April 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: es8328: Remove extra blank lines
by Narcisa Ana Maria Vasile 07 Apr '17

07 Apr '17

This was reported by checkpatch.pl Signed-off-by: Narcisa Ana Maria Vasile <narcisaanamaria12(a)gmail.com> --- sound/soc/codecs/es8328.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3f84fbd..1967157 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -177,8 +177,6 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, return 0; } - - static const struct snd_kcontrol_new es8328_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Digital Volume", ES8328_ADCCONTROL8, ES8328_ADCCONTROL9, -- 1.9.1

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[alsa-devel] [PATCH v2 1/8] regulator: arizona-micsupp: Avoid potential memory leak reading init_data
by Charles Keepax 06 Apr '17

06 Apr '17

The device argument passed to of_get_regulator_init_data is used to do some devres memory allocation. Currently the driver passes the MFD device pointer to this function, this could result in the init_data allocation being leaked if the regulator is unbound but the MFD isn't. Correct this issue by correctly passing the local platform device. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- New patch in the series since v1. drivers/regulator/arizona-micsupp.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/drivers/regulator/arizona-micsupp.c b/drivers/regulator/arizona-micsupp.c index 22bd714..5e38861 100644 --- a/drivers/regulator/arizona-micsupp.c +++ b/drivers/regulator/arizona-micsupp.c @@ -199,7 +199,8 @@ static const struct regulator_init_data arizona_micsupp_ext_default = { .num_consumer_supplies = 1, }; -static int arizona_micsupp_of_get_pdata(struct arizona *arizona, +static int arizona_micsupp_of_get_pdata(struct device *dev, + struct arizona *arizona, struct regulator_config *config, const struct regulator_desc *desc) { @@ -213,7 +214,7 @@ static int arizona_micsupp_of_get_pdata(struct arizona *arizona, if (np) { config->of_node = np; - init_data = of_get_regulator_init_data(arizona->dev, np, desc); + init_data = of_get_regulator_init_data(dev, np, desc); if (init_data) { init_data->consumer_supplies = &micsupp->supply; @@ -268,8 +269,8 @@ static int arizona_micsupp_probe(struct platform_device *pdev) if (IS_ENABLED(CONFIG_OF)) { if (!dev_get_platdata(arizona->dev)) { - ret = arizona_micsupp_of_get_pdata(arizona, &config, - desc); + ret = arizona_micsupp_of_get_pdata(&pdev->dev, arizona, + &config, desc); if (ret < 0) return ret; } -- 2.1.4

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[alsa-devel] [PATCH 0/2] ASoC: codec: wm8960: Relax bit clock computation when using PLL
by Daniel Baluta 06 Apr '17

06 Apr '17

This is a follow up of patch series sent here: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119517.html where only patch 1/2 was merged. In this new patch series, we fix a problem with patch already merged and then we relax bit clock computation when using PLL the same as we did in patch 2/2 of previous series. Daniel Baluta (2): ASoC: codec: wm8960: Stop when a matching PLL freq is found ASoC: codec: wm8960: Relax bit clock computation when using PLL sound/soc/codecs/wm8960.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 1/3] ASoC: cs35l35: Improve power down time
by Charles Keepax 06 Apr '17

