; - priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); priv_data->hsw = pdata->dsp; priv_data->dev = platform->dev; priv_data->pm_state = HSW_PM_STATE_D0; -- 1.9.1

3 3

[alsa-devel] [PATCH] ALSA: hda - fix mute led problem for three HP laptops
by Hui Wang 05 Nov '14

05 Nov '14

Without the fix, the mute led can't work on these three machines. After apply this fix, these three machines will fall back on the led control quirk as below, and through testing, the mute led works very well. PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED, ALC282_STANDARD_PINS, {0x12, 0x90a60140}, ... BugLink: https://bugs.launchpad.net/bugs/1389497 Tested-by: TieFu Chen <tienfu.chen(a)canonical.com> Cc: Kailang Yang <kailang(a)realtek.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ba72fff..1af917f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4744,9 +4744,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2246, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2247, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2248, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2249, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), -- 1.9.1

2 1

[alsa-devel] Compressed audio support over HDMI
by D K 05 Nov '14

05 Nov '14

Hi, I recently came across the compressed audio framework used with ALSA-SOC. Can the framework be used to send compressed audio to HDMI driver?.Like using tinycompress from usespace . thanks,

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Use specific r/w function for DSP mode
by Oder Chiou 04 Nov '14

04 Nov '14

In DSP mode, the register r/w should use the specific function to access that is invoked by address mapping of the DSP. The MX-65[1] is for switching DSP or codec mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 222 ++++++++++++++++++++++++++++++---------------- sound/soc/codecs/rt5677.h | 2 + 2 files changed, 149 insertions(+), 75 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 413bccb..74f65ff 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -539,53 +539,113 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } } +static int rt5677_hw_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + u8 r[1]; + u8 data[2]; + int ret; + + /* Write register */ + r[0] = reg & 0xff; + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = &r[0]; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return -EIO; + } + + *val = (data[0] << 8) | data[1]; + + dev_dbg(&client->dev, "read %02x => %04x\n", reg, *val); + + return 0; +} + +static int rt5677_hw_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + u8 data[3]; + int ret; + + data[0] = reg & 0xff; + data[1] = (0xff00 & val) >> 8; + data[2] = val & 0xff; + + dev_dbg(&client->dev, "write %02x = %04x\n", reg, val); + + ret = i2c_master_send(client, data, 3); + if (ret == 3) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + /** * rt5677_dsp_mode_i2c_write_addr - Write value to address on DSP mode. - * @codec: SoC audio codec device. + * @rt5677: Private Data. * @addr: Address index. * @value: Address data. * * * Returns 0 for success or negative error code. */ -static int rt5677_dsp_mode_i2c_write_addr(struct snd_soc_codec *codec, +static int rt5677_dsp_mode_i2c_write_addr(struct rt5677_priv *rt5677, unsigned int addr, unsigned int value, unsigned int opcode) { - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt5677->i2c; int ret; mutex_lock(&rt5677->dsp_cmd_lock); - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr msb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_LSB, addr & 0xffff); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr lsb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_DATA_MSB, value >> 16); if (ret < 0) { - dev_err(codec->dev, "Failed to set data msb value: %d\n", ret); + dev_err(&client->dev, "Failed to set data msb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_DATA_LSB, value & 0xffff); if (ret < 0) { - dev_err(codec->dev, "Failed to set data lsb value: %d\n", ret); + dev_err(&client->dev, "Failed to set data lsb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE, opcode); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_OP_CODE, opcode); if (ret < 0) { - dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + dev_err(&client->dev, "Failed to set op code value: %d\n", + ret); goto err; } @@ -597,42 +657,45 @@ err: /** * rt5677_dsp_mode_i2c_read_addr - Read value from address on DSP mode. - * @codec: SoC audio codec device. + * rt5677: Private Data. * @addr: Address index. * @value: Address data. * + * * Returns 0 for success or negative error code. */ static int rt5677_dsp_mode_i2c_read_addr( - struct snd_soc_codec *codec, unsigned int addr, unsigned int *value) + struct rt5677_priv *rt5677, unsigned int addr, unsigned int *value) { - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt5677->i2c; int ret; unsigned int msb, lsb; mutex_lock(&rt5677->dsp_cmd_lock); - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr msb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_LSB, addr & 0xffff); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr lsb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE , 0x0002); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_OP_CODE , 0x0002); if (ret < 0) { - dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + dev_err(&client->dev, "Failed to set op code value: %d\n", ret); goto err; } - regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, &msb); - regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, &lsb); + rt5677_hw_read(client, RT5677_DSP_I2C_DATA_MSB, &msb); + rt5677_hw_read(client, RT5677_DSP_I2C_DATA_LSB, &lsb); *value = (msb << 16) | lsb; err: @@ -643,17 +706,17 @@ err: /** * rt5677_dsp_mode_i2c_write - Write register on DSP mode. - * @codec: SoC audio codec device. + * rt5677: Private Data. * @reg: Register index. * @value: Register data. * * * Returns 0 for success or negative error code. */ -static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, +static int rt5677_dsp_mode_i2c_write(struct rt5677_priv *rt5677, unsigned int reg, unsigned int value) { - return rt5677_dsp_mode_i2c_write_addr(codec, 0x18020000 + reg * 2, + return rt5677_dsp_mode_i2c_write_addr(rt5677, 0x18020000 + reg * 2, value, 0x0001); } @@ -661,57 +724,33 @@ static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, * rt5677_dsp_mode_i2c_read - Read register on DSP mode. * @codec: SoC audio codec device. * @reg: Register index. + * @value: Register data. * * - * Returns Register value. + * Returns 0 for success or negative error code. */ -static unsigned int rt5677_dsp_mode_i2c_read( - struct snd_soc_codec *codec, unsigned int reg) +static int rt5677_dsp_mode_i2c_read( + struct rt5677_priv *rt5677, unsigned int reg, unsigned int *value) { - unsigned int value = 0; + int ret = rt5677_dsp_mode_i2c_read_addr(rt5677, 0x18020000 + reg * 2, + value); - rt5677_dsp_mode_i2c_read_addr(codec, 0x18020000 + reg * 2, &value); + *value &= 0xffff; - return value; + return ret; } -/** - * rt5677_dsp_mode_i2c_update_bits - update register on DSP mode. - * @codec: audio codec - * @reg: register index. - * @mask: register mask - * @value: new value - * - * - * Returns 1 for change, 0 for no change, or negative error code. - */ -static int rt5677_dsp_mode_i2c_update_bits(struct snd_soc_codec *codec, - unsigned int reg, unsigned int mask, unsigned int value) +static void rt5677_set_dsp_mode(struct snd_soc_codec *codec, bool on) { - unsigned int old, new; - int change, ret; - - ret = rt5677_dsp_mode_i2c_read(codec, reg); - if (ret < 0) { - dev_err(codec->dev, "Failed to read reg: %d\n", ret); - goto err; - } + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = rt5677_dsp_mode_i2c_write(codec, reg, new); - if (ret < 0) { - dev_err(codec->dev, - "Failed to write reg: %d\n", ret); - goto err; - } + if (on) { + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x2); + rt5677->is_dsp_mode = true; + } else { + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x0); + rt5677->is_dsp_mode = false; } - return change; - -err: - return ret; } static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) @@ -733,9 +772,14 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_write(rt5677->regmap, RT5677_GLB_CLK2, 0x0080); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); - regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07ff); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); + rt5677_set_dsp_mode(codec, true); ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1, codec->dev); @@ -751,8 +795,7 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) release_firmware(rt5677->fw2); } - rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, - 0x0); + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x0); regcache_cache_bypass(rt5677->regmap, false); regcache_cache_only(rt5677->regmap, true); @@ -762,9 +805,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) regcache_cache_only(rt5677->regmap, false); regcache_cache_bypass(rt5677->regmap, true); - rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, - 0x1); - rt5677_dsp_mode_i2c_write(codec, RT5677_PWR_DSP1, 0x0001); + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x1); + rt5677_set_dsp_mode(codec, false); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x0001); regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); @@ -3804,6 +3847,32 @@ static int rt5677_resume(struct snd_soc_codec *codec) #define rt5677_resume NULL #endif +static int rt5677_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5677_priv *rt5677 = i2c_get_clientdata(client); + + if (rt5677->is_dsp_mode) + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + else + rt5677_hw_read(context, reg, val); + + return 0; +} + +static int rt5677_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5677_priv *rt5677 = i2c_get_clientdata(client); + + if (rt5677->is_dsp_mode) + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + else + rt5677_hw_write(context, reg, val); + + return 0; +} + #define RT5677_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5677_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) @@ -3938,6 +4007,8 @@ static const struct regmap_config rt5677_regmap = { .volatile_reg = rt5677_volatile_register, .readable_reg = rt5677_readable_register, + .reg_read = rt5677_read, + .reg_write = rt5677_write, .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5677_reg, @@ -4063,6 +4134,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, rt5677); + rt5677->i2c = i2c; if (pdata) rt5677->pdata = *pdata; @@ -4094,7 +4166,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, msleep(10); } - rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap); + rt5677->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5677_regmap); if (IS_ERR(rt5677->regmap)) { ret = PTR_ERR(rt5677->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index d2c743c..ab81659f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1599,6 +1599,7 @@ struct rt5677_priv { struct regmap *regmap; const struct firmware *fw1, *fw2; struct mutex dsp_cmd_lock; + struct i2c_client *i2c; int sysclk; int sysclk_src; @@ -1614,6 +1615,7 @@ struct rt5677_priv { #endif bool dsp_vad_en; struct regmap_irq_chip_data *irq_data; + bool is_dsp_mode; }; #endif /* __RT5677_H__ */ -- 1.8.1.1.439.g50a6b54

2 5

[alsa-devel] [PATCH 1/2] ASoC: max98090: Correct pclk divisor settings
by Dylan Reid 04 Nov '14

04 Nov '14

The Baytrail-based chromebooks have a 20MHz mclk, the code was setting the divisor incorrectly in this case. According to the 98090 datasheet, the divisor should be set to DIV1 for 10 <= mclk <= 20. Correct this and the surrounding clock ranges as well to match the datasheet. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98090.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index d519294..1229554 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1941,13 +1941,13 @@ static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, * 0x02 (when master clk is 20MHz to 40MHz).. * 0x03 (when master clk is 40MHz to 60MHz).. */ - if ((freq >= 10000000) && (freq < 20000000)) { + if ((freq >= 10000000) && (freq <= 20000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV1); - } else if ((freq >= 20000000) && (freq < 40000000)) { + } else if ((freq > 20000000) && (freq <= 40000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV2); - } else if ((freq >= 40000000) && (freq < 60000000)) { + } else if ((freq > 40000000) && (freq <= 60000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV4); } else { -- 2.1.2.330.g565301e

2 2

[alsa-devel] [PATCH v6] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 04 Nov '14

04 Nov '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 143 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 146 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6b59471..3ee031e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..51c6c4f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -906,6 +906,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d8..1db2168 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1461,7 +1461,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1473,7 +1474,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index cdc837e..68f6057 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -729,6 +729,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2664,6 +2694,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3142,6 +3175,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config += w->params_select; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3222,8 +3258,35 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3231,14 +3294,48 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = kcalloc(num_params, sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + w_param_text[count] = kmemdup((void *)(config->stream_name), + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3246,6 +3343,25 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + private_value = + (unsigned long) kmemdup((void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + template.kcontrol_news = kmemdup(&kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3253,15 +3369,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dpcm: Fix race between FE/BE updates and trigger
by Takashi Iwai 04 Nov '14

04 Nov '14

DPCM can update the FE/BE connection states totally asynchronously from the FE's PCM state. Most of FE/BE state changes are protected by mutex, so that they won't race, but there are still some actions that are uncovered. For example, suppose to switch a BE while a FE's stream is running. This would call soc_dpcm_runtime_update(), which sets FE's runtime_update flag, then sets up and starts BEs, and clears FE's runtime_update flag again. When a device emits XRUN during this operation, the PCM core triggers snd_pcm_stop(XRUN). Since the trigger action is an atomic ops, this isn't blocked by the mutex, thus it kicks off DPCM's trigger action. It eventually updates and clears FE's runtime_update flag while soc_dpcm_runtime_update() is running concurrently, and it results in confusion. Usually, for avoiding such a race, we take a lock. There is a PCM stream lock for that purpose. However, as already mentioned, the trigger action is atomic, and we can't take the lock for the whole soc_dpcm_runtime_update() or other operations that include the lengthy jobs like hw_params or prepare. This patch provides an alternative solution. This adds a way to defer the conflicting trigger callback to be executed at the end of FE/BE state changes. For doing it, two things are introduced: - Each runtime_update state change of FEs is protected via PCM stream lock. - The FE's trigger callback checks the runtime_update flag. If it's not set, the trigger action is executed there. If set, mark the pending trigger action and returns immediately. - At the exit of runtime_update state change, it checks whether the pending trigger is present. If yes, it executes the trigger action at this point. Reported-and-tested-by: Qiao Zhou <zhouqiao(a)marvell.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc-dpcm.h | 2 ++ sound/soc/soc-pcm.c | 72 +++++++++++++++++++++++++++++++++++++----------- 2 files changed, 58 insertions(+), 16 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 2883a7a6f9f3..98f2ade0266e 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -102,6 +102,8 @@ struct snd_soc_dpcm_runtime { /* state and update */ enum snd_soc_dpcm_update runtime_update; enum snd_soc_dpcm_state state; + + int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; /* can this BE stop and free */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311afdeaa..57277dd79e11 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1522,13 +1522,36 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); } +static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); + +/* Set FE's runtime_update state; the state is protected via PCM stream lock + * for avoiding the race with trigger callback. + * If the state is unset and a trigger is pending while the previous operation, + * process the pending trigger action here. + */ +static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, + int stream, enum snd_soc_dpcm_update state) +{ + struct snd_pcm_substream *substream = + snd_soc_dpcm_get_substream(fe, stream); + + snd_pcm_stream_lock_irq(substream); + if (state == SND_SOC_DPCM_UPDATE_NO && fe->dpcm[stream].trigger_pending) { + dpcm_fe_dai_do_trigger(substream, + fe->dpcm[stream].trigger_pending - 1); + fe->dpcm[stream].trigger_pending = 0; + } + fe->dpcm[stream].runtime_update = state; + snd_pcm_stream_unlock_irq(substream); +} + static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; struct snd_pcm_runtime *runtime = fe_substream->runtime; int stream = fe_substream->stream, ret = 0; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); ret = dpcm_be_dai_startup(fe, fe_substream->stream); if (ret < 0) { @@ -1550,13 +1573,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_runtime(fe_substream); snd_pcm_limit_hw_rates(runtime); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; unwind: dpcm_be_dai_startup_unwind(fe, fe_substream->stream); be_err: - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; } @@ -1603,7 +1626,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ dpcm_be_dai_shutdown(fe, substream->stream); @@ -1617,7 +1640,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; } @@ -1665,7 +1688,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) int err, stream = substream->stream; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); dev_dbg(fe->dev, "ASoC: hw_free FE %s\n", fe->dai_link->name); @@ -1680,7 +1703,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) err = dpcm_be_dai_hw_free(fe, stream); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); mutex_unlock(&fe->card->mutex); return 0; @@ -1773,7 +1796,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, int ret, stream = substream->stream; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); memcpy(&fe->dpcm[substream->stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); @@ -1796,7 +1819,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS; out: - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); mutex_unlock(&fe->card->mutex); return ret; } @@ -1910,7 +1933,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); -static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream, ret; @@ -1984,6 +2007,23 @@ out: return ret; } +static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream; + + /* if FE's runtime_update is already set, we're in race; + * process this trigger later at exit + */ + if (fe->dpcm[stream].runtime_update != SND_SOC_DPCM_UPDATE_NO) { + fe->dpcm[stream].trigger_pending = cmd + 1; + return 0; /* delayed, assuming it's successful */ + } + + /* we're alone, let's trigger */ + return dpcm_fe_dai_do_trigger(substream, cmd); +} + int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -2027,7 +2067,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) dev_dbg(fe->dev, "ASoC: prepare FE %s\n", fe->dai_link->name); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* there is no point preparing this FE if there are no BEs */ if (list_empty(&fe->dpcm[stream].be_clients)) { @@ -2054,7 +2094,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); mutex_unlock(&fe->card->mutex); return ret; @@ -2201,11 +2241,11 @@ static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream) { int ret; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); ret = dpcm_run_update_startup(fe, stream); if (ret < 0) dev_err(fe->dev, "ASoC: failed to startup some BEs\n"); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; } @@ -2214,11 +2254,11 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) { int ret; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); ret = dpcm_run_update_shutdown(fe, stream); if (ret < 0) dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n"); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; } -- 2.1.3

3 2

Re: [alsa-devel] ALSA: echoaudio: remove all snd_printk
by Dan Carpenter 04 Nov '14

04 Nov '14

Hello Sudip Mukherjee, The patch b5b4a41b3929: "ALSA: echoaudio: remove all snd_printk" from Nov 3, 2014, leads to the following static checker warning: sound/pci/echoaudio/echoaudio.c:1908 snd_echo_free() error: dereferencing freed memory 'chip' sound/pci/echoaudio/echoaudio.c 1905 /* release chip data */ 1906 free_firmware_cache(chip); 1907 kfree(chip); ^^^^^^^^^^^ 1908 dev_dbg(chip->card->dev, "Chip freed.\n"); ^^^^^^^^^^ We have lots of printks in this function. We could just delete them all and no one would care. These days you can get the same information from ftrace. 1909 return 0; 1910 } regards, dan carpenter

3 2

[alsa-devel] [PATCH] ASoC: Intel: Fix build with CONFIG_SLEEP enabled.
by Liam Girdwood 04 Nov '14

04 Nov '14

Fix the following build error when CONFIG_SLEEP is enabled and CONFIG_RUNTIME is disabled. The BDW ADSP sleep PM functionality depends on the runtime pm calls for context save/restore. All error/warnings: >> ERROR: "snd_soc_suspend" undefined! >> ERROR: "snd_soc_resume" undefined! Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 3cb9ed7..afbcc9b 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -1011,7 +1011,6 @@ static int hsw_pcm_dev_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM #ifdef CONFIG_PM_RUNTIME static int hsw_pcm_runtime_idle(struct device *dev) @@ -1071,6 +1070,8 @@ static int hsw_pcm_runtime_resume(struct device *dev) #define hsw_pcm_runtime_resume NULL #endif +#if defined(CONFIG_PM_SLEEP) && defined(CONFIG_PM_RUNTIME) + static void hsw_pcm_complete(struct device *dev) { struct hsw_priv_data *pdata = dev_get_drvdata(dev); @@ -1166,6 +1167,11 @@ static int hsw_pcm_prepare(struct device *dev) return 0; } +#else +#define hsw_pcm_prepare NULL +#define hsw_pcm_complete NULL +#endif + static const struct dev_pm_ops hsw_pcm_pm = { .runtime_idle = hsw_pcm_runtime_idle, .runtime_suspend = hsw_pcm_runtime_suspend, @@ -1173,9 +1179,6 @@ static const struct dev_pm_ops hsw_pcm_pm = { .prepare = hsw_pcm_prepare, .complete = hsw_pcm_complete, }; -#else -#define hsw_pcm_pm NULL -#endif static struct platform_driver hsw_pcm_driver = { .driver = { -- 2.1.0

2 1

[alsa-devel] [PATCH 0/5] A few ALSA PCM core update patches
by Takashi Iwai 04 Nov '14

04 Nov '14

Hi, here is a small series of patches for cleaning up / enhancing the PCM core stuff. [PATCH 1/5] ALSA: pcm: Simplify snd_pcm_action_lock_irq() [PATCH 2/5] ALSA: pcm: Refactoring snd_pcm_action() [PATCH 3/5] ALSA: pcm: Correct PCM BUG error message [PATCH 4/5] ALSA: pcm: Replace PCM hwptr tracking with tracepoints [PATCH 5/5] ALSA: pcm: Add xrun_injection proc entry Takashi

1 5

[alsa-devel] ASoC: no dapm match for AMIC
by Fabio Estevam 04 Nov '14

04 Nov '14

Hi, I am running linux-next 20141104 on a imx6q-sabresd board and noticed the following error messages: [ 2.145602] imx-wm8962 sound: wm8962 <-> 202c000.ssi mapping ok [ 2.151654] imx-wm8962 sound: Connecting non-supply widget to supply widget is not supported (AMIC -> MICBIAS) [ 2.161710] imx-wm8962 sound: ASoC: no dapm match for AMIC --> (null) --> MICBIAS [ 2.169258] imx-wm8962 sound: ASoC: Failed to add route AMIC -> direct -> MICBIAS What would be the correct way to fix them? Thanks

2 2

[alsa-devel] [PATCH 1/2] ASoC: Remove card field from snd_soc_dai struct
by Lars-Peter Clausen 04 Nov '14

04 Nov '14

The card field of the snd_soc_dai field is very rarely used. We can use dai->component->card instead and remove the card field from the snd_soc_dai struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dai.h | 2 -- sound/soc/soc-core.c | 7 +------ sound/soc/soc-dapm.c | 2 +- 3 files changed, 2 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 45d0fa1..373d177 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -275,8 +275,6 @@ struct snd_soc_dai { unsigned int tx_mask; unsigned int rx_mask; - struct snd_soc_card *card; - struct list_head list; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3818cf3..d9b7b70 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1315,11 +1315,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, num, order); - /* config components */ - cpu_dai->card = card; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->card = card; - /* set default power off timeout */ rtd->pmdown_time = pmdown_time; @@ -2314,7 +2309,7 @@ EXPORT_SYMBOL_GPL(snd_soc_add_card_controls); int snd_soc_add_dai_controls(struct snd_soc_dai *dai, const struct snd_kcontrol_new *controls, int num_controls) { - struct snd_card *card = dai->card->snd_card; + struct snd_card *card = dai->component->card->snd_card; return snd_soc_add_controls(card, dai->dev, controls, num_controls, NULL, dai); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6bf2c97..c5136bb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1043,7 +1043,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list) { - struct snd_soc_card *card = dai->card; + struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; int paths; -- 1.8.0

2 3

[alsa-devel] [PATCH v5 0/3] Add Phytec pbab01 audio support
by Dmitry Lavnikevich 04 Nov '14

04 Nov '14

By mistake previous patchset versions was sent as in-reply-to to previous patchset versions, so they may have been lost among other mails. One of pathces was already applied by Mark Brown, here is 3 patches implementing audio support on Phytec phyFLEX-i.MX6 boards. This patchset version only renames audio card name to OnboardTLV320AIC3007 (on request from Phytec). Patches are based on branch for-next of git://git.kernel.org/pub/scm/linux/kernel/git/shawnguo/linux.git

3 6

[alsa-devel] [PATCH] ASoC: rt5645: make bias level more reasonale
by bardliao＠realtek.com 04 Nov '14

04 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patah separate bias level off to standby and off. The standby level will provide the necessary power for JD and push button functions. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index bc442f4..0e92e63 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2071,8 +2071,8 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { - case SND_SOC_BIAS_STANDBY: - if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2, @@ -2087,15 +2087,24 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, } break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_FV1 | RT5645_PWR_FV2, + RT5645_PWR_FV1 | RT5645_PWR_FV2); + break; + case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); snd_soc_write(codec, RT5645_GEN_CTRL1, 0x0128); - snd_soc_write(codec, RT5645_PWR_DIG1, 0x0000); - snd_soc_write(codec, RT5645_PWR_DIG2, 0x0000); - snd_soc_write(codec, RT5645_PWR_VOL, 0x0000); - snd_soc_write(codec, RT5645_PWR_MIXER, 0x0000); - snd_soc_write(codec, RT5645_PWR_ANLG1, 0x0000); - snd_soc_write(codec, RT5645_PWR_ANLG2, 0x0000); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2 | + RT5645_PWR_FV1 | RT5645_PWR_FV2, 0x0); break; default: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt286: reduce power consumption
by bardliao＠realtek.com 04 Nov '14

04 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch will optimize the power consumption of rt286. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 211 ++++++++++++++++++++++++++++++++++------------- 1 file changed, 155 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 4aa555c..97daa80 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -36,11 +36,13 @@ struct rt286_priv { struct regmap *regmap; + struct snd_soc_codec *codec; struct rt286_platform_data pdata; struct i2c_client *i2c; struct snd_soc_jack *jack; struct delayed_work jack_detect_work; int sys_clk; + int clk_id; struct reg_default *index_cache; }; @@ -298,7 +300,6 @@ static int rt286_support_power_controls[] = { static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) { unsigned int val, buf; - int i; *hp = false; *mic = false; @@ -309,67 +310,44 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) if (*hp) { /* power on HV,VERF */ regmap_update_bits(rt286->regmap, - RT286_POWER_CTRL1, 0x1001, 0x0); + RT286_DC_GAIN, 0x200, 0x200); + + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, + "HV"); + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, + "VREF"); /* power LDO1 */ - regmap_update_bits(rt286->regmap, - RT286_POWER_CTRL2, 0x4, 0x4); - regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24); - regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val); + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, + "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); - msleep(200); - i = 40; - while (((val & 0x0800) == 0) && (i > 0)) { - regmap_read(rt286->regmap, - RT286_CBJ_CTRL2, &val); - i--; - msleep(20); - } + regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24); + msleep(50); - if (0x0400 == (val & 0x0700)) { - *mic = false; + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xfcc0, 0xd400); + msleep(300); + regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val); - regmap_write(rt286->regmap, - RT286_SET_MIC1, 0x20); - /* power off HV,VERF */ - regmap_update_bits(rt286->regmap, - RT286_POWER_CTRL1, 0x1001, 0x1001); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL3, 0xc000, 0x0000); - regmap_update_bits(rt286->regmap, - RT286_CBJ_CTRL1, 0x0030, 0x0000); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL2, 0xc000, 0x0000); - } else if ((0x0200 == (val & 0x0700)) || - (0x0100 == (val & 0x0700))) { + if (0x0070 == (val & 0x0070)) { *mic = true; - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL3, 0xc000, 0x8000); - regmap_update_bits(rt286->regmap, - RT286_CBJ_CTRL1, 0x0030, 0x0020); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL2, 0xc000, 0x8000); } else { - *mic = false; + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xfcc0, 0xe400); + msleep(300); + regmap_read(rt286->regmap, + RT286_CBJ_CTRL2, &val); + if (0x0070 == (val & 0x0070)) + *mic = true; + else + *mic = false; } - - regmap_update_bits(rt286->regmap, - RT286_MISC_CTRL1, - 0x0060, 0x0000); - } else { - regmap_update_bits(rt286->regmap, - RT286_MISC_CTRL1, - 0x0060, 0x0020); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL3, - 0xc000, 0x8000); regmap_update_bits(rt286->regmap, - RT286_CBJ_CTRL1, - 0x0030, 0x0020); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL2, - 0xc000, 0x8000); + RT286_DC_GAIN, 0x200, 0x0); + } else { *mic = false; + regmap_write(rt286->regmap, RT286_SET_MIC1, 0x20); } } else { regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); @@ -378,6 +356,12 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) *mic = buf & 0x80000000; } + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "HV"); + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "VREF"); + if (!*hp) + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; } @@ -415,6 +399,17 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(rt286_mic_detect); +static int is_mclk_mode(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(source->codec); + + if (rt286->clk_id == RT286_SCLK_S_MCLK) + return 1; + else + return 0; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); @@ -568,7 +563,84 @@ static int rt286_adc_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt286_vref_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0400, 0x0000); + mdelay(50); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x38, 0x08); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x38, 0x30); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_mic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1, + 12, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1, + 0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1, + 13, 1, rt286_ldo2_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("MCLK MODE", RT286_PLL_CTRL1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC1 Input Buffer", SND_SOC_NOPM, + 0, 0, rt286_mic1_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1 Pin"), SND_SOC_DAPM_INPUT("DMIC2 Pin"), @@ -642,6 +714,25 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"ADC 0", NULL, "MCLK MODE", is_mclk_mode}, + {"ADC 1", NULL, "MCLK MODE", is_mclk_mode}, + {"Front", NULL, "MCLK MODE", is_mclk_mode}, + {"Surround", NULL, "MCLK MODE", is_mclk_mode}, + + {"HP Power", NULL, "LDO1"}, + {"HP Power", NULL, "LDO2"}, + + {"MIC1", NULL, "LDO1"}, + {"MIC1", NULL, "LDO2"}, + {"MIC1", NULL, "HV"}, + {"MIC1", NULL, "VREF"}, + {"MIC1", NULL, "MIC1 Input Buffer"}, + + {"SPO", NULL, "LDO1"}, + {"SPO", NULL, "LDO2"}, + {"SPO", NULL, "HV"}, + {"SPO", NULL, "VREF"}, + {"DMIC1", NULL, "DMIC1 Pin"}, {"DMIC2", NULL, "DMIC2 Pin"}, {"DMIC1", NULL, "DMIC Receiver"}, @@ -880,6 +971,7 @@ static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, } rt286->sys_clk = freq; + rt286->clk_id = clk_id; return 0; } @@ -915,13 +1007,18 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: mdelay(10); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0400, 0x0400); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); + break; case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); snd_soc_update_bits(codec, - RT286_DC_GAIN, 0x200, 0x0); + RT286_CBJ_CTRL1, 0x0400, 0x0000); break; default: @@ -962,6 +1059,7 @@ static int rt286_probe(struct snd_soc_codec *codec) { struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + rt286->codec = codec; codec->dapm.bias_level = SND_SOC_BIAS_OFF; if (rt286->i2c->irq) { @@ -1152,7 +1250,6 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (!rt286->pdata.cbj_en) { regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000); regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816); - regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0xb000); } else { @@ -1169,8 +1266,10 @@ static int rt286_i2c_probe(struct i2c_client *i2c, mdelay(10); - /*Power down LDO2*/ - regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + /*Power down LDO, VREF*/ + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0xc, 0x0); + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL1, 0x1001, 0x1001); /*Set depop parameter*/ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] [PATCH TINYCOMPRESS][RESEND] makefile.linux: allow env to override CROSS_COMPILE and CC
by Qais Yousef 04 Nov '14

04 Nov '14

allow env variables to override them without having to modify the makefile. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> --- I'm not sure why but I think the first send didn't reach the list, hence resending. makefile.linux | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/makefile.linux b/makefile.linux index 81782d0af461..c4dab6d3b7d5 100644 --- a/makefile.linux +++ b/makefile.linux @@ -9,8 +9,8 @@ LIBOBJ = ${LIBSRC:.c=.o} PLAYOBJ = ${PLAYSRC:.c=.o} RECOBJ = ${RECSRC:.c=.o} -CC = gcc -CROSS_COMPILE = +CROSS_COMPILE ?= +CC ?= ${CROSS_COMPILE}gcc CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress @@ -20,19 +20,19 @@ all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} + @${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} $(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} + @${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} $(RECBIN): ${RECOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} + @${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< - @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< + @${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} -- 2.1.0

1 0

[alsa-devel] [PATCH v2] snd-es18xx: Add GPO controls
by Ondrej Zary 04 Nov '14

04 Nov '14

Add GPO0 and GPO1 (General Purpose Outputs) controls to mixer. These can be used on some cards to control amplifier mute (seen in ES1868 datasheet) or additional onboard chips such as QX2130 QXpander processor. These GPOs are present on ES1868, ES1869, ES1887 and ES1888 chips. Tested on ES1868 with QX2130. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/isa/es18xx.c | 52 ++++++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 42 insertions(+), 10 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..0ead1c8 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -156,6 +156,7 @@ struct snd_es18xx { #define ES18XX_I2S 0x0200 /* I2S mixer control */ #define ES18XX_MUTEREC 0x0400 /* Record source can be muted */ #define ES18XX_CONTROL 0x0800 /* Has control ports */ +#define ES18XX_GPO_2BIT 0x1000 /* GPO0,1 controlled by PM port */ /* Power Management */ #define ES18XX_PM 0x07 @@ -1136,11 +1137,14 @@ static int snd_es18xx_reg_read(struct snd_es18xx *chip, unsigned char reg) return snd_es18xx_read(chip, reg); } -#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, invert) \ +#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, flags) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_es18xx_info_single, \ .get = snd_es18xx_get_single, .put = snd_es18xx_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } + .private_value = reg | (shift << 8) | (mask << 16) | (flags << 24) } + +#define ES18XX_FL_INVERT (1 << 0) +#define ES18XX_FL_PMPORT (1 << 1) static int snd_es18xx_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1159,10 +1163,14 @@ static int snd_es18xx_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; int val; - - val = snd_es18xx_reg_read(chip, reg); + + if (pm_port) + val = inb(chip->port + ES18XX_PM); + else + val = snd_es18xx_reg_read(chip, reg); ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; @@ -1175,7 +1183,8 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; unsigned char val; val = (ucontrol->value.integer.value[0] & mask); @@ -1183,6 +1192,15 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e val = mask - val; mask <<= shift; val <<= shift; + if (pm_port) { + unsigned char cur = inb(chip->port + ES18XX_PM); + + if ((cur & mask) == val) + return 0; + outb((cur & ~mask) | val, chip->port + ES18XX_PM); + return 1; + } + return snd_es18xx_reg_bits(chip, reg, mask, val) != val; } @@ -1304,7 +1322,7 @@ static struct snd_kcontrol_new snd_es18xx_opt_speaker = ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { -ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), +ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, ES18XX_FL_INVERT), ES18XX_SINGLE("Video Playback Switch", 0, 0x7f, 0, 1, 0), ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) @@ -1363,6 +1381,11 @@ static struct snd_kcontrol_new snd_es18xx_hw_volume_controls[] = { ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), }; +static struct snd_kcontrol_new snd_es18xx_opt_gpo_2bit[] = { +ES18XX_SINGLE("GPO0 Switch", 0, ES18XX_PM, 0, 1, ES18XX_FL_PMPORT), +ES18XX_SINGLE("GPO1 Switch", 0, ES18XX_PM, 1, 1, ES18XX_FL_PMPORT), +}; + static int snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; @@ -1629,10 +1652,10 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, switch (chip->version) { case 0x1868: - chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL; + chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL | ES18XX_GPO_2BIT; break; case 0x1869: - chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV; + chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV | ES18XX_GPO_2BIT; break; case 0x1878: chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL; @@ -1642,7 +1665,7 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, break; case 0x1887: case 0x1888: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; + chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_GPO_2BIT; break; default: snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n", @@ -1944,6 +1967,15 @@ static int snd_es18xx_mixer(struct snd_card *card) return err; } } + if (chip->caps & ES18XX_GPO_2BIT) { + for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_gpo_2bit); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_es18xx_opt_gpo_2bit[idx], + chip)); + if (err < 0) + return err; + } + } return 0; } -- Ondrej Zary

2 1

[alsa-devel] [Question about DPCM] dpcm_run_new_update races again xrun
by Qiao Zhou 04 Nov '14

04 Nov '14

Hi Mark, Liam I met one BE not-function issue when developing DPCM feature, and found dpcm_run_new_update is not protected well against xrun(interrupt context). Could you help to give some suggestions? The scenario is like this, taking audio playback for example: 1, FE1 <-> BE1, working well for some time. 2, disconnect BE1 and connect BE2(FE1 <-> BE2) 3, during FE1 connecting BE2, FE1 is still streaming data normally. Then an under-run happens. Below are the code sequence. soc_dpcm_runtime_update() { ... dpcm_connect_be() // connect FE1 & BE2 dpcm_run_new_update(fe, stream) { fe->dpcm.runtime_update = SND_SOC_DPCM_UPDATE_BE dpcm_run_update_startup(fe, stream) { dpcm_be_dai_startup(fe, stream) _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ | // under-run happens (interrupt context) | | snd_pcm_stop(substream) { | | dpcm_fe_dai_trigger(STOP) { | | fe->dpcm.runtime_update = SND_SOC_DPCM_UPDATE_FE | | // trigger stop FE1, BE2 | | fe->dpcm.runtime_update = SND_SOC_DPCM_UPDATE_NO | | } | | } | |_ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _| dpcm_be_dai_hw_params(fe, stream) dpcm_be_dai_prepare(fe, stream) if(fe state == STOP) return 0; dpcm_be_dai_trigger(fe, ....) fe->dpcm.runtime_update = SND_SOC_DPCM_UPDATE_NO ... } } After xrun handler finishes, the FE runtime update staus is UPDATE_NO. Then the following dpcm_be_dai_hw_params & dpcm_be_dai_prepare skip related driver API excuting with this FE runtime status, and return 0 to end runtime-startup. When user APP(ALSA lib/TinyALSA) detects xrun, usually it will do the substream prepare and write data again. Due to BE dai has not been ready for hw_params, the BE dai can't work properly after substream prepare and trigger start. After that system has no sound and can't be recovered, maybe due to FE1 doesn't know what's going on inside BE2. The under-run is random, and under-run handler does what's necessary to be done, though trigger-stop a BE which has not been started yet is not good in my opinion. I did an experiment to avoid be udpate checking in dpcm_be_dai_hw_params by Commenting out "if(!snd_soc_dpcm_be_can_update()}". This change is verified OK since APP will do the prepare & trigger start to resume the under-run. The patch is also attached. >From current code, dpcm_be_dai_hw_params is only callded by fe_hw_params & this runtime startup. This be update checking might be redundant or unnecessary in current code context. Could you help to give some suggestions? Please help to correct me if anything wrong. Thanks in advance. --- sound/soc/soc-pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311a..dfeb1e7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1697,10 +1697,6 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); - /* is this op for this BE ? */ - if (!snd_soc_dpcm_be_can_update(fe, be, stream)) - continue; - /* only allow hw_params() if no connected FEs are running */ if (!snd_soc_dpcm_can_be_params(fe, be, stream)) continue; -- 1.7.9.5 Best Regards Qiao

4 8

[alsa-devel] [PATCHv4] ASoC: simple-card: binding: update binding to support the new style.
by Xiubo Li 04 Nov '14

04 Nov '14

This patch will split the DT binding into old style and new style, the new style will be easier to add many DAI links from old single DAI link DTs. This patch will maintian compatibility with the old DTs. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Changes in v4: - Follow Jyri's advice. - Fix some small words or sentences errors. - Fix discription of the commit commnet. - Apply against the current code. - Add Notes. .../devicetree/bindings/sound/simple-card.txt | 215 ++++++++++++++------- 1 file changed, 148 insertions(+), 67 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c2e9841..5fe0796 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -1,15 +1,19 @@ -Simple-Card: +Device-Tree bindings for Simple Card Simple-Card specifies audio DAI connections of SoC <-> codec. -Required properties: +=== Top level's properties and subnodes === +*** Required properties *** - compatible : "simple-audio-card" -Optional properties: - +*** Optional properties *** - simple-audio-card,name : User specified audio sound card name, one string property. +- simple-audio-card,format : CPU/CODEC common audio format. + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" + (This is used for single DAI link & old style.) - simple-audio-card,widgets : Please refer to widgets.txt. - simple-audio-card,routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the @@ -17,63 +21,99 @@ Optional properties: source. - simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec mclk. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section bellow. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - +- simple-audio-card,frame-master : Indicates DAI link frame master. One phandle to a + CODEC DAI node. + (This is used for single DAI link & old style.) +- simple-audio-card,bitclock-master : Indicates DAI link bit clock master. One phandle to + a CODEC DAI node. + (This is used for single DAI link & old style.) + +*** Optional subnodes *** +- simple-audio-card,dai-link : Container for DAI link level properties and the CPU + and CODEC nodes. This container may be omitted + when the card has only one DAI link and using the old + style. See the examples and the section bellow. +- simple-audio-card,cpu : CPU DAI node. + (This is used for single DAI link & old style.) +- simple-audio-card,codec : CODEC DAI node. + (This is used for single DAI link & old style.) + +=== DAI link node's properties and its subnodes === + +*** Required subnodes *** +- cpu : CPU DAI node. +- codec : CODEC DAI node. + +*** Optional properties *** - format : CPU/CODEC common audio format. "i2s", "right_j", "left_j" , "dsp_a" "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - +- frame-master : Indicates DAI link frame master. One phandle to a + CODEC DAI node. + (This is One boolean property for old style.) +- bitclock-master : Indicates DAI link bit clock master. One phandle to + a CODEC DAI node. + (This is one boolean property for old style.) + +For backward compatibility, the frame-master and bitclock-master +can be used as boolean properties in CPU/CODEC DAI node indicating +that the CODEC DAI is the DAI link frame master and bit clock master. +In this case there should be no DAI link level nodes or boolean +properties of them. + +For all the properties in DAI link level will be applied to both +CPU and CODEC DAIs. + +=== CPU/CODEC DAI node's properties and its subnodes === + +*** Required properties *** +- sound-dai : One phandle and port of CPU/CODEC DAI device node. + +*** Optional properties *** +- bitclock-inversion : Boolean property. Add this if the DAI device uses bit + clock inversion. +- frame-inversion : Boolean property. Add this if the DAI device uses frame + clock inversion. +- frame-master : Boolean property. Add this if the CODEC DAI device is + frame master. + (This is used for single DAI link & old style.) +- bitclock-master : Boolean property. Add this if the CODEC DAI device is + bit clock master. + (This is used for single DAI link & old style.) - dai-tdm-slot-num : Please refer to tdm-slot.txt. - dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) +- clocks / system-clock-frequency : Specify CPU/CODEC DAI's clock if needed. It can + be specified via "clocks" if system has clock node + (= common clock), or "system-clock-frequency"(if system + doens't support common clock) + +=== Notes === +*** DAI hardware clock masters *** +For [prefix]frame-master and [prefix]bitclock-master, which prefix +is "simple-audio-card,", are both wrt the CODEC DAI. If the CODEC DAI +is bit clock and frame master then the CPU DAI will be bit clock and +frame slave, and vice versa. + +=== Examples === +*** CPU & CODEC DAI DT nodes *** +&i2c0 { + ak4648: ak4648@12 { + #sound-dai-cells = <0>; + compatible = "asahi-kasei,ak4648"; + reg = <0x12>; + }; +}; -Example 1 - single DAI link: +sh_fsi2: sh_fsi2@ec230000 { + #sound-dai-cells = <1>; + compatible = "renesas,sh_fsi2"; + reg = <0xec230000 0x400>; + interrupt-parent = <&gic>; + interrupts = <0 146 0x4>; +}; +Example 1 - single DAI link & old style: +bitclock-master and frame-master as phandles. sound { compatible = "simple-audio-card"; simple-audio-card,name = "VF610-Tower-Sound-Card"; @@ -91,6 +131,7 @@ sound { simple-audio-card,cpu { sound-dai = <&sh_fsi2 0>; + bitclock-inversion; }; dailink0_master: simple-audio-card,codec { @@ -99,24 +140,64 @@ sound { }; }; -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; +Example 2 - single DAI link & old style: +bitclock-master and frame-master as boolean properties. +sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; }; -}; -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; + simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + bitclock-master; + frame-master; + bitclock-inversion; + }; }; -Example 2 - many DAI links: +Example 3 - single DAI link & new style: +sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,dai-link { + format = "i2s"; + bitclock-master = <&dailink0_master>; + frame-master = <&dailink0_master>; + cpu { + sound-dai = <&sh_fsi2 0>; + frame-inversion; + }; + dailink0_master: codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + frame-inversion; + }; + }; +}; +Example 4 - many DAI links: sound { compatible = "simple-audio-card"; simple-audio-card,name = "Cubox Audio"; -- 2.1.0.27.g96db324

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic23: depends on SPI
by Sudip Mukherjee 04 Nov '14

04 Nov '14

Fix build errors when CONFIG_SPI is not enabled by making the driver depend on SPI. ERROR: "spi_setup" [sound/soc/codecs/snd-soc-tlv320aic23-spi.ko] undefined! ERROR: "devm_regmap_init_spi" [sound/soc/codecs/snd-soc-tlv320aic23-spi.ko] undefined! ERROR: "spi_register_driver" [sound/soc/codecs/snd-soc-tlv320aic23-spi.ko] undefined! Reported-by: Jim Davis <jim.epost(a)gmail.com> Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- reference of mail by Jim Davis at https://lkml.org/lkml/2014/11/3/551 sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6801b38..9a8ea56 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -603,6 +603,7 @@ config SND_SOC_TLV320AIC23_I2C config SND_SOC_TLV320AIC23_SPI tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI" + depends on SPI select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC26 -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2 1/2] ALSA: echoaudio: add reference of struct echoaudio
by Sudip Mukherjee 03 Nov '14

03 Nov '14

added reference of struct echoaudio to free_firmware function. this structure will be later used to get a reference of the card when converting snd_printk to dev_* in the next patch of the series. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- same as v1. sound/pci/echoaudio/echoaudio.c | 3 ++- sound/pci/echoaudio/echoaudio.h | 3 ++- sound/pci/echoaudio/echoaudio_dsp.c | 10 +++++----- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index d82321f..db1b247 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -69,7 +69,8 @@ static int get_firmware(const struct firmware **fw_entry, -static void free_firmware(const struct firmware *fw_entry) +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip) { #ifdef CONFIG_PM_SLEEP DE_ACT(("firmware not released (kept in cache)\n")); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index b86b88d..a4f112a 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -468,7 +468,8 @@ static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, struct echoaudio *chip, const short fw_index); -static void free_firmware(const struct firmware *fw_entry); +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip); #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 5a6a217..977b2bd 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,12 +206,12 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) } DE_INIT(("ASIC loaded\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; la_error: DE_INIT(("failed on write_dsp\n")); - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -317,11 +317,11 @@ static int install_resident_loader(struct echoaudio *chip) } DE_INIT(("Resident loader successfully installed\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; irl_error: - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -491,7 +491,7 @@ static int load_firmware(struct echoaudio *chip) if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); - free_firmware(fw); + free_firmware(fw, chip); if (err < 0) return err; -- 1.8.1.2

3 6

[alsa-devel] [PATCH] snd_user_file: fix memory leak
by Andreas Schwab 03 Nov '14

03 Nov '14

Call wordfree if and only if wordfree returns zero or WRDE_NOSPACE Signed-off-by: Andreas Schwab <schwab(a)suse.de> --- src/userfile.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/src/userfile.c b/src/userfile.c index 3a73836..72779da 100644 --- a/src/userfile.c +++ b/src/userfile.c @@ -44,19 +44,20 @@ int snd_user_file(const char *file, char **result) err = wordexp(file, &we, WRDE_NOCMD); switch (err) { case WRDE_NOSPACE: + wordfree(&we); return -ENOMEM; case 0: if (we.we_wordc == 1) break; + wordfree(&we); /* fall thru */ default: - wordfree(&we); return -EINVAL; } *result = strdup(we.we_wordv[0]); + wordfree(&we); if (*result == NULL) return -ENOMEM; - wordfree(&we); return 0; } -- 2.1.3 -- Andreas Schwab, SUSE Labs, schwab(a)suse.de GPG Key fingerprint = 0196 BAD8 1CE9 1970 F4BE 1748 E4D4 88E3 0EEA B9D7 "And now for something completely different."

