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Re: [alsa-devel] ERROR: "isa_io_base" [sound/pci/vx222/snd-vx222.ko] undefined!
by Tomi Valkeinen 07 Mar '16

07 Mar '16

On 06/03/16 14:38, kbuild test robot wrote: > Hi Nicolai, > > FYI, the error/warning still remains. > > tree: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master > head: 67944024c1cdd897e49a09b0d6af3ea38d1388ca > commit: 48fd8ecf29e3c1199432e173858f5ca4bc529738 framebuffer: disable vgacon on microblaze arch > date: 7 months ago > config: microblaze-allmodconfig (attached as .config) > reproduce: > wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross > chmod +x ~/bin/make.cross > git checkout 48fd8ecf29e3c1199432e173858f5ca4bc529738 > # save the attached .config to linux build tree > make.cross ARCH=microblaze > > All errors (new ones prefixed by >>): > >>> ERROR: "isa_io_base" [sound/pci/vx222/snd-vx222.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/trident/snd-trident.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-via82xx.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-via82xx-modem.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-sonicvibes.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-maestro3.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-fm801.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-es1968.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-es1938.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-ens1371.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-ens1370.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-cmipci.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-azt3328.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-als300.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/riptide/snd-riptide.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/pcxhr/snd-pcxhr.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/oxygen/snd-oxygen-lib.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/lx6464es/snd-lx6464es.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/ice1712/snd-ice1724.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/ice1712/snd-ice1712.ko] undefined! I didn't look at this closely, but sounds like disabling vgacon on microblaze made sound drivers fail because they or microblaze does not enable something in the kernel config. Tomi

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[alsa-devel] [PATCH] ASoC: rt5616: add missing mute control for HPVOL
by John Lin 07 Mar '16

07 Mar '16

Add missing kcontrol for HPVOL mute control. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5616.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3704d07..f527b5b 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -309,6 +309,8 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { /* Headphone Output Volume */ SOC_DOUBLE("HP Playback Switch", RT5616_HP_VOL, RT5616_L_MUTE_SFT, RT5616_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("HPVOL Playback Switch", RT5616_HP_VOL, + RT5616_VOL_L_SFT, RT5616_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5616_HP_VOL, RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rsrc-card: tidyup convert-channels explain
by Kuninori Morimoto 07 Mar '16

07 Mar '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt index 5abebf7..255ece3 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt @@ -30,7 +30,7 @@ Optional subnode properties: - frame-inversion : bool property. Add this if the dai-link uses frame clock inversion. - convert-rate : platform specified sampling rate convert -- convert-channels : platform specified channel size convert +- convert-channels : platform specified converted channel size (2 - 8 ch) - audio-prefix : see audio-routing - audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5640: Set PLL src according to source
by Bard Liao 07 Mar '16

07 Mar '16

rt5640_set_dai_pll set pll source according to given source and dai id. However, the pll source should be set according to given source only. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6cd84fb..863c190 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1902,21 +1902,12 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_MCLK); break; case RT5640_PLL1_S_BCLK1: + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); + break; case RT5640_PLL1_S_BCLK2: - dai_sel = get_sdp_info(codec, dai->id); - if (dai_sel < 0) { - dev_err(codec->dev, - "Failed to get sdp info: %d\n", dai_sel); - return -EINVAL; - } - if (dai_sel & RT5640_U_IF1) { - snd_soc_update_bits(codec, RT5640_GLB_CLK, - RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); - } - if (dai_sel & RT5640_U_IF2) { - snd_soc_update_bits(codec, RT5640_GLB_CLK, - RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); - } + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); break; default: dev_err(codec->dev, "Unknown PLL source %d\n", source); -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH 0/7] ASoC: rsnd: update SRC activates conditions
by Kuninori Morimoto 07 Mar '16

07 Mar '16

Hi Mark Current Renesas R-Car sound SRC had cared only Playback, because we had heard that it can't use under Capture. But it was telephone-game. accurately, SRC has Sync/Async mode, and it can't use Sync mode when Capture. And, SRC activates conditions is related to CMD, not DVC. These patches update these. Kuninori Morimoto (7): ASoC: rsnd: don't enable SRC Sync mode when Capture with CMD ASoC: rsnd: SRC sync mode is not related to clock master mode ASoC: rsnd: SRC function parameter uses "mod" instead of "src" ASoC: rsnd: rename rsnd_enable_sync_convert() to rsnd_src_sync_is_enabled() ASoC: rsnd: add rsnd_src_get_in/out_rate() ASoC: rsnd: SRC TIMSEL support for Capture ASoC: rsnd: CMD TIMSEL support for Capture sound/soc/sh/rcar/adg.c | 209 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----------------------------------------------------------- sound/soc/sh/rcar/rsnd.h | 18 ++++++------ sound/soc/sh/rcar/src.c | 95 +++++++++++++++++++++++++++++++------------------------------ sound/soc/sh/rcar/ssi.c | 5 ++-- 4 files changed, 177 insertions(+), 150 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [RFT][PATCH 0/4] ALSA: dice: enabled to handle several streams
by Takashi Sakamoto 07 Mar '16