06 Apr '17

Shorten the time it takes to power down the amp by disabling the volume ramp whilst doing the final shutdown. The driver has already muted the amplifier at this stage so doing the volume ramp serves no purpose. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs35l35.c | 8 ++++++++ sound/soc/codecs/cs35l35.h | 3 +++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 9688274..1db07a6 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -187,6 +187,10 @@ static int cs35l35_sdin_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, CS35L35_PDN_ALL_MASK, 1); + /* Already muted, so disable volume ramp for faster shutdown */ + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_DIG_VOL_CTL, + CS35L35_AMP_DIGSFT_MASK, 0); + reinit_completion(&cs35l35->pdn_done); ret = wait_for_completion_timeout(&cs35l35->pdn_done, @@ -199,6 +203,10 @@ static int cs35l35_sdin_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, CS35L35_MCLK_DIS_MASK, 1 << CS35L35_MCLK_DIS_SHIFT); + + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_DIG_VOL_CTL, + CS35L35_AMP_DIGSFT_MASK, + 1 << CS35L35_AMP_DIGSFT_SHIFT); break; default: dev_err(codec->dev, "Invalid event = 0x%x\n", event); diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index 156d2f0..54e9ac5 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -190,6 +190,9 @@ #define CS35L35_AMP_GAIN_ZC_MASK 0x10 #define CS35L35_AMP_GAIN_ZC_SHIFT 4 +#define CS35L35_AMP_DIGSFT_MASK 0x02 +#define CS35L35_AMP_DIGSFT_SHIFT 1 + /* CS35L35_SP_FMT_CTL3 */ #define CS35L35_SP_I2S_DRV_MASK 0x03 #define CS35L35_SP_I2S_DRV_SHIFT 0 -- 2.1.4

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[alsa-devel] [PATCH v3] ASoC: STI: Fix null ptr deference in IRQ handler
by Arnaud Pouliquen 06 Apr '17