2 1

[alsa-devel] [PATCH] USB-Audio: Add Scarlett 2i4 USB to S/PDIF blacklist
by Panu Matilainen 03 Nov '14

03 Nov '14

Like Scarlett 2i2, the 2i4 does not have any S/PDIF connections. Signed-off-by: Panu Matilainen <pmatilai(a)laiskiainen.org> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 77a48b9..a6754d9 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -48,6 +48,7 @@ USB-Audio.pcm.iec958_device { "Plantronics Wireless Audio" 999 "SB WoW Headset" 999 "Scarlett 2i2 USB" 999 + "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 "USB Device 0x46d:0x992" 999 } -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: Intel: Correct a macro for FW message
by Jie Yang 03 Nov '14

03 Nov '14

For the broadwell official released FW(Since 8.4.1.43), the macro SST_HSW_NO_CHANNELS is changed and fixed to 4, so here change it to 4. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index afd0ae1..387511f 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -21,7 +21,7 @@ #include <linux/kernel.h> #include <linux/platform_device.h> -#define SST_HSW_NO_CHANNELS 2 +#define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 #define SST_HSW_DX_CONTEXT_SIZE (640 * 1024) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Add Merrifield machine driver
by Vinod Koul 03 Nov '14

03 Nov '14

This patch adds machine driver support for Merrifield platform. Patch has dependency on Merrifield DSP driver. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- this is standalone patch with no depends on other code posted sound/soc/intel/Kconfig | 15 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/merr_dpcm_wm8958.c | 430 ++++++++++++++++++++++++++++++++++++ 3 files changed, 447 insertions(+), 0 deletions(-) create mode 100644 sound/soc/intel/merr_dpcm_wm8958.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af88..16d1801 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -76,3 +76,18 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_MRFLD_WM8958_MACH + tristate "SOC Machine Audio driver for Intel Merrifield MID platform" + depends on X86 + select SND_SOC_WM8994 + select MFD_CORE + select MFD_WM8994 + select REGULATOR_WM8994 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC + help + This adds support for ASoC machine driver for Intel(R) MID Merrifield platform + used as alsa device in audio substem in Intel(R) MID devices + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9ab43be..9d4b3d4 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -26,6 +26,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o +snd-merr-dpcm-wm8958-objs := merr_dpcm_wm8958.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -34,3 +35,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ +obj-$(CONFIG_SND_SOC_INTEL_MRFLD_WM8958_MACH) += snd-merr-dpcm-wm8958.o diff --git a/sound/soc/intel/merr_dpcm_wm8958.c b/sound/soc/intel/merr_dpcm_wm8958.c new file mode 100644 index 0000000..4ac6a72 --- /dev/null +++ b/sound/soc/intel/merr_dpcm_wm8958.c @@ -0,0 +1,430 @@ +/* + * merr_dpcm_wm8958.c - ASoc DPCM Machine driver for Intel Merrfield MID platform + * + * Copyright (C) 2013-14 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/async.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <linux/input.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include <linux/mfd/wm8994/pdata.h> +#include "../codecs/wm8994.h" +#include "sst-atom-controls.h" + +/* Codec PLL output clk rate */ +#define CODEC_SYSCLK_RATE 24576000 +/* Input clock to codec at MCLK1 PIN */ +#define CODEC_IN_MCLK1_RATE 19200000 +/* Input clock to codec at MCLK2 PIN */ +#define CODEC_IN_MCLK2_RATE 32768 +/* define to select between MCLK1 and MCLK2 input to codec as its clock */ +#define CODEC_IN_MCLK1 1 +#define CODEC_IN_MCLK2 2 +#define CODEC_BE 1 + +struct mrfld_8958_mc_private { + struct snd_soc_jack jack; + int jack_retry; +}; + +/* + * Function to switch the input clock for codec, When audio is in + * progress input clock to codec will be through MCLK1 which is 19.2MHz + * while in off state input clock to codec will be through 32KHz through + * MCLK2 + * card : Sound card structure + * src : Input clock source to codec + */ +static int mrfld_8958_set_codec_clk(struct snd_soc_card *card, int src) +{ + int ret; + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + + if (!aif1_dai) + return -ENODEV; + + switch (src) { + case CODEC_IN_MCLK1: + /* Turn ON the PLL to generate required sysclk rate + * from MCLK1 + */ + ret = snd_soc_dai_set_pll(aif1_dai, + WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + CODEC_IN_MCLK1_RATE, CODEC_SYSCLK_RATE); + if (ret < 0) { + dev_err(card->dev, "Failed to start FLL: %d\n", ret); + return ret; + } + /* Switch to MCLK1 input */ + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_FLL1, + CODEC_SYSCLK_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Failed to set codec sysclk configuration %d\n", + ret); + return ret; + } + break; + case CODEC_IN_MCLK2: + /* Switch to MCLK2 */ + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Failed to switch to MCLK2: %d\n", ret); + return ret; + } + /* Turn off PLL for MCLK1 */ + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, 0, 0, 0); + if (ret < 0) { + dev_err(card->dev, "Failed to stop the FLL: %d", ret); + return ret; + } + break; + default: + return -EINVAL; + } + return 0; +} + +static int mrfld_wm8958_set_clk(struct snd_soc_dai *codec_dai) +{ + int ret = 0; + struct snd_soc_card *card = codec_dai->card; + + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf0, 0xf0, 4, 24); + if (ret < 0) { + dev_err(card->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + ret = mrfld_8958_set_codec_clk(card, CODEC_IN_MCLK1); + + return ret; +} + +static int mrfld_8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + return mrfld_wm8958_set_clk(codec_dai); +} + +static const struct snd_soc_pcm_stream mrfld_wm8958_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static int merr_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int mrfld_8958_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + int ret = 0; + /* "wm8994-aif1" */ + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + + if (!aif1_dai) + return -ENODEV; + + if (dapm->dev != aif1_dai->dev) + return 0; + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + + ret = mrfld_wm8958_set_clk(aif1_dai); + break; + default: + break; + } + return ret; +} + +static int mrfld_8958_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + int ret = 0; + /* "wm8994-aif1" */ + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + + if (!aif1_dai) + return -ENODEV; + + if (dapm->dev != aif1_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + /* + * We are in standby down so */ + /* Switch to 32KHz MCLK2 input clock for codec + */ + ret = mrfld_8958_set_codec_clk(card, CODEC_IN_MCLK2); + break; + default: + break; + } + if (&card->dapm == dapm) + card->dapm.bias_level = level; + return ret; +} + +static const struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static const struct snd_soc_dapm_route map[] = { + { "Headphones", NULL, "HPOUT1L" }, + { "Headphones", NULL, "HPOUT1R" }, + + /* + * This machine uses 2 DMICs, other configs may use more so change + * below accordingly + */ + { "DMIC1DAT", NULL, "DMIC" }, + { "DMIC2DAT", NULL, "DMIC" }, + + /* + * MICBIAS2 is connected as Bias for AMIC so we link it + * here. Also AMIC wires up to IN1LP pin. + * DMIC is externally connected to 1.8V rail, so no link rqd. + */ + { "AMIC", NULL, "MICBIAS2" }, + { "IN1LP", NULL, "AMIC" }, + + /* dsp map link the SWM outs to codec AIF */ + { "AIF1 Playback", NULL, "ssp2 Tx"}, + { "ssp2 Tx", NULL, "codec_out0"}, + { "ssp2 Tx", NULL, "codec_out1"}, + { "codec_in0", NULL, "ssp2 Rx" }, + { "codec_in1", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture"}, + +}; + +static const struct wm8958_micd_rate micdet_rates[] = { + { 32768, true, 1, 4 }, + { 32768, false, 1, 1 }, + { 44100 * 256, true, 7, 10 }, + { 44100 * 256, false, 7, 10 }, +}; + +static int mrfld_8958_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + struct mrfld_8958_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + struct snd_soc_codec *codec = aif1_dai->codec; + + if (!aif1_dai) + return -ENODEV; + + ret = snd_soc_dai_set_tdm_slot(aif1_dai, 0xf0, 0xf0, 4, 24); + if (ret < 0) { + dev_err(card->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + if (!codec) { + dev_err(card->dev, "%s: we didnt find the codec pointer!\n", __func__); + return 0; + } + + ctx->jack_retry = 0; + ret = snd_soc_jack_new(codec, "Intel MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1, + &ctx->jack); + if (ret) { + dev_err(card->dev, "jack creation failed: %d\n", ret); + return ret; + } + + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_1, KEY_MEDIA); + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + + wm8958_mic_detect(codec, &ctx->jack, NULL, NULL, NULL, NULL); + + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; +static int mrfld_8958_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops mrfld_8958_ops = { + .startup = mrfld_8958_startup, +}; + +static struct snd_soc_ops mrfld_8958_be_ssp2_ops = { + .hw_params = mrfld_8958_hw_params, +}; + +struct snd_soc_dai_link mrfld_8958_msic_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Merrifield Audio Port", + .stream_name = "Merrifield Audio", + .cpu_dai_name = "Headset-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "sst-platform", + .init = mrfld_8958_init, + .ignore_suspend = 1, + .dynamic = 1, + .ops = &mrfld_8958_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Merrifield Compress Port", + .stream_name = "Merrifield Compress", + .platform_name = "sst-platform", + .cpu_dai_name = "Compress-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dynamic = 1, + .init = mrfld_8958_init, + }, + + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-platform", + .no_pcm = 1, + .codec_dai_name = "wm8994-aif1", + .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = merr_codec_fixup, + .ignore_suspend = 1, + .ops = &mrfld_8958_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_mrfld = { + .name = "wm8958-audio", + .dai_link = mrfld_8958_msic_dailink, + .num_links = ARRAY_SIZE(mrfld_8958_msic_dailink), + .set_bias_level = mrfld_8958_set_bias_level, + .set_bias_level_post = mrfld_8958_set_bias_level_post, + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = map, + .num_dapm_routes = ARRAY_SIZE(map), + .fully_routed = true, +}; + +static int snd_mrfld_8958_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct mrfld_8958_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (!drv) + return -ENOMEM; + + /* register the soc card */ + snd_soc_card_mrfld.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_mrfld, drv); + ret_val = snd_soc_register_card(&snd_soc_card_mrfld); + if (ret_val) { + dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret_val); + goto unalloc; + } + platform_set_drvdata(pdev, &snd_soc_card_mrfld); + return ret_val; + +unalloc: + return ret_val; +} + +static int snd_mrfld_8958_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_mrfld_8958_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "mrfld_wm8958", + }, + .probe = snd_mrfld_8958_mc_probe, + .remove = snd_mrfld_8958_mc_remove, +}; + +module_platform_driver(snd_mrfld_8958_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Merrifield MID Machine driver"); +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:mrfld_wm8958"); -- 1.7.0.4

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[alsa-devel] Query - PyAlsa Docs
by Chirag Ajmera 03 Nov '14

03 Nov '14

Dear, Where can I get documentation for pyalsa bindings ? The test files (eg cardtest1.py) included in the package fail with the below error ImportError: No module named alsacard import pyalsa - this package is found and loaded Kindly guide Thank you Best, Chirag A.

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[alsa-devel] [PATCH 1/1] PCI: EMU10K1: Deletion of unnecessary checks before three function calls
by SF Markus Elfring 03 Nov '14

03 Nov '14

The functions kfree(), release_firmware() and snd_util_memhdr_free() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/emu10k1/emu10k1_main.c | 9 +++------ sound/pci/emu10k1/emufx.c | 3 +-- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 2292697..b4458a6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1289,10 +1289,8 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) } if (emu->emu1010.firmware_thread) kthread_stop(emu->emu1010.firmware_thread); - if (emu->firmware) - release_firmware(emu->firmware); - if (emu->dock_fw) - release_firmware(emu->dock_fw); + release_firmware(emu->firmware); + release_firmware(emu->dock_fw); if (emu->irq >= 0) free_irq(emu->irq, emu); /* remove reserved page */ @@ -1301,8 +1299,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) (struct snd_util_memblk *)emu->reserved_page); emu->reserved_page = NULL; } - if (emu->memhdr) - snd_util_memhdr_free(emu->memhdr); + snd_util_memhdr_free(emu->memhdr); if (emu->silent_page.area) snd_dma_free_pages(&emu->silent_page); if (emu->ptb_pages.area) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 745f062..eb5c0ab 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -777,8 +777,7 @@ static void snd_emu10k1_ctl_private_free(struct snd_kcontrol *kctl) kctl->private_value = 0; list_del(&ctl->list); kfree(ctl); - if (kctl->tlv.p) - kfree(kctl->tlv.p); + kfree(kctl->tlv.p); } static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu, -- 2.1.3

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[alsa-devel] [PATCH 0/3] Trivial usb-audio cleanup patches
by Takashi Iwai 03 Nov '14

03 Nov '14

Hi, here are a few trivial cleanup patches for usb-audio driver. [PATCH 1/3] ALSA: usb-audio: Flatten probe and disconnect functions [PATCH 2/3] ALSA: usb-audio: Pass direct struct pointer instead of [PATCH 3/3] ALSA: usb-audio: Use strim() instead of open code Takashi

1 3

[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex.
by Peter Rosin 03 Nov '14

03 Nov '14

>From 86be84c4de4e7b21cfda9656a02a902c543210af Mon Sep 17 00:00:00 2001 From: Peter Rosin <peda(a)axentia.se> Date: Wed, 22 Oct 2014 16:45:29 +0200 Subject: [PATCH v2] ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex. The CMR divider register is shared by playback and capture. The SSC driver therefore tries to enforce rules so that the needed register content do not conflict during simultaneous playback/capture. However, the implementation also prevents changing the register content in half-duplex scenarios, which is needed when using the OSS API. Thus, only lock the divider if there is a stream in the other direction. Fixes the below program to not fail with the atmel ssc dai in master mode. #include <sys/ioctl.h> #include <unistd.h> #include <fcntl.h> #include <sys/soundcard.h> int main(void) { int fd; int format; int channels; int speed; if ((fd = open("/dev/dsp", O_WRONLY, 0)) == -1) { perror("open"); return 1; } format = AFMT_S16_LE; if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1) { perror("SNDCTL_DSP_SETFMT"); return 1; } channels = 2; if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { perror("SNDCTL_DSP_CHANNELS"); return 1; } speed = 22025; if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { perror("SNDCTL_DSP_SPEED"); return 1; } return 0; } Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index de433cfd..9ae8475 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -310,7 +310,10 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, * transmit and receive, so if a value has already * been set, it must match this value. */ - if (ssc_p->cmr_div == 0) + if (ssc_p->dir_mask != + (SSC_DIR_MASK_PLAYBACK | SSC_DIR_MASK_CAPTURE)) + ssc_p->cmr_div = div; + else if (ssc_p->cmr_div == 0) ssc_p->cmr_div = div; else if (div != ssc_p->cmr_div) -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: simple-card: add "invert" property for detect GPIOs
by Jianqun 03 Nov '14

03 Nov '14

Since hardware may invert detect GPIO of headphone or mic, add one property to support software invert. Signed-off-by: Jianqun <jay.xu(a)rock-chips.com> --- sound/soc/generic/simple-card.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d1b7293..f1bb537 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -29,7 +29,9 @@ struct simple_card_data { } *dai_props; unsigned int mclk_fs; int gpio_hp_det; + int gpio_hp_det_invert; int gpio_mic_det; + int gpio_mic_det_invert; struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; @@ -148,6 +150,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) simple_card_hp_jack_pins); simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; + simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert; snd_soc_jack_add_gpios(&simple_card_hp_jack, 1, &simple_card_hp_jack_gpio); } @@ -159,6 +162,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(simple_card_mic_jack_pins), simple_card_mic_jack_pins); simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; + simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert; snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, &simple_card_mic_jack_gpio); } @@ -381,6 +385,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); + enum of_gpio_flags flags; u32 val; int ret; @@ -436,13 +441,15 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } - priv->gpio_hp_det = of_get_named_gpio(node, - "simple-audio-card,hp-det-gpio", 0); + priv->gpio_hp_det = of_get_named_gpio_flags(node, + "simple-audio-card,hp-det-gpio", 0, &flags); + priv->gpio_hp_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); if (priv->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; - priv->gpio_mic_det = of_get_named_gpio(node, - "simple-audio-card,mic-det-gpio", 0); + priv->gpio_mic_det = of_get_named_gpio_flags(node, + "simple-audio-card,mic-det-gpio", 0, &flags); + priv->gpio_mic_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); if (priv->gpio_mic_det == -EPROBE_DEFER) return -EPROBE_DEFER; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Remove snd_soc_platform_driver suspend/resume callbacks
by Lars-Peter Clausen 03 Nov '14

03 Nov '14

Those are unused and new drivers should use device driver suspend/resume. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 4 ---- sound/soc/soc-core.c | 10 ---------- 2 files changed, 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ad47e96..edbb07b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -857,8 +857,6 @@ struct snd_soc_platform_driver { int (*probe)(struct snd_soc_platform *); int (*remove)(struct snd_soc_platform *); - int (*suspend)(struct snd_soc_dai *dai); - int (*resume)(struct snd_soc_dai *dai); struct snd_soc_component_driver component_driver; /* pcm creation and destruction */ @@ -891,8 +889,6 @@ struct snd_soc_platform { struct device *dev; const struct snd_soc_platform_driver *driver; - unsigned int suspended:1; /* platform is suspended */ - struct list_head list; struct snd_soc_component component; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dd8234d..00e944e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -592,17 +592,12 @@ int snd_soc_suspend(struct device *dev) for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_platform *platform = card->rtd[i].platform; if (card->rtd[i].dai_link->ignore_suspend) continue; if (cpu_dai->driver->suspend && !cpu_dai->driver->ac97_control) cpu_dai->driver->suspend(cpu_dai); - if (platform->driver->suspend && !platform->suspended) { - platform->driver->suspend(cpu_dai); - platform->suspended = 1; - } } /* close any waiting streams and save state */ @@ -775,17 +770,12 @@ static void soc_resume_deferred(struct work_struct *work) for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_platform *platform = card->rtd[i].platform; if (card->rtd[i].dai_link->ignore_suspend) continue; if (cpu_dai->driver->resume && !cpu_dai->driver->ac97_control) cpu_dai->driver->resume(cpu_dai); - if (platform->driver->resume && platform->suspended) { - platform->driver->resume(cpu_dai); - platform->suspended = 0; - } } if (card->resume_post) -- 1.8.0

2 2

[alsa-devel] [PATCH] ASoC: rockchip-i2s: add rockchip-max98090 driver support
by Jianqun 03 Nov '14

03 Nov '14

This patch is to add support for rockchip board with max98090 codec. It also needs sound node in dt file. Tested on rockchip board based on rk3288. Signed-off-by: Jianqun <jay.xu(a)rock-chips.com> --- sound/soc/rockchip/Kconfig | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 78fc159..482cbba 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -9,3 +9,16 @@ config SND_SOC_ROCKCHIP config SND_SOC_ROCKCHIP_I2S tristate + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP + help + Say Y or M if you want to add support for I2S driver for Rockchip + I2S device. + +config SND_SOC_ROCKCHIP_MAX98090 + tristate "ASoC support Rockchip board with max98090" + depends on I2C && GPIOLIB + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98090 + help + Say Y or M here if you want to add support for Rockchio board using + a max98090 codec. -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: tlv320aic23: add dependencies on I2C/SPI_MASTER
by Max Filippov 03 Nov '14

03 Nov '14

This fixes build errors in configurations with I2C/SPI master disabled. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7dad805..b0bcd0d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -553,10 +553,12 @@ config SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_I2C tristate "Texas Instruments TLV320AIC23 audio CODEC - I2C" + depends on I2C select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_SPI tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI" + depends on SPI_MASTER select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC26 -- 1.8.1.4

2 1

[alsa-devel] [PATCH] [RESEND] snd-es18xx: Add GPO controls
by Ondrej Zary 03 Nov '14

03 Nov '14

Add GPO0 and GPO1 (General Purpose Outputs) controls to mixer. These can be used on some cards to control amplifier mute (seen in ES1868 datasheet) or additional onboard chips such as QX2130 QXpander processor. These GPOs are present on ES1868, ES1869, ES1887 and ES1888 chips. Tested on ES1868 with QX2130. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/isa/es18xx.c | 49 +++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 39 insertions(+), 10 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..8966ed2 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -156,6 +156,7 @@ struct snd_es18xx { #define ES18XX_I2S 0x0200 /* I2S mixer control */ #define ES18XX_MUTEREC 0x0400 /* Record source can be muted */ #define ES18XX_CONTROL 0x0800 /* Has control ports */ +#define ES18XX_GPO_2BIT 0x1000 /* GPO0,1 controlled by PM port */ /* Power Management */ #define ES18XX_PM 0x07 @@ -1136,11 +1137,14 @@ static int snd_es18xx_reg_read(struct snd_es18xx *chip, unsigned char reg) return snd_es18xx_read(chip, reg); } -#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, invert) \ +#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, flags) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_es18xx_info_single, \ .get = snd_es18xx_get_single, .put = snd_es18xx_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } + .private_value = reg | (shift << 8) | (mask << 16) | (flags << 24) } + +#define ES18XX_FL_INVERT (1 << 0) +#define ES18XX_FL_PMPORT (1 << 1) static int snd_es18xx_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1159,10 +1163,14 @@ static int snd_es18xx_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; int val; - - val = snd_es18xx_reg_read(chip, reg); + + if (pm_port) + val = inb(chip->port + ES18XX_PM); + else + val = snd_es18xx_reg_read(chip, reg); ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; @@ -1175,7 +1183,8 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; unsigned char val; val = (ucontrol->value.integer.value[0] & mask); @@ -1183,6 +1192,12 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e val = mask - val; mask <<= shift; val <<= shift; + if (pm_port) { + val = (inb(chip->port + ES18XX_PM) & ~mask) | val; + outb(val, chip->port + ES18XX_PM); + return 0; + } + return snd_es18xx_reg_bits(chip, reg, mask, val) != val; } @@ -1304,7 +1319,7 @@ static struct snd_kcontrol_new snd_es18xx_opt_speaker = ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { -ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), +ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, ES18XX_FL_INVERT), ES18XX_SINGLE("Video Playback Switch", 0, 0x7f, 0, 1, 0), ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) @@ -1363,6 +1378,11 @@ static struct snd_kcontrol_new snd_es18xx_hw_volume_controls[] = { ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), }; +static struct snd_kcontrol_new snd_es18xx_opt_gpo_2bit[] = { +ES18XX_SINGLE("GPO0 Switch", 0, ES18XX_PM, 0, 1, ES18XX_FL_PMPORT), +ES18XX_SINGLE("GPO1 Switch", 0, ES18XX_PM, 1, 1, ES18XX_FL_PMPORT), +}; + static int snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; @@ -1629,10 +1649,10 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, switch (chip->version) { case 0x1868: - chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL; + chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL | ES18XX_GPO_2BIT; break; case 0x1869: - chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV; + chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV | ES18XX_GPO_2BIT; break; case 0x1878: chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL; @@ -1642,7 +1662,7 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, break; case 0x1887: case 0x1888: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; + chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_GPO_2BIT; break; default: snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n", @@ -1944,6 +1964,15 @@ static int snd_es18xx_mixer(struct snd_card *card) return err; } } + if (chip->caps & ES18XX_GPO_2BIT) { + for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_gpo_2bit); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_es18xx_opt_gpo_2bit[idx], + chip)); + if (err < 0) + return err; + } + } return 0; } -- Ondrej Zary

2 1

[alsa-devel] [PATCH 1/2] ALSA: echoaudio: add reference of struct echoaudio
by Sudip Mukherjee 03 Nov '14

03 Nov '14

added reference of struct echoaudio to free_firmware function. this structure will be later used to get a reference of the card when converting snd_printk to dev_* in the next patch of the series. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/echoaudio/echoaudio.c | 3 ++- sound/pci/echoaudio/echoaudio.h | 3 ++- sound/pci/echoaudio/echoaudio_dsp.c | 10 +++++----- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index d82321f..db1b247 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -69,7 +69,8 @@ static int get_firmware(const struct firmware **fw_entry, -static void free_firmware(const struct firmware *fw_entry) +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip) { #ifdef CONFIG_PM_SLEEP DE_ACT(("firmware not released (kept in cache)\n")); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index b86b88d..a4f112a 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -468,7 +468,8 @@ static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, struct echoaudio *chip, const short fw_index); -static void free_firmware(const struct firmware *fw_entry); +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip); #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 5a6a217..977b2bd 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,12 +206,12 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) } DE_INIT(("ASIC loaded\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; la_error: DE_INIT(("failed on write_dsp\n")); - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -317,11 +317,11 @@ static int install_resident_loader(struct echoaudio *chip) } DE_INIT(("Resident loader successfully installed\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; irl_error: - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -491,7 +491,7 @@ static int load_firmware(struct echoaudio *chip) if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); - free_firmware(fw); + free_firmware(fw, chip); if (err < 0) return err; -- 1.8.1.2

2 2

[alsa-devel] [PATCH 0/3] Kill the rest snd_print*() usages
by Takashi Iwai 03 Nov '14

03 Nov '14

Hi, it's a small series to get rid of snd_print*() usages in a few PCI sound drivers. There remain still a few, I know, but this is just a matter of cleanup, so let them be there... [PATCH 1/3] ALSA: ctxfi: Kill the rest snd_print*() [PATCH 2/3] ALSA: pcxhr: Kill the rest snd_print*() [PATCH 3/3] ALSA: au88x0: Kill the rest snd_print*() Takashi

3 8

Re: [alsa-devel] ASoC: Intel: mrfld: add DSP core controls
by Dan Carpenter 03 Nov '14

03 Nov '14

Hello Vinod Koul, The patch 24c8d14192cc: "ASoC: Intel: mrfld: add DSP core controls" from Oct 15, 2014, leads to the following static checker warning: sound/soc/intel/sst-atom-controls.c:1354 sst_map_modules_to_pipe() warn: variable dereferenced before check 'platform' (see line 1353) sound/soc/intel/sst-atom-controls.c 1348 static int sst_map_modules_to_pipe(struct snd_soc_platform *platform) 1349 { 1350 struct snd_soc_dapm_widget *w; 1351 int ret = 0; 1352 1353 list_for_each_entry(w, &platform->component.card->widgets, list) { ^^^^^^^^ New dereference. 1354 if (platform && is_sst_dapm_widget(w) && (w->priv)) { ^^^^^^^^ New check. 1355 struct sst_ids *ids = w->priv; 1356 1357 dev_dbg(platform->dev, "widget type=%d name=%s\n", 1358 w->id, w->name); 1359 INIT_LIST_HEAD(&ids->algo_list); 1360 INIT_LIST_HEAD(&ids->gain_list); 1361 ret = sst_fill_widget_module_info(w, platform); 1362 1363 if (ret < 0) 1364 return ret; 1365 1366 /* fill linked widgets */ 1367 if (ids->parent_wname != NULL) 1368 sst_fill_linked_widgets(platform, ids); 1369 } 1370 } 1371 return 0; 1372 } regards, dan carpenter

2 1

[alsa-devel] [PATCH 0/4] ASoC: Intel: add compressed support
by Vinod Koul 03 Nov '14

03 Nov '14

This add compressed support in sst ipc driver. Along with it two minor fixes on driver as well Fang, Yang A (1): ASoC: Intel: use correct firmware name Subhransu S. Prusty (1): ASoC: mfld-compress: implement .power callback Vinod Koul (2): ASoC: Intel: sst: load firmware using async callback ASoC: Intel: sst - add compressed ops handling sound/soc/intel/sst-mfld-platform-compress.c | 8 +- sound/soc/intel/sst-mfld-platform.h | 1 + sound/soc/intel/sst/sst.c | 15 ++ sound/soc/intel/sst/sst_drv_interface.c | 285 ++++++++++++++++++++++++++ sound/soc/intel/sst/sst_loader.c | 5 +- 5 files changed, 309 insertions(+), 5 deletions(-)

2 9

[alsa-devel] [PATCH] audio: sai: Add Power Management support
by Alison Wang 03 Nov '14

03 Nov '14

This patch adds Power Management support for SAI. Activate regmap cache with REGCACHE_RBTREE, and use regmap cache code to save and restore registers in suspend and resume. The Transmit Control Register (TCSR) and Receive Control Register(RCSR) should be volatile registers. Signed-off-by: Alison Wang <alison.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 33 ++++++++++++++++++++++++++++++++- 1 file changed, 32 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7eeb1dd..c7dd953 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -509,9 +509,11 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: case FSL_SAI_TFR: - case FSL_SAI_RFR: case FSL_SAI_TDR: + case FSL_SAI_RCSR: + case FSL_SAI_RFR: case FSL_SAI_RDR: return true; default: @@ -553,6 +555,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_RBTREE, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -668,6 +671,33 @@ static int fsl_sai_probe(struct platform_device *pdev) SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); } +#ifdef CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + /* Restore all registers */ + regcache_cache_only(sai->regmap, false); + regcache_sync(sai->regmap); + + return 0; +}; +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, @@ -680,6 +710,7 @@ static struct platform_driver fsl_sai_driver = { .name = "fsl-sai", .owner = THIS_MODULE, .of_match_table = fsl_sai_ids, + .pm = &fsl_sai_pm, }, }; module_platform_driver(fsl_sai_driver); -- 2.1.0.27.g96db324

4 5

[alsa-devel] [PATCH] ASoC: Intel: fix missing mutex
by Sudip Mukherjee 02 Nov '14

02 Nov '14

on error in block prepare, we were returning the error code while still holding the mutex. We are releasing the mutex in this patch before return. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/intel/sst-firmware.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 35788ad..dca6d9a 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -1120,6 +1120,7 @@ int sst_block_alloc_scratch(struct sst_dsp *dsp) ret = block_list_prepare(dsp, &dsp->scratch_block_list); if (ret < 0) { dev_err(dsp->dev, "error: scratch block prepare failed\n"); + mutex_unlock(&dsp->mutex); return ret; } -- 1.8.1.2

2 1

Re: [alsa-devel] ASoC HDMI receiver driver
by Tim Harvey 31 Oct '14

31 Oct '14

On Thu, Oct 30, 2014 at 12:41 AM, Lars-Peter Clausen <lars(a)metafoo.de> wrote: > On 10/30/2014 04:34 AM, Tim Harvey wrote: >> Greetings, >> >> I'm working on an ASoC driver (an mfd driver for an HDMI receiver chip >> with an accompanying codec driver for the audio capture portion, and a >> cpu-dai driver for the specific SoC its hooked to). I can't seem to >> figure out what kind of hook I need in order to fill out the >> snd_pcm_substream fields specifying the rate/channels/sample-format of >> the receiving HDMI audio stream. In this case HDMI audio (and thus the >> codec) supports several rates but it doesn't make sense that the codec >> is told what rate to capture at as the rate is dictated by the current >> stream the receiver is being provided. I'm thinking there is some hook >> I'm missing that allows the codec to fill in the params dynamically >> when the capture device is opened. >> >> Any pointers to how I deal with this? > > In the startup() callback of the CODEC driver you can apply additional > constraints to the available parameter ranges. See > snd_pcm_hw_constraints_... in include/sound/pcm.h > > E.g. to constrain the rate to one fixed value you could do > snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, rate, rate); > > - Lars > Lars, Thanks for the tip! I found that if I add some validation into the hw_params function for the codec, that I can return an -EINVAL if the requested params don't match the incoming stream and that will cause the ALSA application to at least error out. However, when I saw your response I then also tried setting the constraint and found that an invalid sample-rate gets 'refined' to the available rate thus the hw_params sanity check passes, yet the user application apparently does not feedback that the rate has been changed. For example if I do the following on a capture stream which is actually 44.1kHz from the HDMI source with a constraint to set the rate to 44100 the audio is played back at 44100 and has all kinds of pops etc: gst-launch alsasrc device="my-hdmi-receiver-card" ! audio/x-raw-int,rate=48000 ! queue ! alsasink device="playback-card" Where-as the following plays back fine: gst-launch alsasrc device="my-hdmi-receiver-card" ! audio/x-raw-int,rate=44100 ! queue ! alsasink device="playback-card" I find that the value passed to startup in the substream is 48000 which gets refined to 44100 if the constraints were set, which ends up passing hw_params 44100 but the 'application' still has issues. I think its more appropriate to validate the hw_params instead of using the constraints. Does this make sense? Regards, Tim

2 3

[alsa-devel] [PATCH] ASoC: CS42L51_I2C depends on I2C
by Randy Dunlap 31 Oct '14

31 Oct '14

From: Randy Dunlap <rdunlap(a)infradead.org> Fix build errors when CONFIG_I2C is not enabled by making the driver depend on I2C. ../sound/soc/codecs/cs42l51-i2c.c:55:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(cs42l51_i2c_driver); ^ ../sound/soc/codecs/cs42l51-i2c.c:55:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] ../sound/soc/codecs/cs42l51-i2c.c:55:1: warning: parameter names (without types) in function declaration [enabled by default] ../sound/soc/codecs/cs42l51-i2c.c:45:26: warning: 'cs42l51_i2c_driver' defined but not used [-Wunused-variable] static struct i2c_driver cs42l51_i2c_driver = { ^ Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20141031.orig/sound/soc/codecs/Kconfig +++ linux-next-20141031/sound/soc/codecs/Kconfig @@ -338,6 +338,7 @@ config SND_SOC_CS42L51 config SND_SOC_CS42L51_I2C tristate "Cirrus Logic CS42L51 CODEC (I2C)" + depends on I2C select SND_SOC_CS42L51 config SND_SOC_CS42L52

3 2

[alsa-devel] [PATCH 00/14] ASoC: AC'97 driver cleanups
by Lars-Peter Clausen 31 Oct '14

31 Oct '14

Hi, This series does a few AC'97 driver cleanups in preparation of the conversion of the drivers to regmap. Most of the patches are quite trivial. The last two patches remove the virtual registers from the wm9712 and wm9713 drivers, which is a requirement for regmap conversion. The last patch is the only one in this series that is a bit more complex. Currently the drivers have a open-coded implementation of shared controls which is replaced by the patch with the generic one from the ASoC core. Unfortunately this changes the controls that are exposed by the driver, instead of having a "Left HP Mixer ..." and a "Right HP Mixer ..." control for each mixer input there will only be a single "HP Mixer" control. This may or may not be considered breaking the userspace interface. If it is we need to take a different approach at removing the virtual register form the driver. - Lars Lars-Peter Clausen (14): ASoC: ad1980: Remove unused header ASoC: ad1980: Cleanup printk usage ASoC: ad1980: Use table based control setup ASoC: stac9766: Cleanup printk usage ASoC: stac9766: Use table based control setup ASoC: wm9705: Cleanup printk usage ASoC: wm9705: Use table based control setup ASoC: wm9712: Cleanup printk usage ASoC: wm9712: Use table based control setup ASoC: wm9713: Cleanup printk usage ASoC: wm9713: Use table based control setup ASoC: wm9713: Move driver state struct allocation to driver probe ASoC: wm9713: Use virtual control instead of virtual register ASoC: wm9712/wm9713: Use shared controls sound/soc/blackfin/bf5xx-ad1980.c | 2 - sound/soc/codecs/ad1980.c | 23 ++---- sound/soc/codecs/ad1980.h | 26 ------ sound/soc/codecs/stac9766.c | 9 +- sound/soc/codecs/wm9705.c | 14 ++-- sound/soc/codecs/wm9712.c | 136 +++++++----------------------- sound/soc/codecs/wm9713.c | 170 +++++++++++--------------------------- 7 files changed, 98 insertions(+), 282 deletions(-) delete mode 100644 sound/soc/codecs/ad1980.h -- 1.8.0

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[alsa-devel] [PATCH 0/9] ASoC: Intel: more modulartion of init and add ACPI
by Vinod Koul 31 Oct '14

31 Oct '14

The first patch is a fix of mfld-pcm driver Rest of the series is devoted to ACPI support with 7 patch moving the probe routine and sperating into common and PCI and later moving PCI code to sst-pci file so that it can be shared with ACPI Lats patch adds support for ACPI with BYT driver support. This series is dependent upon the PM and compressed series sent earlier today Subhransu S. Prusty (9): ASoC: Intel: mfld-pcm: Fix to Store device context in sst_data ASoC: Intel: move the driver wq init to a routine ASoC: Intel: move the lock and wq initialization to routine ASoC: Intel: move the driver context allocation to routine ASoC: Intel: modularize driver probe and remove ASoC: Intel: more probe modularization for sst ASoC: Intel: move PCI probe to a seprate file ASoC: Intel: add shim save context and restore routines ASoC: Intel: Add ACPI driver for SST sound/soc/intel/Kconfig | 7 + sound/soc/intel/sst-mfld-platform-pcm.c | 1 + sound/soc/intel/sst/Makefile | 8 + sound/soc/intel/sst/sst.c | 391 ++++++++++++------------------- sound/soc/intel/sst/sst.h | 7 + sound/soc/intel/sst/sst_acpi.c | 387 ++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pci.c | 225 ++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 2 + 8 files changed, 793 insertions(+), 235 deletions(-) create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_pci.c

3 24

[alsa-devel] [PATCH] ASoC: Intel: Add jack detection for Broadwell
by Jie Yang 31 Oct '14

31 Oct '14

Add jack dectection and event reporting for Broadwell. It use combo jack on BDW platform, which including Mic Jack pin and Headphone jack pin. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/broadwell.c | 49 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 46 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 0e550f1..52cb764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -19,6 +19,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/jack.h> #include <sound/pcm_params.h> #include "sst-dsp.h" @@ -26,8 +27,26 @@ #include "../codecs/rt286.h" +static struct snd_soc_jack broadwell_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin broadwell_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new broadwell_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), +}; + static const struct snd_soc_dapm_widget broadwell_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_MIC("DMIC1", NULL), @@ -42,7 +61,7 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { {"Speaker", NULL, "SPOL"}, /* HP jack connectors - unknown if we have jack deteck */ - {"Headphones", NULL, "HPO Pin"}, + {"Headphone Jack", NULL, "HPO Pin"}, /* other jacks */ {"MIC1", NULL, "Mic Jack"}, @@ -57,6 +76,27 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, }; +static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret = 0; + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset); + + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&broadwell_headset, + ARRAY_SIZE(broadwell_headset_pins), + broadwell_headset_pins); + if (ret) + return ret; + + rt286_mic_detect(codec, &broadwell_headset); + return 0; +} + + static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -116,7 +156,7 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) } /* always connected - check HP for jack detect */ - snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Line Jack"); @@ -196,6 +236,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .no_pcm = 1, .codec_name = "i2c-INT343A:00", .codec_dai_name = "rt286-aif1", + .init = broadwell_rt286_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_suspend = 1, @@ -213,6 +254,8 @@ static struct snd_soc_card broadwell_rt286 = { .owner = THIS_MODULE, .dai_link = broadwell_rt286_dais, .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .controls = broadwell_controls, + .num_controls = ARRAY_SIZE(broadwell_controls), .dapm_widgets = broadwell_widgets, .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), .dapm_routes = broadwell_rt286_map, -- 1.9.1

2 1

[alsa-devel] [PATCH TINYCOMPRESS] makefile.linux: allow env to override CROSS_COMPILE and CC
by Qais Yousef 31 Oct '14

31 Oct '14

allow env variables to override them without having to modify the makefile. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> --- makefile.linux | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/makefile.linux b/makefile.linux index 81782d0af461..c4dab6d3b7d5 100644 --- a/makefile.linux +++ b/makefile.linux @@ -9,8 +9,8 @@ LIBOBJ = ${LIBSRC:.c=.o} PLAYOBJ = ${PLAYSRC:.c=.o} RECOBJ = ${RECSRC:.c=.o} -CC = gcc -CROSS_COMPILE = +CROSS_COMPILE ?= +CC ?= ${CROSS_COMPILE}gcc CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress @@ -20,19 +20,19 @@ all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} + @${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} $(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} + @${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} $(RECBIN): ${RECOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} + @${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< - @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< + @${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} -- 2.1.0

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: Intel: add runtime power management support
by Vinod Koul 31 Oct '14

31 Oct '14

This series fixes the device open pm handling and adds the runtime pm support. Also we fix lock usage while changing the driver state Changes in v2: - fix style in comments - fix error prints and remove noisy debug Mythri P K (1): Audio: SST: use lock when changing SST state. Subhransu S. Prusty (1): ASoC: Intel: mrfld: Fix runtime pm calls in sst_open_pcm_stream Vinod Koul (1): ASoC: Intel: sst: add runtime power management handling sound/soc/intel/sst/sst.c | 65 +++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_drv_interface.c | 12 ++---- sound/soc/intel/sst/sst_ipc.c | 2 +- 3 files changed, 70 insertions(+), 9 deletions(-)

2 8

[alsa-devel] [PATCH 0/3] ASoC: Intel: add runtime power management support
by Vinod Koul 31 Oct '14

31 Oct '14

This series fixes the device opne pm handlinga dn adds the runtime pm support. Also we fix lock usage while changing teh driver state Mythri P K (1): Audio: SST: use lock when changing SST state. Subhransu S. Prusty (1): ASoC: Intel: mrfld: Fix runtime pm calls in sst_open_pcm_stream Vinod Koul (1): ASoC: Intel: sst: add runtime power management handling sound/soc/intel/sst/sst.c | 71 ++++++++++++++++++++++++++++++- sound/soc/intel/sst/sst_drv_interface.c | 12 ++---- sound/soc/intel/sst/sst_ipc.c | 2 +- 3 files changed, 75 insertions(+), 10 deletions(-)

2 6

[alsa-devel] [PATCH] ASoC: TLV320AIC23: make codecs selectable in Kconfig
by Max Filippov 31 Oct '14

31 Oct '14

Now that manual selection of drivers for audio subsystem components is preferred AIC23 codec must be selectable in Kconfig to make it possible. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..7dad805 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -552,11 +552,11 @@ config SND_SOC_TLV320AIC23 tristate config SND_SOC_TLV320AIC23_I2C - tristate + tristate "Texas Instruments TLV320AIC23 audio CODEC - I2C" select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_SPI - tristate + tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI" select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC26 -- 1.8.1.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix block is enabled multiple times issue
by Jie Yang 31 Oct '14

31 Oct '14

During FW parsing and loading, block_list_prepare() may be called for each raw data block copying and this may made the hsw_block_enable() called mutiple times, which increase block->users many times. The result of this is hsw_block_disable() can't power gated the related block when trying to free the blocks during suspend, and the power gating status also confused. Here check the block user status, only calling enable() for those blocks who has no user yet. Remember that this works correctlly on current case, where there are enough SRAM memory so different module won't share a memory block. For further usage, we may need restructure the struct sst_mem_block to save the module list who is using it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 35788ad..c451398 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -149,7 +149,7 @@ static int block_list_prepare(struct sst_dsp *dsp, /* enable each block so that's it'e ready for data */ list_for_each_entry(block, block_list, module_list) { - if (block->ops && block->ops->enable) { + if (block->ops && block->ops->enable && !block->users) { ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, -- 1.9.1

3 3

Re: [alsa-devel] Restore hidden register for default value
by Kailang 31 Oct '14

31 Oct '14

Hi Takashi, It's just the default value refill in initial state. It's only for ALC668(ALC3661). The patch was tested pass with issue machine in ODM site. Many Thanks. BR, Kailang > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, October 22, 2014 4:00 PM > To: Kailang > Subject: Re: Restore hidden register for default value > > At Wed, 22 Oct 2014 07:56:43 +0000, > Kailang wrote: > > > > Hi Takashi, > > > > We want to post ALC668 for hidden register issue. > > > > Could I post like below array for codec initial state? > > > > WORD coefIndexDefault[][2] = {{0x1, > 0xbebe},{0x2,0xaaaa},{0x3,0x0},{0x4,0x180},{0x6,0x0}, > > > {0x7,0xf80},{0x8,0x31},{0xa,0x60},{0xb,0x0},{0xc,0x7cf7},{0xd, > 0x1080},{0xe,0x7f7f}, > > > {0xf,0xcccc},{0x10,0xddcc},{0x11,0x1},{0x13,0x0},{0x14,0x2aa0} > ,{0x17,0xA940}, > > > {0x19,0},{0x1A,0},{0x1B,0},{0x1c,0x0},{0x1d,0x0},{0x1e,0x7418} > ,{0x1f,0x804}, > > > {0x20,0x4200},{0x21,0x468},{0x22,0x8ccc},{0x23,0x250},{0x24,0x7418}, > > > > > {0x27,0x0000},{0x28,0x8ccc},{0x2a,0xff00},{0x2b,0x8000},{0xa7,0xff00}} > > ; Because codec sometimes has wrong value for hidden register after > > resume back or reboot. > > It needed to refill the default value with upper array in > initial state. > > But the hidden registers was more for 59 records. > > Could I post it with no remark? Because it just the default > value. Thanks. > > Some comments would be nice, of course, but not mandatory. > > One concern is exactly which codecs should this be applied. > There are a bunch of codecs compatible with alc668. It's > only for ALC668, and not for others (ALC668, 670, whatever). > > > Takashi > > ------Please consider the environment before printing this e-mail. >

1 0

[alsa-devel] [PATCH] usb-audio: Duplex support for Digidesign Mbox 1 and clock source selector
by Damien Zammit 31 Oct '14

31 Oct '14

Hi, Please review this patch and my previous one. This patch provides duplex support for the Digidesign Mbox 1 sound card. This also depends on my previous patch for enabling the clock source selector because the mixer is enabled in this patch. I reduced the supported sampling rates to just 48kHz because the device works better in this mode with the SPDIF external sync. Damien

1 0

[alsa-devel] [PATCH] usb-audio: Add mixer control for Digidesign Mbox 1 clock source
by Damien Zammit 31 Oct '14

31 Oct '14

This patch provides a mixer control for selecting the clock source of the Digidesign Mbox 1 to either internal clock or S/PDIF external. Trial and error and bus snooping were the only way to get this information, but it works on the hardware.

2 5

[alsa-devel] [PATCH v2 2/2] [media] sound: Update au0828 quirks table
by Mauro Carvalho Chehab 30 Oct '14

30 Oct '14

The au0828 quirks table is currently not in sync with the au0828 media driver. Syncronize it and put them on the same order as found at au0828 driver, as all the au0828 devices with analog TV need the same quirks. Cc: stable(a)vger.kernel.org Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8f3e2bf100eb..83bddbdb90e9 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2827,14 +2827,22 @@ YAMAHA_DEVICE(0x7010, "UB99"), } AU0828_DEVICE(0x2040, 0x7200, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), AU0828_DEVICE(0x2040, 0x7210, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x7217, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721b, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721e, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721f, "Hauppauge", "HVR-950Q"), -AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), AU0828_DEVICE(0x2040, 0x7280, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7201, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x2040, 0x7211, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x2040, 0x7281, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x05e1, 0x0480, "Hauppauge", "Woodbury"), +AU0828_DEVICE(0x2040, 0x8200, "Hauppauge", "Woodbury"), +AU0828_DEVICE(0x2040, 0x7260, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7213, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), /* Digidesign Mbox */ { -- 1.9.3

3 5

[alsa-devel] [PATCH] ASoC: Intel: fix build with runtime PM disabled.
by Liam Girdwood 30 Oct '14

30 Oct '14

Fix the following errors: All error/warnings: >> sound/soc/intel/sst-haswell-pcm.c:1168:13: error: 'hsw_pcm_prepare' undeclared here (not in a function) .prepare = hsw_pcm_prepare, ^ >> sound/soc/intel/sst-haswell-pcm.c:1169:14: error: 'hsw_pcm_complete' undeclared here (not in a function) .complete = hsw_pcm_complete, ^ Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 14 ++++++-------- 2 files changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 0ea7c3d..ffd5728 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1711,7 +1711,7 @@ void sst_hsw_runtime_module_free(struct sst_module_runtime *runtime) sst_module_runtime_free(runtime); } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int sst_hsw_dx_state_dump(struct sst_hsw *hsw) { struct sst_dsp *sst = hsw->dsp; diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 4489a35..cd54dd9 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -1058,6 +1058,12 @@ static int hsw_pcm_runtime_resume(struct device *dev) return ret; } +#else +#define hsw_pcm_runtime_idle NULL +#define hsw_pcm_runtime_suspend NULL +#define hsw_pcm_runtime_resume NULL +#endif + static void hsw_pcm_complete(struct device *dev) { struct hsw_priv_data *pdata = dev_get_drvdata(dev); @@ -1153,14 +1159,6 @@ static int hsw_pcm_prepare(struct device *dev) return 0; } -#else -#define hsw_pcm_runtime_idle NULL -#define hsw_pcm_runtime_suspend NULL -#define hsw_pcm_runtime_resume NULL -#define hsw_pcm_runtime_complete NULL -#define hsw_pcm_runtime_prepare NULL -#endif - static const struct dev_pm_ops hsw_pcm_pm = { .runtime_idle = hsw_pcm_runtime_idle, .runtime_suspend = hsw_pcm_runtime_suspend, -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: Intel; Add debug output when boot fails.
by Liam Girdwood 30 Oct '14

30 Oct '14

Add the debug output from IPCD and IPCX when booting fails. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index b37d3ee..0ea7c3d 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1927,7 +1927,9 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, msecs_to_jiffies(IPC_BOOT_MSECS)); if (ret == 0) { - dev_err(hsw->dev, "error: audio DSP boot timeout\n"); + dev_err(hsw->dev, "error: audio DSP boot timeout IPCD 0x%x IPCX 0x%x\n", + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCD), + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX)); return -EIO; } @@ -2038,7 +2040,9 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) msecs_to_jiffies(IPC_BOOT_MSECS)); if (ret == 0) { ret = -EIO; - dev_err(hsw->dev, "error: ADSP boot timeout\n"); + dev_err(hsw->dev, "error: audio DSP boot timeout IPCD 0x%x IPCX 0x%x\n", + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCD), + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX)); goto boot_err; } -- 2.1.0

2 1

[alsa-devel] opengl 'leaking' sound to input.
by dE 30 Oct '14

30 Oct '14

I'm trying to record audio from the analog input device of my card, but whenever there seems to be some graphics intensive activity like glxgears, OpenGL assisted window manager, audacity graphs etc... a noise appears in the recording which sounds like some kind of sparking disturbance. I've recorded audio many times on this box before, but this started to happen suddenly (maybe after an ugprade, but old live CDs with proprietary nvidia drivers installed or newer nouveau drivers seem to have the same issue). Just to confirm, is this a hardware issue?