07 Mar '16

Hi, This patchset adds support for whole available isochronous streams to ALSA dice driver. As a result, for some models, additional PCM substreams are available via subdevice of ALSA PCM character device. In configuration space of alsa-lib, it's represented like 'hw:0,0,1'(card/device/subdevice). It's soon in the last week of this developing cycle, while I hope to finish my work for ALSA dice driver by this patchset. Unfortunately, this patchset adds huge changes and includes possibilities of regression. I'm happy to receive your reports to test with any of your Dice units. For testers, I push whole patches to 'dice-several-streams' branch in my private repository. The changed driver is available from this branch. I recommend to use DKMS to maintain the driver, instead of installing by your own. https://github.com/takaswie/snd-firewire-improve/tree/dice-several-streams To check whether your units support several isochronous streams, please read output from /proc/asound/cardX/dice. When seeing 'tx1' (input) or 'rx1' (output), the additional subdevices are available. For example, $ arecord -l ... card 1: Pro26000f5a [Pro26-000f5a], device 0: DICE [Pro26-000f5a] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 In above case, you can use two subdevices for PCM capture, thus: * arecord -D plughw:1,0,0 file1.wav * arecord -D plughw:1,0,1 file2.wav In the middle of next week, I plan to post this patchset again for merging. So it's nice to test till then. Regards Takashi Sakamoto (4): ALSA: dice: have two sets of isochronous resources/streams ALSA: dice: handle whole available isochronous streams ALSA: dice: handle several PCM substreams when any isochronous streams are available ALSA: dice: force to add two pcm devices for listed models sound/firewire/dice/dice-midi.c | 8 +- sound/firewire/dice/dice-pcm.c | 110 +++++++---- sound/firewire/dice/dice-stream.c | 400 ++++++++++++++++++++++++-------------- sound/firewire/dice/dice.c | 41 ++++ sound/firewire/dice/dice.h | 33 +++- 5 files changed, 401 insertions(+), 191 deletions(-) -- 2.7.0

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[alsa-devel] [PATCH] ASoC: rt5616: add kcontrol for headphone channel switch
by John Lin 07 Mar '16

07 Mar '16

Add kcontrol for headphone channel switch. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5616.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3704d07..b370993 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -309,6 +309,8 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { /* Headphone Output Volume */ SOC_DOUBLE("HP Playback Switch", RT5616_HP_VOL, RT5616_L_MUTE_SFT, RT5616_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("HP Channel Switch", RT5616_HP_VOL, + RT5616_VOL_L_SFT, RT5616_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5616_HP_VOL, RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ -- 1.8.1.1.439.g50a6b54

3 6

[alsa-devel] [PATCH] Revert "ASoC: fsl_ssi: remove explicit register defaults"
by Fabio Estevam 07 Mar '16