06 Apr '17

With RTlinux a race condition has been found that leads to NULL ptr crash: - On CPU 0: uni_player_irq_handler is called to treat XRUN "(player->state == UNIPERIF_STATE_STOPPED)" is FALSE so status is checked, dev_err(player->dev, "FIFO underflow error detected") is printed and then snd_pcm_stream_lock should be called to lock stream for stopping. - On CPU 1: application stop and close the stream. Issue is that the stop and shutdown functions are executed while "FIFO underflow error detected" is printed. So when CPU 0 calls snd_pcm_stream_lock, player->substream is already null. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- V3: - Lower case for labels, - Fix lock-unlock balance. V2: Add spinlock to protect player/reader->substream --- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 35 ++++++++++++++++++++++++----------- sound/soc/sti/uniperif_reader.c | 24 ++++++++++++++++++++---- 3 files changed, 45 insertions(+), 15 deletions(-) diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index d487dd2..cfcb0ea 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1299,6 +1299,7 @@ struct uniperif { int ver; /* IP version, used by register access macros */ struct regmap_field *clk_sel; struct regmap_field *valid_sel; + spinlock_t irq_lock; /* use to prevent race condition with IRQ */ /* capabilities */ const struct snd_pcm_hardware *hw; diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 60ae31a..d7e8dd4 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -65,10 +65,13 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) unsigned int status; unsigned int tmp; - if (player->state == UNIPERIF_STATE_STOPPED) { - /* Unexpected IRQ: do nothing */ - return IRQ_NONE; - } + spin_lock(&player->irq_lock); + if (!player->substream) + goto irq_spin_unlock; + + snd_pcm_stream_lock(player->substream); + if (player->state == UNIPERIF_STATE_STOPPED) + goto stream_unlock; /* Get interrupt status & clear them immediately */ status = GET_UNIPERIF_ITS(player); @@ -88,9 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); } ret = IRQ_HANDLED; @@ -104,9 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); ret = IRQ_HANDLED; } @@ -116,7 +115,8 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) if (!player->underflow_enabled) { dev_err(player->dev, "unexpected Underflow recovering\n"); - return -EPERM; + ret = -EPERM; + goto stream_unlock; } /* Read the underflow recovery duration */ tmp = GET_UNIPERIF_STATUS_1_UNDERFLOW_DURATION(player); @@ -138,13 +138,16 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); ret = IRQ_HANDLED; } +stream_unlock: + snd_pcm_stream_unlock(player->substream); +irq_spin_unlock: + spin_unlock(&player->irq_lock); + return ret; } @@ -588,6 +591,7 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; struct snd_aes_iec958 *iec958 = &player->stream_settings.iec958; + unsigned long flags; mutex_lock(&player->ctrl_lock); iec958->status[0] = ucontrol->value.iec958.status[0]; @@ -596,12 +600,14 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, iec958->status[3] = ucontrol->value.iec958.status[3]; mutex_unlock(&player->ctrl_lock); + spin_lock_irqsave(&player->irq_lock, flags); if (player->substream && player->substream->runtime) uni_player_set_channel_status(player, player->substream->runtime); else uni_player_set_channel_status(player, NULL); + spin_unlock_irqrestore(&player->irq_lock, flags); return 0; } @@ -686,9 +692,12 @@ static int uni_player_startup(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; + unsigned long flags; int ret; + spin_lock_irqsave(&player->irq_lock, flags); player->substream = substream; + spin_unlock_irqrestore(&player->irq_lock, flags); player->clk_adj = 0; @@ -986,12 +995,15 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; + unsigned long flags; + spin_lock_irqsave(&player->irq_lock, flags); if (player->state != UNIPERIF_STATE_STOPPED) /* Stop the player */ uni_player_stop(player); player->substream = NULL; + spin_unlock_irqrestore(&player->irq_lock, flags); } static int uni_player_parse_dt_audio_glue(struct platform_device *pdev, @@ -1096,6 +1108,7 @@ int uni_player_init(struct platform_device *pdev, } mutex_init(&player->ctrl_lock); + spin_lock_init(&player->irq_lock); /* Ensure that disabled by default */ SET_UNIPERIF_CONFIG_BACK_STALL_REQ_DISABLE(player); diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 93a8df6..ee0055e 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -46,10 +46,15 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) struct uniperif *reader = dev_id; unsigned int status; + spin_lock(&reader->irq_lock); + if (!reader->substream) + goto irq_spin_unlock; + + snd_pcm_stream_lock(reader->substream); if (reader->state == UNIPERIF_STATE_STOPPED) { /* Unexpected IRQ: do nothing */ dev_warn(reader->dev, "unexpected IRQ\n"); - return IRQ_HANDLED; + goto stream_unlock; } /* Get interrupt status & clear them immediately */ @@ -60,13 +65,16 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stream_lock(reader->substream); snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(reader->substream); - return IRQ_HANDLED; + ret = IRQ_HANDLED; } +stream_unlock: + snd_pcm_stream_unlock(reader->substream); +irq_spin_unlock: + spin_unlock(&reader->irq_lock); + return ret; } @@ -347,9 +355,12 @@ static int uni_reader_startup(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *reader = priv->dai_data.uni; + unsigned long flags; int ret; + spin_lock_irqsave(&reader->irq_lock, flags); reader->substream = substream; + spin_unlock_irqrestore(&reader->irq_lock, flags); if (!UNIPERIF_TYPE_IS_TDM(reader)) return 0; @@ -375,12 +386,15 @@ static void uni_reader_shutdown(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *reader = priv->dai_data.uni; + unsigned long flags; + spin_lock_irqsave(&reader->irq_lock, flags); if (reader->state != UNIPERIF_STATE_STOPPED) { /* Stop the reader */ uni_reader_stop(reader); } reader->substream = NULL; + spin_unlock_irqrestore(&reader->irq_lock, flags); } static const struct snd_soc_dai_ops uni_reader_dai_ops = { @@ -415,6 +429,8 @@ int uni_reader_init(struct platform_device *pdev, return -EBUSY; } + spin_lock_init(&reader->irq_lock); + return 0; } EXPORT_SYMBOL_GPL(uni_reader_init); -- 1.9.1

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[alsa-devel] No Headphone Jack Audio from Lenovo Docking Station with a T470 connected
by dmummenschanz＠web.de 06 Apr '17