2 2

[alsa-devel] [ALSA 0/8] Yet more documentation updates
by Takashi Iwai 30 Oct '14

30 Oct '14

Here are more updates of patches to complete kerneldoc comments. At this time, HD-audio generic parser, control PCM and core stuff are covered. Also, ASoC files are included in alsa-driver-api documents now. [PATCH 1/8] ALSA: hda - Add kerneldoc comments to hda_generic.c [PATCH 2/8] ALSA: doc: Add ASoC codes into API documentation [PATCH 3/8] ALSA: control: Add missing kerneldoc comments to exported [PATCH 4/8] ALSA: pcm: Use static inline for [PATCH 5/8] ALSA: pcm: More kerneldoc updates [PATCH 6/8] ALSA: pcm: Convert params_*() with static inline [PATCH 7/8] ALSA: doc: More kerneldoc comments on core components [PATCH 8/8] ALSA: doc: Fix missing "I" for kerneldoc inclusion Takashi

1 8

[alsa-devel] [PATCH] ASoC: Intel: Work around to fix HW D3 potential crash issue
by Jie Yang 30 Oct '14

30 Oct '14

When using clock gatings to save power, there are some known issues: 1. core clock gating (DCLCGE) must be disabled during D0 and D3 entry and updating SRAM banks (VDRTCTL0). 2. DSP trunk clock gating (DTCGE) can cause FW crashes, disable it in D0. To align with the new W/A flow from FW team, we must set VDRTCTL0.D3PGD to 1 (D3 power gating disabled) at first startup and keep it all the time. ADSP will be in D0 on first boot by BIOS part of WA. Required delays must be preserved (waiting for HW to stabilize, after enabling CCG, changing SRAM PG, D3PG). D3->D0: 1. Disable core clock gating (VDRTCTL2.DCLCGE = 0) 2. Enable other CG apart from DTCG and DCLCG (VDRTCTL2. DCLCGE and DTCGE = 0) 3. Disable D3PG (VDRTCTL0.D3PGD = 1) 4. Power up necessary SRAM and wait at least for 18 clock cycles for every bank you have powered up 5. Set D0 state(PMCS.PS = 0), wait for HW 6. Restore MCLK (clkctl.smos, disabled in D3 entry point 4) 7. Stall and reset core, set CSR 8. Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us 9. Unreset core 10.Load FW, configure PLL and other necessary things 11.Unstall core Changing SRAM PG during D0: 1. Disable core clock gating (VDRTCTL2.DCLCGE = 0) 2. Set PG mask 3. Wait at least for 18 clock cycles for every bank you have powered up 4. Enable core clock gating, delay 50 us D0->D3: 1. Disable core clock gating (DCLCGE = 0) 2. Stall and reset core 3. Power down entire SRAM and wait at least for 18 clock cycles for every bank (Enable SRAM PG (ISRAMPGE = 0x3FF, DSRAMPGE = 0xFFFFF, D3SRAMPGD = 0), remember about preserving VDRTCTL0.D3PGD = 1) 4. Shutdown PLL, disable MCLK(clkctl.smos = 0), Enable DTCG to save power 5. Set D3 state(PMCS.PS = 3), delay 50 us 6. Enable core clock gating, delay 50 us Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-dsp.h | 14 ++++-- sound/soc/intel/sst-haswell-dsp.c | 102 ++++++++++++++++++++++++++++++++------ 2 files changed, 98 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 2753b85..f291e32 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -159,12 +159,18 @@ #define SST_VDRTCTL3 0xaC /* VDRTCTL0 */ -#define SST_VDRTCL0_APLLSE_MASK 1 -#define SST_VDRTCL0_DSRAMPGE_SHIFT 16 -#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) -#define SST_VDRTCL0_ISRAMPGE_SHIFT 6 +#define SST_VDRTCL0_D3PGD (1 << 0) +#define SST_VDRTCL0_D3SRAMPGD (1 << 1) +#define SST_VDRTCL0_DSRAMPGE_SHIFT 12 +#define SST_VDRTCL0_DSRAMPGE_MASK (0xfffff << SST_VDRTCL0_DSRAMPGE_SHIFT) +#define SST_VDRTCL0_ISRAMPGE_SHIFT 2 #define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) +/* VDRTCTL2 */ +#define SST_VDRTCL2_DCLCGE (1 << 1) +#define SST_VDRTCL2_DTCGE (1 << 10) +#define SST_VDRTCL2_APLLSE_MASK (1 << 31) + /* PMCS */ #define SST_PMCS 0x84 #define SST_PMCS_PS_MASK 0x3 diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 86aea34..57039b0 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -250,17 +250,42 @@ static irqreturn_t hsw_irq(int irq, void *context) static void hsw_set_dsp_D3(struct sst_dsp *sst) { u32 val; + u32 reg; + + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* switch off audio PLL, DRAM & IRAM blocks */ + /* enable power gating and switch off DRAM & IRAM blocks */ val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - val |= SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | + val |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; + val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD); writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); - /* Set D3 state */ + /* switch off audio PLL */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_APLLSE_MASK; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* disable MCLK(clkctl.smos = 0) */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, + SST_CLKCTL_MASK, 0); + + /* Set D3 state, delay 50 us */ val = readl(sst->addr.pci_cfg + SST_PMCS); val |= SST_PMCS_PS_MASK; writel(val, sst->addr.pci_cfg + SST_PMCS); + udelay(50); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + } static void hsw_reset(struct sst_dsp *sst) @@ -283,6 +308,16 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) int tries = 10; u32 reg; + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* Disable D3PG (VDRTCTL0.D3PGD = 1) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* Set D0 state */ reg = readl(sst->addr.pci_cfg + SST_PMCS); reg &= ~SST_PMCS_PS_MASK; @@ -300,14 +335,6 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) return -ENODEV; finish: - hsw_reset(sst); - - /* switch on audio PLL, DRAM & IRAM blocks */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg &= ~(SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | - SST_VDRTCL0_ISRAMPGE_MASK); - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); @@ -322,6 +349,28 @@ finish: SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); + /* Stall and reset core, set CSR */ + hsw_reset(sst); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + + /* switch on audio PLL */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg &= ~SST_VDRTCL2_APLLSE_MASK; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* disable DMA finish function for SSP0 & SSP1 */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, SST_CSR2_SDFD_SSP1); @@ -343,9 +392,6 @@ finish: sst_dsp_shim_write(sst, 0x80, 0x6); sst_dsp_shim_write(sst, 0xe0, 0x300a); - /* disable all clock gating */ - writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); - return 0; } @@ -497,6 +543,11 @@ static int hsw_block_enable(struct sst_mem_block *block) dev_dbg(block->dsp->dev, " enabled block %d:%d at offset 0x%x\n", block->type, block->index, block->offset); + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val &= ~SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); bit = hsw_block_get_bit(block); writel(val & ~bit, sst->addr.pci_cfg + SST_VDRTCTL0); @@ -504,6 +555,13 @@ static int hsw_block_enable(struct sst_mem_block *block) /* wait 18 DSP clock ticks */ udelay(10); + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + /*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/ sst_mem_block_dummy_read(block); return 0; @@ -521,10 +579,26 @@ static int hsw_block_disable(struct sst_mem_block *block) dev_dbg(block->dsp->dev, " disabled block %d:%d at offset 0x%x\n", block->type, block->index, block->offset); + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val &= ~SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); bit = hsw_block_get_bit(block); writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + /* wait 18 DSP clock ticks */ + udelay(10); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + return 0; } -- 1.9.1

1 0

[alsa-devel] Build error on Fedora 21
by Mark van Rossum 30 Oct '14

30 Oct '14

Hi alsa_utils_alsa-compile.sh fails with: make[1]: Entering directory '/usr/src/kernels/3.17.0-301.fc21.x86_64' CC [M] /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.o /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.c: In function ‘snd_info_version_read’: /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.c:1065:22: error: macro "__DATE__" might prevent reproducible builds [-Werror=date-time] "Compiled on " __DATE__ " for kernel %s" ^ cc1: some warnings being treated as errors scripts/Makefile.build:257: recipe for target '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.o' failed make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.o] Error 1 scripts/Makefile.build:404: recipe for target '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore' failed make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore] Error 2 Makefile:1373: recipe for target '_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d' failed make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d] Error 2 make[1]: Leaving directory '/usr/src/kernels/3.17.0-301.fc21.x86_64' Makefile:167: recipe for target 'compile' failed make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. -- The University of Edinburgh is a charitable body, registered in Scotland, with registration number SC005336.

2 1

[alsa-devel] [PATCH 1/2] [media] sound: simplify au0828 quirk table
by Mauro Carvalho Chehab 30 Oct '14

30 Oct '14

From: Mauro Carvalho Chehab <m.chehab(a)samsung.com> Add a macro to simplify au0828 quirk table. That makes easier to check it against the USB IDs at drivers/media/usb/au0828-card.c Cc: stable(a)vger.kernel.org Signed-off-by: Mauro Carvalho Chehab <m.chehab(a)samsung.com> Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752a420c..5ae1d02d17a3 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2804,133 +2804,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, -/* Hauppauge HVR-950Q and HVR-850 */ -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7200), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-850", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7280), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x0fd9, 0x0008), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, +/* + * Auvitek au0828 devices with audio interface. + * This should be kept in sync with drivers/media/usb/au0828-card.c + * Please notice that some drivers are DVB only, and don't need to be + * here. That's the case, for example, of DVICO_FUSIONHDTV7. + */ + +#define AU0828_DEVICE(vid, pid, vname, pname) { \ + USB_DEVICE_VENDOR_SPEC(vid, pid), \ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \ + USB_DEVICE_ID_MATCH_INT_CLASS | \ + USB_DEVICE_ID_MATCH_INT_SUBCLASS, \ + .bInterfaceClass = USB_CLASS_AUDIO, \ + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, \ + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { \ + .vendor_name = vname, \ + .product_name = pname, \ + .ifnum = QUIRK_ANY_INTERFACE, \ + .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + } \ +} + +AU0828_DEVICE(0x2040, 0x7200, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7210, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7217, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721b, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721e, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721f, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), +AU0828_DEVICE(0x2040, 0x7280, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"), /* Digidesign Mbox */ { -- 1.9.3

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[alsa-devel] [PATCH v2 1/2] [media] sound: simplify au0828 quirk table
by Mauro Carvalho Chehab 30 Oct '14

30 Oct '14

Add a macro to simplify au0828 quirk table. That makes easier to check it against the USB IDs at drivers/media/usb/au0828/au0828-cards.c. Cc: stable(a)vger.kernel.org Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> diff --git a/drivers/media/usb/au0828/au0828-cards.c b/drivers/media/usb/au0828/au0828-cards.c index 9eb77ac2153b..da87f1cc31a9 100644 --- a/drivers/media/usb/au0828/au0828-cards.c +++ b/drivers/media/usb/au0828/au0828-cards.c @@ -36,6 +36,11 @@ static void hvr950q_cs5340_audio(void *priv, int enable) au0828_clear(dev, REG_000, 0x10); } +/* + * WARNING: There's a quirks table at sound/usb/quirks-table.h + * that should also be updated every time a new device with V4L2 support + * is added here. + */ struct au0828_board au0828_boards[] = { [AU0828_BOARD_UNKNOWN] = { .name = "Unknown board", diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752a420c..8f3e2bf100eb 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2804,133 +2804,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, -/* Hauppauge HVR-950Q and HVR-850 */ -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7200), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-850", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7280), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x0fd9, 0x0008), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, +/* + * Auvitek au0828 devices with audio interface. + * This should be kept in sync with drivers/media/usb/au0828/au0828-cards.c + * Please notice that some drivers are DVB only, and don't need to be + * here. That's the case, for example, of DVICO_FUSIONHDTV7. + */ + +#define AU0828_DEVICE(vid, pid, vname, pname) { \ + USB_DEVICE_VENDOR_SPEC(vid, pid), \ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \ + USB_DEVICE_ID_MATCH_INT_CLASS | \ + USB_DEVICE_ID_MATCH_INT_SUBCLASS, \ + .bInterfaceClass = USB_CLASS_AUDIO, \ + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, \ + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { \ + .vendor_name = vname, \ + .product_name = pname, \ + .ifnum = QUIRK_ANY_INTERFACE, \ + .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + } \ +} + +AU0828_DEVICE(0x2040, 0x7200, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7210, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7217, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721b, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721e, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721f, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), +AU0828_DEVICE(0x2040, 0x7280, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"), /* Digidesign Mbox */ { -- 1.9.3

1 0

[alsa-devel] [PATCH 2/2] [media] sound: Update au0828 quirks table
by Mauro Carvalho Chehab 30 Oct '14

30 Oct '14

From: Mauro Carvalho Chehab <m.chehab(a)samsung.com> The au0828 quirks table is currently not in sync with the au0828 media driver. Syncronize it and put them on the same order as found at au0828 driver, as all the au0828 devices with analog TV need the same quirks. Cc: stable(a)vger.kernel.org Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 5ae1d02d17a3..8a6b366f2925 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2827,14 +2827,22 @@ YAMAHA_DEVICE(0x7010, "UB99"), } AU0828_DEVICE(0x2040, 0x7200, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), AU0828_DEVICE(0x2040, 0x7210, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x7217, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721b, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721e, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721f, "Hauppauge", "HVR-950Q"), -AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), AU0828_DEVICE(0x2040, 0x7280, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7201, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x2040, 0x7211, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x2040, 0x7281, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x05e1, 0x0480, "Hauppauge", "Woodbury"), +AU0828_DEVICE(0x2040, 0x8200, "Hauppauge", "Woodbury"), +AU0828_DEVICE(0x2040, 0x7260, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7213, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), /* Digidesign Mbox */ { -- 1.9.3

1 0

[alsa-devel] Selecting correct input device port.
by dE 30 Oct '14

30 Oct '14

As I learn't in this mailng list, a device can typically handle only one stream at a time unless it supports hardware mixing which's exposed via subdevices in ALSA What does subdevice mean for input device? Do they mark physical ports for input? If not how do I record from a specific input port? Is it automatically selected by the card? And what's 'input source' in ALSA mixer?

3 6

Re: [alsa-devel] Update Initial AMP for EAPD control
by Takashi Iwai 30 Oct '14

30 Oct '14

At Thu, 30 Oct 2014 07:38:08 +0000, Kailang wrote: > > Hi Takashi, > > Attach patch is update initial AMP for EAPD control. > > Change log: > - alc889_coef_init() is replaced with alc_update_coef_idx() with a > correct COEF value. > - for ALC262, ALC887 and ALC900, the EAPD setup via the hidden > register is removed because this rather conflicts with the EAPD verb > setup. > - For ALC888-VC, also the hidden register access is removed in > alc888_coef_init(). > - Remove the dead #if 0 code for ALC267/ALC268. > > This update will avoid wrong behavior for some codec. > This patch will fix double setup for EAPD. > It just need to turn on by one site for verb command or hidden register controlled. I found that there is one caller of alc889_coef_init(), so I modified it accordingly, too. Also fixed typos in the changelog, and edited a bit, then applied to git tree. thanks, Takashi > > Many Thanks. > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0001-update-init-amp-for-eapd.patch <application/octet-stream (base64)>] >

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Set GPIO 4 low for a few HP machines
by David Henningsson 30 Oct '14

30 Oct '14

These HP machines needs GPIO 4 low to enable the headphone amplifier. In addition, we still need to control LEDs via vref and GPIO. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1387128 Tested-by: TienFu Chen <tienfu.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bc0301..d16d8ec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3350,6 +3350,27 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec, } } +static void alc280_fixup_hp_gpio4(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Like hp_gpio_mic1_led, but also needs GPIO4 low to enable headphone amp */ + struct alc_spec *spec = codec->spec; + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x18 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 }, + {} + }; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook; + spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; + spec->gpio_led = 0; + spec->cap_mute_led_nid = 0x18; + snd_hda_add_verbs(codec, gpio_init); + codec->power_filter = led_power_filter; + } +} + static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4217,6 +4238,7 @@ enum { ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, + ALC280_FIXUP_HP_GPIO4, }; static const struct hda_fixup alc269_fixups[] = { @@ -4680,7 +4702,10 @@ static const struct hda_fixup alc269_fixups[] = { { }, }, }, - + [ALC280_FIXUP_HP_GPIO4] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc280_fixup_hp_gpio4, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4728,7 +4753,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x8004, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x8004, "HP", ALC269_FIXUP_HP_GPIO4), /* ALC290 */ SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), @@ -4742,7 +4767,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2256, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2258, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2258, "HP", ALC269_FIXUP_HP_GPIO4), SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -4751,7 +4776,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2277, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2277, "HP", ALC269_FIXUP_HP_GPIO4), SND_PCI_QUIRK(0x103c, 0x2278, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2282, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.9.1

2 2

[alsa-devel] [PATCH - hda 1/1] Add ultra dock support for Thinkpad X240.
by Lukas 30 Oct '14

30 Oct '14

Adding ultra doch support for Lenovo Thinkpad X240 (17aa:2214). Signed-off-by: Lukas Bossard <mr.bobukas(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34b7bdb..528bb2d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4803,6 +4803,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: es1968: Replace timeval with ktime_t
by Tina Ruchandani 30 Oct '14

30 Oct '14

es1968_measure_clock uses struct timeval, which on 32-bit systems will overflow in 2038, leading to incorrect interpretation of time.This patch changes the function to use ktime_t instead of struct timeval, which implies: - no y2038: ktime_t uses a 64-bit datatype explicitly. - efficent subtraction: The earlier version computes the difference in usecs while dealing with secs and nsecs. It requires checks to see if the nsecs of stop is less than start. This patch uses a direct subtract of ktime_t and converts to usecs. - use of monotonic clock (ktime_get) over real time (do_gettimeofday), which simplifies timekeeping, as it does not have to deal with cases where stop_time is less than start_time. Signed-off-by: Tina Ruchandani <ruchandani.tina(a)gmail.com> Reviewed-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/es1968.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a9956a7..6039700 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1710,7 +1710,8 @@ static void es1968_measure_clock(struct es1968 *chip) int i, apu; unsigned int pa, offset, t; struct esm_memory *memory; - struct timeval start_time, stop_time; + ktime_t start_time, stop_time; + ktime_t diff; if (chip->clock == 0) chip->clock = 48000; /* default clock value */ @@ -1761,12 +1762,12 @@ static void es1968_measure_clock(struct es1968 *chip) snd_es1968_bob_inc(chip, ESM_BOB_FREQ); __apu_set_register(chip, apu, 5, pa & 0xffff); snd_es1968_trigger_apu(chip, apu, ESM_APU_16BITLINEAR); - do_gettimeofday(&start_time); + start_time = ktime_get(); spin_unlock_irq(&chip->reg_lock); msleep(50); spin_lock_irq(&chip->reg_lock); offset = __apu_get_register(chip, apu, 5); - do_gettimeofday(&stop_time); + stop_time = ktime_get(); snd_es1968_trigger_apu(chip, apu, 0); /* stop */ snd_es1968_bob_dec(chip); chip->in_measurement = 0; @@ -1777,12 +1778,8 @@ static void es1968_measure_clock(struct es1968 *chip) offset &= 0xfffe; offset += chip->measure_count * (CLOCK_MEASURE_BUFSIZE/2); - t = stop_time.tv_sec - start_time.tv_sec; - t *= 1000000; - if (stop_time.tv_usec < start_time.tv_usec) - t -= start_time.tv_usec - stop_time.tv_usec; - else - t += stop_time.tv_usec - start_time.tv_usec; + diff = ktime_sub(stop_time, start_time); + t = ktime_to_us(diff); if (t == 0) { dev_err(chip->card->dev, "?? calculation error..\n"); } else { -- 2.1.0.rc2.206.gedb03e5

2 1

[alsa-devel] Shared volume control again
by Takashi Iwai 30 Oct '14

30 Oct '14

Hi, thinking of the problem about shared volume controls again, I stumbled on a case where your suggested solution doesn't work well. Namely, when a headphone volume is shared for two headphones (one on the laptop and one on the dock), how should it be named? One is "Headphone" and another is "Dock Headphone". That's a bit tricky. So, in the end, we need the mapping representation somehow more explicitly than the control name. Thoughts? Takashi

4 7

[alsa-devel] ASoC HDMI receiver driver
by Tim Harvey 30 Oct '14

30 Oct '14

Greetings, I'm working on an ASoC driver (an mfd driver for an HDMI receiver chip with an accompanying codec driver for the audio capture portion, and a cpu-dai driver for the specific SoC its hooked to). I can't seem to figure out what kind of hook I need in order to fill out the snd_pcm_substream fields specifying the rate/channels/sample-format of the receiving HDMI audio stream. In this case HDMI audio (and thus the codec) supports several rates but it doesn't make sense that the codec is told what rate to capture at as the rate is dictated by the current stream the receiver is being provided. I'm thinking there is some hook I'm missing that allows the codec to fill in the params dynamically when the capture device is opened. Any pointers to how I deal with this? Thanks, Tim

1 0

[alsa-devel] [PATCHv5 1/9] ASOC: wm8971: clean for checkpatch --strict
by Xavier Hsu 30 Oct '14

30 Oct '14

Before adding wm8973 support, clean up existing wm8971 style. Signed-off-by: Xavier Hsu <xavier.hsu(a)linaro.org> Signed-off-by: Andy Green <andy.green(a)linaro.org> --- sound/soc/codecs/wm8971.c | 110 +++++++++++++++++++++++---------------------- sound/soc/codecs/wm8971.h | 30 ++++++------- 2 files changed, 72 insertions(+), 68 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 39ddb9b..01c76a4 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -29,9 +29,9 @@ #include "wm8971.h" -#define WM8971_REG_COUNT 43 +#define WM8971_REG_COUNT 43 -static struct workqueue_struct *wm8971_workq = NULL; +static struct workqueue_struct *wm8971_workq; /* codec private data */ struct wm8971_priv { @@ -92,25 +92,28 @@ static const struct reg_default wm8971_reg_defaults[] = { #define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0) /* WM8971 Controls */ -static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; -static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz", - "200Hz @ 48kHz" }; -static const char *wm8971_treble[] = { "8kHz", "4kHz" }; -static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" }; -static const char *wm8971_ng_type[] = { "Constant PGA Gain", - "Mute ADC Output" }; -static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; -static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", - "Mono (Right)", "Digital Mono"}; -static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" }; -static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA", - "Differential"}; -static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA", - "Differential"}; -static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"}; -static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"}; -static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert", - "L + R Invert"}; +static const char const *wm8971_bass[] = {"Linear Control", "Adaptive Boost"}; +static const char const *wm8971_bass_filter[] = {"130Hz @ 48kHz", + "200Hz @ 48kHz"}; +static const char const *wm8971_treble[] = {"8kHz", "4kHz"}; +static const char const *wm8971_alc_func[] = {"Off", "Right", + "Left", "Stereo"}; +static const char const *wm8971_ng_type[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const char const *wm8971_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"}; +static const char const *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static const char const *wm8971_dac_phase[] = {"Non Inverted", "Inverted"}; +static const char const *wm8971_lline_mux[] = {"Line", "NC", "NC", + "PGA", "Differential"}; +static const char const *wm8971_rline_mux[] = {"Line", "Mic", "NC", + "PGA", "Differential"}; +static const char const *wm8971_lpga_sel[] = {"Line", "NC", "NC", + "Differential"}; +static const char const *wm8971_rpga_sel[] = {"Line", "Mic", "NC", + "Differential"}; +static const char const *wm8971_adcpol[] = {"Normal", "L Invert", + "R Invert", "L + R Invert"}; static const struct soc_enum wm8971_enum[] = { SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass), /* 0 */ @@ -136,24 +139,24 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1), SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V, - WM8971_ROUT1V, 7, 1, 0), + WM8971_ROUT1V, 7, 1, 0), SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V, - WM8971_ROUT2V, 7, 1, 0), + WM8971_ROUT2V, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0), SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0), SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1, - WM8971_LOUTM2, 4, 7, 1), + WM8971_LOUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1, - WM8971_ROUTM2, 4, 7, 1), + WM8971_ROUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1, - WM8971_MOUTM2, 4, 7, 1), + WM8971_MOUTM2, 4, 7, 1), SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V, - WM8971_ROUT1V, 0, 127, 0), + WM8971_ROUT1V, 0, 127, 0), SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V, - WM8971_ROUT2V, 0, 127, 0), + WM8971_ROUT2V, 0, 127, 0), SOC_ENUM("Bass Boost", wm8971_enum[0]), SOC_ENUM("Bass Filter", wm8971_enum[1]), @@ -238,14 +241,14 @@ SOC_DAPM_ENUM("Route", wm8971_enum[13]); static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_left_mixer_controls[0], - ARRAY_SIZE(wm8971_left_mixer_controls)), + &wm8971_left_mixer_controls[0], + ARRAY_SIZE(wm8971_left_mixer_controls)), SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_right_mixer_controls[0], - ARRAY_SIZE(wm8971_right_mixer_controls)), + &wm8971_right_mixer_controls[0], + ARRAY_SIZE(wm8971_right_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0, - &wm8971_mono_mixer_controls[0], - ARRAY_SIZE(wm8971_mono_mixer_controls)), + &wm8971_mono_mixer_controls[0], + ARRAY_SIZE(wm8971_mono_mixer_controls)), SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0), @@ -260,18 +263,18 @@ static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0), SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0, - &wm8971_left_pga_controls), + &wm8971_left_pga_controls), SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0, - &wm8971_right_pga_controls), + &wm8971_right_pga_controls), SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_left_line_controls), + &wm8971_left_line_controls), SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_right_line_controls), + &wm8971_right_line_controls), SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_OUTPUT("LOUT1"), SND_SOC_DAPM_OUTPUT("ROUT1"), @@ -431,7 +434,7 @@ static int get_coeff(int mclk, int rate) } static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) + int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -449,7 +452,7 @@ static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, } static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) + unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = 0; @@ -507,8 +510,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -553,7 +556,7 @@ static int wm8971_mute(struct snd_soc_dai *dai, int mute) } static int wm8971_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) + enum snd_soc_bias_level level) { u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; @@ -580,11 +583,12 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, } #define WM8971_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm8971_dai_ops = { .hw_params = wm8971_pcm_hw_params, @@ -616,6 +620,7 @@ static void wm8971_work(struct work_struct *work) container_of(work, struct snd_soc_dapm_context, delayed_work.work); struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + wm8971_set_bias_level(codec, codec->dapm.bias_level); } @@ -637,7 +642,7 @@ static int wm8971_resume(struct snd_soc_codec *codec) snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_ON; queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); + msecs_to_jiffies(1000)); } return 0; @@ -660,7 +665,7 @@ static int wm8971_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); + msecs_to_jiffies(1000)); /* set the update bits */ snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100); @@ -675,7 +680,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) return ret; } - /* power down chip */ static int wm8971_remove(struct snd_soc_codec *codec) { @@ -729,8 +733,8 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8971); - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8971, &wm8971_dai, 1); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8971, + &wm8971_dai, 1); return ret; } diff --git a/sound/soc/codecs/wm8971.h b/sound/soc/codecs/wm8971.h index f31c38f..dc38e21 100644 --- a/sound/soc/codecs/wm8971.h +++ b/sound/soc/codecs/wm8971.h @@ -22,25 +22,25 @@ #define WM8971_ADCDAC 0x05 #define WM8971_IFACE 0x07 #define WM8971_SRATE 0x08 -#define WM8971_LDAC 0x0a -#define WM8971_RDAC 0x0b -#define WM8971_BASS 0x0c +#define WM8971_LDAC 0x0a +#define WM8971_RDAC 0x0b +#define WM8971_BASS 0x0c #define WM8971_TREBLE 0x0d #define WM8971_RESET 0x0f -#define WM8971_ALC1 0x11 -#define WM8971_ALC2 0x12 -#define WM8971_ALC3 0x13 +#define WM8971_ALC1 0x11 +#define WM8971_ALC2 0x12 +#define WM8971_ALC3 0x13 #define WM8971_NGATE 0x14 -#define WM8971_LADC 0x15 -#define WM8971_RADC 0x16 -#define WM8971_ADCTL1 0x17 -#define WM8971_ADCTL2 0x18 -#define WM8971_PWR1 0x19 -#define WM8971_PWR2 0x1a -#define WM8971_ADCTL3 0x1b +#define WM8971_LADC 0x15 +#define WM8971_RADC 0x16 +#define WM8971_ADCTL1 0x17 +#define WM8971_ADCTL2 0x18 +#define WM8971_PWR1 0x19 +#define WM8971_PWR2 0x1a +#define WM8971_ADCTL3 0x1b #define WM8971_ADCIN 0x1f -#define WM8971_LADCIN 0x20 -#define WM8971_RADCIN 0x21 +#define WM8971_LADCIN 0x20 +#define WM8971_RADCIN 0x21 #define WM8971_LOUTM1 0x22 #define WM8971_LOUTM2 0x23 #define WM8971_ROUTM1 0x24 -- 1.7.9.5

2 10

[alsa-devel] [PATCH v6] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 29 Oct '14

29 Oct '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 143 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 146 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index aac04ff..20863f2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -532,6 +533,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index c83a334..a97848f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -926,6 +926,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d074aa9..93850be 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1436,7 +1436,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1448,7 +1449,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 177bd86..5f7a342 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -750,6 +750,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2701,6 +2731,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3185,6 +3218,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config += w->params_select; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3253,8 +3289,35 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3262,14 +3325,48 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = kcalloc(num_params, sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + w_param_text[count] = kmemdup((void *)(config->stream_name), + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3277,6 +3374,25 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + private_value = + (unsigned long) kmemdup((void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + template.kcontrol_news = kmemdup(&kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3284,15 +3400,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 29 Oct '14

29 Oct '14

XTFPGA boards providean audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 14 + sound/soc/xtensa/Makefile | 1 + sound/soc/xtensa/xtfpga-i2s.c | 647 +++++++++++++++++++++ 7 files changed, 683 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index acb4523..d7bfeec 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10204,6 +10204,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0e96233..9ca12e7 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -56,6 +56,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 534714a..99d7222 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -33,3 +33,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..98d6e0c --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,14 @@ +config SND_SOC_XTFPGA_I2S + tristate + select REGMAP_MMIO + +config SND_SOC_XTENSA_XTFPGA + tristate "SoC Audio for xtensa xtfpga" + depends on XTENSA_PLATFORM_XTFPGA + select SND_SOC_XTFPGA_I2S + select SND_SOC_TLV320AIC23_SPI + select SND_SIMPLE_CARD + help + Say Y or M if you want to add support for codecs attached to + xtfpga daughter board. You will also need to select the platform + to support below. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..97b8eb5 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1 @@ +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..393d5c3 --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,647 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 +#define XTFPGA_I2S_CONFIG_DEFAULT_LEVEL 3 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + bool clk_enabled; + struct regmap *regmap; + void __iomem *regs; + + struct snd_pcm_substream *tx_substream; + unsigned tx_ptr; /* next frame index in the sample buffer */ + unsigned tx_fifo_sz; /* current fifo level estimate */ + unsigned tx_fifo_low; /* FIFO level when level interrupt occurs */ + unsigned tx_fifo_high; /* maximal FIFO level */ + struct tasklet_struct refill_fifo; /* FIFO refilling tasklet */ +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static void xtfpga_i2s_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + struct snd_pcm_runtime *runtime; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + runtime = tx_substream->runtime; +#define xtfpga_i2s_tx_loop(i2s, runtime, channels, sample_bits) \ + do { \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ + for (; i2s->tx_fifo_sz < i2s->tx_fifo_high; \ + i2s->tx_fifo_sz += 2) { \ + iowrite32(p[i2s->tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[i2s->tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++i2s->tx_ptr >= runtime->buffer_size) \ + i2s->tx_ptr = 0; \ + } \ + } while (0) + + switch (runtime->sample_bits | (runtime->channels << 8)) { + case 0x110: + xtfpga_i2s_tx_loop(i2s, runtime, 1, 16); + break; + case 0x210: + xtfpga_i2s_tx_loop(i2s, runtime, 2, 16); + break; + case 0x120: + xtfpga_i2s_tx_loop(i2s, runtime, 1, 32); + break; + case 0x220: + xtfpga_i2s_tx_loop(i2s, runtime, 2, 32); + break; + } + rcu_read_unlock(); +} + +static void xtfpga_i2s_irq_update_config(struct xtfpga_i2s *i2s, + unsigned int_status) +{ + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); +} + +static void xtfpga_i2s_refill_fifo(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + unsigned int_status; + unsigned i; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + if (tx_substream && snd_pcm_running(tx_substream)) { + for (i = 0; i < 2; ++i) { + xtfpga_i2s_push_tx(i2s); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + i2s->tx_fifo_sz = i2s->tx_fifo_low; + } + } else { + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + } + rcu_read_unlock(); + xtfpga_i2s_irq_update_config(i2s, int_status); +} + +static void xtfpga_i2s_refill_fifo_tasklet(unsigned long p) +{ + xtfpga_i2s_refill_fifo((struct xtfpga_i2s *)p); +} + +static irqreturn_t xtfpga_i2s_interrupt(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + int tx_active; + unsigned int_status; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + if (!(int_status & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_sz = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + i2s->tx_fifo_sz = i2s->tx_fifo_low; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE, 0); + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + } + rcu_read_unlock(); + + if (tx_active) { + if (i2s->tx_fifo_high < 256) + xtfpga_i2s_refill_fifo(i2s); + else + tasklet_hi_schedule(&i2s->refill_fifo); + } else { + xtfpga_i2s_irq_update_config(i2s, int_status); + } + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_start(struct xtfpga_i2s *i2s, + struct snd_pcm_substream *substream) +{ + i2s->tx_ptr = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + + xtfpga_i2s_push_tx(i2s); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_VALID); + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + return 0; +} + +static int xtfpga_i2s_stop(struct xtfpga_i2s *i2s, + struct snd_pcm_substream *substream) +{ + rcu_assign_pointer(i2s->tx_substream, NULL); + return 0; +} + +static int xtfpga_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret; + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = xtfpga_i2s_start(i2s, substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = xtfpga_i2s_stop(i2s, substream); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S32_LE: + break; + + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + if (i2s->clk_enabled) { + clk_disable_unprepare(i2s->clk); + i2s->clk_enabled = false; + } + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + err = clk_prepare_enable(i2s->clk); + if (err < 0) + return err; + i2s->clk_enabled = true; + + ratio = (freq - srate * sample_size * 8) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + for (level = 1; + /* period_size * 2: FIFO always gets 2 samples per frame */ + i2s->tx_fifo_low / 2 >= period_size * 2; + ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + i2s->tx_fifo_sz = 0; + i2s->tx_ptr = 0; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + if (i2s->clk_enabled) { + clk_disable_unprepare(i2s->clk); + i2s->clk_enabled = false; + } + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = 0; + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = i2s->tx_ptr; + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .trigger = xtfpga_i2s_trigger, + .hw_params = xtfpga_i2s_hw_params, + .hw_free = xtfpga_i2s_hw_free, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem, *irq; + int err; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + err = -ENODEV; + goto err; + } + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + irq = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!irq) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = -ENODEV; + goto err; + } + err = devm_request_irq(&pdev->dev, irq->start, xtfpga_i2s_interrupt, + IRQF_SHARED, pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + tasklet_init(&i2s->refill_fifo, xtfpga_i2s_refill_fifo_tasklet, + (unsigned long)i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) + dev_err(&pdev->dev, "couldn't register component\n"); +err: + if (err) + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + if (i2s) { + tasklet_kill(&i2s->refill_fifo); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + if (i2s->clk_enabled) + clk_disable_unprepare(i2s->clk); + } + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

4 18

[alsa-devel] [PATCH] ASoC: Fix snd_soc_find_dai() matching component by name
by Lars-Peter Clausen 29 Oct '14

29 Oct '14

Commit 14621c7e5e72 ("ASoC: Consolidate CPU and CODEC DAI lookup") consolidated the lookup of CPU DAIs and CODEC DAIs into a single function. When matching a component by name for CODEC DAIs the code previous to the patch compared the name in the DAI link table with component->name. For CPU DAIs the code compared to dev_name(component->dev). The newly introduced function ended up using the later as well. For most components dev_name(component->dev) and component->name are the same. The main notable exception are I2C devices where the driver name and the device name are concatenated to form the component name. By using dev_name(component->dev) instead of component->name the patch broke the matching of I2C CODECs by name. This patch restores the original behavior by using component->name instead of dev_name(component->dev). This will be safe even for CPU DAIs since for CPU DAIs both are the same. Fixes: 14621c7e5e72 ("ASoC: Consolidate CPU and CODEC DAI lookup") Reported-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 62fbb63..b900f0d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -884,7 +884,7 @@ static struct snd_soc_dai *snd_soc_find_dai( list_for_each_entry(component, &component_list, list) { if (dlc->of_node && component->dev->of_node != dlc->of_node) continue; - if (dlc->name && strcmp(dev_name(component->dev), dlc->name)) + if (dlc->name && strcmp(component->name, dlc->name)) continue; list_for_each_entry(dai, &component->dai_list, list) { if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)) -- 1.8.0

2 1

[alsa-devel] [PATCH 1/3] ASoC: Intel: Add PM support to the HSW/BDW DSP core.
by Liam Girdwood 29 Oct '14

29 Oct '14

Add support for PM wake, sleep and stall calls to the core HSW/BDW driver. This includes reworking the reset and boot code and adding new calls for setting D3/D0 state. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-dsp.c | 158 ++++++++++++++++++++++++++++---------- 1 file changed, 119 insertions(+), 39 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 5058dc8..86aea34 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -247,8 +247,67 @@ static irqreturn_t hsw_irq(int irq, void *context) return ret; } -static void hsw_boot(struct sst_dsp *sst) +static void hsw_set_dsp_D3(struct sst_dsp *sst) +{ + u32 val; + + /* switch off audio PLL, DRAM & IRAM blocks */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + val |= SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | + SST_VDRTCL0_ISRAMPGE_MASK; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* Set D3 state */ + val = readl(sst->addr.pci_cfg + SST_PMCS); + val |= SST_PMCS_PS_MASK; + writel(val, sst->addr.pci_cfg + SST_PMCS); +} + +static void hsw_reset(struct sst_dsp *sst) +{ + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, + SST_CSR_RST | SST_CSR_STALL); + + /* keep in reset for 10ms */ + mdelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +} + +static int hsw_set_dsp_D0(struct sst_dsp *sst) { + int tries = 10; + u32 reg; + + /* Set D0 state */ + reg = readl(sst->addr.pci_cfg + SST_PMCS); + reg &= ~SST_PMCS_PS_MASK; + writel(reg, sst->addr.pci_cfg + SST_PMCS); + + /* check that ADSP shim is enabled */ + while (tries--) { + reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK; + if (reg == 0) + goto finish; + + msleep(1); + } + + return -ENODEV; + +finish: + hsw_reset(sst); + + /* switch on audio PLL, DRAM & IRAM blocks */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg &= ~(SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | + SST_VDRTCL0_ISRAMPGE_MASK); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); @@ -267,30 +326,73 @@ static void hsw_boot(struct sst_dsp *sst) sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, SST_CSR2_SDFD_SSP1); - /* enable DMA engine 0,1 all channels to access host memory */ - sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC, - SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff), - SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff)); + /* set on-demond mode on engine 0,1 for all channels */ + sst_dsp_shim_update_bits(sst, SST_HMDC, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH); + + /* Enable Interrupt from both sides */ + sst_dsp_shim_update_bits(sst, SST_IMRX, (SST_IMRX_BUSY | SST_IMRX_DONE), + 0x0); + sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY | + SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0); + + /* clear IPC registers */ + sst_dsp_shim_write(sst, SST_IPCX, 0x0); + sst_dsp_shim_write(sst, SST_IPCD, 0x0); + sst_dsp_shim_write(sst, 0x80, 0x6); + sst_dsp_shim_write(sst, 0xe0, 0x300a); /* disable all clock gating */ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); + return 0; +} + +static void hsw_boot(struct sst_dsp *sst) +{ + /* set oportunistic mode on engine 0,1 for all channels */ + sst_dsp_shim_update_bits(sst, SST_HMDC, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH, 0); + /* set DSP to RUN */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_STALL, 0x0); } -static void hsw_reset(struct sst_dsp *sst) +static void hsw_stall(struct sst_dsp *sst) { + /* stall DSP */ + sst_dsp_shim_update_bits(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_STALL, + SST_CSR_STALL | SST_CSR_24MHZ_LPCS); +} + +static void hsw_sleep(struct sst_dsp *sst) +{ + dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n"); + /* put DSP into reset and stall */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_RST | SST_CSR_STALL, SST_CSR_RST | SST_CSR_STALL); + sst_dsp_shim_update_bits(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL, + SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS); - /* keep in reset for 10ms */ - mdelay(10); + hsw_set_dsp_D3(sst); + dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n"); +} - /* take DSP out of reset and keep stalled for FW loading */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +static int hsw_wake(struct sst_dsp *sst) +{ + int ret; + + dev_dbg(sst->dev, "HSW_PM dsp runtime resume\n"); + + ret = hsw_set_dsp_D0(sst); + if (ret < 0) + return ret; + + dev_dbg(sst->dev, "HSW_PM dsp runtime resume exit\n"); + + return 0; } struct sst_adsp_memregion { @@ -431,27 +533,6 @@ static struct sst_block_ops sst_hsw_ops = { .disable = hsw_block_disable, }; -static int hsw_enable_shim(struct sst_dsp *sst) -{ - int tries = 10; - u32 reg; - - /* enable shim */ - reg = readl(sst->addr.pci_cfg + SST_SHIM_PM_REG); - writel(reg & ~0x3, sst->addr.pci_cfg + SST_SHIM_PM_REG); - - /* check that ADSP shim is enabled */ - while (tries--) { - reg = sst_dsp_shim_read_unlocked(sst, SST_CSR); - if (reg != 0xffffffff) - return 0; - - msleep(1); - } - - return -ENODEV; -} - static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) { const struct sst_adsp_memregion *region; @@ -490,7 +571,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } /* enable the DSP SHIM */ - ret = hsw_enable_shim(sst); + ret = hsw_set_dsp_D0(sst); if (ret < 0) { dev_err(dev, "error: failed to set DSP D0 and reset SHIM\n"); return ret; @@ -500,10 +581,6 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) if (ret) return ret; - /* Enable Interrupt from both sides */ - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 0x3, 0x0); - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRD, - (0x3 | 0x1 << 16 | 0x3 << 21), 0x0); /* register DSP memory blocks - ideally we should get this from ACPI */ for (i = 0; i < region_count; i++) { @@ -535,6 +612,9 @@ static void hsw_free(struct sst_dsp *sst) struct sst_ops haswell_ops = { .reset = hsw_reset, .boot = hsw_boot, + .stall = hsw_stall, + .wake = hsw_wake, + .sleep = hsw_sleep, .write = sst_shim32_write, .read = sst_shim32_read, .write64 = sst_shim32_write64, -- 2.1.0

2 3

[alsa-devel] [PATCH] ALSA: hda - Add workaround for CMI8888 snoop behavior
by Takashi Iwai 29 Oct '14

29 Oct '14

CMI8888 shows the stuttering playback when the snooping is disabled on the audio buffer. Meanwhile, we've got reports that CORB/RIRB doesn't work in the snooped mode. So, as a compromise, disable the snoop only for CORB/RIRB and enable the snoop for the stream buffers. The resultant patch became a bit ugly, unfortunately, but we still can live with it. Reported-and-tested-by: Geoffrey McRae <geoff(a)spacevs.com> Cc: <stable(a)vger.kernel.org> # 3.17+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index cfcca4c30d4d..9ab1e631cb32 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -374,6 +374,8 @@ static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool #ifdef CONFIG_SND_DMA_SGBUF if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) { struct snd_sg_buf *sgbuf = dmab->private_data; + if (chip->driver_type == AZX_DRIVER_CMEDIA) + return; /* deal with only CORB/RIRB buffers */ if (on) set_pages_array_wc(sgbuf->page_table, sgbuf->pages); else @@ -1769,7 +1771,7 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, #ifdef CONFIG_X86 struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx *chip = apcm->chip; - if (!azx_snoop(chip)) + if (!azx_snoop(chip) && chip->driver_type != AZX_DRIVER_CMEDIA) area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); #endif } -- 2.1.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: dw_pdata can be static
by Liam Girdwood 29 Oct '14

29 Oct '14

From: kbuild test robot <fengguang.wu(a)intel.com> sound/soc/intel/sst-firmware.c:172:29: sparse: symbol 'dw_pdata' was not declared. Should it be static? Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 692a6ae..35788ad 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -169,7 +169,7 @@ err: return ret; } -struct dw_dma_platform_data dw_pdata = { +static struct dw_dma_platform_data dw_pdata = { .is_private = 1, .chan_allocation_order = CHAN_ALLOCATION_ASCENDING, .chan_priority = CHAN_PRIORITY_ASCENDING, -- 2.1.0

2 2

[alsa-devel] [PATCH] ALSA: hdspm: remove unused variable
by Sudip Mukherjee 29 Oct '14

29 Oct '14

removed the unused variables. These variables were only being assigned some value, but the values were never being used. it has been build tested after removing the variables. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/rme9652/hdspm.c | 23 +++-------------------- 1 file changed, 3 insertions(+), 20 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 7f7277b..e09348c1 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1257,14 +1257,13 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate) /* check for external sample rate, returns the sample rate in Hz*/ static int hdspm_external_sample_rate(struct hdspm *hdspm) { - unsigned int status, status2, timecode; + unsigned int status, status2; int syncref, rate = 0, rate_bits; switch (hdspm->io_type) { case AES32: status2 = hdspm_read(hdspm, HDSPM_statusRegister2); status = hdspm_read(hdspm, HDSPM_statusRegister); - timecode = hdspm_read(hdspm, HDSPM_timecodeRegister); syncref = hdspm_autosync_ref(hdspm); switch (syncref) { @@ -4862,18 +4861,15 @@ snd_hdspm_proc_read_madi(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct hdspm *hdspm = entry->private_data; - unsigned int status, status2, control, freq; + unsigned int status, status2; char *pref_sync_ref; char *autosync_ref; char *system_clock_mode; - char *insel; int x, x2; status = hdspm_read(hdspm, HDSPM_statusRegister); status2 = hdspm_read(hdspm, HDSPM_statusRegister2); - control = hdspm->control_register; - freq = hdspm_read(hdspm, HDSPM_timecodeRegister); snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n", hdspm->card_name, hdspm->card->number + 1, @@ -4936,17 +4932,6 @@ snd_hdspm_proc_read_madi(struct snd_info_entry *entry, snd_iprintf(buffer, "Line out: %s\n", (hdspm->control_register & HDSPM_LineOut) ? "on " : "off"); - switch (hdspm->control_register & HDSPM_InputMask) { - case HDSPM_InputOptical: - insel = "Optical"; - break; - case HDSPM_InputCoaxial: - insel = "Coaxial"; - break; - default: - insel = "Unknown"; - } - snd_iprintf(buffer, "ClearTrackMarker = %s, Transmit in %s Channel Mode, " "Auto Input %s\n", @@ -5191,15 +5176,13 @@ snd_hdspm_proc_read_raydat(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct hdspm *hdspm = entry->private_data; - unsigned int status1, status2, status3, control, i; + unsigned int status1, status2, status3, i; unsigned int lock, sync; status1 = hdspm_read(hdspm, HDSPM_RD_STATUS_1); /* s1 */ status2 = hdspm_read(hdspm, HDSPM_RD_STATUS_2); /* freq */ status3 = hdspm_read(hdspm, HDSPM_RD_STATUS_3); /* s2 */ - control = hdspm->control_register; - snd_iprintf(buffer, "STATUS1: 0x%08x\n", status1); snd_iprintf(buffer, "STATUS2: 0x%08x\n", status2); snd_iprintf(buffer, "STATUS3: 0x%08x\n", status3); -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Correct kerneldoc comments
by Takashi Iwai 29 Oct '14