07 Mar '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 5c408fee2546 ("ASoC: fsl_ssi: remove explicit register defaults") causes the driver to fail to probe: fsl-ssi-dai 2028000.ssi: No cache defaults, reading back from HW fsl-ssi-dai 2028000.ssi: Failed to init register map fsl-ssi-dai: probe of 2028000.ssi failed with error -22 , so revert this commit. Cc: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Reported-by: Mika Penttilä <mika.penttila(a)nextfour.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Maciej, As this currently breaks SSI on 4.5-rc we should better revert it for this cycle. We can then add it back to 4.7 when Mark's patch 3245d460a1eb55 ("regmap: cache: Fall back to register by register read for cache defaults") will be available and then the driver will work fine with your patch. sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++++++---------------------- 1 file changed, 20 insertions(+), 22 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ed8de10..40dfd8a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -112,6 +112,20 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val tx; }; +static const struct reg_default fsl_ssi_reg_defaults[] = { + {CCSR_SSI_SCR, 0x00000000}, + {CCSR_SSI_SIER, 0x00003003}, + {CCSR_SSI_STCR, 0x00000200}, + {CCSR_SSI_SRCR, 0x00000200}, + {CCSR_SSI_STCCR, 0x00040000}, + {CCSR_SSI_SRCCR, 0x00040000}, + {CCSR_SSI_SACNT, 0x00000000}, + {CCSR_SSI_STMSK, 0x00000000}, + {CCSR_SSI_SRMSK, 0x00000000}, + {CCSR_SSI_SACCEN, 0x00000000}, + {CCSR_SSI_SACCDIS, 0x00000000}, +}; + static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -176,7 +190,8 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .num_reg_defaults_raw = CCSR_SSI_SACCDIS / sizeof(uint32_t) + 1, + .reg_defaults = fsl_ssi_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg, @@ -186,7 +201,6 @@ static const struct regmap_config fsl_ssi_regconfig = { struct fsl_ssi_soc_data { bool imx; - bool imx21regs; /* imx21-class SSI - no SACC{ST,EN,DIS} regs */ bool offline_config; u32 sisr_write_mask; }; @@ -289,7 +303,6 @@ static struct fsl_ssi_soc_data fsl_ssi_mpc8610 = { static struct fsl_ssi_soc_data fsl_ssi_imx21 = { .imx = true, - .imx21regs = true, .offline_config = true, .sisr_write_mask = 0, }; @@ -573,12 +586,8 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) */ regmap_write(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV); - - /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ - if (!ssi_private->soc->imx21regs) { - regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); - regmap_write(regs, CCSR_SSI_SACCEN, 0x300); - } + regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); + regmap_write(regs, CCSR_SSI_SACCEN, 0x300); /* * Enable SSI, Transmit and Receive. AC97 has to communicate with the @@ -1388,7 +1397,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct resource *res; void __iomem *iomem; char name[64]; - struct regmap_config regconfig = fsl_ssi_regconfig; of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) @@ -1436,25 +1444,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(iomem); ssi_private->ssi_phys = res->start; - if (ssi_private->soc->imx21regs) { - /* - * According to datasheet imx21-class SSI - * don't have SACC{ST,EN,DIS} regs. - */ - regconfig.max_register = CCSR_SSI_SRMSK; - regconfig.num_reg_defaults_raw = - CCSR_SSI_SRMSK / sizeof(uint32_t) + 1; - } - ret = of_property_match_string(np, "clock-names", "ipg"); if (ret < 0) { ssi_private->has_ipg_clk_name = false; ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, - &regconfig); + &fsl_ssi_regconfig); } else { ssi_private->has_ipg_clk_name = true; ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev, - "ipg", iomem, &regconfig); + "ipg", iomem, &fsl_ssi_regconfig); } if (IS_ERR(ssi_private->regs)) { dev_err(&pdev->dev, "Failed to init register map\n"); -- 1.9.1

3 6

[alsa-devel] Trouble about Roland UA-11 UA-33 UA-55 and snd-usb-audio module
by FirebUntU 07 Mar '16

07 Mar '16

Hi guys, I write you about support of snd-usb-audio module and a relatively-new gamma of devices from Roland/Edirol: DUO CAPTURE, TRI CAPTURE and QUAD CAPTURE respectively: UA-11, UA-33, UA-55 These aren't not listed in the Alsa db. I have searched a solution for a week and I haven't found a trick to make UA-11 to work under Alsa 1.25.0 then I downloaded the driver for try to patch it and I found these devices is not listed in the tricks table In despair I have played with the switch button called "EXT * **" in the bottom of the unit of UA-11 and magically the device ID is transformed from * 0582:012b to ** 0582:0137 Image: https://dl.dropbox.com/u/32759898/output.jpg (Rear of the UA-11 unit, from the manual) *Before* (Recognized from udev - Unknown from alsa): http://www.alsa-project.org/db/?f=502ee68cf31882ba9ef23525fd66452c34d329e1 *After* (Recognized from udev - Recognized from alsa): http://www.alsa-project.org/db/?f=b2ae3e4331fb3fe613ba71e7aba1ad2e0e6dcd29 In the tricks table header of ALSA the 0582:0137 stay for PM-1000 in /TODO/ statement. I understand the EXT switch, undiscovered in the manual of the DUO CAPTURE is - as the fact - the Advanced Mode. Now the peripherical running both in playback and recording in 16 and 24 bit and without any mixer options in alsamixer. Can you update future releases with this gamma of devices? Awaiting for your positive response :) Thanks. Best regards, Leonardo

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[alsa-devel] [PATCH 1/2] ASoC: cs35l32: avoid uninitialized variable access
by Arnd Bergmann 06 Mar '16