06 Apr '17

Hello, I'm unable to get any sound from the Headphone Jack of a Lenovo Ultra Docking Station with a new Lenovo T470 connected. With a T440p however I get sound from the Docking Station Headphone Jack. I did some research on my alsa configuration and found out that there is no pin configuration supplied from the Laptop BIOS that connects to the Docking Headphone Jack. The hdajacksensetest tool only shows this when the Laptop is docked and a Headphone is pluged in: Pin 0x18 (Black Mic, Left side): present = No Pin 0x21 (Black Headphone, Left side): present = No So I did some patching using a early firmware load patch floating around the web targeting the same docking station for my older T440p laptop: echo 0x17 0x21211010 > /sys/class/sound/hwC0D0/user_pin_configs echo 1 > /sys/class/sound/hwC0D0/reconfig Now my Docking Station Headphone Jack is being detected! :) With the Headphone Connected: Pin 0x17 (Black Headphone, Docking station, Rear side): present = Yes Pin 0x18 (Black Mic, Left side): present = No Pin 0x21 (Black Headphone, Left side): present = No And disconnected: Pin 0x17 (Black Headphone, Docking station, Rear side): present = No Pin 0x18 (Black Mic, Left side): present = No Pin 0x21 (Black Headphone, Left side): present = No However there is STILL no sound even after unmuting everything in alsamixer! I'm pretty sure 0x17 is the right pincfg but I'm not sure about 0x21211010 and how I obtain such an address for my laptop. Could someone please help me out here or give me a hint? Here is a alsa-info.sh dump when the Laptop is not connected to the Dock: https://pastebin.com/eTNnfnzL And here when the Laptop is connected to the Dock (No changes): https://pastebin.com/ySBsKUeH And here after setting the pin config 0x17 0x21211010: https://pastebin.com/xmDfsNtQ I'm sure I'm very close. But something is still missing. Any help is much apprechiated! Thanks in advance Dieter

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Re: [alsa-devel] [PATCH 4/5] vfio: platform: Convert to use devm_*()
by Leo Yan 06 Apr '17

06 Apr '17

On Sun, Apr 02, 2017 at 04:45:28PM +0200, Auger Eric wrote: > Hi Leo, > > On 26/03/2017 16:41, Leo Yan wrote: > > Convert driver to use devm_*() APIs so rely on driver model core layer > > to manage resources. This eliminates error path boilerplate and makes > > code neat. > > > > This patch also renames amba_id structure, the old code used some code > > which directly copied from other driver. > > > > Signed-off-by: Leo Yan <leo.yan(a)linaro.org> > Looks good to me > Reviewed-by: Eric Auger <eric.auger(a)redhat.com> Thanks for reviewing, Eric. > May be interesting to go further converting as well the vfio-platform > driver but this can be done later on. Looked a bit for vfio-platform driver and I'm not sure if need some converting within function vfio_platform_probe_common(). So I'd like to leave this to who really understand the subsystem :) [...] Thanks, Leo Yan

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[alsa-devel] ALSA, SB1240 (Sound Blaster HD USB DAC) - need help :(
by Самохин Александр Вален тинович 06 Apr '17