29 Oct '14

Complete the missing parameters and fix anything wrong there. Just comment changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 15 +++++- sound/pci/hda/hda_codec.c | 114 ++++++++++++++++++++++++++++++++++++++-- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_jack.c | 21 ++++++++ 4 files changed, 146 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index fcc5e478c9a1..7388958b01af 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -464,8 +464,12 @@ EXPORT_SYMBOL_GPL(snd_hda_get_input_pin_attr); /** * hda_get_input_pin_label - Give a label for the given input pin + * @codec: the HDA codec + * @item: ping config item to refer + * @pin: the pin NID + * @check_location: flag to add the jack location prefix * - * When check_location is true, the function checks the pin location + * When @check_location is true, the function checks the pin location * for mic and line-in pins, and set an appropriate prefix like "Front", * "Rear", "Internal". */ @@ -550,6 +554,9 @@ static int check_mic_location_need(struct hda_codec *codec, /** * hda_get_autocfg_input_label - Get a label for the given input + * @codec: the HDA codec + * @cfg: the parsed pin configuration + * @input: the input index number * * Get a label for the given input pin defined by the autocfg item. * Unlike hda_get_input_pin_label(), this function checks all inputs @@ -677,6 +684,12 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, /** * snd_hda_get_pin_label - Get a label for the given I/O pin + * @codec: the HDA codec + * @nid: pin NID + * @cfg: the parsed pin configuration + * @label: the string buffer to store + * @maxlen: the max length of string buffer (including termination) + * @indexp: the pointer to return the index number (for multiple ctls) * * Get a label for the given pin. This function works for both input and * output pins. When @cfg is given as non-NULL, the function tries to get diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0025bf4c2f44..e15254204c72 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -416,7 +416,6 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) * snd_hda_get_conn_list - get connection list * @codec: the HDA codec * @nid: NID to parse - * @len: number of connection list entries * @listp: the pointer to store NID list * * Parses the connection list of the given widget and stores the pointer @@ -2004,6 +2003,7 @@ EXPORT_SYMBOL_GPL(query_amp_caps); * @codec: the HD-audio codec * @nid: the NID to query * @dir: either #HDA_INPUT or #HDA_OUTPUT + * @bits: bit mask to check the result * * Check whether the widget has the given amp capability for the direction. */ @@ -2023,7 +2023,7 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_caps); * snd_hda_override_amp_caps - Override the AMP capabilities * @codec: the CODEC to clean up * @nid: the NID to clean up - * @direction: either #HDA_INPUT or #HDA_OUTPUT + * @dir: either #HDA_INPUT or #HDA_OUTPUT * @caps: the capability bits to set * * Override the cached AMP caps bits value by the given one. @@ -2320,6 +2320,8 @@ static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir, /** * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer + * @kcontrol: referred ctl element + * @uinfo: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -2381,6 +2383,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, /** * snd_hda_mixer_amp_volume_get - Get callback for a standard AMP mixer volume + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -2406,6 +2410,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_get); /** * snd_hda_mixer_amp_volume_put - Put callback for a standard AMP mixer volume + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -2436,6 +2442,10 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); /** * snd_hda_mixer_amp_volume_put - TLV callback for a standard AMP mixer volume + * @kcontrol: ctl element + * @op_flag: operation flag + * @size: byte size of input TLV + * @_tlv: TLV data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3012,6 +3022,8 @@ EXPORT_SYMBOL_GPL(snd_hda_sync_vmaster_hook); /** * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch + * @kcontrol: referred ctl element + * @uinfo: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3031,6 +3043,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_info); /** * snd_hda_mixer_amp_switch_get - Get callback for a standard AMP mixer switch + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3057,6 +3071,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_get); /** * snd_hda_mixer_amp_switch_put - Put callback for a standard AMP mixer switch + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3100,6 +3116,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); /** * snd_hda_mixer_bind_switch_get - Get callback for a bound volume control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_MUTE*() macros. @@ -3123,6 +3141,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_get); /** * snd_hda_mixer_bind_switch_put - Put callback for a bound volume control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_MUTE*() macros. @@ -3153,6 +3173,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_put); /** * snd_hda_mixer_bind_ctls_info - Info callback for a generic bound control + * @kcontrol: referred ctl element + * @uinfo: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. @@ -3176,6 +3198,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_info); /** * snd_hda_mixer_bind_ctls_get - Get callback for a generic bound control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. @@ -3199,6 +3223,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_get); /** * snd_hda_mixer_bind_ctls_put - Put callback for a generic bound control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. @@ -3228,6 +3254,10 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_put); /** * snd_hda_mixer_bind_tlv - TLV callback for a generic bound control + * @kcontrol: ctl element + * @op_flag: operation flag + * @size: byte size of input TLV + * @tlv: TLV data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() macro. @@ -4305,6 +4335,7 @@ static struct hda_rate_tbl rate_bits[] = { * @channels: the number of channels * @format: the PCM format (SNDRV_PCM_FORMAT_XXX) * @maxbps: the max. bps + * @spdif_ctls: HD-audio SPDIF status bits (0 if irrelevant) * * Calculate the format bitset from the given rate, channels and th PCM format. * @@ -4980,6 +5011,7 @@ static void __snd_hda_power_down(struct hda_codec *codec) * snd_hda_power_save - Power-up/down/sync the codec * @codec: HD-audio codec * @delta: the counter delta to change + * @d3wait: sync for D3 transition complete * * Change the power-up counter via @delta, and power up or down the hardware * appropriately. For the power-down, queue to the delayed action. @@ -5055,6 +5087,10 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); /** * snd_hda_ch_mode_info - Info callback helper for the channel mode enum + * @codec: the HDA codec + * @uinfo: pointer to get/store the data + * @chmode: channel mode array + * @num_chmodes: channel mode array size */ int snd_hda_ch_mode_info(struct hda_codec *codec, struct snd_ctl_elem_info *uinfo, @@ -5074,6 +5110,11 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_info); /** * snd_hda_ch_mode_get - Get callback helper for the channel mode enum + * @codec: the HDA codec + * @ucontrol: pointer to get/store the data + * @chmode: channel mode array + * @num_chmodes: channel mode array size + * @max_channels: max number of channels */ int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -5095,6 +5136,11 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_get); /** * snd_hda_ch_mode_put - Put callback helper for the channel mode enum + * @codec: the HDA codec + * @ucontrol: pointer to get/store the data + * @chmode: channel mode array + * @num_chmodes: channel mode array size + * @max_channelsp: pointer to store the max channels */ int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -5123,6 +5169,8 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_put); /** * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + * @imux: imux helper object + * @uinfo: pointer to get/store the data */ int snd_hda_input_mux_info(const struct hda_input_mux *imux, struct snd_ctl_elem_info *uinfo) @@ -5144,6 +5192,11 @@ EXPORT_SYMBOL_GPL(snd_hda_input_mux_info); /** * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + * @codec: the HDA codec + * @imux: imux helper object + * @ucontrol: pointer to get/store the data + * @nid: input mux NID + * @cur_val: pointer to get/store the current imux value */ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, @@ -5168,7 +5221,13 @@ int snd_hda_input_mux_put(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_input_mux_put); -/* +/** + * snd_hda_enum_helper_info - Helper for simple enum ctls + * @kcontrol: ctl element + * @uinfo: pointer to get/store the data + * @num_items: number of enum items + * @texts: enum item string array + * * process kcontrol info callback of a simple string enum array * when @num_items is 0 or @texts is NULL, assume a boolean enum array */ @@ -5257,6 +5316,8 @@ EXPORT_SYMBOL_GPL(snd_hda_bus_reboot_notify); /** * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5273,6 +5334,11 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_open); /** * snd_hda_multi_out_dig_prepare - prepare the digital out stream + * @codec: the HDA codec + * @mout: hda_multi_out object + * @stream_tag: stream tag to assign + * @format: format id to assign + * @substream: PCM substream to assign */ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, struct hda_multi_out *mout, @@ -5289,6 +5355,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_prepare); /** * snd_hda_multi_out_dig_cleanup - clean-up the digital out stream + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5302,6 +5370,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_cleanup); /** * snd_hda_multi_out_dig_close - release the digital out stream + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5315,6 +5385,10 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_close); /** * snd_hda_multi_out_analog_open - open analog outputs + * @codec: the HDA codec + * @mout: hda_multi_out object + * @substream: PCM substream to assign + * @hinfo: PCM information to assign * * Open analog outputs and set up the hw-constraints. * If the digital outputs can be opened as slave, open the digital @@ -5365,6 +5439,11 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_open); /** * snd_hda_multi_out_analog_prepare - Preapre the analog outputs. + * @codec: the HDA codec + * @mout: hda_multi_out object + * @stream_tag: stream tag to assign + * @format: format id to assign + * @substream: PCM substream to assign * * Set up the i/o for analog out. * When the digital out is available, copy the front out to digital out, too. @@ -5442,6 +5521,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_prepare); /** * snd_hda_multi_out_analog_cleanup - clean up the setting for analog out + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5473,6 +5554,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_cleanup); /** * snd_hda_get_default_vref - Get the default (mic) VREF pin bits + * @codec: the HDA codec + * @pin: referred pin NID * * Guess the suitable VREF pin bits to be set as the pin-control value. * Note: the function doesn't set the AC_PINCTL_IN_EN bit. @@ -5498,7 +5581,12 @@ unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin) } EXPORT_SYMBOL_GPL(snd_hda_get_default_vref); -/* correct the pin ctl value for matching with the pin cap */ +/** + * snd_hda_correct_pin_ctl - correct the pin ctl value for matching with the pin cap + * @codec: the HDA codec + * @pin: referred pin NID + * @val: pin ctl value to audit + */ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val) { @@ -5549,6 +5637,19 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); +/** + * _snd_hda_pin_ctl - Helper to set pin ctl value + * @codec: the HDA codec + * @pin: referred pin NID + * @val: pin control value to set + * @cached: access over codec pinctl cache or direct write + * + * This function is a helper to set a pin ctl value more safely. + * It corrects the pin ctl value via snd_hda_correct_pin_ctl(), stores the + * value in pin target array via snd_hda_codec_set_pin_target(), then + * actually writes the value via either snd_hda_codec_update_cache() or + * snd_hda_codec_write() depending on @cached flag. + */ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val, bool cached) { @@ -5565,6 +5666,11 @@ EXPORT_SYMBOL_GPL(_snd_hda_set_pin_ctl); /** * snd_hda_add_imux_item - Add an item to input_mux + * @codec: the HDA codec + * @imux: imux helper object + * @label: the name of imux item to assign + * @index: index number of imux item to assign + * @type_idx: pointer to store the resultant label index * * When the same label is used already in the existing items, the number * suffix is appended to the label. This label index number is stored diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index e1cd34d9011d..0e6d7534f491 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -371,7 +371,7 @@ error: return ret; } -/** +/* * SNDRV_PCM_RATE_* and AC_PAR_PCM values don't match, print correct rates with * hdmi-specific routine. */ diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index f56765ae73a7..b2d81ab22fb0 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -57,6 +57,8 @@ static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid) /** * snd_hda_jack_tbl_get - query the jack-table entry for the given NID + * @codec: the HDA codec + * @nid: pin NID to refer to */ struct hda_jack_tbl * snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid) @@ -75,6 +77,8 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get); /** * snd_hda_jack_tbl_get_from_tag - query the jack-table entry for the given tag + * @codec: the HDA codec + * @tag: tag value to refer to */ struct hda_jack_tbl * snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag) @@ -93,6 +97,8 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get_from_tag); /** * snd_hda_jack_tbl_new - create a jack-table entry for the given NID + * @codec: the HDA codec + * @nid: pin NID to assign */ static struct hda_jack_tbl * snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) @@ -162,6 +168,7 @@ static void jack_detect_update(struct hda_codec *codec, /** * snd_hda_set_dirty_all - Mark all the cached as dirty + * @codec: the HDA codec * * This function sets the dirty flag to all entries of jack table. * It's called from the resume path in hda_codec.c. @@ -218,6 +225,9 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state); /** * snd_hda_jack_detect_enable - enable the jack-detection + * @codec: the HDA codec + * @nid: pin NID to enable + * @func: callback function to register * * In the case of error, the return value will be a pointer embedded with * errno. Check and handle the return value appropriately with standard @@ -266,6 +276,9 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable); /** * snd_hda_jack_set_gating_jack - Set gating jack. + * @codec: the HDA codec + * @gated_nid: gated pin NID + * @gating_nid: gating pin NID * * Indicates the gated jack is only valid when the gating jack is plugged. */ @@ -287,6 +300,7 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_set_gating_jack); /** * snd_hda_jack_report_sync - sync the states of all jacks and report if changed + * @codec: the HDA codec */ void snd_hda_jack_report_sync(struct hda_codec *codec) { @@ -349,6 +363,11 @@ static void hda_free_jack_priv(struct snd_jack *jack) /** * snd_hda_jack_add_kctl - Add a kctl for the given pin + * @codec: the HDA codec + * @nid: pin NID to assign + * @name: string name for the jack + * @idx: index number for the jack + * @phantom_jack: flag to deal as a phantom jack * * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. @@ -456,6 +475,8 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, /** * snd_hda_jack_add_kctls - Add kctls for all pins included in the given pincfg + * @codec: the HDA codec + * @cfg: pin config table to parse */ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) -- 2.1.2

1 1

[alsa-devel] [PATCH 00/11] Add AXD Audio Processing IP driver
by Qais Yousef 29 Oct '14

29 Oct '14

AXD Audio Processing IP performs audio decoding, encoding, mixing, equalisation, synchronisation and playback. This series adds the driver and DT binding document. I tried to split it into 'logical' units as there are many files and some of them are big (necessarily I believe). If it is still difficult to review as is I'd be happy to take on any suggestion to make things easier. I added it to drivers/char as this is the best location I could think of at the moment. It doesn't fit in alsa subsystem but I Cced them to confirm. Also I'm not sure if new drivers should go to staging first then get promoted or they can be accepted directly. This is based on Linus' 3.18-rc2 Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: <devicetree(a)vger.kernel.org> Cc: <alsa-devel(a)alsa-project.org> Qais Yousef (11): MAINTANERS: Add AXD as a supported driver dt: bindings: add AXD Audio Processing IP binding document drivers: char: add AXD Audio Processing IP driver drivers: char: axd: add fw binary header manipulation files drivers: char: axd: add buffers manipulation files drivers: char: axd: add basic files for sending/receiving axd cmds drivers: char: axd: add cmd interfce helper functions drivers: char: axd: add low level AXD platform setup files drivers: char: axd: add sysfs setup files drivers: char: axd: add ts interface file drivers: char: axd: add Kconfig and Makefile Documentation/devicetree/bindings/axd.txt | 39 + MAINTAINERS | 6 + drivers/char/Kconfig | 2 + drivers/char/Makefile | 2 + drivers/char/axd/Kconfig | 19 + drivers/char/axd/Makefile | 20 + drivers/char/axd/axd_api.h | 641 ++++++ drivers/char/axd/axd_buffers.c | 245 +++ drivers/char/axd/axd_buffers.h | 70 + drivers/char/axd/axd_cmds.c | 98 + drivers/char/axd/axd_cmds.h | 531 +++++ drivers/char/axd/axd_cmds_codec_internal.c | 58 + drivers/char/axd/axd_cmds_codec_internal.h | 28 + drivers/char/axd/axd_cmds_config.c | 1194 ++++++++++ drivers/char/axd/axd_cmds_decoder_config.c | 1806 ++++++++++++++++ drivers/char/axd/axd_cmds_encoder_config.c | 519 +++++ drivers/char/axd/axd_cmds_info.c | 1409 ++++++++++++ drivers/char/axd/axd_cmds_internal.c | 3237 ++++++++++++++++++++++++++++ drivers/char/axd/axd_cmds_internal.h | 306 +++ drivers/char/axd/axd_cmds_pipes.c | 1368 ++++++++++++ drivers/char/axd/axd_hdr.c | 69 + drivers/char/axd/axd_hdr.h | 20 + drivers/char/axd/axd_module.c | 1064 +++++++++ drivers/char/axd/axd_module.h | 99 + drivers/char/axd/axd_platform.h | 31 + drivers/char/axd/axd_platform_mips.c | 270 +++ drivers/char/axd/axd_sysfs.h | 37 + drivers/char/axd/axd_sysfs_ctrl.c | 539 +++++ drivers/char/axd/axd_sysfs_input.c | 406 ++++ drivers/char/axd/axd_sysfs_output.c | 1373 ++++++++++++ drivers/char/axd/axd_ts_driver.h | 24 + include/linux/axd.h | 32 + 32 files changed, 15562 insertions(+) create mode 100644 Documentation/devicetree/bindings/axd.txt create mode 100644 drivers/char/axd/Kconfig create mode 100644 drivers/char/axd/Makefile create mode 100644 drivers/char/axd/axd_api.h create mode 100644 drivers/char/axd/axd_buffers.c create mode 100644 drivers/char/axd/axd_buffers.h create mode 100644 drivers/char/axd/axd_cmds.c create mode 100644 drivers/char/axd/axd_cmds.h create mode 100644 drivers/char/axd/axd_cmds_codec_internal.c create mode 100644 drivers/char/axd/axd_cmds_codec_internal.h create mode 100644 drivers/char/axd/axd_cmds_config.c create mode 100644 drivers/char/axd/axd_cmds_decoder_config.c create mode 100644 drivers/char/axd/axd_cmds_encoder_config.c create mode 100644 drivers/char/axd/axd_cmds_info.c create mode 100644 drivers/char/axd/axd_cmds_internal.c create mode 100644 drivers/char/axd/axd_cmds_internal.h create mode 100644 drivers/char/axd/axd_cmds_pipes.c create mode 100644 drivers/char/axd/axd_hdr.c create mode 100644 drivers/char/axd/axd_hdr.h create mode 100644 drivers/char/axd/axd_module.c create mode 100644 drivers/char/axd/axd_module.h create mode 100644 drivers/char/axd/axd_platform.h create mode 100644 drivers/char/axd/axd_platform_mips.c create mode 100644 drivers/char/axd/axd_sysfs.h create mode 100644 drivers/char/axd/axd_sysfs_ctrl.c create mode 100644 drivers/char/axd/axd_sysfs_input.c create mode 100644 drivers/char/axd/axd_sysfs_output.c create mode 100644 drivers/char/axd/axd_ts_driver.h create mode 100644 include/linux/axd.h -- 2.1.0

6 37

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix probing and stuttering on CMI8888 HD-audio controller
by Takashi Iwai 29 Oct '14

29 Oct '14

ASUS Phoebus with CMI8888 HD-audio chip (PCI id 13f6:5011) doesn't work with HD-audio driver as is because of some weird nature. For making DMA properly working, we need to disable MSI. The position report buffer doesn't work, thus we need to force reading LPIB instead. And yet, the codec CORB/RIRB communication gives errors unless we disable the snooping (caching). In this patch, all these workarounds are added as a quirk for the device. The HD-audio *codec* chip needs yet another workaround, but it'll be provided in the succeeding patch. Reported-and-tested-by: Vincent Lejeune <vljn(a)ovi.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5db1948699d8..aa302fb03fc5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -265,6 +265,7 @@ enum { AZX_DRIVER_TERA, AZX_DRIVER_CTX, AZX_DRIVER_CTHDA, + AZX_DRIVER_CMEDIA, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -330,6 +331,7 @@ static char *driver_short_names[] = { [AZX_DRIVER_TERA] = "HDA Teradici", [AZX_DRIVER_CTX] = "HDA Creative", [AZX_DRIVER_CTHDA] = "HDA Creative", + [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -1373,6 +1375,7 @@ static void azx_check_snoop_available(struct azx *chip) snoop = false; break; case AZX_DRIVER_CTHDA: + case AZX_DRIVER_CMEDIA: snoop = false; break; } @@ -2154,6 +2157,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB }, #endif + /* CM8888 */ + { PCI_DEVICE(0x13f6, 0x5011), + .driver_data = AZX_DRIVER_CMEDIA | + AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB }, /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, /* VMware HDAudio */ -- 2.0.3

2 6

[alsa-devel] [PATCH v03 0/4] ASoC: davinci-mcasp: Correct start/stop sequences
by Peter Ujfalusi 29 Oct '14

29 Oct '14

Hi, For some reason these patches did not made it to mainline kernel. V2 was sent on 28.04.2014: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074871.html and got applied the same day: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074878.html I just noticed that they are missing. Changes since v2: - rebased on next Changes since v1: - Correction for the stop sequence is added - start code cleanup patch added The startup sequence for TX and RX was not correct, not following the programming sequence from the TRM. This could cause initial channel swap when starting TX. When stopping the stream the AFIFO should be stopped as the last step to avoid unpredictable behavior. The change has been tested on AM335x and AM437x but it should be valid for all devices. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Correct TX start sequence ASoC: davinci-mcasp: Correct RX start sequence ASoC: davinci-mcasp: When stopping TX/RX stop the AFIFO as the last step ASoC: davinci-mcasp: Move the AFIFO related code under start_tx/rx functions sound/soc/davinci/davinci-mcasp.c | 95 ++++++++++++++++++--------------------- sound/soc/davinci/davinci-mcasp.h | 6 +++ 2 files changed, 49 insertions(+), 52 deletions(-) -- 2.1.2

2 5

[alsa-devel] Kerneldoc warnings in ASoC source codes
by Takashi Iwai 29 Oct '14

29 Oct '14

Hi, I tried to process kerneldoc to sound/soc/soc-*.c and include/sound/soc.h, and hit quite lots of warnings. Many of them are trivial, typos, but some are unclear. I Cc'ed relevant people: could you guys fix these properly for 3.19 kernel? FYI, you can get the errors, e.g. scripts/kernel-doc sound/soc/soc-core.c > /dev/null thanks, Takashi Warning(.//include/sound/soc.h:595): No description found for parameter 'list' Warning(.//include/sound/soc.h:613): No description found for parameter 'list' Warning(.//include/sound/soc.h:648): No description found for parameter 'gpiod_dev' Warning(.//include/sound/soc.h:648): No description found for parameter 'debounce_time' Warning(.//include/sound/soc.h:648): No description found for parameter 'jack' Warning(.//include/sound/soc.h:648): No description found for parameter 'work' Warning(.//include/sound/soc.h:648): No description found for parameter 'desc' Warning(.//include/sound/soc.h:648): No description found for parameter 'data' Warning(.//include/sound/soc.h:648): Excess struct/union/enum/typedef member 'debouce_time' description in 'snd_soc_jack_gpio' Warning(.//sound/soc/soc-core.c:2122): No description found for parameter 'ops' Warning(.//sound/soc/soc-core.c:2122): No description found for parameter 'pdev' Warning(.//sound/soc/soc-core.c:2696): No description found for parameter 'ucontrol' Warning(.//sound/soc/soc-core.c:2696): Excess function parameter 'uinfo' description in 'snd_soc_put_volsw_sx' Warning(.//sound/soc/soc-core.c:3466): No description found for parameter 'ratio' Warning(.//sound/soc/soc-core.c:4088): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4126): No description found for parameter 'platform_drv' Warning(.//sound/soc/soc-core.c:4126): Excess function parameter 'platform_driver' description in 'snd_soc_add_platform' Warning(.//sound/soc/soc-core.c:4165): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4165): No description found for parameter 'platform_drv' Warning(.//sound/soc/soc-core.c:4165): Excess function parameter 'platform' description in 'snd_soc_register_platform' Warning(.//sound/soc/soc-core.c:4221): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4221): Excess function parameter 'platform' description in 'snd_soc_unregister_platform' Warning(.//sound/soc/soc-core.c:4315): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4315): No description found for parameter 'codec_drv' Warning(.//sound/soc/soc-core.c:4315): No description found for parameter 'dai_drv' Warning(.//sound/soc/soc-core.c:4315): No description found for parameter 'num_dai' Warning(.//sound/soc/soc-core.c:4315): Excess function parameter 'codec' description in 'snd_soc_register_codec' Warning(.//sound/soc/soc-core.c:4412): No description found for parameter 'dev' Warning(.//sound/soc/soc-core.c:4412): Excess function parameter 'codec' description in 'snd_soc_unregister_codec' Warning(.//sound/soc/soc-devres.c:70): No description found for parameter 'platform_drv' Warning(.//sound/soc/soc-devres.c:70): Excess function parameter 'platform' description in 'devm_snd_soc_register_platform' Warning(.//sound/soc/soc-dapm.c:2690): No description found for parameter 'card' Warning(.//sound/soc/soc-dapm.c:2690): Excess function parameter 'dapm' description in 'snd_soc_dapm_new_widgets' Warning(.//sound/soc/soc-jack.c:126): No description found for parameter 'zones' Warning(.//sound/soc/soc-jack.c:126): Excess function parameter 'zone' description in 'snd_soc_jack_add_zones'

1 0

[alsa-devel] [PATCH] ASoC: rsnd: remove unsupported PAUSE flag
by Kuninori Morimoto 29 Oct '14

29 Oct '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car sound doesn't support PAUSE. Remove SNDRV_PCM_INFO_PAUSE flags from snd_pcm_hardware info Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1922ec5..7004219 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -886,8 +886,7 @@ static int rsnd_dai_probe(struct platform_device *pdev, static struct snd_pcm_hardware rsnd_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE, + SNDRV_PCM_INFO_MMAP_VALID, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 32, .period_bytes_max = 8192, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: fsi: remove unsupported PAUSE flag
by Kuninori Morimoto 29 Oct '14

29 Oct '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> FSI doesn't support PAUSE. Remove SNDRV_PCM_INFO_PAUSE flags from snd_pcm_hardware info Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 66fddec..88e5df4 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1711,8 +1711,7 @@ static const struct snd_soc_dai_ops fsi_dai_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, + SNDRV_PCM_INFO_MMAP_VALID, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 32, .period_bytes_max = 8192, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rt5645:Mark RT5645_TDM_CTRL_3 as readable
by yang.a.fang＠intel.com 29 Oct '14

29 Oct '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> amixer query fails due to it is not readable reigster Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3fb83bf..d16331e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -139,6 +139,7 @@ static const struct reg_default rt5645_reg[] = { { 0x76, 0x000a }, { 0x77, 0x0c00 }, { 0x78, 0x0000 }, + { 0x79, 0x0123 }, { 0x80, 0x0000 }, { 0x81, 0x0000 }, { 0x82, 0x0000 }, @@ -334,6 +335,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_DMIC_CTRL2: case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: + case RT5645_TDM_CTRL_3: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: Intel: Add dependency on DesignWare DMA controller
by Mark Brown 29 Oct '14

29 Oct '14

We have calls into the controller so we need to ensure it is being built. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af880868b..ae7f87221a3c 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -20,6 +20,7 @@ config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI depends on (X86 || COMPILE_TEST) + depends on DW_DMAC_CORE help This adds support for Intel(R) Smart Sound Technology (SST). Say Y if you have such a device -- 2.1.1

1 0

[alsa-devel] - snd-usb-audio - SPDIF clock source mixer control for Digidesign Mbox
by Damien Zammit 29 Oct '14

29 Oct '14

Hi all, As some of you may remember, the Digidesign Mbox has a multiple endpoint interface and does not follow the standard USB audio spec. I am pretty sure that the first interface 0 is for firmware upgrades only, and the second interface has 2 endpoints one for playback and one for capture. I have heard that the SPDIF output is always on, but there is a special command to change the clock source to external sync with SPDIF. I think I have found the correct USB command for changing the clock source of the device to SPDIF external sync. I tried to use a pair of the devices in JACK using a pcm_multi configuration and syncing one of them to the other with a RCA cable, but unfortunately I get lots of XRUNs in JACK and I am wondering if there is a reason for this? Can someone who knows about the pcm_multi stuff shed some light on why I might be getting thousands of XRUNs when I use two identical USB soundcards that are (hopefully) synced to each other? I have written a non-standard mixer control for this so alsamixer can select the external clock source. Basically, the command to set SPDIF sync seems to include a sample rate command, but the sample rate is set to zero for SPDIF sync. Could JACK be resetting the sample rate and overriding my command perhaps? How would I override the ALSA driver code so that the sample rate is not reset by JACK to 48000 instead of 0 but only when the mixer control sets the flag? Damien

1 0

[alsa-devel] [PATCH] ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_rxctrl
by Jianqun 29 Oct '14

29 Oct '14

We can get into an infinite loop if the I2S_CLR register fails to clear due to a missing break statement, so add that. Signed-off-by: Jianqun <jay.xu(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index f373e37..c74ba37 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -154,8 +154,10 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) while (val) { regmap_read(i2s->regmap, I2S_CLR, &val); retry--; - if (!retry) + if (!retry) { dev_warn(i2s->dev, "fail to clear\n"); + break; + } } } } -- 1.9.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: sgtl5000: Use the preferred form for passing a size of a struct
by Fabio Estevam 29 Oct '14

29 Oct '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> According to Documentation/CodingStyle - Chapter 14: "The preferred form for passing a size of a struct is the following: p = kmalloc(sizeof(*p), ...); The alternative form where struct name is spelled out hurts readability and introduces an opportunity for a bug when the pointer variable type is changed but the corresponding sizeof that is passed to a memory allocator is not." So do it as recommeded. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6bb77d7..3f8c05b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1419,8 +1419,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, int ret, reg, rev; unsigned int mclk; - sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), - GFP_KERNEL); + sgtl5000 = devm_kzalloc(&client->dev, sizeof(*sgtl5000), GFP_KERNEL); if (!sgtl5000) return -ENOMEM; -- 1.9.1

3 4

[alsa-devel] [PATCH - hda 1/1] Add ultra dock support for Thinkpad X240.
by Lukas Bossard 29 Oct '14

29 Oct '14

Adding ultra doch support for Lenovo Thinkpad X240 (17aa:2214). In the same process also rename ALC292_FIXUP_TPT440_DOCK to a more generic ALC292_FIXUP_LENOVO_ULTRA_DOCK Signed-off-by: Lukas Bossard <mr.bobukas(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34b7bdb..693f083 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4212,7 +4212,7 @@ enum { ALC255_FIXUP_HEADSET_MODE, ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_LENOVO_ULTRA_DOCK, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -4640,7 +4640,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, - [ALC292_FIXUP_TPT440_DOCK] = { + [ALC292_FIXUP_LENOVO_ULTRA_DOCK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x16, 0x21211010 }, /* dock headphone */ @@ -4799,16 +4799,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_LENOVO_ULTRA_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_LENOVO_ULTRA_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_LENOVO_ULTRA_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_LENOVO_ULTRA_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_LENOVO_ULTRA_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), - SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_LENOVO_ULTRA_DOCK), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), @@ -4887,7 +4888,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, - {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, + {.id = ALC292_FIXUP_LENOVO_ULTRA_DOCK, .name = "lenovo-ultra-dock"}, {} }; -- 2.1.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: doc: Add missing headers and compress stuff to alsa-driver-api.tmpl
by Takashi Iwai 29 Oct '14

29 Oct '14

Some header files have kereldoc comments but are not referred properly. Let's add them. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/DocBook/alsa-driver-api.tmpl | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index 0230a96f0564..13f8b241f042 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -57,6 +57,7 @@ !Esound/core/pcm.c !Esound/core/pcm_lib.c !Esound/core/pcm_native.c +!include/sound/pcm.h </sect1> <sect1><title>PCM Format Helpers</title> !Esound/core/pcm_misc.c @@ -64,6 +65,10 @@ <sect1><title>PCM Memory Management</title> !Esound/core/pcm_memory.c </sect1> + <sect1><title>PCM DMA Engine API</title> +!Esound/core/pcm_dmaengine.c +!Iinclude/sound/dmaengine_pcm.h + </sect1> </chapter> <chapter><title>Control/Mixer API</title> <sect1><title>General Control Interface</title> @@ -91,12 +96,19 @@ !Esound/core/info.c </sect1> </chapter> + <chapter><title>Compress Offload</title> + <sect1><title>Compress Offload API</title> +!Esound/core/compress_offload.c +!Iinclude/sound/compress_driver.h + </sect1> + </chapter> <chapter><title>Miscellaneous Functions</title> <sect1><title>Hardware-Dependent Devices API</title> !Esound/core/hwdep.c </sect1> <sect1><title>Jack Abstraction Layer API</title> !Esound/core/jack.c +!Iinclude/sound/jack.h </sect1> <sect1><title>ISA DMA Helpers</title> !Esound/core/isadma.c -- 2.1.2

2 3

[alsa-devel] [PATCH 0/3] documentation fixes
by Takashi Iwai 29 Oct '14

29 Oct '14

Hi, here is a few more minor fixes for kerneldoc comments. Per request from Vinod, uapid/sound/compress_*.h are included, too. [PATCH 1/3] ALSA: doc: Fix enum snd_jack_types comments [PATCH 2/3] ALSA: doc: Fix uapi/sound/compress_offload.h kerneldoc [PATCH 3/3] ALSA: doc: Include uapi/sound/compress_*.h, too Takashi

1 3

[alsa-devel] [PATCH] 6fire: Convert byte_rev_table uses to bitrev8
by Joe Perches 29 Oct '14

29 Oct '14

Use the inline function instead of directly indexing the array. This allows some architectures with hardware instructions for bit reversals to eliminate the array. Signed-off-by: Joe Perches <joe(a)perches.com> --- On Sun, 2014-10-26 at 23:46 -0700, Joe Perches wrote: > On Mon, 2014-10-27 at 14:37 +0800, Wang, Yalin wrote: > > this change add CONFIG_HAVE_ARCH_BITREVERSE config option, > > so that we can use arm/arm64 rbit instruction to do bitrev operation > > by hardware. [] > > diff --git a/include/linux/bitrev.h b/include/linux/bitrev.h > > index 7ffe03f..ef5b2bb 100644 > > --- a/include/linux/bitrev.h > > +++ b/include/linux/bitrev.h > > @@ -3,6 +3,14 @@ > > > > #include <linux/types.h> > > > > +#ifdef CONFIG_HAVE_ARCH_BITREVERSE > > +#include <asm/bitrev.h> > > + > > +#define bitrev32 __arch_bitrev32 > > +#define bitrev16 __arch_bitrev16 > > +#define bitrev8 __arch_bitrev8 > > + > > +#else > > extern u8 const byte_rev_table[256]; sound/usb/6fire/firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 3b02e54..62c25e7 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -316,7 +316,7 @@ static int usb6fire_fw_fpga_upload( while (c != end) { for (i = 0; c != end && i < FPGA_BUFSIZE; i++, c++) - buffer[i] = byte_rev_table[(u8) *c]; + buffer[i] = bitrev8((u8)*c); ret = usb6fire_fw_fpga_write(device, buffer, i); if (ret < 0) {

2 3

[alsa-devel] [PATCH v2] ASoC: rt5677: Add TDM channel mapping function
by Oder Chiou 28 Oct '14

28 Oct '14

It is for channel to slot mapping, and it is not only for 8 channels mapping, but also in 2, 4 and 6 channels mapping. If we want to use the 2 channels in the stereo2 adc path, we need to set the item "2/1/3/4" or "2/3/1/4". It also adds for stereo channel swap. It can map the sterero channels "L/R" to "R/L", "L/L" or R/R. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 239 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5677.h | 38 +++++++- 2 files changed, 262 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index d17d079..5a75552 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1834,6 +1834,93 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if4_adc_mux = SOC_DAPM_ENUM("IF4 ADC Source", rt5677_if4_adc_enum); +/* TDM IF1/2 ADC Data Selection */ /* MX-3B MX-40 [7:6][5:4][3:2][1:0] */ +static const char * const rt5677_if12_adc_swap_src[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc1_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC1_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc1_swap_mux = + SOC_DAPM_ENUM("IF1 ADC1 Swap Source", rt5677_if1_adc1_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc2_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc2_swap_mux = + SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if1_adc2_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc3_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc3_swap_mux = + SOC_DAPM_ENUM("IF1 ADC3 Swap Source", rt5677_if1_adc3_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc4_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc4_swap_mux = + SOC_DAPM_ENUM("IF1 ADC4 Swap Source", rt5677_if1_adc4_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc1_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc1_swap_mux = + SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if2_adc1_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc2_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc2_swap_mux = + SOC_DAPM_ENUM("IF2 ADC2 Swap Source", rt5677_if2_adc2_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc3_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc3_swap_mux = + SOC_DAPM_ENUM("IF2 ADC3 Swap Source", rt5677_if2_adc3_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc4_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc4_swap_mux = + SOC_DAPM_ENUM("IF2 ADC4 Swap Source", rt5677_if2_adc4_swap_enum); + +/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */ +static const char * const rt5677_if1_adc_tdm_swap_src[] = { + "1/2/3/4", "2/1/3/4", "2/3/1/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", + "3/1/2/4", "3/4/1/2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc_tdm_swap_enum, RT5677_TDM1_CTRL2, + RT5677_IF1_ADC_CTRL_SFT, rt5677_if1_adc_tdm_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc_tdm_swap_mux = + SOC_DAPM_ENUM("IF1 ADC TDM Swap Source", rt5677_if1_adc_tdm_swap_enum); + +/* TDM IF2 ADC Data Selection */ /* MX-41[2:0] */ +static const char * const rt5677_if2_adc_tdm_swap_src[] = { + "1/2/3/4", "2/1/3/4", "3/1/2/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", + "2/3/1/4", "3/4/1/2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc_tdm_swap_enum, RT5677_TDM2_CTRL2, + RT5677_IF2_ADC_CTRL_SFT, rt5677_if2_adc_tdm_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc_tdm_swap_mux = + SOC_DAPM_ENUM("IF2 ADC TDM Swap Source", rt5677_if2_adc_tdm_swap_enum); + static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1945,6 +2032,52 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5677_if1_adc_tdm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_read(rt5677->regmap, RT5677_TDM1_CTRL2, &value); + if (value & RT5677_IF1_ADC_CTRL_MASK) + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC_MODE_MASK, + RT5677_IF1_ADC_MODE_TDM); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_read(rt5677->regmap, RT5677_TDM2_CTRL2, &value); + if (value & RT5677_IF2_ADC_CTRL_MASK) + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC_MODE_MASK, + RT5677_IF2_ADC_MODE_TDM); + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), @@ -2104,10 +2237,8 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_PGA("Stereo4 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), /* DSP */ SND_SOC_DAPM_MUX("IB9 Mux", SND_SOC_NOPM, 0, 0, @@ -2230,6 +2361,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { &rt5677_if1_adc3_mux), SND_SOC_DAPM_MUX("IF1 ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if1_adc4_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc1_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc2_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc3_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc4_swap_mux), + SND_SOC_DAPM_MUX_E("IF1 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc_tdm_swap_mux, rt5677_if1_adc_tdm_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("IF2 ADC1 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if2_adc1_mux), SND_SOC_DAPM_MUX("IF2 ADC2 Mux", SND_SOC_NOPM, 0, 0, @@ -2238,6 +2380,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { &rt5677_if2_adc3_mux), SND_SOC_DAPM_MUX("IF2 ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if2_adc4_mux), + SND_SOC_DAPM_MUX("IF2 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc1_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc2_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc3_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc4_swap_mux), + SND_SOC_DAPM_MUX_E("IF2 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc_tdm_swap_mux, rt5677_if2_adc_tdm_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("IF3 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5677_if3_adc_mux), SND_SOC_DAPM_MUX("IF4 ADC Mux", SND_SOC_NOPM, 0, 0, @@ -2621,11 +2774,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF1 ADC4 Mux", "OB67", "OB67" }, { "IF1 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF1 ADC1 Swap Mux", "L/R", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "R/L", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "L/L", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "R/R", "IF1 ADC1 Mux" }, + + { "IF1 ADC2 Swap Mux", "L/R", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "R/L", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "L/L", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "R/R", "IF1 ADC2 Mux" }, + + { "IF1 ADC3 Swap Mux", "L/R", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "R/L", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "L/L", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "R/R", "IF1 ADC3 Mux" }, + + { "IF1 ADC4 Swap Mux", "L/R", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "R/L", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "L/L", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "R/R", "IF1 ADC4 Mux" }, + + { "IF1 ADC", NULL, "IF1 ADC1 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC2 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC3 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC4 Swap Mux" }, + + { "IF1 ADC TDM Swap Mux", "1/2/3/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "2/1/3/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "2/3/1/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "4/1/2/3", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "1/3/2/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "1/4/2/3", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "3/1/2/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "3/4/1/2", "IF1 ADC" }, + { "AIF1TX", NULL, "I2S1" }, - { "AIF1TX", NULL, "IF1 ADC1 Mux" }, - { "AIF1TX", NULL, "IF1 ADC2 Mux" }, - { "AIF1TX", NULL, "IF1 ADC3 Mux" }, - { "AIF1TX", NULL, "IF1 ADC4 Mux" }, + { "AIF1TX", NULL, "IF1 ADC TDM Swap Mux" }, { "IF2 ADC1 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, { "IF2 ADC1 Mux", "OB01", "OB01 Bypass Mux" }, @@ -2642,11 +2826,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF2 ADC4 Mux", "OB67", "OB67" }, { "IF2 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF2 ADC1 Swap Mux", "L/R", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "R/L", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "L/L", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "R/R", "IF2 ADC1 Mux" }, + + { "IF2 ADC2 Swap Mux", "L/R", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "R/L", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "L/L", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "R/R", "IF2 ADC2 Mux" }, + + { "IF2 ADC3 Swap Mux", "L/R", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "R/L", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "L/L", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "R/R", "IF2 ADC3 Mux" }, + + { "IF2 ADC4 Swap Mux", "L/R", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "R/L", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "L/L", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "R/R", "IF2 ADC4 Mux" }, + + { "IF2 ADC", NULL, "IF2 ADC1 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC2 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC3 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC4 Swap Mux" }, + + { "IF2 ADC TDM Swap Mux", "1/2/3/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "2/1/3/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "3/1/2/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "4/1/2/3", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "1/3/2/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "1/4/2/3", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "2/3/1/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "3/4/1/2", "IF2 ADC" }, + { "AIF2TX", NULL, "I2S2" }, - { "AIF2TX", NULL, "IF2 ADC1 Mux" }, - { "AIF2TX", NULL, "IF2 ADC2 Mux" }, - { "AIF2TX", NULL, "IF2 ADC3 Mux" }, - { "AIF2TX", NULL, "IF2 ADC4 Mux" }, + { "AIF2TX", NULL, "IF2 ADC TDM Swap Mux" }, { "IF3 ADC Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, { "IF3 ADC Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 20efa4a..a0ae32f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -799,7 +799,21 @@ #define RT5677_PDM2_I2C_EXE (0x1 << 1) #define RT5677_PDM2_I2C_BUSY (0x1 << 0) -/* MX3C TDM1 control 1 (0x3c) */ +/* MX3B TDM1 control 1 (0x3b) */ +#define RT5677_IF1_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF1_ADC_MODE_SFT 12 +#define RT5677_IF1_ADC_MODE_I2S (0x0 << 12) +#define RT5677_IF1_ADC_MODE_TDM (0x1 << 12) +#define RT5677_IF1_ADC1_SWAP_MASK (0x3 << 6) +#define RT5677_IF1_ADC1_SWAP_SFT 6 +#define RT5677_IF1_ADC2_SWAP_MASK (0x3 << 4) +#define RT5677_IF1_ADC2_SWAP_SFT 4 +#define RT5677_IF1_ADC3_SWAP_MASK (0x3 << 2) +#define RT5677_IF1_ADC3_SWAP_SFT 2 +#define RT5677_IF1_ADC4_SWAP_MASK (0x3 << 0) +#define RT5677_IF1_ADC4_SWAP_SFT 0 + +/* MX3C TDM1 control 2 (0x3c) */ #define RT5677_IF1_ADC4_MASK (0x3 << 10) #define RT5677_IF1_ADC4_SFT 10 #define RT5677_IF1_ADC3_MASK (0x3 << 8) @@ -808,8 +822,24 @@ #define RT5677_IF1_ADC2_SFT 6 #define RT5677_IF1_ADC1_MASK (0x3 << 4) #define RT5677_IF1_ADC1_SFT 4 - -/* MX41 TDM2 control 1 (0x41) */ +#define RT5677_IF1_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF1_ADC_CTRL_SFT 0 + +/* MX40 TDM2 control 1 (0x40) */ +#define RT5677_IF2_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF2_ADC_MODE_SFT 12 +#define RT5677_IF2_ADC_MODE_I2S (0x0 << 12) +#define RT5677_IF2_ADC_MODE_TDM (0x1 << 12) +#define RT5677_IF2_ADC1_SWAP_MASK (0x3 << 6) +#define RT5677_IF2_ADC1_SWAP_SFT 6 +#define RT5677_IF2_ADC2_SWAP_MASK (0x3 << 4) +#define RT5677_IF2_ADC2_SWAP_SFT 4 +#define RT5677_IF2_ADC3_SWAP_MASK (0x3 << 2) +#define RT5677_IF2_ADC3_SWAP_SFT 2 +#define RT5677_IF2_ADC4_SWAP_MASK (0x3 << 0) +#define RT5677_IF2_ADC4_SWAP_SFT 0 + +/* MX41 TDM2 control 2 (0x41) */ #define RT5677_IF2_ADC4_MASK (0x3 << 10) #define RT5677_IF2_ADC4_SFT 10 #define RT5677_IF2_ADC3_MASK (0x3 << 8) @@ -818,6 +848,8 @@ #define RT5677_IF2_ADC2_SFT 6 #define RT5677_IF2_ADC1_MASK (0x3 << 4) #define RT5677_IF2_ADC1_SFT 4 +#define RT5677_IF2_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF2_ADC_CTRL_SFT 0 /* Digital Microphone Control 1 (0x50) */ #define RT5677_DMIC_1_EN_MASK (0x1 << 15) -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: Update HSW/BDW SST driver to new FW architecture
by Liam Girdwood 28 Oct '14

28 Oct '14

The following patch series updates the current upstream driver to use the new more feature rich ADSP FW architecture. The new FW arch allows for multiple instantiation of objects in dynamic memory locations. In summary :- o Modules can be created multiple times. o Modules persistent data can be allocated in any free blocks o Global scratch data can be allocated into any free blocks. o DMA can be used to load the module data. The driver now has to track and manage the new multiple module resource allocations and pass this data to FW on stream initialisations. Changes since V1 :- o Rebased on top of Vinods latest changes to remove a merge conflict. o Added mutex unlock in block error path. The series is sent as 7 patches for ease of review but should be applied as a single patch to preserve git bisect build. The following changes since commit 9a80e8f597f3bde0e1d4a4abb021d475520005a5: ASoC: Intel: mrfld: Define sst_res_info for acpi (2014-10-27 18:02:38 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc.git intel/new-fw-arch for you to fetch changes up to 86ce414b582c79fa2e5a5f19e046eaee5bfa84e1: ASoC: Intel: Update HSW/BDW SST driver to new FW architecture (2014-10-28 17:05:46 +0000) ---------------------------------------------------------------- Liam Girdwood (1): ASoC: Intel: Update HSW/BDW SST driver to new FW architecture sound/soc/intel/sst-baytrail-dsp.c | 24 +- sound/soc/intel/sst-dsp-priv.h | 133 ++++++--- sound/soc/intel/sst-dsp.c | 8 + sound/soc/intel/sst-dsp.h | 7 + sound/soc/intel/sst-firmware.c | 929 +++++++++++++++++++++++++++++++++++++++++++++++++------------ sound/soc/intel/sst-haswell-dsp.c | 59 ++-- sound/soc/intel/sst-haswell-ipc.c | 112 ++++---- sound/soc/intel/sst-haswell-ipc.h | 11 +- sound/soc/intel/sst-haswell-pcm.c | 83 ++++-- 9 files changed, 1025 insertions(+), 341 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: core: Split ops out of soc-core.c
by Mark Brown 28 Oct '14

