November 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2] Digidesign Mbox 1 duplex mode and mixer
by Damien Zammit 10 Nov '14

10 Nov '14

Takashi asked me to rewrite the patches and send them without attachments. I hope this email series works and they can be reviewed. These two patches provide the following for Digidesign Mbox 1: o Mixer control for clock source selection: Internal/SPDIF o Duplex mode (playback and capture) Damien Zammit (2): snd-usb-audio: Add mixer control for Digidesign Mbox 1 clock source snd-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 | 177 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 73 ++++++++++++------- sound/usb/quirks.c | 30 ++++++++ sound/usb/usbaudio.h | 1 + 6 files changed, 271 insertions(+), 24 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 0/2] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 09 Nov '14

09 Nov '14

This patch set adds multi-CODEC support to the simple card. v3: rebase on broonie git 2014/11/03 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 | 135 +++++++++++++-------- sound/soc/soc-core.c | 75 +++++++++--- 4 files changed, 157 insertions(+), 84 deletions(-) -- 2.1.1

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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

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[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

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[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

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[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

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[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.

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[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

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[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

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