06 Apr '17

Hi, I’m trying adding support for Creative SB1240(Sound Blaster HD, USB DAC).. Player Volume control is working like a charm, but.. “Capture” interface (Mic, Line, Phonograph) – not working at all. Can you help me to add support for this card? I’m trying to sniff USB traffic between USB DAC & Windows Driver(with Wireshark usbdump). But.. I need more theory, to do it.. I cannot get work any capturing interface ☹ I’t my environment: # uname -a Linux kernel 4.9.20 #1 SMP Wed Apr 5 18:42:37 MSK 2017 x86_64 GNU/Linux # cat /etc/group audio:x:63:root # dmesg *************** [ 1.670745] usb 1-1.1: New USB device found, idVendor=041e, idProduct=3232 [ 1.670748] usb 1-1.1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 1.670750] usb 1-1.1: Product: USB Sound Blaster HD [ 1.670752] usb 1-1.1: Manufacturer: Creative Technology [ 1.670753] usb 1-1.1: SerialNumber: 000003II [ 1.673435] input: Creative Technology USB Sound Blaster HD as /devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.1/1-1.1:1.5/0003:041E:3232.0001/input/input6 *************** # ls -la /dev/snd/ total 0 drwxr-xr-x 4 root root 220 Apr 5 21:33 . drwxr-xr-x 15 root root 3360 Apr 5 21:34 .. drwxr-xr-x 2 root root 60 Apr 5 21:33 by-id drwxr-xr-x 2 root root 60 Apr 5 21:33 by-path crw-rw---- 1 root audio 116, 2 Apr 5 21:33 controlC0 crw-rw---- 1 root audio 116, 4 Apr 5 21:33 pcmC0D0c crw-rw---- 1 root audio 116, 3 Apr 5 21:34 pcmC0D0p crw-rw---- 1 root audio 116, 6 Apr 5 22:57 pcmC0D1c crw-rw---- 1 root audio 116, 5 Apr 5 21:33 pcmC0D1p crw-rw---- 1 root audio 116, 7 Apr 5 21:33 pcmC0D2p crw-rw---- 1 root audio 116, 33 Apr 5 21:33 timer # amixer info Card default 'HD'/'Creative Technology USB Sound Blaster HD at usb-0000:00:1a.0-1.1, full speed' Mixer name : 'USB Mixer' Components : 'USB041e:3232' Controls : 8 Simple ctrls : 2 # amixer contents numid=6,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=128,step=0 : values=128,128 | dBminmax-min=-64.00dB,max=0.00dB numid=8,iface=MIXER,name='PCM Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Mic' ; Item #1 'Line' ; Item #2 'Phonograph' : values=1 numid=4,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=1,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=3,4 | container | chmap-fixed=FL,FR numid=5,iface=PCM,name='Capture Channel Map',device=1 ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=2,iface=PCM,name='Playback Channel Map',device=1 ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=3,iface=PCM,name='Playback Channel Map',device=2 ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR С уважением, Александр Самохин Заместитель начальника отдела Отдел информационных технологий и компьютерного обеспечения ФГУП «Большой Московский цирк» Email: a.samokhin(a)greatcircus.ru<mailto:a.samokhin@greatcircus.ru> , it(a)greatcircus.ru<mailto:it@greatcircus.ru> Tel: +7 (495) 930-22-25, вн. 2002 Mob: +7 (903) 296-77-98

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[alsa-devel] [resend][PATCH v5 00/10] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 06 Apr '17

06 Apr '17

Hi Rob, Mark 2 weeks passed. I will re-post these patches. I added already posted "base: export symbol of_phandle_iterator_init/next" as [01/10] patch which is needed on this series. Kuninori Morimoto (10): of_graph: export symbol of_phandle_iterator_init/next of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() ASoC: add snd_soc_get_dai_id() ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 10 files changed, 604 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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[alsa-devel] [PATCH v4 0/2] ASoC: add driver support for CX2092X DSP
by simon.ho.cnxt＠gmail.com 06 Apr '17

06 Apr '17

From: Simon Ho <simon.ho(a)conexant.com> Hi Pierre, Mark and Takashi I corrected the driver based on your input as below. - Corrected coding style. - Removed ".owner" assignment. - Removed SPI interface support since SPI fearure is removed from this new chip. Thanks, Simon Simon Ho (2): doc: cx2092x: Add DT bingings doc for CX2092X DSP ASoC: Add support for Conexant CX2092X DSP .../devicetree/bindings/sound/cnxt,cx2092x.txt | 25 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2092x.c | 436 +++++++++++++++++++++ sound/soc/codecs/cx2092x.h | 19 + 5 files changed, 488 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cnxt,cx2092x.txt create mode 100644 sound/soc/codecs/cx2092x.c create mode 100644 sound/soc/codecs/cx2092x.h -- 2.7.4

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