28 Oct '14

The main ASoC source file is getting quite large and the standard ops don't really have anything to do with the rest of the file so split them out into a separate file. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/Makefile | 2 +- sound/soc/soc-core.c | 1006 ------------------------------------------------ sound/soc/soc-ops.c | 1039 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1040 insertions(+), 1007 deletions(-) create mode 100644 sound/soc/soc-ops.c diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d88edfced8c4..2a043804a2bc 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96ecdc30eb60..a2b51edf6d83 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2334,1012 +2334,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls); /** - * snd_soc_info_enum_double - enumerated double mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a double enumerated - * mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - - return snd_ctl_enum_info(uinfo, e->shift_l == e->shift_r ? 1 : 2, - e->items, e->texts); -} -EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); - -/** - * snd_soc_get_enum_double - enumerated double mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a double enumerated mixer. - * - * Returns 0 for success. - */ -int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, item; - unsigned int reg_val; - int ret; - - ret = snd_soc_component_read(component, e->reg, &reg_val); - if (ret) - return ret; - val = (reg_val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); - ucontrol->value.enumerated.item[0] = item; - if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); - ucontrol->value.enumerated.item[1] = item; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); - -/** - * snd_soc_put_enum_double - enumerated double mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a double enumerated mixer. - * - * Returns 0 for success. - */ -int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int *item = ucontrol->value.enumerated.item; - unsigned int val; - unsigned int mask; - - if (item[0] >= e->items) - return -EINVAL; - val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; - if (e->shift_l != e->shift_r) { - if (item[1] >= e->items) - return -EINVAL; - val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; - } - - return snd_soc_component_update_bits(component, e->reg, mask, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); - -/** - * snd_soc_read_signed - Read a codec register and interprete as signed value - * @component: component - * @reg: Register to read - * @mask: Mask to use after shifting the register value - * @shift: Right shift of register value - * @sign_bit: Bit that describes if a number is negative or not. - * @signed_val: Pointer to where the read value should be stored - * - * This functions reads a codec register. The register value is shifted right - * by 'shift' bits and masked with the given 'mask'. Afterwards it translates - * the given registervalue into a signed integer if sign_bit is non-zero. - * - * Returns 0 on sucess, otherwise an error value - */ -static int snd_soc_read_signed(struct snd_soc_component *component, - unsigned int reg, unsigned int mask, unsigned int shift, - unsigned int sign_bit, int *signed_val) -{ - int ret; - unsigned int val; - - ret = snd_soc_component_read(component, reg, &val); - if (ret < 0) - return ret; - - val = (val >> shift) & mask; - - if (!sign_bit) { - *signed_val = val; - return 0; - } - - /* non-negative number */ - if (!(val & BIT(sign_bit))) { - *signed_val = val; - return 0; - } - - ret = val; - - /* - * The register most probably does not contain a full-sized int. - * Instead we have an arbitrary number of bits in a signed - * representation which has to be translated into a full-sized int. - * This is done by filling up all bits above the sign-bit. - */ - ret |= ~((int)(BIT(sign_bit) - 1)); - - *signed_val = ret; - - return 0; -} - -/** - * snd_soc_info_volsw - single mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a single mixer control, or a double - * mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - - uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - mc->min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw); - -/** - * snd_soc_get_volsw - single mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a single mixer control, or a double mixer - * control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - int val; - int ret; - - if (sign_bit) - mask = BIT(sign_bit + 1) - 1; - - ret = snd_soc_read_signed(component, reg, mask, shift, sign_bit, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = val - min; - if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; - - if (snd_soc_volsw_is_stereo(mc)) { - if (reg == reg2) - ret = snd_soc_read_signed(component, reg, mask, rshift, - sign_bit, &val); - else - ret = snd_soc_read_signed(component, reg2, mask, shift, - sign_bit, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[1] = val - min; - if (invert) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw); - -/** - * snd_soc_put_volsw - single mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a single mixer control, or a double mixer - * control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - unsigned int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - int err; - bool type_2r = false; - unsigned int val2 = 0; - unsigned int val, val_mask; - - if (sign_bit) - mask = BIT(sign_bit + 1) - 1; - - val = ((ucontrol->value.integer.value[0] + min) & mask); - if (invert) - val = max - val; - val_mask = mask << shift; - val = val << shift; - if (snd_soc_volsw_is_stereo(mc)) { - val2 = ((ucontrol->value.integer.value[1] + min) & mask); - if (invert) - val2 = max - val2; - if (reg == reg2) { - val_mask |= mask << rshift; - val |= val2 << rshift; - } else { - val2 = val2 << shift; - type_2r = true; - } - } - err = snd_soc_component_update_bits(component, reg, val_mask, val); - if (err < 0) - return err; - - if (type_2r) - err = snd_soc_component_update_bits(component, reg2, val_mask, - val2); - - return err; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw); - -/** - * snd_soc_get_volsw_sx - single mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a single mixer control, or a double mixer - * control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - int mask = (1 << (fls(min + max) - 1)) - 1; - unsigned int val; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret < 0) - return ret; - - ucontrol->value.integer.value[0] = ((val >> shift) - min) & mask; - - if (snd_soc_volsw_is_stereo(mc)) { - ret = snd_soc_component_read(component, reg2, &val); - if (ret < 0) - return ret; - - val = ((val >> rshift) - min) & mask; - ucontrol->value.integer.value[1] = val; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); - -/** - * snd_soc_put_volsw_sx - double mixer set callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to set the value of a double mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - int mask = (1 << (fls(min + max) - 1)) - 1; - int err = 0; - unsigned int val, val_mask, val2 = 0; - - val_mask = mask << shift; - val = (ucontrol->value.integer.value[0] + min) & mask; - val = val << shift; - - err = snd_soc_component_update_bits(component, reg, val_mask, val); - if (err < 0) - return err; - - if (snd_soc_volsw_is_stereo(mc)) { - val_mask = mask << rshift; - val2 = (ucontrol->value.integer.value[1] + min) & mask; - val2 = val2 << rshift; - - err = snd_soc_component_update_bits(component, reg2, val_mask, - val2); - } - return err; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); - -/** - * snd_soc_info_volsw_s8 - signed mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); - -/** - * snd_soc_get_volsw_s8 - signed mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int val; - int min = mc->min; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = - ((signed char)(val & 0xff))-min; - ucontrol->value.integer.value[1] = - ((signed char)((val >> 8) & 0xff))-min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); - -/** - * snd_soc_put_volsw_sgn - signed mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - int min = mc->min; - unsigned int val; - - val = (ucontrol->value.integer.value[0]+min) & 0xff; - val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; - - return snd_soc_component_update_bits(component, reg, 0xffff, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); - -/** - * snd_soc_info_volsw_range - single mixer info callback with range. - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information, within a range, about a single - * mixer control. - * - * returns 0 for success. - */ -int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); - -/** - * snd_soc_put_volsw_range - single mixer put value callback with range. - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value, within a range, for a single mixer control. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int rreg = mc->rreg; - unsigned int shift = mc->shift; - int min = mc->min; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - unsigned int val, val_mask; - int ret; - - if (invert) - val = (max - ucontrol->value.integer.value[0]) & mask; - else - val = ((ucontrol->value.integer.value[0] + min) & mask); - val_mask = mask << shift; - val = val << shift; - - ret = snd_soc_component_update_bits(component, reg, val_mask, val); - if (ret < 0) - return ret; - - if (snd_soc_volsw_is_stereo(mc)) { - if (invert) - val = (max - ucontrol->value.integer.value[1]) & mask; - else - val = ((ucontrol->value.integer.value[1] + min) & mask); - val_mask = mask << shift; - val = val << shift; - - ret = snd_soc_component_update_bits(component, rreg, val_mask, - val); - } - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); - -/** - * snd_soc_get_volsw_range - single mixer get callback with range - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value, within a range, of a single mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int rreg = mc->rreg; - unsigned int shift = mc->shift; - int min = mc->min; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - unsigned int val; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = (val >> shift) & mask; - if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; - else - ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[0] - min; - - if (snd_soc_volsw_is_stereo(mc)) { - ret = snd_soc_component_read(component, rreg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[1] = (val >> shift) & mask; - if (invert) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - else - ucontrol->value.integer.value[1] = - ucontrol->value.integer.value[1] - min; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); - -/** - * snd_soc_limit_volume - Set new limit to an existing volume control. - * - * @codec: where to look for the control - * @name: Name of the control - * @max: new maximum limit - * - * Return 0 for success, else error. - */ -int snd_soc_limit_volume(struct snd_soc_codec *codec, - const char *name, int max) -{ - struct snd_card *card = codec->component.card->snd_card; - struct snd_kcontrol *kctl; - struct soc_mixer_control *mc; - int found = 0; - int ret = -EINVAL; - - /* Sanity check for name and max */ - if (unlikely(!name || max <= 0)) - return -EINVAL; - - list_for_each_entry(kctl, &card->controls, list) { - if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { - found = 1; - break; - } - } - if (found) { - mc = (struct soc_mixer_control *)kctl->private_value; - if (max <= mc->max) { - mc->platform_max = max; - ret = 0; - } - } - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_limit_volume); - -int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_bytes *params = (void *)kcontrol->private_value; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; - uinfo->count = params->num_regs * component->val_bytes; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_info); - -int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_bytes *params = (void *)kcontrol->private_value; - int ret; - - if (component->regmap) - ret = regmap_raw_read(component->regmap, params->base, - ucontrol->value.bytes.data, - params->num_regs * component->val_bytes); - else - ret = -EINVAL; - - /* Hide any masked bytes to ensure consistent data reporting */ - if (ret == 0 && params->mask) { - switch (component->val_bytes) { - case 1: - ucontrol->value.bytes.data[0] &= ~params->mask; - break; - case 2: - ((u16 *)(&ucontrol->value.bytes.data))[0] - &= cpu_to_be16(~params->mask); - break; - case 4: - ((u32 *)(&ucontrol->value.bytes.data))[0] - &= cpu_to_be32(~params->mask); - break; - default: - return -EINVAL; - } - } - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_get); - -int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_bytes *params = (void *)kcontrol->private_value; - int ret, len; - unsigned int val, mask; - void *data; - - if (!component->regmap || !params->num_regs) - return -EINVAL; - - len = params->num_regs * component->val_bytes; - - data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); - if (!data) - return -ENOMEM; - - /* - * If we've got a mask then we need to preserve the register - * bits. We shouldn't modify the incoming data so take a - * copy. - */ - if (params->mask) { - ret = regmap_read(component->regmap, params->base, &val); - if (ret != 0) - goto out; - - val &= params->mask; - - switch (component->val_bytes) { - case 1: - ((u8 *)data)[0] &= ~params->mask; - ((u8 *)data)[0] |= val; - break; - case 2: - mask = ~params->mask; - ret = regmap_parse_val(component->regmap, - &mask, &mask); - if (ret != 0) - goto out; - - ((u16 *)data)[0] &= mask; - - ret = regmap_parse_val(component->regmap, - &val, &val); - if (ret != 0) - goto out; - - ((u16 *)data)[0] |= val; - break; - case 4: - mask = ~params->mask; - ret = regmap_parse_val(component->regmap, - &mask, &mask); - if (ret != 0) - goto out; - - ((u32 *)data)[0] &= mask; - - ret = regmap_parse_val(component->regmap, - &val, &val); - if (ret != 0) - goto out; - - ((u32 *)data)[0] |= val; - break; - default: - ret = -EINVAL; - goto out; - } - } - - ret = regmap_raw_write(component->regmap, params->base, - data, len); - -out: - kfree(data); - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_put); - -int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *ucontrol) -{ - struct soc_bytes_ext *params = (void *)kcontrol->private_value; - - ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES; - ucontrol->count = params->max; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); - -int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *tlv) -{ - struct soc_bytes_ext *params = (void *)kcontrol->private_value; - unsigned int count = size < params->max ? size : params->max; - int ret = -ENXIO; - - switch (op_flag) { - case SNDRV_CTL_TLV_OP_READ: - if (params->get) - ret = params->get(tlv, count); - break; - case SNDRV_CTL_TLV_OP_WRITE: - if (params->put) - ret = params->put(tlv, count); - break; - } - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); - -/** - * snd_soc_info_xr_sx - signed multi register info callback - * @kcontrol: mreg control - * @uinfo: control element information - * - * Callback to provide information of a control that can - * span multiple codec registers which together - * forms a single signed value in a MSB/LSB manner. - * - * Returns 0 for success. - */ -int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mreg_control *mc = - (struct soc_mreg_control *)kcontrol->private_value; - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = mc->min; - uinfo->value.integer.max = mc->max; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); - -/** - * snd_soc_get_xr_sx - signed multi register get callback - * @kcontrol: mreg control - * @ucontrol: control element information - * - * Callback to get the value of a control that can span - * multiple codec registers which together forms a single - * signed value in a MSB/LSB manner. The control supports - * specifying total no of bits used to allow for bitfields - * across the multiple codec registers. - * - * Returns 0 for success. - */ -int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mreg_control *mc = - (struct soc_mreg_control *)kcontrol->private_value; - unsigned int regbase = mc->regbase; - unsigned int regcount = mc->regcount; - unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; - unsigned int invert = mc->invert; - unsigned long mask = (1UL<<mc->nbits)-1; - long min = mc->min; - long max = mc->max; - long val = 0; - unsigned int regval; - unsigned int i; - int ret; - - for (i = 0; i < regcount; i++) { - ret = snd_soc_component_read(component, regbase+i, &regval); - if (ret) - return ret; - val |= (regval & regwmask) << (regwshift*(regcount-i-1)); - } - val &= mask; - if (min < 0 && val > max) - val |= ~mask; - if (invert) - val = max - val; - ucontrol->value.integer.value[0] = val; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); - -/** - * snd_soc_put_xr_sx - signed multi register get callback - * @kcontrol: mreg control - * @ucontrol: control element information - * - * Callback to set the value of a control that can span - * multiple codec registers which together forms a single - * signed value in a MSB/LSB manner. The control supports - * specifying total no of bits used to allow for bitfields - * across the multiple codec registers. - * - * Returns 0 for success. - */ -int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mreg_control *mc = - (struct soc_mreg_control *)kcontrol->private_value; - unsigned int regbase = mc->regbase; - unsigned int regcount = mc->regcount; - unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; - unsigned int invert = mc->invert; - unsigned long mask = (1UL<<mc->nbits)-1; - long max = mc->max; - long val = ucontrol->value.integer.value[0]; - unsigned int i, regval, regmask; - int err; - - if (invert) - val = max - val; - val &= mask; - for (i = 0; i < regcount; i++) { - regval = (val >> (regwshift*(regcount-i-1))) & regwmask; - regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; - err = snd_soc_component_update_bits(component, regbase+i, - regmask, regval); - if (err < 0) - return err; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); - -/** - * snd_soc_get_strobe - strobe get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback get the value of a strobe mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = 1 << shift; - unsigned int invert = mc->invert != 0; - unsigned int val; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - val &= mask; - - if (shift != 0 && val != 0) - val = val >> shift; - ucontrol->value.enumerated.item[0] = val ^ invert; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_strobe); - -/** - * snd_soc_put_strobe - strobe put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback strobe a register bit to high then low (or the inverse) - * in one pass of a single mixer enum control. - * - * Returns 1 for success. - */ -int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = 1 << shift; - unsigned int invert = mc->invert != 0; - unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; - unsigned int val1 = (strobe ^ invert) ? mask : 0; - unsigned int val2 = (strobe ^ invert) ? 0 : mask; - int err; - - err = snd_soc_component_update_bits(component, reg, mask, val1); - if (err < 0) - return err; - - return snd_soc_component_update_bits(component, reg, mask, val2); -} -EXPORT_SYMBOL_GPL(snd_soc_put_strobe); - -/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c new file mode 100644 index 000000000000..c326452a8f8e --- /dev/null +++ b/sound/soc/soc-ops.c @@ -0,0 +1,1039 @@ +/* + * soc-ops.c -- Generic ASoC operations + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. + * + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> + * with code, comments and ideas from :- + * Richard Purdie <richard(a)openedhand.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/ctype.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dpcm.h> +#include <sound/initval.h> + +/** + * snd_soc_info_enum_double - enumerated double mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a double enumerated + * mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + return snd_ctl_enum_info(uinfo, e->shift_l == e->shift_r ? 1 : 2, + e->items, e->texts); +} +EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); + +/** + * snd_soc_get_enum_double - enumerated double mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a double enumerated mixer. + * + * Returns 0 for success. + */ +int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val, item; + unsigned int reg_val; + int ret; + + ret = snd_soc_component_read(component, e->reg, &reg_val); + if (ret) + return ret; + val = (reg_val >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[0] = item; + if (e->shift_l != e->shift_r) { + val = (reg_val >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[1] = item; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); + +/** + * snd_soc_put_enum_double - enumerated double mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a double enumerated mixer. + * + * Returns 0 for success. + */ +int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val; + unsigned int mask; + + if (item[0] >= e->items) + return -EINVAL; + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + mask = e->mask << e->shift_l; + if (e->shift_l != e->shift_r) { + if (item[1] >= e->items) + return -EINVAL; + val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; + mask |= e->mask << e->shift_r; + } + + return snd_soc_component_update_bits(component, e->reg, mask, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); + +/** + * snd_soc_read_signed - Read a codec register and interprete as signed value + * @component: component + * @reg: Register to read + * @mask: Mask to use after shifting the register value + * @shift: Right shift of register value + * @sign_bit: Bit that describes if a number is negative or not. + * @signed_val: Pointer to where the read value should be stored + * + * This functions reads a codec register. The register value is shifted right + * by 'shift' bits and masked with the given 'mask'. Afterwards it translates + * the given registervalue into a signed integer if sign_bit is non-zero. + * + * Returns 0 on sucess, otherwise an error value + */ +static int snd_soc_read_signed(struct snd_soc_component *component, + unsigned int reg, unsigned int mask, unsigned int shift, + unsigned int sign_bit, int *signed_val) +{ + int ret; + unsigned int val; + + ret = snd_soc_component_read(component, reg, &val); + if (ret < 0) + return ret; + + val = (val >> shift) & mask; + + if (!sign_bit) { + *signed_val = val; + return 0; + } + + /* non-negative number */ + if (!(val & BIT(sign_bit))) { + *signed_val = val; + return 0; + } + + ret = val; + + /* + * The register most probably does not contain a full-sized int. + * Instead we have an arbitrary number of bits in a signed + * representation which has to be translated into a full-sized int. + * This is done by filling up all bits above the sign-bit. + */ + ret |= ~((int)(BIT(sign_bit) - 1)); + + *signed_val = ret; + + return 0; +} + +/** + * snd_soc_info_volsw - single mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a single mixer control, or a double + * mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - mc->min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw); + +/** + * snd_soc_get_volsw - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int sign_bit = mc->sign_bit; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + int val; + int ret; + + if (sign_bit) + mask = BIT(sign_bit + 1) - 1; + + ret = snd_soc_read_signed(component, reg, mask, shift, sign_bit, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[0] = val - min; + if (invert) + ucontrol->value.integer.value[0] = + max - ucontrol->value.integer.value[0]; + + if (snd_soc_volsw_is_stereo(mc)) { + if (reg == reg2) + ret = snd_soc_read_signed(component, reg, mask, rshift, + sign_bit, &val); + else + ret = snd_soc_read_signed(component, reg2, mask, shift, + sign_bit, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[1] = val - min; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw); + +/** + * snd_soc_put_volsw - single mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + unsigned int sign_bit = mc->sign_bit; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + int err; + bool type_2r = false; + unsigned int val2 = 0; + unsigned int val, val_mask; + + if (sign_bit) + mask = BIT(sign_bit + 1) - 1; + + val = ((ucontrol->value.integer.value[0] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + if (snd_soc_volsw_is_stereo(mc)) { + val2 = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val2 = max - val2; + if (reg == reg2) { + val_mask |= mask << rshift; + val |= val2 << rshift; + } else { + val2 = val2 << shift; + type_2r = true; + } + } + err = snd_soc_component_update_bits(component, reg, val_mask, val); + if (err < 0) + return err; + + if (type_2r) + err = snd_soc_component_update_bits(component, reg2, val_mask, + val2); + + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw); + +/** + * snd_soc_get_volsw_sx - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret < 0) + return ret; + + ucontrol->value.integer.value[0] = ((val >> shift) - min) & mask; + + if (snd_soc_volsw_is_stereo(mc)) { + ret = snd_soc_component_read(component, reg2, &val); + if (ret < 0) + return ret; + + val = ((val >> rshift) - min) & mask; + ucontrol->value.integer.value[1] = val; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); + +/** + * snd_soc_put_volsw_sx - double mixer set callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a double mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + int err = 0; + unsigned int val, val_mask, val2 = 0; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] + min) & mask; + val = val << shift; + + err = snd_soc_component_update_bits(component, reg, val_mask, val); + if (err < 0) + return err; + + if (snd_soc_volsw_is_stereo(mc)) { + val_mask = mask << rshift; + val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = val2 << rshift; + + err = snd_soc_component_update_bits(component, reg2, val_mask, + val2); + } + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); + +/** + * snd_soc_info_volsw_s8 - signed mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + int min = mc->min; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); + +/** + * snd_soc_get_volsw_s8 - signed mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int val; + int min = mc->min; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[0] = + ((signed char)(val & 0xff))-min; + ucontrol->value.integer.value[1] = + ((signed char)((val >> 8) & 0xff))-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); + +/** + * snd_soc_put_volsw_sgn - signed mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + int min = mc->min; + unsigned int val; + + val = (ucontrol->value.integer.value[0]+min) & 0xff; + val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + + return snd_soc_component_update_bits(component, reg, 0xffff, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); + +/** + * snd_soc_info_volsw_range - single mixer info callback with range. + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information, within a range, about a single + * mixer control. + * + * returns 0 for success. + */ +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + int min = mc->min; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); + +/** + * snd_soc_put_volsw_range - single mixer put value callback with range. + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value, within a range, for a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val, val_mask; + int ret; + + if (invert) + val = (max - ucontrol->value.integer.value[0]) & mask; + else + val = ((ucontrol->value.integer.value[0] + min) & mask); + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_component_update_bits(component, reg, val_mask, val); + if (ret < 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + if (invert) + val = (max - ucontrol->value.integer.value[1]) & mask; + else + val = ((ucontrol->value.integer.value[1] + min) & mask); + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_component_update_bits(component, rreg, val_mask, + val); + } + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); + +/** + * snd_soc_get_volsw_range - single mixer get callback with range + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value, within a range, of a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[0] = (val >> shift) & mask; + if (invert) + ucontrol->value.integer.value[0] = + max - ucontrol->value.integer.value[0]; + else + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[0] - min; + + if (snd_soc_volsw_is_stereo(mc)) { + ret = snd_soc_component_read(component, rreg, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[1] = (val >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + else + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); + +/** + * snd_soc_limit_volume - Set new limit to an existing volume control. + * + * @codec: where to look for the control + * @name: Name of the control + * @max: new maximum limit + * + * Return 0 for success, else error. + */ +int snd_soc_limit_volume(struct snd_soc_codec *codec, + const char *name, int max) +{ + struct snd_card *card = codec->component.card->snd_card; + struct snd_kcontrol *kctl; + struct soc_mixer_control *mc; + int found = 0; + int ret = -EINVAL; + + /* Sanity check for name and max */ + if (unlikely(!name || max <= 0)) + return -EINVAL; + + list_for_each_entry(kctl, &card->controls, list) { + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { + found = 1; + break; + } + } + if (found) { + mc = (struct soc_mixer_control *)kctl->private_value; + if (max <= mc->max) { + mc->platform_max = max; + ret = 0; + } + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_limit_volume); + +int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = params->num_regs * component->val_bytes; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_info); + +int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + int ret; + + if (component->regmap) + ret = regmap_raw_read(component->regmap, params->base, + ucontrol->value.bytes.data, + params->num_regs * component->val_bytes); + else + ret = -EINVAL; + + /* Hide any masked bytes to ensure consistent data reporting */ + if (ret == 0 && params->mask) { + switch (component->val_bytes) { + case 1: + ucontrol->value.bytes.data[0] &= ~params->mask; + break; + case 2: + ((u16 *)(&ucontrol->value.bytes.data))[0] + &= cpu_to_be16(~params->mask); + break; + case 4: + ((u32 *)(&ucontrol->value.bytes.data))[0] + &= cpu_to_be32(~params->mask); + break; + default: + return -EINVAL; + } + } + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_get); + +int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + int ret, len; + unsigned int val, mask; + void *data; + + if (!component->regmap || !params->num_regs) + return -EINVAL; + + len = params->num_regs * component->val_bytes; + + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + /* + * If we've got a mask then we need to preserve the register + * bits. We shouldn't modify the incoming data so take a + * copy. + */ + if (params->mask) { + ret = regmap_read(component->regmap, params->base, &val); + if (ret != 0) + goto out; + + val &= params->mask; + + switch (component->val_bytes) { + case 1: + ((u8 *)data)[0] &= ~params->mask; + ((u8 *)data)[0] |= val; + break; + case 2: + mask = ~params->mask; + ret = regmap_parse_val(component->regmap, + &mask, &mask); + if (ret != 0) + goto out; + + ((u16 *)data)[0] &= mask; + + ret = regmap_parse_val(component->regmap, + &val, &val); + if (ret != 0) + goto out; + + ((u16 *)data)[0] |= val; + break; + case 4: + mask = ~params->mask; + ret = regmap_parse_val(component->regmap, + &mask, &mask); + if (ret != 0) + goto out; + + ((u32 *)data)[0] &= mask; + + ret = regmap_parse_val(component->regmap, + &val, &val); + if (ret != 0) + goto out; + + ((u32 *)data)[0] |= val; + break; + default: + ret = -EINVAL; + goto out; + } + } + + ret = regmap_raw_write(component->regmap, params->base, + data, len); + +out: + kfree(data); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_put); + +int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + + ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES; + ucontrol->count = params->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); + +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + unsigned int count = size < params->max ? size : params->max; + int ret = -ENXIO; + + switch (op_flag) { + case SNDRV_CTL_TLV_OP_READ: + if (params->get) + ret = params->get(tlv, count); + break; + case SNDRV_CTL_TLV_OP_WRITE: + if (params->put) + ret = params->put(tlv, count); + break; + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); + +/** + * snd_soc_info_xr_sx - signed multi register info callback + * @kcontrol: mreg control + * @uinfo: control element information + * + * Callback to provide information of a control that can + * span multiple codec registers which together + * forms a single signed value in a MSB/LSB manner. + * + * Returns 0 for success. + */ +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = mc->min; + uinfo->value.integer.max = mc->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); + +/** + * snd_soc_get_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to get the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long min = mc->min; + long max = mc->max; + long val = 0; + unsigned int regval; + unsigned int i; + int ret; + + for (i = 0; i < regcount; i++) { + ret = snd_soc_component_read(component, regbase+i, &regval); + if (ret) + return ret; + val |= (regval & regwmask) << (regwshift*(regcount-i-1)); + } + val &= mask; + if (min < 0 && val > max) + val |= ~mask; + if (invert) + val = max - val; + ucontrol->value.integer.value[0] = val; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); + +/** + * snd_soc_put_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to set the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long max = mc->max; + long val = ucontrol->value.integer.value[0]; + unsigned int i, regval, regmask; + int err; + + if (invert) + val = max - val; + val &= mask; + for (i = 0; i < regcount; i++) { + regval = (val >> (regwshift*(regcount-i-1))) & regwmask; + regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; + err = snd_soc_component_update_bits(component, regbase+i, + regmask, regval); + if (err < 0) + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); + +/** + * snd_soc_get_strobe - strobe get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback get the value of a strobe mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret) + return ret; + + val &= mask; + + if (shift != 0 && val != 0) + val = val >> shift; + ucontrol->value.enumerated.item[0] = val ^ invert; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_strobe); + +/** + * snd_soc_put_strobe - strobe put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback strobe a register bit to high then low (or the inverse) + * in one pass of a single mixer enum control. + * + * Returns 1 for success. + */ +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; + unsigned int val1 = (strobe ^ invert) ? mask : 0; + unsigned int val2 = (strobe ^ invert) ? 0 : mask; + int err; + + err = snd_soc_component_update_bits(component, reg, mask, val1); + if (err < 0) + return err; + + return snd_soc_component_update_bits(component, reg, mask, val2); +} +EXPORT_SYMBOL_GPL(snd_soc_put_strobe); -- 2.1.1

1 0

[alsa-devel] [PATCH] ASoC: soc-pcm: always cancel any pending delayed stream shutdown
by Hector Palacios 28 Oct '14

28 Oct '14

After playback a shutdown is scheduled to happen within 5 seconds. If another playback takes place before the scheduled work times out, the work is cancelled, but if a recording takes place, the work will shutdown in the middle of the recording, causing problems. To reproduce the issue chain the following commands: arecord -M --duration 5 -c 2 /tmp/test.wav; aplay /tmp/test.wav; sleep 2; arecord -M --duration 5 -c 2 /tmp/test.wav; aplay /tmp/test.wav; The fist command records 5 seconds of audio. The second command plays the recorded audio. At the end of the playback a work will be scheduled to shutdown in 5 seconds, but only 2 seconds after the first playback, we run another recording. When we have recorded 3 seconds, the shutdown will occur which can cause incorrect behavior and bad recorded data. As a consequence, the last playback reproduces correct audio for the first 3 seconds and incorrect audio after that. Signed-off-by: Hector Palacios <hector.palacios(a)digi.com> --- sound/soc/soc-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 642c86240752..d19a926872e8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -760,8 +760,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } /* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - rtd->pop_wait) { + if (rtd->pop_wait) { rtd->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); }

4 5

[alsa-devel] [PATCH v2 1/7] ASoC: Intel: Make ADSP memory block allocation more generic
by Liam Girdwood 28 Oct '14

28 Oct '14

Current block allocation is tied to block type and requestor type. Make the allocation more generic by removing the struct module parameter and adding a generic block allocator structure. Also pass in the list that the blocks have to be added too in order to remove dependence on block requestor type. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-dsp.c | 24 ++- sound/soc/intel/sst-dsp-priv.h | 83 +++++----- sound/soc/intel/sst-firmware.c | 317 +++++++++++++++++++++++++------------ sound/soc/intel/sst-haswell-dsp.c | 41 ++--- 4 files changed, 288 insertions(+), 177 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c index fc58876..5a9e567 100644 --- a/sound/soc/intel/sst-baytrail-dsp.c +++ b/sound/soc/intel/sst-baytrail-dsp.c @@ -67,17 +67,12 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; int count; memset(&template, 0, sizeof(template)); template.id = module->type; template.entry = module->entry_point; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -94,19 +89,19 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_BYT_IRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_BYT_DRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.dram_offset; - block_data.type = SST_MEM_DRAM; + mod->type = SST_MEM_DRAM; break; case SST_BYT_CACHE: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe); - block_data.type = SST_MEM_CACHE; + mod->type = SST_MEM_CACHE; break; default: dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n", @@ -114,11 +109,10 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); - sst_module_insert_fixed_block(mod, &block_data); + sst_module_alloc_blocks(mod); block = (void *)block + sizeof(*block) + block->size; } diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index ffb308b..f58a049 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -84,15 +84,6 @@ struct sst_mailbox { }; /* - * Audio DSP Firmware data types. - */ -enum sst_data_type { - SST_DATA_M = 0, /* module block data */ - SST_DATA_P = 1, /* peristant data (text, data) */ - SST_DATA_S = 2, /* scratch data (usually buffers) */ -}; - -/* * Audio DSP memory block types. */ enum sst_mem_type { @@ -125,23 +116,6 @@ struct sst_fw { }; /* - * Audio DSP Generic Module data. - * - * This is used to dsecribe any sections of persistent (text and data) and - * scratch (buffers) of module data in ADSP memory space. - */ -struct sst_module_data { - - enum sst_mem_type type; /* destination memory type */ - enum sst_data_type data_type; /* type of module data */ - - u32 size; /* size in bytes */ - int32_t offset; /* offset in FW file */ - u32 data_offset; /* offset in ADSP memory space */ - void *data; /* module data */ -}; - -/* * Audio DSP Generic Module Template. * * Used to define and register a new FW module. This data is extracted from @@ -150,15 +124,42 @@ struct sst_module_data { struct sst_module_template { u32 id; u32 entry; /* entry point */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; + u32 persistent_size; +}; + +/* + * Block Allocator - Used to allocate blocks of DSP memory. + */ +struct sst_block_allocator { + u32 id; + u32 offset; + int size; + enum sst_mem_type type; +}; + +/* + * Runtime Module Instance - A module object can be instanciated multiple + * times within the DSP FW. + */ +struct sst_module_runtime { + struct sst_dsp *dsp; + int id; + struct sst_module *module; /* parent module we belong too */ + + u32 persistent_offset; /* private memory size */ + void *private; + + struct list_head list; + struct list_head block_list; /* list of blocks used */ }; /* * Audio DSP Generic Module. * * Each Firmware file can consist of 1..N modules. A module can span multiple - * ADSP memory blocks. The simplest FW will be a file with 1 module. + * ADSP memory blocks. The simplest FW will be a file with 1 module. A module + * can be instanciated multiple times in the DSP. */ struct sst_module { struct sst_dsp *dsp; @@ -167,10 +168,13 @@ struct sst_module { /* module configuration */ u32 id; u32 entry; /* module entry point */ - u32 offset; /* module offset in firmware file */ + s32 offset; /* module offset in firmware file */ u32 size; /* module size */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; /* global scratch memory required */ + u32 persistent_size; /* private memory required */ + enum sst_mem_type type; /* destination memory type */ + u32 data_offset; /* offset in ADSP memory space */ + void *data; /* module data */ /* runtime */ u32 usage_count; /* can be unloaded if count == 0 */ @@ -180,6 +184,7 @@ struct sst_module { struct list_head block_list; /* Module list of blocks in use */ struct list_head list; /* DSP list of modules */ struct list_head list_fw; /* FW list of modules */ + struct list_head runtime_list; /* list of runtime module objects*/ }; /* @@ -208,7 +213,6 @@ struct sst_mem_block { struct sst_block_ops *ops; /* block operations, if any */ /* block status */ - enum sst_data_type data_type; /* data type held in this block */ u32 bytes_used; /* bytes in use by modules */ void *private; /* generic core does not touch this */ int users; /* number of modules using this block */ @@ -290,19 +294,20 @@ void sst_fw_unload(struct sst_fw *sst_fw); /* Create/Free firmware modules */ struct sst_module *sst_module_new(struct sst_fw *sst_fw, struct sst_module_template *template, void *private); -void sst_module_free(struct sst_module *sst_module); -int sst_module_insert(struct sst_module *sst_module); -int sst_module_remove(struct sst_module *sst_module); -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data); +void sst_module_free(struct sst_module *module); struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); /* allocate/free pesistent/scratch memory regions managed by drv */ struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); void sst_mem_block_free_scratch(struct sst_dsp *dsp, struct sst_module *scratch); -int sst_block_module_remove(struct sst_module *module); +int sst_module_alloc_blocks(struct sst_module *module); +int sst_module_free_blocks(struct sst_module *module); +/* generic block allocation */ +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list); +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list); /* Register the DSPs memory blocks - would be nice to read from ACPI */ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index cf3d199..ca90278 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,7 +30,7 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void block_module_remove(struct sst_module *module); +#define SST_HSW_BLOCK_ANY 0xffffffff static inline void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { @@ -38,6 +38,65 @@ static inline void sst_memcpy32(volatile void __iomem *dest, void *src, u32 byte __iowrite32_copy((void *)dest, src, bytes/4); } + +/* remove module from memory - callers hold locks */ +static void block_list_remove(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block, *tmp; + int err; + + /* disable each block */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->disable) { + err = block->ops->disable(block); + if (err < 0) + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + } + } + + /* mark each block as free */ + list_for_each_entry_safe(block, tmp, block_list, module_list) { + list_del(&block->module_list); + list_move(&block->list, &dsp->free_block_list); + dev_dbg(dsp->dev, "block freed %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } +} + +/* prepare the memory block to receive data from host - callers hold locks */ +static int block_list_prepare(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block; + int ret = 0; + + /* enable each block so that's it'e ready for data */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->enable) { + ret = block->ops->enable(block); + if (ret < 0) { + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + goto err; + } + } + } + return ret; + +err: + list_for_each_entry(block, block_list, module_list) { + if (block->ops && block->ops->disable) + block->ops->disable(block); + } + return ret; +} + /* create new generic firmware object */ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, const struct firmware *fw, void *private) @@ -199,73 +258,122 @@ void sst_module_free(struct sst_module *sst_module) } EXPORT_SYMBOL_GPL(sst_module_free); -static struct sst_mem_block *find_block(struct sst_dsp *dsp, int type, - u32 offset) +struct sst_module_runtime *sst_module_runtime_new(struct sst_module *module, + int id, void *private) +{ + struct sst_dsp *dsp = module->dsp; + struct sst_module_runtime *runtime; + + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (runtime == NULL) + return NULL; + + runtime->id = id; + runtime->dsp = dsp; + runtime->module = module; + INIT_LIST_HEAD(&runtime->block_list); + + mutex_lock(&dsp->mutex); + list_add(&runtime->list, &module->runtime_list); + mutex_unlock(&dsp->mutex); + + return runtime; +} +EXPORT_SYMBOL_GPL(sst_module_runtime_new); + +void sst_module_runtime_free(struct sst_module_runtime *runtime) +{ + struct sst_dsp *dsp = runtime->dsp; + + mutex_lock(&dsp->mutex); + list_del(&runtime->list); + mutex_unlock(&dsp->mutex); + + kfree(runtime); +} +EXPORT_SYMBOL_GPL(sst_module_runtime_free); + +static struct sst_mem_block *find_block(struct sst_dsp *dsp, + struct sst_block_allocator *ba) { struct sst_mem_block *block; list_for_each_entry(block, &dsp->free_block_list, list) { - if (block->type == type && block->offset == offset) + if (block->type == ba->type && block->offset == ba->offset) return block; } return NULL; } -static int block_alloc_contiguous(struct sst_module *module, - struct sst_module_data *data, u32 offset, int size) +/* Block allocator must be on block boundary */ +static int block_alloc_contiguous(struct sst_dsp *dsp, + struct sst_block_allocator *ba, struct list_head *block_list) { struct list_head tmp = LIST_HEAD_INIT(tmp); - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block; + u32 block_start = SST_HSW_BLOCK_ANY; + int size = ba->size, offset = ba->offset; + + while (ba->size > 0) { - while (size > 0) { - block = find_block(dsp, data->type, offset); + block = find_block(dsp, ba); if (!block) { list_splice(&tmp, &dsp->free_block_list); + + ba->size = size; + ba->offset = offset; return -ENOMEM; } list_move_tail(&block->list, &tmp); - offset += block->size; - size -= block->size; + ba->offset += block->size; + ba->size -= block->size; } + ba->size = size; + ba->offset = offset; + + list_for_each_entry(block, &tmp, list) { + + if (block->offset < block_start) + block_start = block->offset; - list_for_each_entry(block, &tmp, list) - list_add(&block->module_list, &module->block_list); + list_add(&block->module_list, block_list); + + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } list_splice(&tmp, &dsp->used_block_list); return 0; } -/* allocate free DSP blocks for module data - callers hold locks */ -static int block_alloc(struct sst_module *module, - struct sst_module_data *data) +/* allocate first free DSP blocks for data - callers hold locks */ +static int block_alloc(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; int ret = 0; - if (data->size == 0) + if (ba->size == 0) return 0; /* find first free whole blocks that can hold module */ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; - if (data->size > block->size) + if (ba->size > block->size) continue; - data->offset = block->offset; - block->data_type = data->data_type; - block->bytes_used = data->size % block->size; - list_add(&block->module_list, &module->block_list); + ba->offset = block->offset; + block->bytes_used = ba->size % block->size; + list_add(&block->module_list, block_list); list_move(&block->list, &dsp->used_block_list); - dev_dbg(dsp->dev, " *module %d added block %d:%d\n", - module->id, block->type, block->index); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } @@ -273,15 +381,19 @@ static int block_alloc(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; /* do we span > 1 blocks */ - if (data->size > block->size) { - ret = block_alloc_contiguous(module, data, - block->offset, data->size); + if (ba->size > block->size) { + + /* align ba to block boundary */ + ba->offset = block->offset; + + ret = block_alloc_contiguous(dsp, ba, block_list); if (ret == 0) return ret; + } } @@ -289,93 +401,74 @@ static int block_alloc(struct sst_module *module, return -ENOMEM; } -/* remove module from memory - callers hold locks */ -static void block_module_remove(struct sst_module *module) +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_mem_block *block, *tmp; - struct sst_dsp *dsp = module->dsp; - int err; + int ret; - /* disable each block */ - list_for_each_entry(block, &module->block_list, module_list) { + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba->size, ba->offset, ba->type); - if (block->ops && block->ops->disable) { - err = block->ops->disable(block); - if (err < 0) - dev_err(dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - } - } + mutex_lock(&dsp->mutex); - /* mark each block as free */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { - list_del(&block->module_list); - list_move(&block->list, &dsp->free_block_list); + ret = block_alloc(dsp, ba, block_list); + if (ret < 0) { + dev_err(dsp->dev, "error: can't alloc blocks %d\n", ret); + goto out; } -} - -/* prepare the memory block to receive data from host - callers hold locks */ -static int block_module_prepare(struct sst_module *module) -{ - struct sst_mem_block *block; - int ret = 0; - /* enable each block so that's it'e ready for module P/S data */ - list_for_each_entry(block, &module->block_list, module_list) { + /* prepare DSP blocks for module usage */ + ret = block_list_prepare(dsp, block_list); + if (ret < 0) + dev_err(dsp->dev, "error: prepare failed\n"); - if (block->ops && block->ops->enable) { - ret = block->ops->enable(block); - if (ret < 0) { - dev_err(module->dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - goto err; - } - } - } +out: + mutex_unlock(&dsp->mutex); return ret; +} +EXPORT_SYMBOL_GPL(sst_alloc_blocks); -err: - list_for_each_entry(block, &module->block_list, module_list) { - if (block->ops && block->ops->disable) - block->ops->disable(block); - } - return ret; +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list) +{ + mutex_lock(&dsp->mutex); + block_list_remove(dsp, block_list); + mutex_unlock(&dsp->mutex); + return 0; } +EXPORT_SYMBOL_GPL(sst_free_blocks); /* allocate memory blocks for static module addresses - callers hold locks */ -static int block_alloc_fixed(struct sst_module *module, - struct sst_module_data *data) +static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; - u32 end = data->offset + data->size, block_end; + u32 end = ba->offset + ba->size, block_end; int err; /* only IRAM/DRAM blocks are managed */ - if (data->type != SST_MEM_IRAM && data->type != SST_MEM_DRAM) + if (ba->type != SST_MEM_IRAM && ba->type != SST_MEM_DRAM) return 0; /* are blocks already attached to this module */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { + list_for_each_entry_safe(block, tmp, block_list, module_list) { - /* force compacting mem blocks of the same data_type */ - if (block->data_type != data->data_type) + /* ignore blocks with wrong type */ + if (block->type != ba->type) continue; block_end = block->offset + block->size; /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) + if (ba->offset >= block->offset && end <= block_end) return 0; /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { - err = block_alloc_contiguous(module, data, - block->offset + block->size, - data->size - block->size); + /* align ba to block boundary */ + ba->size -= block_end - ba->offset; + ba->offset = block_end; + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; @@ -388,53 +481,67 @@ static int block_alloc_fixed(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { block_end = block->offset + block->size; + /* ignore blocks with wrong type */ + if (block->type != ba->type) + continue; + /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) { + if (ba->offset >= block->offset && end <= block_end) { /* add block */ - block->data_type = data->data_type; list_move(&block->list, &dsp->used_block_list); - list_add(&block->module_list, &module->block_list); + list_add(&block->module_list, block_list); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { + + /* align ba to block boundary */ + ba->offset = block->offset; - err = block_alloc_contiguous(module, data, - block->offset, data->size); + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; return 0; } - } return -ENOMEM; } /* Load fixed module data into DSP memory blocks */ -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data) +int sst_module_alloc_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; + struct sst_fw *sst_fw = module->sst_fw; + struct sst_block_allocator ba; int ret; + ba.size = module->size; + ba.type = module->type; + ba.offset = module->offset; + + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba.size, ba.offset, ba.type); + mutex_lock(&dsp->mutex); /* alloc blocks that includes this section */ - ret = block_alloc_fixed(module, data); + ret = block_alloc_fixed(dsp, &ba, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: no free blocks for section at offset 0x%x size 0x%x\n", - data->offset, data->size); + module->offset, module->size); mutex_unlock(&dsp->mutex); return -ENOMEM; } /* prepare DSP blocks for module copy */ - ret = block_module_prepare(module); + ret = block_list_prepare(dsp, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: fw module prepare failed\n"); goto err; @@ -447,19 +554,23 @@ int sst_module_insert_fixed_block(struct sst_module *module, return ret; err: - block_module_remove(module); + block_list_remove(dsp, &module->block_list); mutex_unlock(&dsp->mutex); return ret; } -EXPORT_SYMBOL_GPL(sst_module_insert_fixed_block); +EXPORT_SYMBOL_GPL(sst_module_alloc_blocks); /* Unload entire module from DSP memory */ -int sst_block_module_remove(struct sst_module *module) +int sst_module_free_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; mutex_lock(&dsp->mutex); - block_module_remove(module); + block_list_remove(dsp, &module->block_list); + mutex_unlock(&dsp->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(sst_module_free_blocks); mutex_unlock(&dsp->mutex); return 0; } diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 4b6c163..22cc697 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -86,9 +86,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; - int count; + int count, ret; void __iomem *ram; /* TODO: allowed module types need to be configurable */ @@ -109,13 +108,9 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, memset(&template, 0, sizeof(template)); template.id = module->type; - template.entry = module->entry_point; - template.p.size = module->info.persistent_size; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.size = module->info.scratch_size; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; + template.entry = module->entry_point - 4; + template.persistent_size = module->info.persistent_size; + template.scratch_size = module->info.scratch_size; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -135,14 +130,14 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_HSW_IRAM: ram = dsp->addr.lpe; - block_data.offset = + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: ram = dsp->addr.lpe; - block_data.offset = block->ram_offset; - block_data.type = SST_MEM_DRAM; + mod->offset = block->ram_offset; + mod->type = SST_MEM_DRAM; break; default: dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n", @@ -151,20 +146,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); - block_data.data_offset = block_data.data - fw->dma_buf; + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); + mod->data_offset = mod->data - fw->dma_buf; - dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + dev_dbg(dsp->dev, "module block %d type 0x%x " "size 0x%x ==> ram %p offset 0x%x\n", - count, block->type, block->size, ram, + count, mod->type, block->size, ram, block->ram_offset); - sst_module_insert_fixed_block(mod, &block_data); + ret = sst_module_alloc_blocks(mod); + if (ret < 0) { + dev_err(dsp->dev, "error: could not allocate blocks for module %d\n", + count); + sst_module_free(mod); + return ret; + } block = (void *)block + sizeof(*block) + block->size; } + return 0; } -- 1.9.1

1 6

[alsa-devel] [PATCH] ALSA: compress: fix documentation errors
by Vinod Koul 28 Oct '14

28 Oct '14

Some structure documentation was not right so fix it now Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/compress_driver.h | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index ae6c3b8..396e8f7 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -42,12 +42,11 @@ struct snd_compr_ops; * @buffer_size: size of the above buffer * @fragment_size: size of buffer fragment in bytes * @fragments: number of such fragments - * @hw_pointer: offset of last location in buffer where DSP copied data - * @app_pointer: offset of last location in buffer where app wrote data * @total_bytes_available: cumulative number of bytes made available in * the ring buffer * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep + * @private_data: driver private data pointer */ struct snd_compr_runtime { snd_pcm_state_t state; @@ -94,6 +93,8 @@ struct snd_compr_stream { * This can be called in during stream creation only to set codec params * and the stream properties * @get_params: retrieve the codec parameters, mandatory + * @set_metadata: Set the metadata values for a stream + * @get_metadata: retreives the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: fsl_asrc: Add reg_defaults for regmap to fix kernel dump
by Nicolin Chen 28 Oct '14

28 Oct '14

Kernel dump (WARN_ON) ocurred during system boot-up inside regmap_write(): ------------[ cut here ]------------ WARNING: CPU: 0 PID: 47 at kernel/locking/lockdep.c:2744 lockdep_trace_alloc+0xe8/0x108() DEBUG_LOCKS_WARN_ON(irqs_disabled_flags(flags)) Modules linked in: CPU: 0 PID: 47 Comm: kworker/u2:2 Not tainted 3.18.0-rc1-10245-gb75d289-dirty #56 Workqueue: deferwq deferred_probe_work_func Backtrace: [<80012294>] (dump_backtrace) from [<80012578>] (show_stack+0x18/0x1c) r6:8097c73c r5:8097c73c r4:00000000 r3:be33ba80 [<80012560>] (show_stack) from [<806aac48>] (dump_stack+0x8c/0xa4) [<806aabbc>] (dump_stack) from [<8002a694>] (warn_slowpath_common+0x70/0x94) r6:80062838 r5:00000009 r4:bd827b30 r3:be33ba80 [<8002a624>] (warn_slowpath_common) from [<8002a6f0>] (warn_slowpath_fmt+0x38/0x40) r8:00000004 r7:00000001 r6:000080d0 r5:60000193 r4:bd826010 [<8002a6bc>] (warn_slowpath_fmt) from [<80062838>] (lockdep_trace_alloc+0xe8/0x108) r3:80831590 r2:8082e160 [<80062750>] (lockdep_trace_alloc) from [<800ea5dc>] (kmem_cache_alloc+0x28/0x134) r5:000080d0 r4:be001f00 [<800ea5b4>] (kmem_cache_alloc) from [<8038d72c>] (regcache_rbtree_write+0x15c/0x648) r10:00000000 r9:0000001c r8:00000004 r7:00000001 r6:00000000 r5:bd819a00 r4:00000000 r3:811aea88 [<8038d5d0>] (regcache_rbtree_write) from [<8038c4d8>] (regcache_write+0x5c/0x64) r10:be3f9f88 r9:00000000 r8:00000004 r7:00000001 r6:00000000 r5:00000001 r4:bd819a00 [<8038c47c>] (regcache_write) from [<8038b0dc>] (_regmap_raw_write+0x134/0x5f4) r6:be3f9f84 r5:00000001 r4:bd819a00 r3:00000001 [<8038afa8>] (_regmap_raw_write) from [<8038b610>] (_regmap_bus_raw_write+0x74/0x94) r10:00000000 r9:00000001 r8:be3fb080 r7:bd819a00 r6:00000001 r5:00000000 r4:bd819a00 [<8038b59c>] (_regmap_bus_raw_write) from [<8038a8b4>] (_regmap_write+0x60/0x9c) r6:00000001 r5:00000000 r4:bd819a00 r3:8038b59c [<8038a854>] (_regmap_write) from [<8038ba24>] (regmap_write+0x48/0x68) r7:bd81ad80 r6:00000001 r5:00000000 r4:bd819a00 [<8038b9dc>] (regmap_write) from [<80528f30>] (fsl_asrc_dai_probe+0x34/0x104) r6:bd888628 r5:be3fb080 r4:be3b4410 r3:be3b442c ------------[ dump end ]------------ ============================================================================= 2741 /* 2742 * Oi! Can't be having __GFP_FS allocations with IRQs disabled. 2743 */ 2744 if (DEBUG_LOCKS_WARN_ON(irqs_disabled_flags(flags))) 2745 return; ============================================================================= By looking at 2744 line, we can get that it's because regcache_rbtree_write() would call kmalloc() with GFP flag if it couldn't find an existing block to insert nodes while this kmalloc() call is inside a spin_lock_irq_save pair, i.e. IRQs disabled. Even though this may be a bug that should be fixed, I still try to send this patch as a quick fix (work around) since it does no harm to assign default values of every registers when using regcache. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_asrc.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index f2d8652..817d304 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -684,12 +684,38 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg) } } +static struct reg_default fsl_asrc_reg[] = { + { REG_ASRCTR, 0x0000 }, { REG_ASRIER, 0x0000 }, + { REG_ASRCNCR, 0x0000 }, { REG_ASRCFG, 0x0000 }, + { REG_ASRCSR, 0x0000 }, { REG_ASRCDR1, 0x0000 }, + { REG_ASRCDR2, 0x0000 }, { REG_ASRSTR, 0x0000 }, + { REG_ASRRA, 0x0000 }, { REG_ASRRB, 0x0000 }, + { REG_ASRRC, 0x0000 }, { REG_ASRPM1, 0x0000 }, + { REG_ASRPM2, 0x0000 }, { REG_ASRPM3, 0x0000 }, + { REG_ASRPM4, 0x0000 }, { REG_ASRPM5, 0x0000 }, + { REG_ASRTFR1, 0x0000 }, { REG_ASRCCR, 0x0000 }, + { REG_ASRDIA, 0x0000 }, { REG_ASRDOA, 0x0000 }, + { REG_ASRDIB, 0x0000 }, { REG_ASRDOB, 0x0000 }, + { REG_ASRDIC, 0x0000 }, { REG_ASRDOC, 0x0000 }, + { REG_ASRIDRHA, 0x0000 }, { REG_ASRIDRLA, 0x0000 }, + { REG_ASRIDRHB, 0x0000 }, { REG_ASRIDRLB, 0x0000 }, + { REG_ASRIDRHC, 0x0000 }, { REG_ASRIDRLC, 0x0000 }, + { REG_ASR76K, 0x0A47 }, { REG_ASR56K, 0x0DF3 }, + { REG_ASRMCRA, 0x0000 }, { REG_ASRFSTA, 0x0000 }, + { REG_ASRMCRB, 0x0000 }, { REG_ASRFSTB, 0x0000 }, + { REG_ASRMCRC, 0x0000 }, { REG_ASRFSTC, 0x0000 }, + { REG_ASRMCR1A, 0x0000 }, { REG_ASRMCR1B, 0x0000 }, + { REG_ASRMCR1C, 0x0000 }, +}; + static const struct regmap_config fsl_asrc_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, .max_register = REG_ASRMCR1C, + .reg_defaults = fsl_asrc_reg, + .num_reg_defaults = ARRAY_SIZE(fsl_asrc_reg), .readable_reg = fsl_asrc_readable_reg, .volatile_reg = fsl_asrc_volatile_reg, .writeable_reg = fsl_asrc_writeable_reg, -- 1.9.1

2 4

[alsa-devel] [PATCH][resend] ASoC: simple-card / rsnd resend
by Kuninori Morimoto 28 Oct '14

28 Oct '14

Hi Mark These are resend patches which are still not applied, and, no response. The original patch-set had sent in... Subject: [PATCH 0/3] ASoC: simple-card: cleanup simple-card daifmt Date: Tue, 09 Sep 2014 21:36:43 -0700 (PDT) Subject: [PATCH 0/8] ASoC: rsnd: add Volume Ramp support Date: Tue, 21 Oct 2014 18:13:17 -0700 (PDT) Best regards --- Kuninori Morimoto

2 8

[alsa-devel] [PATCH v2] sound: pci: pcxhr: convert timeval to ktime_t
by Aya Mahfouz 28 Oct '14

28 Oct '14

This patch is concerned with migrating the time variables in the pcxhr module found in the sound driver. The changes are concerend with the y2038 problem where timeval will overflow in the year 2038. ktime_t was used instead of timeval to get the wall time. The difference is displayed now in nanoseconds instead of microseconds. Signed-off-by: Aya Mahfouz <mahfouz.saif.elyazal(a)gmail.com> Reviewed-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: changed variables names to avoid redeclaration conflicts. sound/pci/pcxhr/pcxhr.c | 10 ++++++---- sound/pci/pcxhr/pcxhr_core.c | 10 ++++++---- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index b854fc5..7c33c97 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -711,8 +711,9 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) int playback_mask = 0; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start_time, stop_time, diff_time; + + start_time = ktime_get(); #endif mutex_lock(&mgr->setup_mutex); @@ -823,9 +824,10 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) mutex_unlock(&mgr->setup_mutex); #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop_time = ktime_get(); + diff_time = ktime_sub(stop_time, start_time); dev_dbg(&mgr->pci->dev, "***TRIGGER START*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff_time)), err); #endif } diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index a584acb..181f772 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -910,8 +910,9 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int audio_mask; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start_time, stop_time, diff_time; + + start_time = ktime_get(); #endif audio_mask = (playback_mask | (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET)); @@ -960,9 +961,10 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, return err; } #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop_time = ktime_get(); + diff_time = ktime_sub(stop_time, start_time); dev_dbg(&mgr->pci->dev, "***SET PIPE STATE*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff_time)), err); #endif return 0; } -- 1.9.3

2 1

[alsa-devel] [PATCH] sound: pci: pcxhr: convert timeval to ktime_t
by Aya Mahfouz 28 Oct '14

28 Oct '14

This patch is concerned with migrating the time variables in the pcxhr module found in the sound driver. The changes are concerend with the y2038 problem where timeval will overflow in the year 2038. ktime_t was used instead of timeval to get the wall time. The difference is displayed now in nanoseconds instead of microseconds. Signed-off-by: Aya Mahfouz <mahfouz.saif.elyazal(a)gmail.com> Reviewed-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/pcxhr/pcxhr.c | 10 ++++++---- sound/pci/pcxhr/pcxhr_core.c | 10 ++++++---- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index b854fc5..3f4607c 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -711,8 +711,9 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) int playback_mask = 0; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start, stop, diff; + + start = ktime_get(); #endif mutex_lock(&mgr->setup_mutex); @@ -823,9 +824,10 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) mutex_unlock(&mgr->setup_mutex); #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop = ktime_get(); + diff = ktime_sub(stop, start); dev_dbg(&mgr->pci->dev, "***TRIGGER START*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff)), err); #endif } diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index a584acb..be3c003 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -910,8 +910,9 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int audio_mask; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start, stop, diff; + + start = ktime_get(); #endif audio_mask = (playback_mask | (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET)); @@ -960,9 +961,10 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, return err; } #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop = ktime_get(); + diff = ktime_sub(stop, start); dev_dbg(&mgr->pci->dev, "***SET PIPE STATE*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff)), err); #endif return 0; } -- 1.9.3

2 3

[alsa-devel] [PATCH] ALSA: pcm: Zero-clear reserved fields of PCM status ioctl in compat mode
by Takashi Iwai 28 Oct '14

28 Oct '14

In compat mode, we copy each field of snd_pcm_status struct but don't touch the reserved fields, and this leaves uninitialized values there. Meanwhile the native ioctl does zero-clear the whole structure, so we should follow the same rule in compat mode, too. Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_compat.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 102e8fd1d450..2d957ba63557 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -210,6 +210,8 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, if (err < 0) return err; + if (clear_user(src, sizeof(*src))) + return -EFAULT; if (put_user(status.state, &src->state) || compat_put_timespec(&status.trigger_tstamp, &src->trigger_tstamp) || compat_put_timespec(&status.tstamp, &src->tstamp) || -- 2.1.2

1 0

[alsa-devel] [PATCH] ASoC: es8328-i2c: Fix i2c_device_id name field in es8328_id
by Axel Lin 28 Oct '14

28 Oct '14

The convention for i2c_device_id name does not need to have company prefix. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/es8328-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c index aae410d..2d05b5d 100644 --- a/sound/soc/codecs/es8328-i2c.c +++ b/sound/soc/codecs/es8328-i2c.c @@ -19,7 +19,7 @@ #include "es8328.h" static const struct i2c_device_id es8328_id[] = { - { "everest,es8328", 0 }, + { "es8328", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, es8328_id); -- 1.9.1

2 1

[alsa-devel] What are subdevices?
by dE 28 Oct '14

28 Oct '14

Hi! I tried this question in the alsa-users mailing list but it doesn't appear to be much active. So I ask here. I'm facing problems with my system, so decided to learn ALSA to see what's the problem. My first question -- what's a device and a sub device? A device seems like the output streams the sound card supports. Like HDMI, DP, analog etc... So what's a subdevice? Is this related to multichannel? Does it designate physical ports on the card? Thanks.

2 5

[alsa-devel] [PATCH V3 1/2] ARM: pxa: spitz: register spitz-audio device
by Dmitry Eremin-Solenikov 28 Oct '14

28 Oct '14

Register spitz-audio device to be used by ASoC driver to bind ASoC platform driver. Currently old 'soc-audio' approach is used, which needs to be replaced with proper device. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- arch/arm/mach-pxa/spitz.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 840c3a4..962a7f3 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -924,6 +924,14 @@ static inline void spitz_i2c_init(void) {} #endif /****************************************************************************** + * Audio devices + ******************************************************************************/ +static inline void spitz_audio_init(void) +{ + platform_device_register_simple("spitz-audio", -1, NULL, 0); +} + +/****************************************************************************** * Machine init ******************************************************************************/ static void spitz_poweroff(void) @@ -970,6 +978,7 @@ static void __init spitz_init(void) spitz_nor_init(); spitz_nand_init(); spitz_i2c_init(); + spitz_audio_init(); } static void __init spitz_fixup(struct tag *tags, char **cmdline) -- 2.1.1

2 2

[alsa-devel] [PATCH] ASoC: Use generic control handlers for S8 control
by Lars-Peter Clausen 28 Oct '14

28 Oct '14

Commit f227b88f0fce ("ASoC: core: Add signed register volume control logic") added support for signed control to the generic volsw control handler. This makes it possible to use them for the S8 control as well, rather than having to use a custom control handler implementation. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 19 +++++------- sound/soc/soc-core.c | 87 ---------------------------------------------------- 2 files changed, 8 insertions(+), 98 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..ad47e96 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -36,6 +36,11 @@ {.reg = xreg, .rreg = xreg, .shift = shift_left, \ .rshift = shift_right, .max = xmax, .platform_max = xmax, \ .invert = xinvert, .autodisable = xautodisable}) +#define SOC_DOUBLE_S_VALUE(xreg, shift_left, shift_right, xmin, xmax, xsign_bit, xinvert, xautodisable) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = shift_left, \ + .rshift = shift_right, .min = xmin, .max = xmax, .platform_max = xmax, \ + .sign_bit = xsign_bit, .invert = xinvert, .autodisable = xautodisable}) #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \ SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ @@ -171,11 +176,9 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ - .put = snd_soc_put_volsw_s8, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .min = xmin, .max = xmax, \ - .platform_max = xmax} } + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ + .put = snd_soc_put_volsw, \ + .private_value = SOC_DOUBLE_S_VALUE(xreg, 0, 8, xmin, xmax, 7, 0, 0) } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ @@ -545,12 +548,6 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); -int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96ecdc3..47c378a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2721,93 +2721,6 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); /** - * snd_soc_info_volsw_s8 - signed mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); - -/** - * snd_soc_get_volsw_s8 - signed mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int val; - int min = mc->min; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = - ((signed char)(val & 0xff))-min; - ucontrol->value.integer.value[1] = - ((signed char)((val >> 8) & 0xff))-min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); - -/** - * snd_soc_put_volsw_sgn - signed mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - int min = mc->min; - unsigned int val; - - val = (ucontrol->value.integer.value[0]+min) & 0xff; - val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; - - return snd_soc_component_update_bits(component, reg, 0xffff, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); - -/** * snd_soc_info_volsw_range - single mixer info callback with range. * @kcontrol: mixer control * @uinfo: control element information -- 1.8.0

2 1

[alsa-devel] [PATCH v2 0/8] ASoC: dapm: Cleanups and speed-ups
by Lars-Peter Clausen 28 Oct '14

28 Oct '14

Only as few minor cleanup changes since v1. No change in functionallity. Original cover letter from v1: This series contains a few cleanups and speed-ups for the DAPM implementation. It starts with removing some dead or outdated code and optimizing some lesser used operations. The first larger change introduced by this series is the categorization of widgets into sinks, sources, supplies and normal widgets. This is all the information the main DAPM algorithm needs and using these categories makes the code simpler and faster and hopefully easier to understand. The second major change is to use more aggressive caching on the connected input and output paths of a widget. It takes advantage of the fact that typically only as subsection of the DAPM graph changes per operation. This allows to drastically reduce the number of neighbor and path checks for some DAPM operations. The change is most effective when the system is idle and when changing the graph routing for a widget with multiple inputs. This e.g. happens a lot when switching the audio profile. It still has quite good improvements for other operations as well and typically leads to at least a 50% reduction of path and neighbor checks. Some before and after DAPM stats. All testing was done with the ADAU1761. Before is asoc/for-next, After~1 is asoc/for-next + this series except the last patch, After is asoc/for-next + this series. Changing a single control on a 5 input mixer while the system is idle: Power Path Neighbour Before: 2 25 32 After~1: 2 12 30 After: 2 1 2 Cumulative numbers for switching the audio profile which changes 7 controls while the system is idle: Power Path Neighbour Before: 16 141 188 After~1: 16 80 170 After: 16 7 23 Cumulative numbers for switching the audio profile which changes 7 controls while playback is active: Power Path Neighbour Before: 52 207 297 After~1: 51 123 273 After: 51 29 109 Starting (or stopping) the playback stream: Power Path Neighbour Before: 35 80 125 After~1: 34 34 117 After: 34 17 69 - Lars Lars-Peter Clausen (8): ASoC: dapm: Only mark paths dirty when the connection status changed ASoC: dapm: Do not add un-muxed paths to MUX control ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets ASoC: dapm: Introduce toplevel widget categories ASoC: dapm: Add a flag to mark paths connected to supply widgets ASoC: dapm: Mark endpoints instead of IO widgets dirty during suspend/resume ASoC: dapm: Add a few supply widget sanity checks ASoC: dapm: Use more aggressive caching include/sound/soc-dapm.h | 8 +- sound/soc/soc-core.c | 8 +- sound/soc/soc-dapm.c | 576 ++++++++++++++++++++++++++++------------------- 3 files changed, 353 insertions(+), 239 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH 00/12] ASoC: dapm: Cleanups and speed-ups
by Lars-Peter Clausen 28 Oct '14

28 Oct '14

Hi, This series contains a few cleanups and speed-ups for the DAPM implementation. It starts with removing some dead or outdated code and optimizing some lesser used operations. The first larger change introduced by this series is the categorization of widgets into sinks, sources, supplies and normal widgets. This is all the information the main DAPM algorithm needs and using these categories makes the code simpler and faster and hopefully easier to understand. The second major change is to use more aggressive caching on the connected input and output paths of a widget. It takes advantage of the fact that typically only as subsection of the DAPM graph changes per operation. This allows to drastically reduce the number of neighbor and path checks for some DAPM operations. The change is most effective when the system is idle and when changing the graph routing for a widget with multiple inputs. This e.g. happens a lot when switching the audio profile. It still has quite good improvements for other operations as well and typically leads to at least a 50% reduction of path and neighbor checks. Some before and after DAPM stats. All testing was done with the ADAU1761. Before is asoc/for-next, After~1 is asoc/for-next + this series except the last patch, After is asoc/for-next + this series. Changing a single control on a 5 input mixer while the system is idle: Power Path Neighbour Before: 2 25 32 After~1: 2 12 30 After: 2 1 2 Cumulative numbers for switching the audio profile which changes 7 controls while the system is idle: Power Path Neighbour Before: 16 141 188 After~1: 16 80 170 After: 16 7 23 Cumulative numbers for switching the audio profile which changes 7 controls while playback is active: Power Path Neighbour Before: 52 207 297 After~1: 51 123 273 After: 51 29 109 Starting (or stopping) the playback stream: Power Path Neighbour Before: 35 80 125 After~1: 34 34 117 After: 34 17 69 - Lars Lars-Peter Clausen (12): ASoC: dapm: Reduce number of checked paths in dapm_widget_in_card_paths() ASoC: dapm: Remove always true path source/sink checks ASoC: dapm: Only mark paths dirty when the connection status changed ASoC: dapm: Do not add un-muxed paths to MUX control ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets ASoC: dapm: Remove special DAI widget power check functions ASoC: dapm: Remove path 'walked' flag ASoC: dapm: Introduce toplevel widget categories ASoC: dapm: Add a flag to mark paths connected to supply widgets ASoC: dapm: Mark endpoints instead of IO widgets dirty during suspend/resume ASoC: dapm: Add a few supply widget sanity checks ASoC: dapm: Use more aggressive caching include/sound/soc-dapm.h | 9 +- sound/soc/soc-core.c | 8 +- sound/soc/soc-dapm.c | 755 +++++++++++++++++++++++++---------------------- 3 files changed, 409 insertions(+), 363 deletions(-) -- 1.8.0

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[alsa-devel] 2014 ALSA micro-conference meeting notes - 10/14 Dussedorf
by Patrick Lai 27 Oct '14

27 Oct '14

Attendees: Takashi Iwai, Mark Brown, Vinod Koul, Patrick Lai, Lars-Peter Clausen, Eric Laurent, David Henningsson, Tanu Kaskinen, Arun Raghavan, Paul Handrigan, Daniel Mack, Dylan Reid, Charles Keepax, Nariman Poushin, Peter Meerwald, Alexander Patrakov, Hui Wang, Tim Cussins, Nick Stoughton Topic: ASoC componentization Motivation - hardware is becoming more complex as CPU and CODEC functionality are overlapping. Codec now has DSP. DSP on CPU need DAPM and IO. Current ASoC APIs are based on clearly distinguished component type, platform, CPU, CODEC. It makes difficult to support complex configuration such as CPU -> CODEC -> CODEC or DSP with DAC, ADC, SRC Points discussed - Componentization is coming on V 3.18. - Lars-Peter is cleaning up suspend/resume logic in ASoC framework as conflicting with individual driver suspend/resume. Goal is to move toward bus suspend/resume - Grand scheme is to treat hardware module as component instead of clearly distinguish component type(platform/cpu/codec). Russel King introduced component framework recently - one master with many slave components. ASoC could adopt component framework. Sound card would be the master while individual drivers register slave components. Moving to use component framework would help unbinding of drivers cleanly as well. - Discuss possible phasing out/refactor DPCM, DAI-LINK and add support to DAPM for PCM configuration propagation. Topic: Dynamic FW framework Motivation: It is very common for SoC/CODEC vendors putting DSP onto their hardware. Topology/functionality can change based on DSP image being loaded. Dynamic FW framework is designed to address the need to run-time create DSP topology graph without hard coding into the driver - For DSP with large memory case, swapping DSP image at run-time might not be a concern so firmware download can happen once after boot up but CODEC DSP tend to be memory constraint. Run-time swapping requiring deregister of existing controls and register with new one. ALSA framework does not have good support for this requirement currently. - DFW tool allows each solution vendor to develop its own plug-in and can convert static description to DFW binary format. Through this way, solution vendor can specify its own data and associate with ASoC component such as DAPM widget. Topic: Mapping channels through DAI links in ASoC Motivation: Currently, it is difficult to track how each channel get propagated from one component to another i.e 8 channel streams from CPU to CODECs with stereo channels in TDM interface - Consensus that there is value to track the channel mapping - Possible solution is to create widget per channel of PCM object Topic: ALSA test - Discussed test ALSA core framework on virtual platform and patch submission should go through testing. For embedded system, the benefit may not be significant as bug tend to show up during use case transition - David demonstrated HDA CODEC test. Perhaps, this test can be leveraged for ALSA core test framework - There is no ALSA compliance test right now. There is small tool called alsa-time-test which could potentially be part of ALSA test suite. Another way to validate conformance is to run PulseAudio as PulseAudio exercises a lot of alsa-lib API - Vinod suggested a test framework Liam has been developing, to compare the sign wave form with the captured one. Contribution is welcome and can be merged to any public repo. In addition, for *any* other test programs or metrics, contribution is welcomed. Topic: ALSA documentation - For mixer control related query, Documentation/sound/alsa/ControlNames.txt in kernel tree is a good resource to read - Each kernel function can have a Javadoc style comment. This can be formated through DocBook directory in kernel tree as PDF or HTML (or anything else). So, from now on, the javadoc style comment is mandatory for each patch including exported symbols. - the root DocBook template file is missing for ASoC. Presentation given by Mark in ELC few years back could be a good supplement for someone to get a quick intro to develop drivers to plug into ASoC framework Topic: ALSA control names - Even though in embedded world, SoC architecture from different vendors are not so consistent enough to come up with common name. There is still value to come up with standard name for different types of controls such as Volume, Ramp up/down time (in unit of DB per ms or sample count) - Media controller discussion came up again as it can be solution to help user-space recognize mixer control for the topology - General consensus is that each vendor should have control definition document to describe what each control do - As part of user case transition, user-space often execute mute as well. Instead of ASoC framework handles digital mute as part of startup/shutdown sequence, let user-space have the control through mixer control command. - Suggest to have volume, mute control per PCM object which takes the same approach as channel MAP API by exposing mixer control with name "<pcm device id> volume" Topic: ALSA code removal - Can OSS be removed? Anyone is still using OSS? Perhaps, OSS emulation can be done in user-space - Add warning to notify people to contact usage of APIs to prevent deprecation - Discuss removal of old driver but does not look safe to remove - Suggestion to remove ADPCM support in ALSA core framework - Removal in alsa-library: aserver, shm, and share plugins were discussed. The decision seems to be putting them under a ./configure switch now and remove later Topic: DSP effect library/more DSP support - For DSP effect, Intel goes through mixer control approach which controls are defined in the dynamic firmware. It is not clear how Android effect would be associate stream to corresponding mixer controls for the DSP effect - Discuss couple issues when dealing with DSP architecture. These problems surfaces on PulseAudio 1. DSP DMA data as soon as it can put into its internal memory. Hence, hw_ptr does not reflect actual consumption point in DSP. This causes rewinding logic not function properly. It would be ideal if ALSA could allow driver expose granularity of hardware pointer 2. DSP continuous latency can vary. ALSA should have mechanism to allow driver expose latency information to user-space Topic: consolidate jack report mechanism Currently, there are K-control, input event, extcon mechanism to report headset jack events. Intel would like to see k-control as final solution in order to align with DSP notification through k-control Discussed to consolidate into one mechanism. One proposal from Mark is making a common in-kernel framework which all driver would use this jack framework, the framework would in turn expose this jack through different user-space interfaces depending on kernel configuration Topic: Convergence of single configuration file format UCM vs mixer_path Single format would make vendors who supports both UCM and mixer_path systems life easier. PulseAudio extract use cases list from UCM while mixer_path does not have such support yet. Action item: Arun to look into the convergence effort Topic: scheduled trigger start of PCM - Nick to submit snd_pcm_start_at() patch. The change is software based solution but margin of error is acceptable for the use case in mind - Brought up the use case which time stamp is sent along with every buffer so DSP knows to drop when the buffer is given pass the deadline. Topic: ACPI Intel is moving system-level device configuration file toward ACPI. Device specific data(_DSD) section defined in ACPI 5.1 specification allows device vendor to pass device specific data. (i.e CODEC device data). Topic: synchronization of HDMI/USB hot-plug and configuration events - Instead of waiting for USB hotplug event, Android should wait for udev event to notify new USB sound card is created - Discuss making HDMI a separate sound card. However, HDMI could simply be another routing path in a sound card for embedded architecture. - Suggest to return error when reading EDID info is HDMI is not ready Topic: Sharing hardware component between ALSA framework and other drivers - When hot word is detected, go through DAPM to enable mic path. Topic: Rewindng in ALSA PCM plugins 1. Discussion of the method to calculate the amount of samples that should not be rewound over after the hardware pointer. Result: the audience demonstrated general understanding of the issue. Lars suggested having two hardware pointers, because there are two boundaries that are unsafe to cross when overwriting samples: one from the left (normal write path), and one from the right (rewind path). No action items. 2. Discussion of the criteria when to enable timer-based scheduling. Suggestion: do not enumerate badness [*] when deciding whether to enable tsched in PulseAudio, because the bad things tend to go unreported by the drivers and people treat these as PulseAudio bugs. Move to a whitelist-based approach. The suggestion was rejected. [*] BLOCK_TRANSFER (unimplemented), BATCH (implemented), huge FIFO (unimplemented), completely unrewindable (unimplemented, but a workaround for ioplug exists), what else? 3. Suggestion to revive the snd_pcm_hw_params_can_rewind() API function. Approved by Takashi. 4. Discussion whether the snd_pcm_rewindable() function is useful at all. The uselessness comes from the need to call snd_pcm_avail_update() beforehand, and the position of Clemens Ladisch that the result of snd_pcm_rewindable() must not take pointer granularity and other reasons why non-rewindable samples exist into account. That way, the caller can just deduce the result from the snd_pcm_avail_update() return value. David Henningsson does not like this situation. Mark Brown suggests reporting the DMA transfer size, with the possibility for architectures to override the returned value with something larger such as the cache line size. 5. Status report on rewind support in PCM plugins. Summary: at least 10 plugins out of 26 have known problems and should have rewind support either fixed (which is not always possible) or disabled. Thanks Patrick -- Patrick Lai Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH 0/3] ASoC: intel: add support for future acpi platforms
by Vinod Koul 27 Oct '14

27 Oct '14

We have some SoCs which are ACPI based. Current driver supports Merrifield type PCI SoC. SO make some minor changes required for ACPI support in subsequent series Subhransu S. Prusty (3): ASoC: Intel: mrfld: Replace pci_id with unique device id ASoC: Intel: mrfld - Define ipc_info structure ASoC: Intel: mrfld: Define sst_res_info for acpi arch/x86/include/asm/platform_sst_audio.h | 28 ++++++++++++++++++++++++++++ sound/soc/intel/sst/sst.c | 10 +++++----- sound/soc/intel/sst/sst.h | 5 +++-- sound/soc/intel/sst/sst_pvt.c | 4 ++-- 4 files changed, 38 insertions(+), 9 deletions(-)

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[alsa-devel] [PATCH] ALSA: hda - Pass printf argument directly to request_module()
by Takashi Iwai 27 Oct '14

27 Oct '14

request_module() handles the printf style arguments, so we don't have to render strings in the caller side. Not only it reduces the unnecessary temporary string buffer, it's even safer from the security POV. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 259fbeaa37bd..0025bf4c2f44 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -966,14 +966,12 @@ find_codec_preset(struct hda_codec *codec) mutex_unlock(&preset_mutex); if (mod_requested < HDA_MODREQ_MAX_COUNT) { - char name[32]; if (!mod_requested) - snprintf(name, sizeof(name), "snd-hda-codec-id:%08x", - codec->vendor_id); + request_module("snd-hda-codec-id:%08x", + codec->vendor_id); else - snprintf(name, sizeof(name), "snd-hda-codec-id:%04x*", - (codec->vendor_id >> 16) & 0xffff); - request_module(name); + request_module("snd-hda-codec-id:%04x*", + (codec->vendor_id >> 16) & 0xffff); mod_requested++; goto again; } -- 2.1.2

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[alsa-devel] Uninitialized id returned by saffirepro_both_clk_src_get.
by Christian Vogel 27 Oct '14

27 Oct '14

Hi, there's a possibility to get a Oops caused by an uninitialized value in snd_bebob_stream_check_internal_clock for a SaffirePro running on the internal clock. [ 88.100531] BUG: unable to handle kernel paging request at 8a3c85fc [ 88.103808] IP: [<e8553aa0>] snd_bebob_stream_check_internal_clock+0x66/0x11e [snd_bebob] ...which is dereferencing of clk_spec->labels[id] in... sound/firewire/bebob/bebob_stream.c : /* 1.The device has its own operation to switch source of clock */ if (clk_spec) { err = clk_spec->get(bebob, &id); if (err < 0) dev_err(&bebob->unit->device, "fail to get clock source: %d\n", err); --> else if (strncmp(clk_spec->labels[id], SND_BEBOB_CLOCK_INTERNAL, strlen(SND_BEBOB_CLOCK_INTERNAL)) == 0) *internal = true; goto end; } id is uninitialized, and will not be set by clk_spec->get (which is saffirepro_both_clk_src_get(), even if it returns ok(0). Attached patch tries to clean up the logic in saffirepro_both_clk_src_get() and also adds a safety check to snd_bebob_stream_check_internal_clock(). Thanks for Takashi Sakamoto to whom I sent the patch initially and who suggested some cleanup to my code, reviewed the patch and suggested I send it to alsa-dev. Chris

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[alsa-devel] [PATCH] pcm: Fix snd_pcm_avail_update description
by David Henningsson 27 Oct '14

27 Oct '14

While reading alsa-lib code, I discovered that snd_pcm_avail_update does require a user-kernel context switch: snd_pcm_avail_update() -> snd_pcm_hw_avail_update() -> sync_ptr() -> sync_ptr1() -> ioctl() The documentation says it does not, so let's fix the documentation. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 4a7be6c..e57e362 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -403,10 +403,12 @@ The function #snd_pcm_avail_update() updates the current available count of samples for writing (playback) or filled samples for reading (capture). This call is mandatory for updating actual r/w pointer. Using standalone, it is a light method to obtain current stream position, -because it does not require the user <-> kernel context switch, but the value -is less accurate, because ring buffer pointers are updated in kernel drivers +because while it does require a user <-> kernel context switch, the value is +just copied from the kernel, and not updated by synchronously asking the hardware +about its current position. The ring buffer pointers are updated in kernel drivers only when an interrupt occurs. If you want to get accurate stream state, -use functions #snd_pcm_avail(), #snd_pcm_delay() or #snd_pcm_avail_delay(). +use functions #snd_pcm_avail(), #snd_pcm_delay(), #snd_pcm_avail_delay() or +#snd_pcm_status(). </p> <p> The function #snd_pcm_avail() reads the current hardware pointer -- 1.9.1

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[alsa-devel] [GIT PULL] ASoC fixes for v3.18
by Mark Brown 27 Oct '14

27 Oct '14

The following changes since commit cac7f2429872d3733dc3f9915857b1691da2eb2f: Linux 3.18-rc2 (2014-10-26 16:48:41 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.18-rc2 for you to fetch changes up to c0d018bd5b1aabad59dffdec568e189359d93a14: Merge remote-tracking branches 'asoc/fix/adau1761', 'asoc/fix/fsl', 'asoc/fix/intel', 'asoc/fix/s6000' and 'asoc/fix/sgtl5000' into asoc-linus (2014-10-27 11:17:41 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.18 A few small driver fixes for v3.18 plus the removal of the s6000 support since the relevant chip is no longer supported in mainline. ---------------------------------------------------------------- Daniel Glöckner (1): ASoC: s6000: remove driver Daniel Mack (1): ASoC: fsl: use strncpy() to prevent copying of over-long names Fabio Estevam (1): ASoC: sgtl500: Document the required supplies Lars-Peter Clausen (1): ASoC: adau1761: Fix input PGA volume Liam Girdwood (1): ASoC: Intel: HSW/BDW only support S16 and S24 formats. Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1761', 'asoc/fix/fsl', 'asoc/fix/intel', 'asoc/fix/s6000' and 'asoc/fix/sgtl5000' into asoc-linus .../devicetree/bindings/sound/sgtl5000.txt | 10 + sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/codecs/adau1761.c | 4 + sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 4 +- sound/soc/s6000/Kconfig | 26 - sound/soc/s6000/Makefile | 11 - sound/soc/s6000/s6000-i2s.c | 617 --------------------- sound/soc/s6000/s6000-i2s.h | 23 - sound/soc/s6000/s6000-pcm.c | 521 ----------------- sound/soc/s6000/s6000-pcm.h | 33 -- sound/soc/s6000/s6105-ipcam.c | 221 -------- 14 files changed, 17 insertions(+), 1459 deletions(-) delete mode 100644 sound/soc/s6000/Kconfig delete mode 100644 sound/soc/s6000/Makefile delete mode 100644 sound/soc/s6000/s6000-i2s.c delete mode 100644 sound/soc/s6000/s6000-i2s.h delete mode 100644 sound/soc/s6000/s6000-pcm.c delete mode 100644 sound/soc/s6000/s6000-pcm.h delete mode 100644 sound/soc/s6000/s6105-ipcam.c

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Re: [alsa-devel] [PATCH v2 8/8] ASoC: dapm: Use more aggressive caching
by noman pouigt 27 Oct '14

27 Oct '14

>E.g. imagine the following example: > > A --> B ----. > v > M --> N --> Z <-- S <-- R > | > v > X> > >Widget Z has multiple input paths, if any change was made that cause Z to be >marked as dirty the power state of Z has to be re-computed. This requires to >know the number of inputs and outputs of Z, which requires to know the >number of inputs and outputs of all widgets on all paths from or to Z. >Previously this meant re-computing all inputs and outputs of all the path >going into or out of Z. With this patch in place only paths that actually >have changed need to be re-computed. so suppose "B" was the widget which caused "Z" to be dirtied then with your patch only you will calculate A->B->N->Z ? Is my understanding right? A->B->C | v G->P v | D->E->F What will happen in the above diagram if P gets dirtied?

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[alsa-devel] [PATCH v2] ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672
by mengdong.lin＠intel.com 27 Oct '14

27 Oct '14

From: Mengdong Lin <mengdong.lin(a)intel.com> Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5672 codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af88..7479ce0 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -16,6 +16,9 @@ config SND_SST_MFLD_PLATFORM config SND_SST_IPC tristate +config SND_SST_MACHINE + tristate + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI @@ -76,3 +79,20 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5672_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5670 + select SND_SST_MFLD_PLATFORM + select SND_SOC_INTEL_SST + select SND_SST_IPC + select SND_SST_MACHINE + default n + + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y if you have such a device + If unsure select "N". + diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9ab43be..4069d3f 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -31,6 +31,7 @@ obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SST_MACHINE) += board/ # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/board/Makefile b/sound/soc/intel/board/Makefile new file mode 100644 index 0000000..9ecc227 --- /dev/null +++ b/sound/soc/intel/board/Makefile @@ -0,0 +1,2 @@ +snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o diff --git a/sound/soc/intel/board/cht_bsw_rt5672.c b/sound/soc/intel/board/cht_bsw_rt5672.c new file mode 100644 index 0000000..789d1a0 --- /dev/null +++ b/sound/soc/intel/board/cht_bsw_rt5672.c @@ -0,0 +1,246 @@ +/* + * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5672 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> + * Mengdong Lin <mengdong.lin(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/rt5670.h" +#include "../sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (strncmp(codec_dai->name, "rt5670-aif1", 11)) + return 0; + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5670-aif1", + .codec_name = "i2c-10EC5670:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailcraudio", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = snd_soc_register_card(&snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5672", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5672"); -- 1.9.1

1 0

[alsa-devel] Customizing simple audio card?
by Max Filippov 27 Oct '14

27 Oct '14

Hello, I'm developing ALSA driver for sound hardware on Xtensa FPGA boards. The hardware is very simple, we have AIC23 codec connected to I2S bus and I2S bus master that pumps data pushed by CPU into its FIFO. Both are connected to the same clock synthesizer output. There are no other devices that need to be supported now. Due to its simplicity the setup could happily use the simple sound card driver. But to support both 44100 and 48000 frequency ranges I need to change master clock, setting it either to 11289600 or to 12288000 Hz. I can do it in the I2S driver, but I see no way to inform audio codec of that change. I used the following hack to debug everything with the simple audio card: diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index cef7776..abef5ed 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -43,6 +44,22 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, SND_SOC_CLOCK_IN); } + switch (params_rate(params)) { + case 48000: + case 32000: + case 16000: + case 8000: + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, + SND_SOC_CLOCK_IN); + break; + case 44100: + case 22050: + case 11025: + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, + SND_SOC_CLOCK_IN); + break; + + } return ret; } but that's of course not a solution. So my question is is there any way I can subclass or inherit from the simple audio card driver to add such functionality to it? -- Thanks. -- Max

2 2

[alsa-devel] [PATCH 1/7] ASoC: Intel: Make ADSP memory block allocation more generic
by Liam Girdwood 27 Oct '14

27 Oct '14

Current block allocation is tied to block type and requestor type. Make the allocation more generic by removing the struct module parameter and adding a generic block allocator structure. Also pass in the list that the blocks have to be added too in order to remove dependence on block requestor type. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-dsp.c | 24 ++- sound/soc/intel/sst-dsp-priv.h | 83 +++++----- sound/soc/intel/sst-firmware.c | 317 +++++++++++++++++++++++++------------ sound/soc/intel/sst-haswell-dsp.c | 41 ++--- 4 files changed, 288 insertions(+), 177 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c index fc58876..5a9e567 100644 --- a/sound/soc/intel/sst-baytrail-dsp.c +++ b/sound/soc/intel/sst-baytrail-dsp.c @@ -67,17 +67,12 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; int count; memset(&template, 0, sizeof(template)); template.id = module->type; template.entry = module->entry_point; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -94,19 +89,19 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_BYT_IRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_BYT_DRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.dram_offset; - block_data.type = SST_MEM_DRAM; + mod->type = SST_MEM_DRAM; break; case SST_BYT_CACHE: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe); - block_data.type = SST_MEM_CACHE; + mod->type = SST_MEM_CACHE; break; default: dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n", @@ -114,11 +109,10 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); - sst_module_insert_fixed_block(mod, &block_data); + sst_module_alloc_blocks(mod); block = (void *)block + sizeof(*block) + block->size; } diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index ffb308b..f58a049 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -84,15 +84,6 @@ struct sst_mailbox { }; /* - * Audio DSP Firmware data types. - */ -enum sst_data_type { - SST_DATA_M = 0, /* module block data */ - SST_DATA_P = 1, /* peristant data (text, data) */ - SST_DATA_S = 2, /* scratch data (usually buffers) */ -}; - -/* * Audio DSP memory block types. */ enum sst_mem_type { @@ -125,23 +116,6 @@ struct sst_fw { }; /* - * Audio DSP Generic Module data. - * - * This is used to dsecribe any sections of persistent (text and data) and - * scratch (buffers) of module data in ADSP memory space. - */ -struct sst_module_data { - - enum sst_mem_type type; /* destination memory type */ - enum sst_data_type data_type; /* type of module data */ - - u32 size; /* size in bytes */ - int32_t offset; /* offset in FW file */ - u32 data_offset; /* offset in ADSP memory space */ - void *data; /* module data */ -}; - -/* * Audio DSP Generic Module Template. * * Used to define and register a new FW module. This data is extracted from @@ -150,15 +124,42 @@ struct sst_module_data { struct sst_module_template { u32 id; u32 entry; /* entry point */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; + u32 persistent_size; +}; + +/* + * Block Allocator - Used to allocate blocks of DSP memory. + */ +struct sst_block_allocator { + u32 id; + u32 offset; + int size; + enum sst_mem_type type; +}; + +/* + * Runtime Module Instance - A module object can be instanciated multiple + * times within the DSP FW. + */ +struct sst_module_runtime { + struct sst_dsp *dsp; + int id; + struct sst_module *module; /* parent module we belong too */ + + u32 persistent_offset; /* private memory size */ + void *private; + + struct list_head list; + struct list_head block_list; /* list of blocks used */ }; /* * Audio DSP Generic Module. * * Each Firmware file can consist of 1..N modules. A module can span multiple - * ADSP memory blocks. The simplest FW will be a file with 1 module. + * ADSP memory blocks. The simplest FW will be a file with 1 module. A module + * can be instanciated multiple times in the DSP. */ struct sst_module { struct sst_dsp *dsp; @@ -167,10 +168,13 @@ struct sst_module { /* module configuration */ u32 id; u32 entry; /* module entry point */ - u32 offset; /* module offset in firmware file */ + s32 offset; /* module offset in firmware file */ u32 size; /* module size */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; /* global scratch memory required */ + u32 persistent_size; /* private memory required */ + enum sst_mem_type type; /* destination memory type */ + u32 data_offset; /* offset in ADSP memory space */ + void *data; /* module data */ /* runtime */ u32 usage_count; /* can be unloaded if count == 0 */ @@ -180,6 +184,7 @@ struct sst_module { struct list_head block_list; /* Module list of blocks in use */ struct list_head list; /* DSP list of modules */ struct list_head list_fw; /* FW list of modules */ + struct list_head runtime_list; /* list of runtime module objects*/ }; /* @@ -208,7 +213,6 @@ struct sst_mem_block { struct sst_block_ops *ops; /* block operations, if any */ /* block status */ - enum sst_data_type data_type; /* data type held in this block */ u32 bytes_used; /* bytes in use by modules */ void *private; /* generic core does not touch this */ int users; /* number of modules using this block */ @@ -290,19 +294,20 @@ void sst_fw_unload(struct sst_fw *sst_fw); /* Create/Free firmware modules */ struct sst_module *sst_module_new(struct sst_fw *sst_fw, struct sst_module_template *template, void *private); -void sst_module_free(struct sst_module *sst_module); -int sst_module_insert(struct sst_module *sst_module); -int sst_module_remove(struct sst_module *sst_module); -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data); +void sst_module_free(struct sst_module *module); struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); /* allocate/free pesistent/scratch memory regions managed by drv */ struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); void sst_mem_block_free_scratch(struct sst_dsp *dsp, struct sst_module *scratch); -int sst_block_module_remove(struct sst_module *module); +int sst_module_alloc_blocks(struct sst_module *module); +int sst_module_free_blocks(struct sst_module *module); +/* generic block allocation */ +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list); +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list); /* Register the DSPs memory blocks - would be nice to read from ACPI */ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 3bb43da..f2f1d5f 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,7 +30,7 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void block_module_remove(struct sst_module *module); +#define SST_HSW_BLOCK_ANY 0xffffffff static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { @@ -41,6 +41,65 @@ static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) memcpy_toio(dest + i, src + i, 4); } + +/* remove module from memory - callers hold locks */ +static void block_list_remove(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block, *tmp; + int err; + + /* disable each block */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->disable) { + err = block->ops->disable(block); + if (err < 0) + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + } + } + + /* mark each block as free */ + list_for_each_entry_safe(block, tmp, block_list, module_list) { + list_del(&block->module_list); + list_move(&block->list, &dsp->free_block_list); + dev_dbg(dsp->dev, "block freed %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } +} + +/* prepare the memory block to receive data from host - callers hold locks */ +static int block_list_prepare(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block; + int ret = 0; + + /* enable each block so that's it'e ready for data */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->enable) { + ret = block->ops->enable(block); + if (ret < 0) { + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + goto err; + } + } + } + return ret; + +err: + list_for_each_entry(block, block_list, module_list) { + if (block->ops && block->ops->disable) + block->ops->disable(block); + } + return ret; +} + /* create new generic firmware object */ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, const struct firmware *fw, void *private) @@ -202,73 +261,122 @@ void sst_module_free(struct sst_module *sst_module) } EXPORT_SYMBOL_GPL(sst_module_free); -static struct sst_mem_block *find_block(struct sst_dsp *dsp, int type, - u32 offset) +struct sst_module_runtime *sst_module_runtime_new(struct sst_module *module, + int id, void *private) +{ + struct sst_dsp *dsp = module->dsp; + struct sst_module_runtime *runtime; + + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (runtime == NULL) + return NULL; + + runtime->id = id; + runtime->dsp = dsp; + runtime->module = module; + INIT_LIST_HEAD(&runtime->block_list); + + mutex_lock(&dsp->mutex); + list_add(&runtime->list, &module->runtime_list); + mutex_unlock(&dsp->mutex); + + return runtime; +} +EXPORT_SYMBOL_GPL(sst_module_runtime_new); + +void sst_module_runtime_free(struct sst_module_runtime *runtime) +{ + struct sst_dsp *dsp = runtime->dsp; + + mutex_lock(&dsp->mutex); + list_del(&runtime->list); + mutex_unlock(&dsp->mutex); + + kfree(runtime); +} +EXPORT_SYMBOL_GPL(sst_module_runtime_free); + +static struct sst_mem_block *find_block(struct sst_dsp *dsp, + struct sst_block_allocator *ba) { struct sst_mem_block *block; list_for_each_entry(block, &dsp->free_block_list, list) { - if (block->type == type && block->offset == offset) + if (block->type == ba->type && block->offset == ba->offset) return block; } return NULL; } -static int block_alloc_contiguous(struct sst_module *module, - struct sst_module_data *data, u32 offset, int size) +/* Block allocator must be on block boundary */ +static int block_alloc_contiguous(struct sst_dsp *dsp, + struct sst_block_allocator *ba, struct list_head *block_list) { struct list_head tmp = LIST_HEAD_INIT(tmp); - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block; + u32 block_start = SST_HSW_BLOCK_ANY; + int size = ba->size, offset = ba->offset; + + while (ba->size > 0) { - while (size > 0) { - block = find_block(dsp, data->type, offset); + block = find_block(dsp, ba); if (!block) { list_splice(&tmp, &dsp->free_block_list); + + ba->size = size; + ba->offset = offset; return -ENOMEM; } list_move_tail(&block->list, &tmp); - offset += block->size; - size -= block->size; + ba->offset += block->size; + ba->size -= block->size; } + ba->size = size; + ba->offset = offset; + + list_for_each_entry(block, &tmp, list) { + + if (block->offset < block_start) + block_start = block->offset; - list_for_each_entry(block, &tmp, list) - list_add(&block->module_list, &module->block_list); + list_add(&block->module_list, block_list); + + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } list_splice(&tmp, &dsp->used_block_list); return 0; } -/* allocate free DSP blocks for module data - callers hold locks */ -static int block_alloc(struct sst_module *module, - struct sst_module_data *data) +/* allocate first free DSP blocks for data - callers hold locks */ +static int block_alloc(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; int ret = 0; - if (data->size == 0) + if (ba->size == 0) return 0; /* find first free whole blocks that can hold module */ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; - if (data->size > block->size) + if (ba->size > block->size) continue; - data->offset = block->offset; - block->data_type = data->data_type; - block->bytes_used = data->size % block->size; - list_add(&block->module_list, &module->block_list); + ba->offset = block->offset; + block->bytes_used = ba->size % block->size; + list_add(&block->module_list, block_list); list_move(&block->list, &dsp->used_block_list); - dev_dbg(dsp->dev, " *module %d added block %d:%d\n", - module->id, block->type, block->index); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } @@ -276,15 +384,19 @@ static int block_alloc(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; /* do we span > 1 blocks */ - if (data->size > block->size) { - ret = block_alloc_contiguous(module, data, - block->offset, data->size); + if (ba->size > block->size) { + + /* align ba to block boundary */ + ba->offset = block->offset; + + ret = block_alloc_contiguous(dsp, ba, block_list); if (ret == 0) return ret; + } } @@ -292,93 +404,74 @@ static int block_alloc(struct sst_module *module, return -ENOMEM; } -/* remove module from memory - callers hold locks */ -static void block_module_remove(struct sst_module *module) +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_mem_block *block, *tmp; - struct sst_dsp *dsp = module->dsp; - int err; + int ret; - /* disable each block */ - list_for_each_entry(block, &module->block_list, module_list) { + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba->size, ba->offset, ba->type); - if (block->ops && block->ops->disable) { - err = block->ops->disable(block); - if (err < 0) - dev_err(dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - } - } + mutex_lock(&dsp->mutex); - /* mark each block as free */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { - list_del(&block->module_list); - list_move(&block->list, &dsp->free_block_list); + ret = block_alloc(dsp, ba, block_list); + if (ret < 0) { + dev_err(dsp->dev, "error: can't alloc blocks %d\n", ret); + goto out; } -} - -/* prepare the memory block to receive data from host - callers hold locks */ -static int block_module_prepare(struct sst_module *module) -{ - struct sst_mem_block *block; - int ret = 0; - /* enable each block so that's it'e ready for module P/S data */ - list_for_each_entry(block, &module->block_list, module_list) { + /* prepare DSP blocks for module usage */ + ret = block_list_prepare(dsp, block_list); + if (ret < 0) + dev_err(dsp->dev, "error: prepare failed\n"); - if (block->ops && block->ops->enable) { - ret = block->ops->enable(block); - if (ret < 0) { - dev_err(module->dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - goto err; - } - } - } +out: + mutex_unlock(&dsp->mutex); return ret; +} +EXPORT_SYMBOL_GPL(sst_alloc_blocks); -err: - list_for_each_entry(block, &module->block_list, module_list) { - if (block->ops && block->ops->disable) - block->ops->disable(block); - } - return ret; +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list) +{ + mutex_lock(&dsp->mutex); + block_list_remove(dsp, block_list); + mutex_unlock(&dsp->mutex); + return 0; } +EXPORT_SYMBOL_GPL(sst_free_blocks); /* allocate memory blocks for static module addresses - callers hold locks */ -static int block_alloc_fixed(struct sst_module *module, - struct sst_module_data *data) +static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; - u32 end = data->offset + data->size, block_end; + u32 end = ba->offset + ba->size, block_end; int err; /* only IRAM/DRAM blocks are managed */ - if (data->type != SST_MEM_IRAM && data->type != SST_MEM_DRAM) + if (ba->type != SST_MEM_IRAM && ba->type != SST_MEM_DRAM) return 0; /* are blocks already attached to this module */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { + list_for_each_entry_safe(block, tmp, block_list, module_list) { - /* force compacting mem blocks of the same data_type */ - if (block->data_type != data->data_type) + /* ignore blocks with wrong type */ + if (block->type != ba->type) continue; block_end = block->offset + block->size; /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) + if (ba->offset >= block->offset && end <= block_end) return 0; /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { - err = block_alloc_contiguous(module, data, - block->offset + block->size, - data->size - block->size); + /* align ba to block boundary */ + ba->size -= block_end - ba->offset; + ba->offset = block_end; + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; @@ -391,53 +484,67 @@ static int block_alloc_fixed(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { block_end = block->offset + block->size; + /* ignore blocks with wrong type */ + if (block->type != ba->type) + continue; + /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) { + if (ba->offset >= block->offset && end <= block_end) { /* add block */ - block->data_type = data->data_type; list_move(&block->list, &dsp->used_block_list); - list_add(&block->module_list, &module->block_list); + list_add(&block->module_list, block_list); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { + + /* align ba to block boundary */ + ba->offset = block->offset; - err = block_alloc_contiguous(module, data, - block->offset, data->size); + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; return 0; } - } return -ENOMEM; } /* Load fixed module data into DSP memory blocks */ -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data) +int sst_module_alloc_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; + struct sst_fw *sst_fw = module->sst_fw; + struct sst_block_allocator ba; int ret; + ba.size = module->size; + ba.type = module->type; + ba.offset = module->offset; + + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba.size, ba.offset, ba.type); + mutex_lock(&dsp->mutex); /* alloc blocks that includes this section */ - ret = block_alloc_fixed(module, data); + ret = block_alloc_fixed(dsp, &ba, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: no free blocks for section at offset 0x%x size 0x%x\n", - data->offset, data->size); + module->offset, module->size); mutex_unlock(&dsp->mutex); return -ENOMEM; } /* prepare DSP blocks for module copy */ - ret = block_module_prepare(module); + ret = block_list_prepare(dsp, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: fw module prepare failed\n"); goto err; @@ -450,19 +557,23 @@ int sst_module_insert_fixed_block(struct sst_module *module, return ret; err: - block_module_remove(module); + block_list_remove(dsp, &module->block_list); mutex_unlock(&dsp->mutex); return ret; } -EXPORT_SYMBOL_GPL(sst_module_insert_fixed_block); +EXPORT_SYMBOL_GPL(sst_module_alloc_blocks); /* Unload entire module from DSP memory */ -int sst_block_module_remove(struct sst_module *module) +int sst_module_free_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; mutex_lock(&dsp->mutex); - block_module_remove(module); + block_list_remove(dsp, &module->block_list); + mutex_unlock(&dsp->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(sst_module_free_blocks); mutex_unlock(&dsp->mutex); return 0; } diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 4b6c163..22cc697 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -86,9 +86,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; - int count; + int count, ret; void __iomem *ram; /* TODO: allowed module types need to be configurable */ @@ -109,13 +108,9 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, memset(&template, 0, sizeof(template)); template.id = module->type; - template.entry = module->entry_point; - template.p.size = module->info.persistent_size; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.size = module->info.scratch_size; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; + template.entry = module->entry_point - 4; + template.persistent_size = module->info.persistent_size; + template.scratch_size = module->info.scratch_size; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -135,14 +130,14 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_HSW_IRAM: ram = dsp->addr.lpe; - block_data.offset = + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: ram = dsp->addr.lpe; - block_data.offset = block->ram_offset; - block_data.type = SST_MEM_DRAM; + mod->offset = block->ram_offset; + mod->type = SST_MEM_DRAM; break; default: dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n", @@ -151,20 +146,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); - block_data.data_offset = block_data.data - fw->dma_buf; + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); + mod->data_offset = mod->data - fw->dma_buf; - dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + dev_dbg(dsp->dev, "module block %d type 0x%x " "size 0x%x ==> ram %p offset 0x%x\n", - count, block->type, block->size, ram, + count, mod->type, block->size, ram, block->ram_offset); - sst_module_insert_fixed_block(mod, &block_data); + ret = sst_module_alloc_blocks(mod); + if (ret < 0) { + dev_err(dsp->dev, "error: could not allocate blocks for module %d\n", + count); + sst_module_free(mod); + return ret; + } block = (void *)block + sizeof(*block) + block->size; } + return 0; } -- 1.9.1

3 11

[alsa-devel] New SSID for Dell Headset mode
by Kailang 27 Oct '14

27 Oct '14

Hi Takashi, Dell has new machine. That SSID was 0x069f. It is the lite version of 0x0626. The audio layout was the same. Attach patch was add the SSID for the fixup table. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH 00/11] treewide: mask then shift defects and style updates
by Joe Perches 27 Oct '14

27 Oct '14

logical mask has lower precedence than shift but should be done before the shift so parentheses are generally required. And when masking with a fixed value after a shift, normal kernel style has the shift on the left, then the shift on the right so convert a few non-conforming uses. Joe Perches (11): block: nvme-scsi: Fix probable mask then right shift defects radeon: evergreen: Fix probable mask then right shift defects aiptek: Fix probable mask then right shift defects dvb-net: Fix probable mask then right shift defects cx25840/cx18: Use standard ordering of mask and shift wm8350-core: Fix probable mask then right shift defect iwlwifi: dvm: Fix probable mask then right shift defect ssb: driver_chip_comon_pmu: Fix probable mask then right shift defect tty: ipwireless: Fix probable mask then right shift defects hwa-hc: Fix probable mask then right shift defect sound: ad1889: Fix probable mask then right shift defects drivers/block/nvme-scsi.c | 12 ++++++------ drivers/gpu/drm/radeon/evergreen.c | 3 ++- drivers/input/tablet/aiptek.c | 6 +++--- drivers/media/dvb-core/dvb_net.c | 4 +++- drivers/media/i2c/cx25840/cx25840-core.c | 12 ++++++------ drivers/media/pci/cx18/cx18-av-core.c | 16 ++++++++-------- drivers/mfd/wm8350-core.c | 2 +- drivers/net/wireless/iwlwifi/dvm/lib.c | 4 ++-- drivers/ssb/driver_chipcommon_pmu.c | 4 ++-- drivers/tty/ipwireless/hardware.c | 12 ++++++------ drivers/usb/host/hwa-hc.c | 2 +- sound/pci/ad1889.c | 8 ++++---- 12 files changed, 44 insertions(+), 41 deletions(-) -- 2.1.2

2 2

[alsa-devel] [PATCH][fix] ALSA: bebob: fix wrong decoding of clock information for Terratec PHASE 88 Rack FW
by Takashi Sakamoto 27 Oct '14

27 Oct '14

Terratec PHASE 88 rack fw has two registers for source of clock, one is for internal/external, and another is for wordclock/spdif for external. When clock source is internal, information in another register has no meaning. Thus it must be ignored, but current implementation decodes it. This causes over-indexing reference to labels. Reported-by: András Murányi <muranyia(a)gmail.com> Tested-by: András Murányi <muranyia(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_terratec.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index 0e4c0bf..9940611 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -24,7 +24,12 @@ phase88_rack_clk_src_get(struct snd_bebob *bebob, unsigned int *id) if (err < 0) goto end; - *id = (enable_ext & 0x01) | ((enable_word & 0x01) << 1); + if (enable_ext == 0) + *id = 0; + else if (enable_word == 0) + *id = 1; + else + *id = 2; end: return err; } -- 1.9.1

3 2

Re: [alsa-devel] ALSA - AD1893
by Raymond Yau 27 Oct '14

27 Oct '14

> I am working on Chromium and trying to get a machine with AD1983 to > function correctly. > > I'm on 3.14 kernel. My issue is that I need a way for CRAS (no pulse) > to tell HDA driver to switch capture from Front mic to rear Line/Mic. > The way alsamixer does it, but I want to add that functionality to Cras. > Could you point my to which IOCTL to call or alsa API to look at? You have to post the outpout of alsa-info.sh It depend on the pin default of the rear jack is line in or mic https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… - line_in_auto_switch (bool): enable/disable the line-in auto-switch feature; default false - add_stereo_mix_input (bool): add the stereo mix (analog-loopback mix) to the input mux if available - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Stereo mix was already added to those desktop using VIA or ADI codec when auto-mic is not enabled Stereo mix also has input from DAC, line in and mic node 0x14 Record Selector can also select input from mono mixer node 0xf (mono mix) For those HDA codec with one ADC and most AC97 codecs amixer -c0 cset "Capture Source" "Line" Any reason to use codec specific ioctl since it won't work on other hda codec

1 0

[alsa-devel] [RFC][PATCH 00/15 v4] OXFW driver, a succesor to firewire-speakers
by Takashi Sakamoto 26 Oct '14

26 Oct '14

This patchset updates a part of my previous series: [alsa-devel] [RFC v3] [PATCH 00/52] Enhancement for support of firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… This patchset is based on this series. (This series is under reviewing.) [alsa-devel] [PATCH 00/49 v4] Enhancement for support of Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075841.html If you try this patchset, please apply the series in advance and apply this patchset. Or just use backport codes in my github repository: https://github.com/takaswie/snd-firewire-improve Or if you use Ubuntu 14.04, DKMS package is available from David's PPA. (Special thanks to David Henningsson.) https://launchpad.net/~diwic/+archive/snd-firewire-improve If you have Griffin FireWave, I would strongly request to test. Please see 'An issue for FireWave' in this message. == Background Currently, in ALSA, 'firewire-speakers' driver supports OXFW970/971 chipset which Oxford Semiconductor produced. But this driver has some restrictions: - Don't support MIDI substreams - Don't support substreams for PCM/MIDI capturing There are several devices which are based on OXFW970/971. Especially, devices which Behringer and Mackie produced support playback/capture PCM samples/MIDI messages. So firewire-speakers needs re-work to support these devices. This patchset renames 'firewire-speakers' to 'oxfw'. Maintainers already agreed to rename firewire-speakers to suitable name, in my first RFC in Jan 2014. == Supported devices Currently these devices are supported: * Griffin Firewave * LaCie Firewire Speakers * Behringer F-Control Audio 202 * Mackie(Loud) Onyx-i series (former models) * Mackie(Loud) Onyx Satellite * Mackie(Loud) Tapco Link.Firewire These devices can be supported but its IDs are unknown: * Mackie(Loud), d.2 pro * Mackie(Loud), d.4 pro * Mackie(Loud), U.420 * Mackie(Loud), U.420d When you have unknown devices, please inform me output of this command: $ hexdump -C /sys/bus/firewire/devices/fwX/config_rom (X is node id for your device. Tipically '1'.) I tested this driver with Behringer F-Control Audio 202. == Features - Capturing/Playbacking PCM samples/MIDI messages at supported sampling rates via ALSA PCM/MIDI interfaces - Retrieving firewire node information via ALSA hwdep interface - Locking/Unlocking kernel streaming via ALSA hwdep interface - Receiving notification at starting/stopping kernel streaming via ALSA hwdep interface == Solutions for reported regressions Two regressions were reported for my previous series: 1.Some names in card structure is trimmed. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073128.h… 2.The number of PCM channels are not detected correctly. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073145.h… The cause of issue 1 is that the device has too long names in its config rom. To solve this issue, for the driver, this patchset uses the same way which firewire-speakers uses, propper alias in device-specific structure. The cause of issue 2 is that the device has several format entries for the same sampling rate: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073561.h… To solve this issue, this patchset includes improved mechanism to detect stream formation. Additionally, this patchset adds new device entry for Mackie(Loud) Tapco Link.Firewire, refering to FFADO bug ticket 236: http://subversion.ffado.org/ticket/236 == An issue for FireWave I have an issue for Griffin Firewave. This device can handle AV/C Stream Format Information command and driver can get to know stream formations via the command. With Stefan's help, responses from FireWave against this command are clear. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073561.h… According to these responses, supported formations are: Input Stream to device: Rate PCM MIDI 32000 6 0 44100 6 0 48000 6 0 96000 6 0 32000 2 0 44100 2 0 48000 2 0 0 0 0 0 0 0 0 0 0 (I also expect to get this output from /proc/asound/FireWave/firewire/formation. I note it's order is not important.) But current 'firewire-speakers' has PCM constraints below: - 6ch at 32/44.1/48/96 kHz - 2ch at 48/96 kHz http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073562.h… So this OXFW driver may lost a support for 2ch/96.0kHz for FireWave and gain 2ch/32.0kHz and 2ch/44.1kHz. I don't know exactly the reason of these gaps. So I strongly hope testers to check these two items: - FireWave really can handle 2ch/96kHz with firewire-speakers driver. - FireWave really can handle 2ch/32.0kHz and 2ch/44.1kHz with oxfw driver. I think a test with aplay is easy to execute: $ aplay -D hw:FireWave -f S32_LE -c 2 -r 96000 /dev/urandom $ aplay -D hw:FireWave -f S32_LE -c 2 -r 48000 /dev/urandom $ aplay -D hw:FireWave -f S32_LE -c 2 -r 44100 /dev/urandom $ aplay -D hw:FireWave -f S32_LE -c 2 -r 32000 /dev/urandom If 2ch/96kHz is apparently available, I'll change codes as a quirk. FYI, when I tested unavailable number of PCM channels with a hack, FCA202 playback PCM substreams but becomes freezed after stopping streaming. $ cat /var/log/syslog ... kernel: [10098.670390] snd_oxfw fw1.0: transaction failed: timeout ... I believe the other OXFW970/971 based devices have a similar behaviour. ================ Takashi Sakamoto (15): speakers: Rename to oxfw and rename some members oxfw: Move to its own directory oxfw: Split stream functionality to a new file and add a header file oxfw: Split PCM functionality to a new file oxfw: Split control functionality to a new file oxfw: Change the way to name card oxfw: Change the way to make PCM rules/constraints oxfw: Add proc interface for debugging purpose oxfw: Change the way to start stream oxfw: Add support for AV/C stream format command to get supported stream formation oxfw: Add a quirk for Griffin FireWave oxfw: Add support for Behringer/Mackie devices oxfw: Add support AMDTP in-stream and PCM capture oxfw: Add support for capture/playback MIDI messages oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 18 +- sound/firewire/Makefile | 3 +- sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw.c | 284 +++++++++++++ sound/firewire/oxfw/oxfw.h | 141 +++++++ sound/firewire/oxfw/oxfw_command.c | 113 ++++++ sound/firewire/oxfw/oxfw_control.c | 283 +++++++++++++ sound/firewire/oxfw/oxfw_hwdep.c | 190 +++++++++ sound/firewire/oxfw/oxfw_midi.c | 168 ++++++++ sound/firewire/oxfw/oxfw_pcm.c | 367 +++++++++++++++++ sound/firewire/oxfw/oxfw_proc.c | 83 ++++ sound/firewire/oxfw/oxfw_stream.c | 658 ++++++++++++++++++++++++++++++ sound/firewire/speakers.c | 792 ------------------------------------- 15 files changed, 2308 insertions(+), 801 deletions(-) create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h create mode 100644 sound/firewire/oxfw/oxfw_command.c create mode 100644 sound/firewire/oxfw/oxfw_control.c create mode 100644 sound/firewire/oxfw/oxfw_hwdep.c create mode 100644 sound/firewire/oxfw/oxfw_midi.c create mode 100644 sound/firewire/oxfw/oxfw_pcm.c create mode 100644 sound/firewire/oxfw/oxfw_proc.c create mode 100644 sound/firewire/oxfw/oxfw_stream.c delete mode 100644 sound/firewire/speakers.c -- 1.8.3.2

3 34

[alsa-devel] Request for comments for SND_SOC_IMX_WM8731
by Jonathan Bennett 26 Oct '14

26 Oct '14

This patch is a continuation of my work on supporting the Utilite devices on the vanilla kernel. I've modified the code slightly to conform to established practice. Namely instead of using src-port and ext-port, we use mux-int-port and mux-ext-port. This driver has support for putting the codec in either master mode or slave mode. So far I've only had success with using slave mode. For trying to submit for kernel inclusion, should I strip out the code for codec master mode? As a kernel dev noob, any other comments are welcome. Thanks, Jonathan Bennett diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 081e406..09c923f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,17 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8731 + tristate "SoC Audio support for i.MX boards with wm8731 + depends on OF && I2C + select SND_SOC_WM8731 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Soc Audio support for i.MX with wm8731 + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..4292fac 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8731-objs := imx-wm8731.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -65,5 +66,6 @@ obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_WM8731) += snd-soc-imx-wm8731.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8731.c b/sound/soc/fsl/imx-wm8731.c new file mode 100644 index 0000000..8ce4e9f --- /dev/null +++ b/sound/soc/fsl/imx-wm8731.c @@ -0,0 +1,687 @@ +/* + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright (C) 2012 Freescale Semiconductor, Inc. + * Copyright (C) 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8731.h" +#include "imx-audmux.h" +#include "imx-ssi.h" + +#define DAI_NAME_SIZE 32 +#define WM8731_MCLK_FREQ (24000000 / 2) + +struct imx_wm8731_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct i2c_client *codec_dev; + /* audio_clocking_data */ + struct clk *pll; + struct clk *clock_root; + long sysclk; + long current_rate; + /* apis */ + int (*clock_enable)(int enable,struct imx_wm8731_data *data); +}; + +static int imx_wm8731_init(struct snd_soc_pcm_runtime *rtd); +static int imx_hifi_hw_params_slv_mode(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); +static void imx_hifi_shutdown(struct snd_pcm_substream *substream); + +struct imx_priv { + struct platform_device *pdev; + struct imx_wm8731_data *data; +}; + +static struct imx_priv card_priv; + +static struct snd_soc_ops imx_hifi_ops = { + .shutdown = imx_hifi_shutdown, +}; + +/* imx card dapm widgets */ +static const struct snd_soc_dapm_widget imx_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +/* imx machine connections to the codec pins */ +static const struct snd_soc_dapm_route audio_map[] = { + { "Headphone Jack", NULL, "LHPOUT" }, + { "Headphone Jack", NULL, "RHPOUT" }, + + { "Ext Spk", NULL, "LOUT" }, + { "Ext Spk", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line Jack" }, + { "RLINEIN", NULL, "Line Jack" }, + + { "MICIN", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Mic Jack"}, +}; + +static int wm8731_slv_mode_init(struct imx_wm8731_data *data) +{ + struct clk *new_parent; + struct clk *ssi_clk; + struct i2c_client *codec_dev = data->codec_dev; + + new_parent = devm_clk_get(&codec_dev->dev, "pll4"); + if (IS_ERR(new_parent)) { + pr_err("Could not get \"pll4\" clock \n"); + return PTR_ERR(new_parent); + } + + ssi_clk = devm_clk_get(&codec_dev->dev, "imx-ssi.1"); + if (IS_ERR(ssi_clk)) { + pr_err("Could not get \"imx-ssi.1\" clock \n"); + return PTR_ERR(ssi_clk); + } + + clk_set_parent(ssi_clk, new_parent); + + data->pll = new_parent; + data->clock_root = ssi_clk; + data->current_rate = 0; + + data->sysclk = 0; + + return 0; +} + +static int wm8731_slv_mode_clock_enable(int enable, struct imx_wm8731_data *data) +{ + long pll_rate; + long rate_req; + long rate_avail; + + if (!enable) + return 0; + + if (data->sysclk == data->current_rate) + return 0; + + switch (data->sysclk) { + case 11289600: + pll_rate = 632217600; + break; + + case 12288000: + pll_rate = 688128000; + break; + + default: + return -EINVAL; + } + + rate_req = pll_rate; + rate_avail = clk_round_rate(data->pll, rate_req); + clk_set_rate(data->pll, rate_avail); + + rate_req = data->sysclk; + rate_avail = clk_round_rate(data->clock_root, + rate_req); + clk_set_rate(data->clock_root, rate_avail); + + pr_info("%s: \"imx-ssi.1\" rate = %ld (= %ld)\n", + __func__, rate_avail, rate_req); + + data->current_rate = data->sysclk; + + return 0; +} + +static int imx_hifi_startup_slv_mode(struct snd_pcm_substream *substream) +{ + /* + * As SSI's sys clock rate depends on sampling rate, + * the clock enabling code is moved to imx_hifi_hw_params(). + */ + return 0; +} + +static int wm8731_mst_mode_init(struct imx_wm8731_data *data) +{ + long rate; + struct clk *new_parent; + struct clk *ssi_clk; + struct i2c_client *codec_dev = data->codec_dev; + + new_parent = devm_clk_get(&codec_dev->dev, "cko2"); + if (IS_ERR(new_parent)) { + pr_err("Could not get \"cko2\" clock \n"); + return PTR_ERR(new_parent); + } + + ssi_clk = devm_clk_get(&codec_dev->dev, "cko"); + if (IS_ERR(ssi_clk)) { + pr_err("Could not get \"cko\" clock \n"); + return PTR_ERR(ssi_clk); + } + + rate = clk_round_rate(new_parent, WM8731_MCLK_FREQ); + clk_set_rate(new_parent, rate); + + clk_set_parent(ssi_clk, new_parent); + + rate = clk_round_rate(ssi_clk, WM8731_MCLK_FREQ); + clk_set_rate(ssi_clk, rate); + + pr_info("%s: \"CLKO\" rate = %ld (= %d)\n", + __func__, rate, WM8731_MCLK_FREQ); + + data->pll = new_parent; + data->clock_root = ssi_clk; + data->sysclk = rate; + + return 0; +} + +static int wm8731_mst_mode_clock_enable(int enable, struct imx_wm8731_data *data) +{ + struct clk *clko = data->clock_root; + + if (enable) + clk_enable(clko); + else + clk_disable(clko); + + return 0; +} + +static int imx_hifi_startup_mst_mode(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + + if (!codec_dai->active) + data->clock_enable(1,data); + + return 0; +} + + +static int imx_hifi_hw_params_slv_mode(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + + u32 dai_format; + snd_pcm_format_t sample_format; + unsigned int channels; + unsigned int tx_mask, rx_mask; + unsigned int sampling_rate; + unsigned int div_2, div_psr, div_pm; + int ret; + + sampling_rate = params_rate(params); + sample_format = params_format(params); + + channels = params_channels(params); + printk("%s:%s sampling rate = %u channels = %u \n", __FUNCTION__, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "Playback" : "Capture"), + sampling_rate, channels); + + /* set CPU DAI configuration */ + switch (sampling_rate) { + case 8000: + case 32000: + case 48000: + case 96000: + data->sysclk = 12288000; + break; + + case 44100: + case 88200: + data->sysclk = 11289600; + break; + + default: + return -EINVAL; + } + + wm8731_slv_mode_clock_enable(1,data); + + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS; + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); + if (ret < 0) + return ret; + + /* set i.MX active slot mask */ + /* S[TR]CCR:DC */ + tx_mask = ~((1 << channels) - 1); + rx_mask = tx_mask; + snd_soc_dai_set_tdm_slot(cpu_dai, tx_mask, rx_mask, 2, 32); + + /* + * SSI sysclk divider: + * div_2: /1 or /2 + * div_psr: /1 or /8 + * div_pm: /1 .. /256 + */ + div_2 = 0; + div_psr = 0; + switch (sampling_rate) { + case 8000: + // 1x1x12 + div_pm = 11; + break; + case 32000: + // 1x1x3 + div_pm = 2; + break; + case 48000: + // 1x1x2 + div_pm = 1; + break; + case 96000: + // 1x1x1 + div_pm = 0; + break; + case 44100: + // 1x1x2 + div_pm = 1; + break; + case 88200: + // 1x1x1 + div_pm = 0; + break; + default: + return -EINVAL; + } + + /* sync mode: a single clock controls both playback and capture */ + snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSI_TX_DIV_2, (div_2 ? SSI_STCCR_DIV2 : 0)); + snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSI_TX_DIV_PSR, (div_psr ? SSI_STCCR_PSR : 0)); + snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSI_TX_DIV_PM, div_pm); + + /* set codec DAI configuration */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_format); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8731_SYSCLK_MCLK, + data->sysclk, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + pr_err("Failed to set codec master clock to %u: %d \n", + data->sysclk, ret); + return ret; + } + + return 0; +} + +static int imx_hifi_hw_params_mst_mode(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + u32 dai_format; + unsigned int channels; + unsigned int tx_mask, rx_mask; + unsigned int sampling_rate; + int ret; + + + sampling_rate = params_rate(params); + channels = params_channels(params); + pr_debug("%s:%s sampling rate = %u channels = %u \n", __FUNCTION__, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "Playback" : "Capture"), + sampling_rate, channels); + + /* set cpu DAI configuration */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM; + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); + if (ret < 0) + return ret; + + /* set i.MX active slot mask */ + /* S[TR]CCR:DC */ + tx_mask = ~((1 << channels) - 1); + rx_mask = tx_mask; + snd_soc_dai_set_tdm_slot(cpu_dai, tx_mask, rx_mask, 2, 32); + + /* set codec DAI configuration */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_format); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8731_SYSCLK_MCLK, + data->sysclk, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + pr_err("Failed to set codec master clock to %u: %d \n", + data->sysclk, ret); + return ret; + } + + return 0; +} + +static void imx_hifi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + + if (!codec_dai->active) + data->clock_enable(0,data); + + return; +} + +static int imx_wm8731_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_codec *codec = rtd->codec; + + /* Add imx specific widgets */ + ret = snd_soc_dapm_new_controls(&codec->dapm, imx_dapm_widgets, + ARRAY_SIZE(imx_dapm_widgets)); + if (ret) + goto out_retcode; + + /* Set up imx specific audio path audio_map */ + ret = snd_soc_dapm_add_routes(&codec->dapm, audio_map, ARRAY_SIZE(audio_map)); + if (ret) + goto out_retcode; + + ret = snd_soc_dapm_enable_pin(&codec->dapm, "Headphone Jack"); + if (ret) + goto out_retcode; + + ret = snd_soc_dapm_nc_pin(&codec->dapm, "Ext Spk"); + if (ret) + goto out_retcode; + +out_retcode: + + if (ret) + pr_err("%s: failed with error code: %d \n", __FUNCTION__, ret); + else + pr_info("%s: success \n", __FUNCTION__); + + return ret; +} + +/** + * Configure AUDMUX interconnection between + * _slave (CPU side) and _master (codec size) + * + * When SSI operates in master mode, 5-wire interconnect with + * audio codec is required: + * TXC - BCLK + * TXD - DAC data + * RXD - ADC data + * TXFS - {DAC|ADC}LRC, i.e. word clock + * RXC - MCLK, i.e. oversampling clock + * Audmux is operated in asynchronous mode to enable 6-wire + * interface (as opposed to 4-wire interface in sync mode). + */ +static int imx_audmux_config_slv_mode(int _slave, int _master) +{ + unsigned int ptcr, pdcr; + int slave = _slave - 1; + int master = _master - 1; + + ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(slave) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(slave | 0x8) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(slave); + + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(slave); + imx_audmux_v2_configure_port(master, ptcr, pdcr); + ptcr = ptcr & ~IMX_AUDMUX_V2_PTCR_SYN; + imx_audmux_v2_configure_port(master, ptcr, pdcr); + + ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(master | 0x8) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(master); + + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(master); + imx_audmux_v2_configure_port(slave, ptcr, pdcr); + ptcr = ptcr & ~IMX_AUDMUX_V2_PTCR_SYN; + imx_audmux_v2_configure_port(slave, ptcr, pdcr); + + return 0; +} + +static int imx_audmux_config_mst_mode(int _slave, int _master) +{ + unsigned int ptcr, pdcr; + int slave = _slave - 1; + int master = _master - 1; + + ptcr = IMX_AUDMUX_V2_PTCR_SYN; + ptcr |= IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(master) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(master); + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(master); + imx_audmux_v2_configure_port(slave, ptcr, pdcr); + + ptcr = IMX_AUDMUX_V2_PTCR_SYN; + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(slave); + imx_audmux_v2_configure_port(master, ptcr, pdcr); + + return 0; +} + +static int imx_wm8731_probe(struct platform_device *pdev) +{ + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8731_data *data; + unsigned int src_port, ext_port; + unsigned int ssi_mode; + const char *ssi_mode_str; + + int ret; + + priv->pdev = pdev; + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + card_priv.data = data; + + data->codec_dev = codec_dev; + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8731-hifi"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.init = &imx_wm8731_init; + + ret = of_property_read_u32(pdev->dev.of_node, "mux-int-port", &src_port); + if (ret) { + dev_err(&pdev->dev, "failed to get \"mux-int-port\" value\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_u32(pdev->dev.of_node, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "failed to get \"mux-ext-port\" value\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_string(ssi_np, "fsl,mode", &ssi_mode_str); + if (ret) { + dev_err(&pdev->dev, "failed to get \"fsl,mode\" value\n"); + ret = -EINVAL; + goto fail; + } + + ssi_mode = strcmp(ssi_mode_str, "i2s-master"); + + if (ssi_mode) { + /* Master Mode */ + imx_audmux_config_mst_mode(src_port, ext_port); + wm8731_mst_mode_init(data); + data->clock_enable = wm8731_mst_mode_clock_enable; + imx_hifi_ops.hw_params = imx_hifi_hw_params_mst_mode; + imx_hifi_ops.startup = imx_hifi_startup_mst_mode; + } else { + /* Slave Mode */ + imx_audmux_config_slv_mode(src_port, ext_port); + wm8731_slv_mode_init(data); + data->clock_enable = wm8731_slv_mode_clock_enable; + imx_hifi_ops.hw_params = imx_hifi_hw_params_slv_mode; + imx_hifi_ops.startup = imx_hifi_startup_slv_mode; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + data->card.num_links = 1; + data->card.dai_link = &data->dai; + + data->card.dapm_widgets = imx_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_dapm_widgets); + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } +/* of_node_put(ssi_np); + of_node_put(codec_np); */ + return 0; +fail: + + if (ssi_np) + of_node_put(ssi_np); + + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8731_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id imx_wm8731_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8731", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8731_dt_ids); + +static struct platform_driver imx_wm8731_driver = { + .driver = { + .name = "imx-wm8731", + .owner = THIS_MODULE, + .of_match_table = imx_wm8731_dt_ids, + }, + .probe = imx_wm8731_probe, + .remove = imx_wm8731_remove, +}; +module_platform_driver(imx_wm8731_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8731 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8731");

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[alsa-devel] [PATCH] ASoC: rt5645:Mark RT5645_TDM_CTRL_3 as readable
by yang.a.fang＠intel.com 26 Oct '14

26 Oct '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> amixer query fails due to it is not readable reigster Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3fb83bf..d16331e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -139,6 +139,7 @@ static const struct reg_default rt5645_reg[] = { { 0x76, 0x000a }, { 0x77, 0x0c00 }, { 0x78, 0x0000 }, + { 0x79, 0x0123 }, { 0x80, 0x0000 }, { 0x81, 0x0000 }, { 0x82, 0x0000 }, @@ -334,6 +335,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_DMIC_CTRL2: case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: + case RT5645_TDM_CTRL_3: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.7.9.5

1 0

[alsa-devel] [PATCH 0/2] Disable the ThinkPad HW mute/level control when possible
by Andy Lutomirski 25 Oct '14

25 Oct '14

This series is being resurrected after several years, and it's quite different from before, so I'm restarting the version number at 1 :) ThinkPad volume and mute controls are a mess. For whatever reason, ThinkPads have mute buttons that send KEY_MUTE *and* control an invisible-to-ALSA mute switch. Some of them have volume controls that interact with this switch as well. This is a perennial source of problems. On most ThinkPads, if you press mute and then unmute using GUI controls, you have no sound, because userspace and ALSA state gets out of sync with the hardware switch. There's a separate "sound card" that exposes the hardware switch, but userspace code generally doesn't understand that. There are already a few _OSI(Linux) overrides to turn all the hardware buttons into regular buttons. Rather than quirking ACPI everywhere, just teach thinkpad-acpi to program the buttons for full software control and to disable hardware controls. That allows us to remove the ACPI quirks and have normal mute controls. This approach should be much simpler than adding even more kludgey ALSA integration for questionable gain. I haven't tested on my X220 (which has a mute light in addition to a mute button), since that laptop's SATA controller died last week. It should work, though, and if someone else could test this on a ThinkPad with a mute light, that would be great. Andy Lutomirski (2): thinkpad-acpi: Try to use full software mute control acpi: Remove _OSI(Linux) for ThinkPads drivers/acpi/blacklist.c | 54 ------------------ drivers/platform/x86/thinkpad_acpi.c | 107 +++++++++++++++++++++++++++++++---- 2 files changed, 97 insertions(+), 64 deletions(-) -- 1.9.3

2 10

[alsa-devel] Recording with asus xonar essence st card
by Vasily Postnicov 25 Oct '14

25 Oct '14

Hello. This is not related to alsa, but I decided to write this message here, because as far as I know alsa has the best open source driver for asus xonar series. I am using DragonFlyBSD and a bit modified driver originally written for FreeBSD by Alexander Polakov https://github.com/shamazmazum/xonar-dragonfly It lacks support for recording. I am trying to add this functionality from alsa/OSS drivers, but I have some unanswered questions. There are they here: 1) How do I initialize my soundcard for recording properly? I think I must enable CS5381 ADC using GPIO registers, but what are OXYGEN_REC_ROUTING, OXYGEN_ADC_MONITOR and OXYGEN_A_MONITOR_ROUTING for? Does MONITOR mean the same thing as in the user manual? If ROUTING registers are just switches for source of a signal to be recorded, is it enough to use OXYGEN_REC_B_ROUTE_I2S_ADC_2 for ST card to record from line-in/mic jack? 2) How do I control volume of sound while recording? Have Xonar Essence cards any controls for this? I cannot find it out looking at oxygen_mixer.c

2 1

[alsa-devel] [PATCH] ASoC: rt5645:Mark RT5645_TDM_CTRL_3 as readable
by yang.a.fang＠intel.com 25 Oct '14

25 Oct '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> amixer query fails due to it is not readable reigster Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3fb83bf..d16331e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -139,6 +139,7 @@ static const struct reg_default rt5645_reg[] = { { 0x76, 0x000a }, { 0x77, 0x0c00 }, { 0x78, 0x0000 }, + { 0x79, 0x0123 }, { 0x80, 0x0000 }, { 0x81, 0x0000 }, { 0x82, 0x0000 }, @@ -334,6 +335,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_DMIC_CTRL2: case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: + case RT5645_TDM_CTRL_3: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ALSA: pcm: Avoid mmap warnings on x86
by Takashi Iwai 24 Oct '14

24 Oct '14

On x86, using dma_mmap_coherent() for the pages allocated via dma_alloc_coherent() results in a warning like: aplay:32536 map pfn RAM range req uncached-minus for [mem 0x21d500000-0x21d51ffff], got write-back Until the issue is addressed in the core side, take back to the old good way in PCM code only for x86. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a915ed07aee1..2f7ad10ee7c3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3324,6 +3324,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_end - area->vm_start, area->vm_page_prot); } #endif /* CONFIG_GENERIC_ALLOCATOR */ +#ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */ if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, @@ -3331,6 +3332,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); +#endif /* CONFIG_X86 */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; -- 2.1.2

1 0

Re: [alsa-devel] Default value change for restore default in initial state
by Takashi Iwai 24 Oct '14

24 Oct '14

At Fri, 24 Oct 2014 08:35:12 +0000, Kailang wrote: > > Hi Takashi, > > We have some changed for restore default for ALC282 and ALC283. Could you give a bit more information, what these updates actually fix? thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - Add missing terminating entry to SND_HDA_PIN_QUIRK macro
by David Henningsson 24 Oct '14

24 Oct '14

Without this terminating entry, the pin matching would continue across random memory until a zero or a non-matching entry was found. The result being that in some cases, the pin quirk would not be applied correctly. Cc: stable(a)vger.kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_local.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 7eb44e7..62658f2 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -419,7 +419,7 @@ struct snd_hda_pin_quirk { .subvendor = _subvendor,\ .name = _name,\ .value = _value,\ - .pins = (const struct hda_pintbl[]) { _pins } \ + .pins = (const struct hda_pintbl[]) { _pins, {0, 0}} \ } #else @@ -427,7 +427,7 @@ struct snd_hda_pin_quirk { { .codec = _codec,\ .subvendor = _subvendor,\ .value = _value,\ - .pins = (const struct hda_pintbl[]) { _pins } \ + .pins = (const struct hda_pintbl[]) { _pins, {0, 0}} \ } #endif -- 1.9.1

2 1

[alsa-devel] [PATCH] hda-emu: Add snd_ctl_enum_info
by David Henningsson 24 Oct '14

24 Oct '14

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/sound/control.h | 2 ++ snd-control.c | 18 ++++++++++++++++++ 2 files changed, 20 insertions(+) diff --git a/include/sound/control.h b/include/sound/control.h index 47b92c5..eaaada4 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -101,6 +101,8 @@ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, + unsigned int items, const char *const names[]); /* * virtual master control diff --git a/snd-control.c b/snd-control.c index 3e3686f..a4a7a14 100644 --- a/snd-control.c +++ b/snd-control.c @@ -153,6 +153,24 @@ int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, } #endif +int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, + unsigned int items, const char *const names[]) +{ + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = channels; + info->value.enumerated.items = items; + if (!items) + return 0; + if (info->value.enumerated.item >= items) + info->value.enumerated.item = items - 1; + if (strlen(names[info->value.enumerated.item]) >= sizeof(info->value.enumerated.name)) + hda_log(HDA_LOG_WARN, "ALSA: too long item name '%s'\n", names[info->value.enumerated.item]); + strlcpy(info->value.enumerated.name, + names[info->value.enumerated.item], + sizeof(info->value.enumerated.name)); + return 0; +} + int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, int active) { -- 1.9.1

2 1

[alsa-devel] ALSA HDMI output not detected
by samuelchemla＠orange.fr 23 Oct '14

23 Oct '14

Hi guys, I opened a bug about HDMI sound output not detected on an NVIDIA optimus laptop. I don't know if HDMI sound card detection is up to ALSA, the kernel or the video driver. Could anyone give me a clue so the bug can be routed to its correct destination? Link to the bug: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1377653 Regards, Sam

2 2

[alsa-devel] [PATCH] ASoC: jack: update calls to gpiod_get*()
by Alexandre Courbot 23 Oct '14

23 Oct '14

Add the new flags argument to calls of (devm_)gpiod_get*() and remove any direction setting code afterwards. Currently both forms (with or without the flags argument) are valid thanks to transitional macros in <linux/gpio/consumer.h>. These macros will be removed once all consumers are updated and the flags argument will become compulsary. Signed-off-by: Alexandre Courbot <acourbot(a)nvidia.com> --- sound/soc/soc-jack.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea997a3..f921d0098518 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -309,7 +309,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, /* GPIO descriptor */ gpios[i].desc = gpiod_get_index(gpios[i].gpiod_dev, gpios[i].name, - gpios[i].idx); + gpios[i].idx, GPIOD_IN); if (IS_ERR(gpios[i].desc)) { ret = PTR_ERR(gpios[i].desc); dev_err(gpios[i].gpiod_dev, @@ -327,17 +327,14 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, goto undo; } - ret = gpio_request(gpios[i].gpio, gpios[i].name); + ret = gpio_request_one(gpios[i].gpio, GPIOF_IN, + gpios[i].name); if (ret) goto undo; gpios[i].desc = gpio_to_desc(gpios[i].gpio); } - ret = gpiod_direction_input(gpios[i].desc); - if (ret) - goto err; - INIT_DELAYED_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; -- 2.1.2

2 1

Re: [alsa-devel] [pulseaudio-discuss] Problem with timing and USB scanning...
by Raymond Yau 23 Oct '14

23 Oct '14

> > Howdy! I have a situation where PulseAudio is "grabbing" a usb headset before > fully initialized, thus not all features are being recognized. > > More details here: > > https://askubuntu.com/questions/537166/usb%2daudio%2ddevice%2dnot%2dfully%2… You have to post output of alsa-info.sh and pactl list before and after the hot plug This look like priority issue if you have usb web cam mic and hotplug usb headset https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/u… Another issue is some usb devices only obtained dB scale at a later time after the control is created This may need some delay for pulseaudio to get dB scale You can add printk statement to find out how much time you need > > and a related, I think, bug here: > > https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/841433 > > Anyone have a pointer to a udev rule I could create (or a PulseAudio config I > could tweak) that would case a delay before the device is scanned by > PulseAudio? > > I'm pretty sure it's a timing issue, because: > > Works OK: > Cold boot (or reboot) > Resume from hibernate > Resume (or hotplug) -> killall/restart pulseaudio > > Fails to work: > Hotplug > Resume from sleep > > Any tips? >

2 3

[alsa-devel] Echo Audio Echo 2 USB Interface
by Tristan Chambers 23 Oct '14

23 Oct '14

I picked up an Echo Audio Echo 2 USB Interface. The product specs list iPad support through camera connection kit, so I assume it is class compliant. http://echoaudio.com/blogs/news/6454472-echo-digital-audio-introduces-the-e… However, on most systems when I plug the device in, it shows up in my sound settings panel under the Input tab, but not under the Output tab. Recording works. I can successfully record with arecord -vv -fdat -D "plughw:CARD=Audio,DEV=0" echo.wav. I also can increase the sample rate by changing the flags. I verified that the sample rate actually increases by recording a 50khz test tone into the input and examining the wave form in an editor. It's there! So this is very promising. (anyone know of a test to verify bit depth? I want to make sure it's actually recording at 24bits.) Playback however, is nothing but trouble. Like I said, the output device doesn't show up in the Output tab in my sound settings (gnome/pulse). However, when I list devices and available PCMs with aplay -l and aplay -L the device shows up on the lists. See specimens #1, #2, and #3. When I try to play though I get about a 1/4 second of audio and then nothing, and subsequent attempts to play sound output nothing. The device then must be unplugged and plugged back in. I'm playing with the command aplay -vv foo.wav --device="plughw:CARD=Audio,DEV=0". See output screen on specimen #4. I installed the latest kernel and had the exact same phenomenon. See specimen #5. To move forward here. What kind of debugging tools should I use to help figure out where the problem is? Any suggestions you have to get it working? To be sure that the device is in good working order I hooked it up to an OSX machine running 10.6.8. Everything works great with no additional drivers installed, or special configurations. Tanks for your help, Tristan --------------- #1 -------------------- tristan@ilya:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC259 Analog [ALC259 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Audio [Echo 2 Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 tristan@ilya:~$ aplay -L default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC259 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, HDMI 0 HDMI Audio Output dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct sample mixing device dmix:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct sample snooping device dsnoop:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct hardware device without any conversions hw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Hardware device with all software conversions plughw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Hardware device with all software conversions sysdefault:CARD=Audio Echo 2 Audio, USB Audio Default Audio Device front:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Front speakers surround40:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct sample mixing device dsnoop:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct sample snooping device hw:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct hardware device without any conversions plughw:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Hardware device with all software conversions --------------- #2 -------------------- tristan@ilya:~$ arecord -l **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC259 Analog [ALC259 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Audio [Echo 2 Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 tristan@ilya:~$ arecord -L default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC259 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC259 Analog Hardware device with all software conversions sysdefault:CARD=Audio Echo 2 Audio, USB Audio Default Audio Device front:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Front speakers surround40:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct sample mixing device dsnoop:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct sample snooping device hw:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Direct hardware device without any conversions plughw:CARD=Audio,DEV=0 Echo 2 Audio, USB Audio Hardware device with all software conversions ---------------- #3 ------------------- tristan@ilya:/proc/asound/card1$ cat stream0 ECHO Digital Audio Echo 2 Audio at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S32_LE Channels: 4 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 2 Endpoint: 2 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us ------------------- #4 ------------------ aplay -vv foo.wav --device="plughw:CARD=Audio,DEV=0" Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 4 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 appl_ptr : 0 hw_ptr : 0 ############+ | 23% ------------------- #5 ------------------ tristan@ilya:~$ uname -a Linux ilya 3.17.0-031700-lowlatency #201410060605 SMP PREEMPT Mon Oct 6 11:06:16 UTC 2014 i686 i686 i686 GNU/Linux tristan@ilya:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k3.17.0-031700-lowlatency.

2 2

[alsa-devel] [PATCH v3 03/59] dmaengine: Make the destination abbreviation coherent
by Maxime Ripard 22 Oct '14

22 Oct '14

The dmaengine header abbreviates destination as at least two different strings. Make a coherent use of a single one. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> Acked-by: Mark Brown <broonie(a)kernel.org> Acked-by: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> Acked-by: Stephen Warren <swarren(a)wwwdotorg.org> --- drivers/dma/bcm2835-dma.c | 2 +- drivers/dma/edma.c | 2 +- drivers/dma/fsl-edma.c | 2 +- drivers/dma/nbpfaxi.c | 2 +- drivers/dma/omap-dma.c | 2 +- drivers/dma/pl330.c | 2 +- drivers/dma/sirf-dma.c | 2 +- include/linux/dmaengine.h | 8 ++++---- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 9 files changed, 12 insertions(+), 12 deletions(-) diff --git a/drivers/dma/bcm2835-dma.c b/drivers/dma/bcm2835-dma.c index 68007974961a..8a6a9c27fe83 100644 --- a/drivers/dma/bcm2835-dma.c +++ b/drivers/dma/bcm2835-dma.c @@ -571,7 +571,7 @@ static int bcm2835_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); - caps->dstn_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->dst_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = false; caps->cmd_terminate = true; diff --git a/drivers/dma/edma.c b/drivers/dma/edma.c index 123f578d6dd3..790054eef76c 100644 --- a/drivers/dma/edma.c +++ b/drivers/dma/edma.c @@ -998,7 +998,7 @@ static int edma_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = EDMA_DMA_BUSWIDTHS; - caps->dstn_addr_widths = EDMA_DMA_BUSWIDTHS; + caps->dst_addr_widths = EDMA_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/drivers/dma/fsl-edma.c b/drivers/dma/fsl-edma.c index 3c5711d5fe97..13df85a16438 100644 --- a/drivers/dma/fsl-edma.c +++ b/drivers/dma/fsl-edma.c @@ -781,7 +781,7 @@ static int fsl_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = FSL_EDMA_BUSWIDTHS; - caps->dstn_addr_widths = FSL_EDMA_BUSWIDTHS; + caps->dst_addr_widths = FSL_EDMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/drivers/dma/nbpfaxi.c b/drivers/dma/nbpfaxi.c index 5aeada56a442..b4a61ccbb443 100644 --- a/drivers/dma/nbpfaxi.c +++ b/drivers/dma/nbpfaxi.c @@ -1076,7 +1076,7 @@ static int nbpf_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = NBPF_DMA_BUSWIDTHS; - caps->dstn_addr_widths = NBPF_DMA_BUSWIDTHS; + caps->dst_addr_widths = NBPF_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = false; caps->cmd_terminate = true; diff --git a/drivers/dma/omap-dma.c b/drivers/dma/omap-dma.c index bbea8243f9e8..8035b8c33cfd 100644 --- a/drivers/dma/omap-dma.c +++ b/drivers/dma/omap-dma.c @@ -1100,7 +1100,7 @@ static int omap_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = OMAP_DMA_BUSWIDTHS; - caps->dstn_addr_widths = OMAP_DMA_BUSWIDTHS; + caps->dst_addr_widths = OMAP_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/drivers/dma/pl330.c b/drivers/dma/pl330.c index 4839bfa74a10..b563bfa6cc4d 100644 --- a/drivers/dma/pl330.c +++ b/drivers/dma/pl330.c @@ -2576,7 +2576,7 @@ static int pl330_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = PL330_DMA_BUSWIDTHS; - caps->dstn_addr_widths = PL330_DMA_BUSWIDTHS; + caps->dst_addr_widths = PL330_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = false; caps->cmd_terminate = true; diff --git a/drivers/dma/sirf-dma.c b/drivers/dma/sirf-dma.c index aac03ab10c54..149d7fddfa75 100644 --- a/drivers/dma/sirf-dma.c +++ b/drivers/dma/sirf-dma.c @@ -652,7 +652,7 @@ static int sirfsoc_dma_device_slave_caps(struct dma_chan *dchan, struct dma_slave_caps *caps) { caps->src_addr_widths = SIRFSOC_DMA_BUSWIDTHS; - caps->dstn_addr_widths = SIRFSOC_DMA_BUSWIDTHS; + caps->dst_addr_widths = SIRFSOC_DMA_BUSWIDTHS; caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); caps->cmd_pause = true; caps->cmd_terminate = true; diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 653a1fd07ae8..4d912575fec0 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -387,7 +387,7 @@ enum dma_residue_granularity { /* struct dma_slave_caps - expose capabilities of a slave channel only * * @src_addr_widths: bit mask of src addr widths the channel supports - * @dstn_addr_widths: bit mask of dstn addr widths the channel supports + * @dst_addr_widths: bit mask of dstn addr widths the channel supports * @directions: bit mask of slave direction the channel supported * since the enum dma_transfer_direction is not defined as bits for each * type of direction, the dma controller should fill (1 << <TYPE>) and same @@ -398,7 +398,7 @@ enum dma_residue_granularity { */ struct dma_slave_caps { u32 src_addr_widths; - u32 dstn_addr_widths; + u32 dst_addr_widths; u32 directions; bool cmd_pause; bool cmd_terminate; @@ -638,10 +638,10 @@ struct dma_device { void (*device_free_chan_resources)(struct dma_chan *chan); struct dma_async_tx_descriptor *(*device_prep_dma_memcpy)( - struct dma_chan *chan, dma_addr_t dest, dma_addr_t src, + struct dma_chan *chan, dma_addr_t dst, dma_addr_t src, size_t len, unsigned long flags); struct dma_async_tx_descriptor *(*device_prep_dma_xor)( - struct dma_chan *chan, dma_addr_t dest, dma_addr_t *src, + struct dma_chan *chan, dma_addr_t dst, dma_addr_t *src, unsigned int src_cnt, size_t len, unsigned long flags); struct dma_async_tx_descriptor *(*device_prep_dma_xor_val)( struct dma_chan *chan, dma_addr_t *src, unsigned int src_cnt, diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index b329b84bc5af..851f7afcd5dc 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -151,7 +151,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea hw.info |= SNDRV_PCM_INFO_BATCH; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - addr_widths = dma_caps.dstn_addr_widths; + addr_widths = dma_caps.dst_addr_widths; else addr_widths = dma_caps.src_addr_widths; } -- 2.1.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: two trivial fixes
by Daniel Mack 22 Oct '14

22 Oct '14

Just two random fixes spotted by Coverity. Daniel Mack (2): ASoC: soc-compress: consolidate two identical branches ASoC: fsl: use strncpy() to prevent copying of over-long names sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/soc-compress.c | 11 ++--------- 3 files changed, 4 insertions(+), 11 deletions(-) -- 2.1.0

4 5

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: Track playback and capture CMR dividers separately.
by Peter Rosin 22 Oct '14

22 Oct '14

>From 1e5621d7b9887c648d1a66238dc82d715c1e2cad Mon Sep 17 00:00:00 2001 From: Peter Rosin <peda(a)axentia.se> Date: Mon, 20 Oct 2014 14:38:04 +0200 Subject: [PATCH] ASoC: atmel_ssc_dai: Track playback and capture CMR dividers separately. The CMR divider register is shared by playback and capture. The SSC driver therefore tries to enforce rules so that the needed register content do not conflict during simultaneous playback/capture. However, the implementation also prevents changing the register content in a half-duplex scenario, which is needed when changing sample rates. Thus, keep track of the desired playback and capture clock dividers separately, and allow changing rates without closing the stream. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 31 ++++++++++++++++++++++--------- sound/soc/atmel/atmel_ssc_dai.h | 10 ++++++---- 2 files changed, 28 insertions(+), 13 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f403f39..fec14fb 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -277,7 +277,8 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, /* Reset the SSC */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); /* Clear the SSC dividers */ - ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + ssc_p->tcmr_div = ssc_p->rcmr_div = 0; + ssc_p->tcmr_period = ssc_p->rcmr_period = 0; } spin_unlock_irq(&ssc_p->lock); } @@ -304,17 +305,27 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; switch (div_id) { - case ATMEL_SSC_CMR_DIV: + case ATMEL_SSC_TCMR_DIV: /* * The same master clock divider is used for both * transmit and receive, so if a value has already - * been set, it must match this value. + * been set for the other direction, it must match + * this value. */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else - if (div != ssc_p->cmr_div) - return -EBUSY; + if (ssc_p->rcmr_div == 0) + ssc_p->tcmr_div = div; + else if (div != ssc_p->rcmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_RCMR_DIV: + /* + * See ATMEL_SSC_TCMR_DIV. + */ + if (ssc_p->tcmr_div == 0) + ssc_p->rcmr_div = div; + else if (div != ssc_p->tcmr_div) + return -EBUSY; break; case ATMEL_SSC_TCMR_PERIOD: @@ -345,6 +356,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; + u16 cmr; int start_event; int ret; int fslen, fslen_ext; @@ -626,7 +638,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, } /* set SSC clock mode register */ - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + cmr = ssc_p->tcmr_div ? ssc_p->tcmr_div : ssc_p->rcmr_div; + ssc_writel(ssc_p->ssc->regs, CMR, cmr); /* set receive clock mode and format */ ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d5..a25df7a 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -39,9 +39,10 @@ #define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ /* SSC divider ids */ -#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +#define ATMEL_SSC_TCMR_DIV 0 /* MCK divider for transmit BCLK */ +#define ATMEL_SSC_RCMR_DIV 1 /* MCK divider for receive BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 2 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 3 /* BCLK divider for receive FS */ /* * SSC direction masks */ @@ -110,7 +111,8 @@ struct atmel_ssc_info { unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ unsigned short initialized; /* true if SSC has been initialized */ unsigned short daifmt; - unsigned short cmr_div; + unsigned short tcmr_div; + unsigned short rcmr_div; unsigned short tcmr_period; unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; -- 1.7.10.4

2 11

[alsa-devel] [PATCH] ASoC: rt5677: add GPIO IRQ support
by Anatol Pomozov 22 Oct '14

22 Oct '14

This allows to enable Mic Jack detection feature Author: Oder Chiou <oder_chiou(a)realtek.com> Modified-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 10 ++ include/sound/rt5677.h | 7 ++ sound/soc/codecs/rt5677.c | 134 +++++++++++++++++++++ sound/soc/codecs/rt5677.h | 49 ++++++++ 4 files changed, 200 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index f82f0e9..740ff77 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -33,6 +33,15 @@ Optional properties: 1 - pull down 2 - pull up +- realtek,jd1-gpio + Configures GPIO Mic Jack detection 1. + Select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively. + +- realtek,jd2-gpio +- realtek,jd3-gpio + Configures GPIO Mic Jack detection 2 and 3. + Select 0 ~ 3 as OFF, GPIO4, GPIO5 and GPIO6 respectively. + Pins on the device (for linking into audio routes): * IN1P @@ -63,4 +72,5 @@ rt5677 { <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; realtek,in1-differential = "true"; realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; /* pull up GPIO6 */ + realtek,jd2-gpio = <3>; /* Enables Jack detection for GPIO6 */ }; diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h index a56b429..d9eb7d8 100644 --- a/include/sound/rt5677.h +++ b/include/sound/rt5677.h @@ -30,6 +30,13 @@ struct rt5677_platform_data { /* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */ u8 gpio_config[6]; + + /* jd1 can select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively */ + unsigned int jd1_gpio; + /* jd2 and jd3 can select 0 ~ 3 as + OFF, GPIO4, GPIO5 and GPIO6 respectively */ + unsigned int jd2_gpio; + unsigned int jd3_gpio; }; #endif diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f6df276..1f5897f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3614,6 +3614,46 @@ static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, } } +static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct regmap_irq_chip_data *data = rt5677->irq_data; + int irq; + + if (offset >= RT5677_GPIO1 && offset <= RT5677_GPIO3) { + if ((rt5677->pdata.jd1_gpio == 1 && offset == RT5677_GPIO1) || + (rt5677->pdata.jd1_gpio == 2 && + offset == RT5677_GPIO2) || + (rt5677->pdata.jd1_gpio == 3 && + offset == RT5677_GPIO3)) { + irq = RT5677_IRQ_JD1; + } else { + return -ENXIO; + } + } + + if (offset >= RT5677_GPIO4 && offset <= RT5677_GPIO6) { + if ((rt5677->pdata.jd2_gpio == 1 && offset == RT5677_GPIO4) || + (rt5677->pdata.jd2_gpio == 2 && + offset == RT5677_GPIO5) || + (rt5677->pdata.jd2_gpio == 3 && + offset == RT5677_GPIO6)) { + irq = RT5677_IRQ_JD2; + } else if ((rt5677->pdata.jd3_gpio == 1 && + offset == RT5677_GPIO4) || + (rt5677->pdata.jd3_gpio == 2 && + offset == RT5677_GPIO5) || + (rt5677->pdata.jd3_gpio == 3 && + offset == RT5677_GPIO6)) { + irq = RT5677_IRQ_JD3; + } else { + return -ENXIO; + } + } + + return regmap_irq_get_virq(data, irq); +} + static struct gpio_chip rt5677_template_chip = { .label = "rt5677", .owner = THIS_MODULE, @@ -3621,6 +3661,7 @@ static struct gpio_chip rt5677_template_chip = { .set = rt5677_gpio_set, .direction_input = rt5677_gpio_direction_in, .get = rt5677_gpio_get, + .to_irq = rt5677_to_irq, .can_sleep = 1, }; @@ -3688,6 +3729,31 @@ static int rt5677_probe(struct snd_soc_codec *codec) for (i = 0; i < RT5677_GPIO_NUM; i++) rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]); + if (rt5677->irq_data) { + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, 0x8000, + 0x8000); + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x0018, + 0x0008); + + if (rt5677->pdata.jd1_gpio) + regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1, + RT5677_SEL_GPIO_JD1_MASK, + rt5677->pdata.jd1_gpio << + RT5677_SEL_GPIO_JD1_SFT); + + if (rt5677->pdata.jd2_gpio) + regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1, + RT5677_SEL_GPIO_JD2_MASK, + rt5677->pdata.jd2_gpio << + RT5677_SEL_GPIO_JD2_SFT); + + if (rt5677->pdata.jd3_gpio) + regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1, + RT5677_SEL_GPIO_JD3_MASK, + rt5677->pdata.jd3_gpio << + RT5677_SEL_GPIO_JD3_SFT); + } + mutex_init(&rt5677->dsp_cmd_lock); return 0; @@ -3918,9 +3984,74 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) of_property_read_u8_array(np, "realtek,gpio-config", rt5677->pdata.gpio_config, RT5677_GPIO_NUM); + of_property_read_u32(np, "realtek,jd1-gpio", &rt5677->pdata.jd1_gpio); + of_property_read_u32(np, "realtek,jd2-gpio", &rt5677->pdata.jd2_gpio); + of_property_read_u32(np, "realtek,jd3-gpio", &rt5677->pdata.jd3_gpio); + return 0; } +static struct regmap_irq rt5677_irqs[] = { + [RT5677_IRQ_JD1] = { + .reg_offset = 0, + .mask = RT5677_EN_IRQ_GPIO_JD1, + }, + [RT5677_IRQ_JD2] = { + .reg_offset = 0, + .mask = RT5677_EN_IRQ_GPIO_JD2, + }, + [RT5677_IRQ_JD3] = { + .reg_offset = 0, + .mask = RT5677_EN_IRQ_GPIO_JD3, + }, +}; + +static struct regmap_irq_chip rt5677_irq_chip = { + .name = "rt5677", + .irqs = rt5677_irqs, + .num_irqs = ARRAY_SIZE(rt5677_irqs), + + .num_regs = 1, + .status_base = RT5677_IRQ_CTRL1, + .mask_base = RT5677_IRQ_CTRL1, + .mask_invert = 1, +}; + +int rt5677_irq_init(struct i2c_client *i2c) +{ + int ret; + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + + if (!rt5677->pdata.jd1_gpio && + !rt5677->pdata.jd2_gpio && + !rt5677->pdata.jd3_gpio) + return 0; + + if (!i2c->irq) { + dev_err(&i2c->dev, "No interrupt specified\n"); + return -EINVAL; + } + + ret = regmap_add_irq_chip(rt5677->regmap, i2c->irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, 0, + &rt5677_irq_chip, &rt5677->irq_data); + + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register IRQ chip: %d\n", ret); + return ret; + } + + return 0; +} + +void rt5677_irq_exit(struct i2c_client *i2c) +{ + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + + if (rt5677->irq_data) + regmap_del_irq_chip(i2c->irq, rt5677->irq_data); +} + static int rt5677_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -4018,6 +4149,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, } rt5677_init_gpio(i2c); + rt5677_irq_init(i2c); return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, rt5677_dai, ARRAY_SIZE(rt5677_dai)); @@ -4025,6 +4157,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, static int rt5677_i2c_remove(struct i2c_client *i2c) { + rt5677_irq_exit(i2c); + snd_soc_unregister_codec(&i2c->dev); rt5677_free_gpio(i2c); diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 20efa4a..d2c743c 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1368,6 +1368,48 @@ #define RT5677_SEL_SRC_IB01 (0x1 << 0) #define RT5677_SEL_SRC_IB01_SFT 0 +/* Jack Detect Control 1 (0xb5) */ +#define RT5677_SEL_GPIO_JD1_MASK (0x3 << 14) +#define RT5677_SEL_GPIO_JD1_SFT 14 +#define RT5677_SEL_GPIO_JD2_MASK (0x3 << 12) +#define RT5677_SEL_GPIO_JD2_SFT 12 +#define RT5677_SEL_GPIO_JD3_MASK (0x3 << 10) +#define RT5677_SEL_GPIO_JD3_SFT 10 + +/* IRQ Control 1 (0xbd) */ +#define RT5677_STA_GPIO_JD1 (0x1 << 15) +#define RT5677_STA_GPIO_JD1_SFT 15 +#define RT5677_EN_IRQ_GPIO_JD1 (0x1 << 14) +#define RT5677_EN_IRQ_GPIO_JD1_SFT 14 +#define RT5677_EN_GPIO_JD1_STICKY (0x1 << 13) +#define RT5677_EN_GPIO_JD1_STICKY_SFT 13 +#define RT5677_INV_GPIO_JD1 (0x1 << 12) +#define RT5677_INV_GPIO_JD1_SFT 12 +#define RT5677_STA_GPIO_JD2 (0x1 << 11) +#define RT5677_STA_GPIO_JD2_SFT 11 +#define RT5677_EN_IRQ_GPIO_JD2 (0x1 << 10) +#define RT5677_EN_IRQ_GPIO_JD2_SFT 10 +#define RT5677_EN_GPIO_JD2_STICKY (0x1 << 9) +#define RT5677_EN_GPIO_JD2_STICKY_SFT 9 +#define RT5677_INV_GPIO_JD2 (0x1 << 8) +#define RT5677_INV_GPIO_JD2_SFT 8 +#define RT5677_STA_MICBIAS1_OVCD (0x1 << 7) +#define RT5677_STA_MICBIAS1_OVCD_SFT 7 +#define RT5677_EN_IRQ_MICBIAS1_OVCD (0x1 << 6) +#define RT5677_EN_IRQ_MICBIAS1_OVCD_SFT 6 +#define RT5677_EN_MICBIAS1_OVCD_STICKY (0x1 << 5) +#define RT5677_EN_MICBIAS1_OVCD_STICKY_SFT 5 +#define RT5677_INV_MICBIAS1_OVCD (0x1 << 4) +#define RT5677_INV_MICBIAS1_OVCD_SFT 4 +#define RT5677_STA_GPIO_JD3 (0x1 << 3) +#define RT5677_STA_GPIO_JD3_SFT 3 +#define RT5677_EN_IRQ_GPIO_JD3 (0x1 << 2) +#define RT5677_EN_IRQ_GPIO_JD3_SFT 2 +#define RT5677_EN_GPIO_JD3_STICKY (0x1 << 1) +#define RT5677_EN_GPIO_JD3_STICKY_SFT 1 +#define RT5677_INV_GPIO_JD3 (0x1 << 0) +#define RT5677_INV_GPIO_JD3_SFT 0 + /* GPIO status (0xbf) */ #define RT5677_GPIO6_STATUS_MASK (0x1 << 5) #define RT5677_GPIO6_STATUS_SFT 5 @@ -1545,6 +1587,12 @@ enum { RT5677_GPIO_NUM, }; +enum { + RT5677_IRQ_JD1, + RT5677_IRQ_JD2, + RT5677_IRQ_JD3, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1565,6 +1613,7 @@ struct rt5677_priv { struct gpio_chip gpio_chip; #endif bool dsp_vad_en; + struct regmap_irq_chip_data *irq_data; }; #endif /* __RT5677_H__ */ -- 2.1.0.rc2.206.gedb03e5

3 5

[alsa-devel] [PATCH 01/12] ASoC: ak4535: Cleanup manual bias level transitions
by Lars-Peter Clausen 22 Oct '14

22 Oct '14

Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4535.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 30e2978..eced46d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -373,36 +373,19 @@ static struct snd_soc_dai_driver ak4535_dai = { .ops = &ak4535_dai_ops, }; -static int ak4535_suspend(struct snd_soc_codec *codec) -{ - ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ak4535_resume(struct snd_soc_codec *codec) { snd_soc_cache_sync(codec); - ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } static int ak4535_probe(struct snd_soc_codec *codec) { - /* power on device */ - ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); return 0; } -/* power down chip */ -static int ak4535_remove(struct snd_soc_codec *codec) -{ - ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static const struct regmap_config ak4535_regmap = { .reg_bits = 8, .val_bits = 8, @@ -417,10 +400,10 @@ static const struct regmap_config ak4535_regmap = { static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .probe = ak4535_probe, - .remove = ak4535_remove, - .suspend = ak4535_suspend, .resume = ak4535_resume, .set_bias_level = ak4535_set_bias_level, + .suspend_bias_off = true, + .dapm_widgets = ak4535_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets), .dapm_routes = ak4535_audio_map, -- 1.8.0

2 12

[alsa-devel] [PATCH next 1/2] ALSA: pcm: Remove arch-dependent mmap kludges
by Takashi Iwai 22 Oct '14

22 Oct '14

Since we have consistently dma_mmap_coherent() for all architectures, the current ifdef and arch-specific codes in pcm core can be cleaned up gracefully. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 30 ------------------------------ 1 file changed, 30 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 815396d8427f..aa6754da7929 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -35,9 +35,6 @@ #include <sound/timer.h> #include <sound/minors.h> #include <asm/io.h> -#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) -#include <dma-coherence.h> -#endif /* * Compatibility @@ -3251,20 +3248,6 @@ static inline struct page * snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) { void *vaddr = substream->runtime->dma_area + ofs; -#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) - return virt_to_page(CAC_ADDR(vaddr)); -#endif -#if defined(CONFIG_PPC32) && defined(CONFIG_NOT_COHERENT_CACHE) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { - dma_addr_t addr = substream->runtime->dma_addr + ofs; - addr -= get_dma_offset(substream->dma_buffer.dev.dev); - /* assume dma_handle set via pfn_to_phys() in - * mm/dma-noncoherent.c - */ - return pfn_to_page(addr >> PAGE_SHIFT); - } -#endif return virt_to_page(vaddr); } @@ -3309,13 +3292,6 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { .fault = snd_pcm_mmap_data_fault, }; -#ifndef ARCH_HAS_DMA_MMAP_COHERENT -/* This should be defined / handled globally! */ -#if defined(CONFIG_ARM) || defined(CONFIG_ARM64) -#define ARCH_HAS_DMA_MMAP_COHERENT -#endif -#endif - /* * mmap the DMA buffer on RAM */ @@ -3331,7 +3307,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_end - area->vm_start, area->vm_page_prot); } #endif /* CONFIG_GENERIC_ALLOCATOR */ -#ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, @@ -3339,11 +3314,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); -#elif defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV && - !plat_device_is_coherent(substream->dma_buffer.dev.dev)) - area->vm_page_prot = pgprot_noncached(area->vm_page_prot); -#endif /* ARCH_HAS_DMA_MMAP_COHERENT */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; -- 2.1.2

1 1

[alsa-devel] [PATCH 0/2] ASoC: spread usage of snd_ctl_elem_info()
by Takashi Iwai 22 Oct '14

22 Oct '14

Hi, just like my previous patch series, this convers the simple open-code for enum info callbacks with the existing snd_ctl_enum_info() helper. They can be applied individually, as far as I know, but I think it'd fit better to the same branch for other similar patches. Takashi

3 5

[alsa-devel] [PATCH V2 1/2] ARM: pxa: spitz: register spitz-audio device
by Dmitry Eremin-Solenikov 22 Oct '14

22 Oct '14

Register spitz-audio device to be used by ASoC driver. [V2: Removed ifdefs around spitz_audio_init]. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- arch/arm/mach-pxa/spitz.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 840c3a4..af1dea3 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -924,6 +924,19 @@ static inline void spitz_i2c_init(void) {} #endif /****************************************************************************** + * Audio devices + ******************************************************************************/ +static struct platform_device spitz_audio_device = { + .name = "spitz-audio", + .id = -1, +}; + +static inline void spitz_audio_init(void) +{ + platform_device_register(&spitz_audio_device); +} + +/****************************************************************************** * Machine init ******************************************************************************/ static void spitz_poweroff(void) @@ -970,6 +983,7 @@ static void __init spitz_init(void) spitz_nor_init(); spitz_nand_init(); spitz_i2c_init(); + spitz_audio_init(); } static void __init spitz_fixup(struct tag *tags, char **cmdline) -- 2.1.1

2 3

[alsa-devel] [PATCH] ASoC: Intel: Fix block is enabled multiple times issue
by Jie Yang 22 Oct '14

22 Oct '14

During FW parsing and loading, block_list_prepare() may be called for each raw data block copying and this may made the hsw_block_enable() called mutiple times, which increase block->users many times. The result of this is hsw_block_disable() can't power gated the related block when trying to free the blocks during suspend, and the power gating status also confused. Here check the block user status, only calling enable() for those blocks who has no user yet. Remember that this works correctlly on current case, where there are enough SRAM memory so different module won't share a memory block. For further usage, we may need restructure the struct sst_mem_block to save the module list who is using it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index f2f1d5f..db60398 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -80,7 +80,7 @@ static int block_list_prepare(struct sst_dsp *dsp, /* enable each block so that's it'e ready for data */ list_for_each_entry(block, block_list, module_list) { - if (block->ops && block->ops->enable) { + if (block->ops && block->ops->enable && !block->users) { ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: adau1761: Fix input PGA volume
by Lars-Peter Clausen 22 Oct '14

22 Oct '14

For the input PGA to work correctly the ALC clock needs to be active. Otherwise volume changes are not applied. Fixes: dab464b60b2 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/adau1761.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 5518ebd..91f6028 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -405,6 +405,7 @@ static const struct snd_soc_dapm_widget adau1761_dapm_widgets[] = { 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Slew Clock", ADAU1761_CLK_ENABLE0, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ALC Clock", ADAU1761_CLK_ENABLE0, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Digital Clock 0", 1, ADAU1761_CLK_ENABLE1, 0, 0, NULL, 0), @@ -436,6 +437,9 @@ static const struct snd_soc_dapm_route adau1761_dapm_routes[] = { { "Right Playback Mixer", NULL, "Slew Clock" }, { "Left Playback Mixer", NULL, "Slew Clock" }, + { "Left Input Mixer", NULL, "ALC Clock" }, + { "Right Input Mixer", NULL, "ALC Clock" }, + { "Digital Clock 0", NULL, "SYSCLK" }, { "Digital Clock 1", NULL, "SYSCLK" }, }; -- 1.8.0

2 1

[alsa-devel] [PATCH 0/5] Add Phytec pbab01 audio support
by Dmitry Lavnikevich 22 Oct '14

22 Oct '14

This patchset implements audio support on Phytec's phyFLEX-i.MX6 boards. This includes creating imx-tlv320aic3x machine driver, fixing tlv320aic3x driver playback speed issue and enabling audio for pbab01 itself.

10 44

[alsa-devel] [PATCH v2 0/2] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 22 Oct '14

22 Oct '14

This patch set adds multi-CODEC support to the simple card. v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (2): ASoC: core: add multi-codec support in DT ASoC: simple-card: add multi-CODECs in DT .../devicetree/bindings/sound/simple-card.txt | 28 +++-- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 114 +++++++++++++-------- sound/soc/soc-core.c | 75 +++++++++++--- 4 files changed, 149 insertions(+), 71 deletions(-) -- 2.1.1

1 2

[alsa-devel] Alsa-lib 32-bit on 64-bit machine
by Alan Horstmann 22 Oct '14

22 Oct '14

Hi, I've tried to find info or examples of this on the web, but to no avail. What is the recommended way to build Alsa-lib on a 64-bit machine to create the additional alternative 32-bit libraries for compatibility with 32-bit applications? The original distro (in this case a Suse Enterprise) has them, but standard './configure make install' only builds the 64-bit variants. I only succeeded by explicitly adding CFLAGS, and wondered if it should work using configure options, or another way? I'm also not sure exactly which flags should be set with that method (m32, march=i686 others?). Does anyone know how it is done in the Suse builds? Thanks, Alan

2 3

[alsa-devel] [RFC PATCH 1/2] ALSA: hda - Fix "PCM" name being used on one DAC when there are two DACs
by David Henningsson 22 Oct '14

22 Oct '14

In the scenario where there is one "Line Out", one "Speaker" and one "Headphone", and there are only two DACs, two outputs will share a DAC. Currently any mixer on such a DAC will get the "PCM" name, which is misleading. Instead use "Headphone+LO" or "Speaker+LO" to better specify what the volume actually controls. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_generic.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 64220c0..1d742ea 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1038,6 +1038,19 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, break; *index = ch; return "Headphone"; + case AUTO_PIN_LINE_OUT: + /* This deals with the case where we have two DACs and + * one LO, one HP and one Speaker */ + if (!ch && cfg->speaker_outs && cfg->hp_outs) { + bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); + bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + if (hp_lo_shared && spk_lo_shared) + return spec->vmaster_mute.hook ? "PCM" : "Master"; + if (hp_lo_shared) + return "Headphone+LO"; + if (spk_lo_shared) + return "Speaker+LO"; + } } /* for a single channel output, we don't have to name the channel */ -- 1.9.1

4 28

[alsa-devel] [PATCH] ASoC: simple-card: Add mic and hp detect gpios.
by Dylan Reid 22 Oct '14

22 Oct '14

Allow Headphone and Microphone jack detect gpios to be specified in device tree. This will allow a few systems including rk3288_max98090 to use simple-card instead of having their own board file. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- .../devicetree/bindings/sound/simple-card.txt | 4 ++ sound/soc/generic/simple-card.c | 73 ++++++++++++++++++++++ 2 files changed, 77 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c2e9841..72d94b7 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -17,6 +17,10 @@ Optional properties: source. - simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec mclk. +- simple-audio-card,hp_det_gpio : Reference to GPIO that signals when + headphones are attached. +- simple-audio-card,mic_det_gpio : Reference to GPIO that signals when + a microphone is attached. Optional subnodes: diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 709ce67..fcb431f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -10,10 +10,13 @@ */ #include <linux/clk.h> #include <linux/device.h> +#include <linux/gpio.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_gpio.h> #include <linux/platform_device.h> #include <linux/string.h> +#include <sound/jack.h> #include <sound/simple_card.h> #include <sound/soc-dai.h> #include <sound/soc.h> @@ -25,6 +28,8 @@ struct simple_card_data { struct asoc_simple_dai codec_dai; } *dai_props; unsigned int mclk_fs; + int gpio_hp_det; + int gpio_mic_det; struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; @@ -54,6 +59,32 @@ static struct snd_soc_ops asoc_simple_card_ops = { .hw_params = asoc_simple_card_hw_params, }; +static struct snd_soc_jack simple_card_hp_jack; +static struct snd_soc_jack_pin simple_card_hp_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, +}; +static struct snd_soc_jack_gpio simple_card_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack simple_card_mic_jack; +static struct snd_soc_jack_pin simple_card_mic_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; +static struct snd_soc_jack_gpio simple_card_mic_jack_gpio = { + .name = "Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set) { @@ -109,6 +140,28 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + if (gpio_is_valid(priv->gpio_hp_det)) { + snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE, + &simple_card_hp_jack); + snd_soc_jack_add_pins(&simple_card_hp_jack, + ARRAY_SIZE(simple_card_hp_jack_pins), + simple_card_hp_jack_pins); + + simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; + snd_soc_jack_add_gpios(&simple_card_hp_jack, 1, + &simple_card_hp_jack_gpio); + } + + if (gpio_is_valid(priv->gpio_mic_det)) { + snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE, + &simple_card_mic_jack); + snd_soc_jack_add_pins(&simple_card_mic_jack, + ARRAY_SIZE(simple_card_mic_jack_pins), + simple_card_mic_jack_pins); + simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; + snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, + &simple_card_mic_jack_gpio); + } return 0; } @@ -383,6 +436,16 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + priv->gpio_hp_det = of_get_named_gpio(node, + "simple-audio-card,hp-det-gpio", 0); + if (priv->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + priv->gpio_mic_det = of_get_named_gpio(node, + "simple-audio-card,mic-det-gpio", 0); + if (priv->gpio_mic_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (!priv->snd_card.name) priv->snd_card.name = priv->snd_card.dai_link->name; @@ -502,6 +565,16 @@ err: static int asoc_simple_card_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(priv->gpio_hp_det)) + snd_soc_jack_free_gpios(&simple_card_hp_jack, 1, + &simple_card_hp_jack_gpio); + if (gpio_is_valid(priv->gpio_mic_det)) + snd_soc_jack_free_gpios(&simple_card_mic_jack, 1, + &simple_card_mic_jack_gpio); + return asoc_simple_card_unref(pdev); } -- 2.1.0.rc2.206.gedb03e5

7 11

[alsa-devel] [PATCH v3 0/5] ASoC: intel: fix to sst driver series
by Vinod Koul 22 Oct '14

22 Oct '14

This series fixes comments observed while applying the sst series last week. changes in v3: - fix the makefile artifact not required changes in v2: - typo fixes - make wrapper inline - add newline on err log Vinod Koul (5): ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: intel: log an error on double free ASoC: intel: fix the kernldoc comment ASoC: intel: explain why block not found isn't error always ASoC: intel: use __iowrite32_copy for 32 bit copy sound/soc/intel/sst-firmware.c | 9 +++------ sound/soc/intel/sst/sst_drv_interface.c | 4 +++- sound/soc/intel/sst/sst_ipc.c | 15 +++++++++++++++ sound/soc/intel/sst/sst_loader.c | 12 +++++------- 4 files changed, 26 insertions(+), 14 deletions(-)

2 6

[alsa-devel] [PATCH 1/2] MAINTAINERS: add atmel ssc driver maintainer entry
by Bo Shen 21 Oct '14

21 Oct '14

Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 3705430..c004e6f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1703,6 +1703,13 @@ M: Nicolas Ferre <nicolas.ferre(a)atmel.com> S: Supported F: drivers/spi/spi-atmel.* +ATMEL SSC DRIVER +M: Bo Shen <voice.shen(a)atmel.com> +L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) +S: Supported +F: drivers/misc/atmel-ssc.c +F: include/linux/atmel-ssc.h + ATMEL Timer Counter (TC) AND CLOCKSOURCE DRIVERS M: Nicolas Ferre <nicolas.ferre(a)atmel.com> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) -- 2.1.0.24.g4109c28

4 6

[alsa-devel] [PATCH] ASoC: rt5677: fix rt5677 spi driver build
by Ben Zhang 21 Oct '14

21 Oct '14

Create a separate module for rt5677 spi driver. Without this patch, the build fails due to multiple defs of 'init_module' and 'cleanup_module'. module_spi_driver() defines its own module, so it can't be part of the rt5677 module. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/Kconfig | 4 ++++ sound/soc/codecs/Makefile | 4 +++- sound/soc/codecs/rt5677-spi.c | 2 ++ 3 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2c7482e..6f21a76 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -504,6 +504,10 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate +config SND_SOC_RT5677_SPI + tristate + default SND_SOC_RT5677 + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4435f9f..3e57edc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -79,7 +79,8 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5670-objs := rt5670.o -snd-soc-rt5677-objs := rt5677.o rt5677-spi.o +snd-soc-rt5677-objs := rt5677.o +snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -256,6 +257,7 @@ obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o +obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 11c38f3..ef6348c 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -52,6 +52,7 @@ int rt5677_spi_write(u8 *txbuf, size_t len) return status; } +EXPORT_SYMBOL_GPL(rt5677_spi_write); /** * rt5677_spi_burst_write - Write data to SPI by rt5677 dsp memory address. @@ -107,6 +108,7 @@ int rt5677_spi_burst_write(u32 addr, const struct firmware *fw) return 0; } +EXPORT_SYMBOL_GPL(rt5677_spi_burst_write); static int rt5677_spi_probe(struct spi_device *spi) { -- 2.1.0.rc2.206.gedb03e5

2 1

[alsa-devel] [PATCH] ASoC: dapm: Remove redundant cast
by Rasmus Villemoes 21 Oct '14

21 Oct '14

Both path->name and e->texts[i] have type const char*, so the cast is slightly confusing and certainly unnecessary. Signed-off-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9c..39f992b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -496,7 +496,7 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); - path->name = (char*)e->texts[i]; + path->name = e->texts[i]; if (i == item) path->connect = 1; else -- 2.0.4

2 1

[alsa-devel] [PATCH] ASoC: tegra: Read and use the GPIO flags of the headphone detect
by Alban Bedel 21 Oct '14

21 Oct '14

The headphone detect was hardcoded to low-active, use the flags from DT to allow high-active as well. Signed-off-by: Alban Bedel <alban.bedel(a)avionic-design.de> --- sound/soc/tegra/tegra_rt5640.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index a689883..4ebe387 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -44,6 +44,7 @@ struct tegra_rt5640 { struct tegra_asoc_utils_data util_data; int gpio_hp_det; + enum of_gpio_flags gpio_hp_det_flags; }; static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, @@ -119,6 +120,8 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; + tegra_rt5640_hp_jack_gpio.invert = + !!(machine->gpio_hp_det_flags & OF_GPIO_ACTIVE_LOW); snd_soc_jack_add_gpios(&tegra_rt5640_hp_jack, 1, &tegra_rt5640_hp_jack_gpio); @@ -180,7 +183,8 @@ static int tegra_rt5640_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + machine->gpio_hp_det = of_get_named_gpio_flags( + np, "nvidia,hp-det-gpios", 0, &machine->gpio_hp_det_flags); if (machine->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; -- 2.1.1

3 2

[alsa-devel] [PATCH 1/2] ARM: pxa: spitz: register spitz-audio device
by Dmitry Eremin-Solenikov 21 Oct '14

21 Oct '14

Register spitz-audio device to be used by ASoC driver. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- arch/arm/mach-pxa/spitz.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 840c3a4..a9f971f 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -924,6 +924,23 @@ static inline void spitz_i2c_init(void) {} #endif /****************************************************************************** + * Audio devices + ******************************************************************************/ +#if defined(CONFIG_SND_PXA2XX_SOC_SPITZ) || defined(SND_PXA2XX_SOC_SPITZ_MODULE) +static struct platform_device spitz_audio_device = { + .name = "spitz-audio", + .id = -1, +}; + +static inline void spitz_audio_init(void) +{ + platform_device_register(&spitz_audio_device); +} +#else +static inline void spitz_audio_init(void) {} +#endif + +/****************************************************************************** * Machine init ******************************************************************************/ static void spitz_poweroff(void) @@ -970,6 +987,7 @@ static void __init spitz_init(void) spitz_nor_init(); spitz_nand_init(); spitz_i2c_init(); + spitz_audio_init(); } static void __init spitz_fixup(struct tag *tags, char **cmdline) -- 2.1.1

2 3

[alsa-devel] 3.18.0-rc1 pulseaudio possible recursive locking
by Arthur Marsh 21 Oct '14

21 Oct '14

Hi, I have been seeing this a few times lately, and noticed that it was still present Is this the right place to report it or should I be reporting a bug against pulseaudio? [ 182.273991] ============================================= [ 182.273996] [ INFO: possible recursive locking detected ] [ 182.274003] 3.18.0-rc1 #1297 Not tainted [ 182.274008] --------------------------------------------- [ 182.274013] pulseaudio/5183 is trying to acquire lock: [ 182.274018] (&(&substream->self_group.lock)->rlock/1){......}, at: [<f8432284>] snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274063] but task is already holding lock: [ 182.274069] (&(&substream->self_group.lock)->rlock/1){......}, at: [<f8432284>] snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274092] other info that might help us debug this: [ 182.274098] Possible unsafe locking scenario: [ 182.274104] CPU0 [ 182.274108] ---- [ 182.274112] lock(&(&substream->self_group.lock)->rlock/1); [ 182.274122] lock(&(&substream->self_group.lock)->rlock/1); [ 182.274132] *** DEADLOCK *** [ 182.274138] May be due to missing lock nesting notation [ 182.274146] 4 locks held by pulseaudio/5183: [ 182.274151] #0: (snd_pcm_link_rwlock){......}, at: [<f8432fe3>] snd_pcm_stream_lock+0x1b/0x43 [snd_pcm] [ 182.274177] #1: (&(&substream->self_group.lock)->rlock){......}, at: [<f8432fee>] snd_pcm_stream_lock+0x26/0x43 [snd_pcm] [ 182.274202] #2: (&(&substream->group->lock)->rlock){......}, at: [<f8432cf4>] snd_pcm_action+0x34/0x10b [snd_pcm] [ 182.274227] #3: (&(&substream->self_group.lock)->rlock/1){......}, at: [<f8432284>] snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274254] stack backtrace: [ 182.274263] CPU: 1 PID: 5183 Comm: pulseaudio Not tainted 3.18.0-rc1 #1297 [ 182.274269] Hardware name: System Manufacturer System Name/P4S800, BIOS ASUS P4S800 ACPI BIOS Revision 1011 Beta 001 08/30/2005 [ 182.274275] c1bfc4e0 00000000 e55c9d50 c145d969 c1bfc4e0 e55c9dd4 c1082b7b c15b7f9a [ 182.274295] e55b630c 0000143f 00000000 e55b6010 e55b6070 00000000 5402a095 00000004 [ 182.274315] c17a4034 c1bfc4e0 e55b6508 e55b6520 00000000 00000008 00000000 e55b6508 [ 182.274335] Call Trace: [ 182.274353] [<c145d969>] dump_stack+0x4b/0x75 [ 182.274366] [<c1082b7b>] __lock_acquire+0x187c/0x19b3 [ 182.274378] [<c10834ad>] lock_acquire+0xa1/0x126 [ 182.274394] [<f8432284>] ? snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274404] [<c1463a9d>] _raw_spin_lock_nested+0x41/0x4e [ 182.274420] [<f8432284>] ? snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274436] [<f8432284>] snd_pcm_action_group+0x96/0x215 [snd_pcm] [ 182.274445] [<c146393b>] ? _raw_spin_trylock+0x6b/0x80 [ 182.274462] [<f8432d0f>] snd_pcm_action+0x4f/0x10b [snd_pcm] [ 182.274479] [<f843332a>] snd_pcm_drop+0x4a/0x7d [snd_pcm] [ 182.274496] [<f84351d9>] snd_pcm_common_ioctl1+0x7b0/0xcc4 [snd_pcm] [ 182.274514] [<f84357ce>] snd_pcm_playback_ioctl1+0xe1/0x3a8 [snd_pcm] [ 182.274532] [<f8435ab7>] snd_pcm_playback_ioctl+0x22/0x35 [snd_pcm] [ 182.274549] [<f8435a95>] ? snd_pcm_playback_ioctl1+0x3a8/0x3a8 [snd_pcm] [ 182.274561] [<c1171e82>] do_vfs_ioctl+0x306/0x576 [ 182.274573] [<c117d3ca>] ? mntput_no_expire+0x61/0x1e7 [ 182.274583] [<c117d377>] ? mntput_no_expire+0xe/0x1e7 [ 182.274594] [<c117d56b>] ? mntput+0x1b/0x29 [ 182.274603] [<c1161438>] ? __fput+0x144/0x1aa [ 182.274614] [<c1172147>] SyS_ioctl+0x55/0x75 [ 182.274624] [<c14646e0>] sysenter_do_call+0x12/0x12 Regards, Arthur.

2 3

[alsa-devel] [PATCH] ALSA: ctxfi: remove unused variable
by Sudip Mukherjee 21 Oct '14

21 Oct '14

As of now the pointer to struct dai is not being used anywhere in the function. So it is safe to remove the variable. If we are ever doing anything with the container_of(daio, struct dai, daio), then at that time we can again add the variable. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/ctxfi/ctatc.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 4546590..632e843 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1145,7 +1145,6 @@ static int atc_release_resources(struct ct_atc *atc) int i; struct daio_mgr *daio_mgr = NULL; struct dao *dao = NULL; - struct dai *dai = NULL; struct daio *daio = NULL; struct sum_mgr *sum_mgr = NULL; struct src_mgr *src_mgr = NULL; @@ -1172,9 +1171,6 @@ static int atc_release_resources(struct ct_atc *atc) dao = container_of(daio, struct dao, daio); dao->ops->clear_left_input(dao); dao->ops->clear_right_input(dao); - } else { - dai = container_of(daio, struct dai, daio); - /* some thing to do for dai ... */ } daio_mgr->put_daio(daio_mgr, daio); } -- 1.8.1.2

2 1

[alsa-devel] [PATCH 00/43] Spread usage of snd_ctl_elem_info()
by Takashi Iwai 21 Oct '14

21 Oct '14

Hi, this is a series of patches to just convert the plain info callback for enum ctl elements to snd_ctl_elem_info(). Also, it includes the extension of snd_ctl_elem_info(), for catching the unexpected string cut-off and handling the zero items. Takashi

1 43

[alsa-devel] bug in alsa-utils 1.0.28 - arecord
by Jordan Kagan 21 Oct '14

21 Oct '14

Hello, I encountered an issue with using the arecord binary in this most recent version. When using the '-d' duration flag, the program will continue to open files AFTER the duration has expired. The issue occurs when the capture() while loop hits the break due to a value returned by pcm_read(). When the while loop is exited, the program hits the do-while loop and continues to open files indefinitely. If rcount is returned from pcm_read() instead of "result", the problem does not occur. Thanks, Jordan

2 1

Re: [alsa-devel] [PATCH v2] ASoC: simple-card: add asoc_simple_card_parse_daifmt()
by Mark Brown 21 Oct '14

21 Oct '14

On Tue, Sep 23, 2014 at 07:29:34PM -0700, Kuninori Morimoto wrote: > > Hi Mark > > Should I re-send this patch ? Don't top post... I've been at Linaro Connect so haven't had a chance to look at it.

2 3

[alsa-devel] [PATCH] ALSA: Update control names documentation
by David Henningsson 21 Oct '14

21 Oct '14

This document was not really up-to-date. Add recent additions to this standard - based on what the HDA driver currently does, which is some kind of a de facto standard. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/ControlNames.txt | 32 ++++++++++++++++++++++++++----- 1 file changed, 27 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index fea65bb..79a6127 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -1,6 +1,6 @@ This document describes standard names of mixer controls. -Syntax: SOURCE [DIRECTION] FUNCTION +Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION DIRECTION: <nothing> (both directions) @@ -14,12 +14,29 @@ FUNCTION: Volume Route (route control, hardware specific) +CHANNEL: + <nothing> (channel independent, or applies to all channels) + Front + Surround (rear left/right in 4.0/5.1 surround) + CLFE + Center + LFE + Side (side left/right for 7.1 surround) + +LOCATION: (physical location of source) + Front + Rear + Dock (docking station) + Internal + SOURCE: Master Master Mono Hardware Master Speaker (internal speaker) + Bass Speaker (internal LFE speaker) Headphone + Line Out Beep (beep generator) Phone Phone Input @@ -27,14 +44,14 @@ SOURCE: Synth FM Mic - Line + Headset Mic (mic part of combined headset jack - 4-pin headphone + mic) + Headphone Mic (mic part of either/or - 3-pin headphone or mic) + Line (input only, use "Line Out" for output) CD Video Zoom Video Aux PCM - PCM Front - PCM Rear PCM Pan Loopback Analog Loopback (D/A -> A/D loopback) @@ -47,8 +64,13 @@ SOURCE: Music I2S IEC958 + HDMI + SPDIF (output only) + SPDIF In + Digital In + HDMI/DP (either HDMI or DisplayPort) -Exceptions: +Exceptions (deprecated): [Digital] Capture Source [Digital] Capture Switch (aka input gain switch) [Digital] Capture Volume (aka input gain volume) -- 1.9.1

3 3

[alsa-devel] Advanced Gtk+ Sequencer 0.4.2
by Joël Krähemann 20 Oct '14

20 Oct '14

Advanced Gtk+ Sequencer 0.4.2 is out Check download at: http://sourceforge.net/projects/ags/files/ags-0_4_2.tar.bz2/download or API Reference by issuing: http://ags.sourceforge.net/api/ags/index.html Advanced GTK+ Sequencer is intended to use for music composition. Use its piano roll, synth, matrix editor, drum machine, soundfont2 player, mixer or the output panel.

1 0

[alsa-devel] [PATCH v4 0/7] ASoC: Intel: sst - add the merrifield IPC driver
by Vinod Koul 20 Oct '14

20 Oct '14

This path series adds the Merrifield IPC driver which is used to load and manage the DSP along with sending and receiving IPCs for PCM and compressed audio. v4 updates: - fix the comment style issues - fix for id allocation handling - modularize the code to remove few repated patterns - comments from previos rev (hopefully didnt miss anything) Vinod Koul (7): ASoC: Intel: mrfld - add the dsp sst driver ASoC: Intel: mrfld - Add DSP load and management ASoC: Intel: sst - add pcm ops handling ASoC: Intel: sst: Add IPC handling ASoC: Intel: sst: add stream operations ASoC: Intel: sst: Add some helper functions ASoC: Intel: sst: Add makefile and kconfig changes arch/x86/include/asm/platform_sst_audio.h | 34 ++ sound/soc/intel/Kconfig | 4 + sound/soc/intel/Makefile | 3 + sound/soc/intel/sst/Makefile | 3 + sound/soc/intel/sst/sst.c | 434 +++++++++++++++++++++++ sound/soc/intel/sst/sst.h | 539 +++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_drv_interface.c | 403 +++++++++++++++++++++ sound/soc/intel/sst/sst_ipc.c | 358 +++++++++++++++++++ sound/soc/intel/sst/sst_loader.c | 461 ++++++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 446 ++++++++++++++++++++++++ sound/soc/intel/sst/sst_stream.c | 437 +++++++++++++++++++++++ 11 files changed, 3122 insertions(+), 0 deletions(-) create mode 100644 sound/soc/intel/sst/Makefile create mode 100644 sound/soc/intel/sst/sst.c create mode 100644 sound/soc/intel/sst/sst.h create mode 100644 sound/soc/intel/sst/sst_drv_interface.c create mode 100644 sound/soc/intel/sst/sst_ipc.c create mode 100644 sound/soc/intel/sst/sst_loader.c create mode 100644 sound/soc/intel/sst/sst_pvt.c create mode 100644 sound/soc/intel/sst/sst_stream.c

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[alsa-devel] [PATCH 1/2] ALSA: snd-usb: drop unused varible assigments
by Daniel Mack 20 Oct '14

20 Oct '14

Don't assign 'len' in cases where we don't make use of the returned value. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/mixer.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2e4a9db..63a8adb 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1290,9 +1290,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, kctl->id.name, sizeof(kctl->id.name), 1); if (!len) - len = snprintf(kctl->id.name, - sizeof(kctl->id.name), - "Feature %d", unitid); + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Feature %d", unitid); } if (!mapped_name) @@ -1305,9 +1304,9 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, */ if (!mapped_name && !(state->oterm.type >> 16)) { if ((state->oterm.type & 0xff00) == 0x0100) - len = append_ctl_name(kctl, " Capture"); + append_ctl_name(kctl, " Capture"); else - len = append_ctl_name(kctl, " Playback"); + append_ctl_name(kctl, " Playback"); } append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); -- 2.1.0

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[alsa-devel] [PATCH 0/7] ASoC: Intel: Update HSW/BDW SST driver to new FW architecture
by Liam Girdwood 20 Oct '14

20 Oct '14

The following patch series updates the current upstream driver to use the new more feature rich ADSP FW architecture. The new FW arch allows for multiple instantiation of objects in dynamic memory locations. In summary :- o Modules can be created multiple times. o Modules persistent data can be allocated in any free blocks o Global scratch data can be allocated into any free blocks. o DMA can be used to load the module data. The driver now has to track and manage the new multiple module resource allocations and pass this data to FW on stream initialisations. Thanks Liam

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