06 Mar '16

gcc warns about the possibilty of accessing a property read from devicetree in cs35l32_i2c_probe() when it has not been initialized because CONFIG_OF is disabled: sound/soc/codecs/cs35l32.c: In function 'cs35l32_i2c_probe': sound/soc/codecs/cs35l32.c:278:2: warning: 'val' may be used uninitialized in this function [-Wmaybe-uninitialized] The code is actually correct because it checks the dev->of_node variable first and we know this is NULL here, but by adding a check for IS_ENABLED(CONFIG_OF), we can let the compiler know as well, and also generate smaller object code. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cs35l32.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 44c30fe3e315..52ffaa8eb02b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -372,7 +372,7 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "could not allocate pdata\n"); return -ENOMEM; } - if (i2c_client->dev.of_node) { + if (IS_ENABLED(CONFIG_OF) && i2c_client->dev.of_node) { ret = cs35l32_handle_of_data(i2c_client, &cs35l32->pdata); if (ret != 0) -- 2.7.0

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[alsa-devel] [PATCH - pcm 1/1] pcm: fix "unable to create IPC shm instance" in some case
by IceBsi 05 Mar '16

05 Mar '16

From: Qing Cai <caiqing(a)neusoft.com> As stated in manpage SHMCTL(2), shm_nattch is "No. of current attaches" (i.e., number of processes attached to the shared memeory). If an application uses alsa-lib and invokes fork() at some point, there should be the following execution sequence: 1. execute the following statement: pcm_direct.c:110: dmix->shmptr = shmat(dmix->shmid, 0, 0) (shm_nattch becomes 1) 2. invoke fork() in some thread. (shm_nattch becomes 2) 3. execute the following statement: pcm_direct.c:122: if (buf.shm_nattch == 1) 4. execute the following statement: pcm_direct.c:131: if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) (As stated in manpage SHMGET(2), "When a new shared memory segment is created, its contents are initialized to zero values", so dmix->shmptr->magic is 0) 5. execute the following statements: pcm_direct.c:132: snd_pcm_direct_shm_discard(dmix) pcm_direct.c:133: return -EINVAL The above execution sequence will cause the following error: unable to create IPC shm instance This error causes multimedia application has no sound. This error rarely occurs, probability is about 1%. Because the first user of the shared memory will get that dmix->shmptr->magic is 0, check dmix->shmptr->magic's value to determine if "we're the first user" is OK. Tests have been made 400+ times after this fix, and the issue no longer exists. Signed-off-by: Qing Cai <bsiice(a)msn.com> Signed-off-by: Qing Cai <caiqing(a)neusoft.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index fd3877c..93d6f3c 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -119,7 +119,7 @@ retryget: snd_pcm_direct_shm_discard(dmix); return err; } - if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ + if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) { /* we're the first user, clear the segment */ memset(dmix->shmptr, 0, sizeof(snd_pcm_direct_share_t)); if (dmix->ipc_gid >= 0) { buf.shm_perm.gid = dmix->ipc_gid; -- 2.7.2

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[alsa-devel] Applied "ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616" to the asoc tree
by Mark Brown 05 Mar '16

05 Mar '16

The patch ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616 has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 288bc356a881fcdce05f9782736a6a3d9db359a9 Mon Sep 17 00:00:00 2001 From: Caesar Wang <wxt(a)rock-chips.com> Date: Fri, 4 Mar 2016 14:54:50 +0800 Subject: [PATCH] ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616 This patch fixes that hasn't built the rt5616 driver with 'CONFIG_SND_SOC_RT5616=y' in .config. The tristate is the prompt on the 'make menuconfig', in other words, that can't show the prompt and select it if we don't say what's the tristate. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 50693c867e71..1de259860bc9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -596,7 +596,7 @@ config SND_SOC_RT298 depends on I2C config SND_SOC_RT5616 - tristate + tristate "Realtek RT5616 CODEC" config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" -- 2.7.0

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[alsa-devel] [PATCH 00/14] drivers: use __maybe_unused to hide pm functions
by Arnd Bergmann 05 Mar '16

05 Mar '16

I found many variations of the bug in these device drivers (and some USB drivers I already send patches for in a separate series). In each case, the power management operations structure conditionally references suspend/resume functions, but the functions are hidden in an incorrect #ifdef or not hidden at all. We could try to correct the #ifdefs, but it seems easier to just mark those functions as __maybe_unused, which has the same effect but provides better compile-time test coverage and (subjectively) looks a bit nicer. I have a patch series that avoids all warnings in ARM randconfig builds, and I have verified that all these patches fix a warning that is still present in today's linux-next, and that they do not introduce new warnings in any configuration I found. Note that all these drivers are ARM specific, so I assume that all portable drivers got fixed already when someone rand into the problem on x86. There are no dependencies between the patches, so I'd appreciate subsystem maintainers to put them directly into their git trees. Arnd Arnd Bergmann (14): pinctrl: at91: use __maybe_unused to hide pm functions irqchip: st: use __maybe_unused to hide st_irq_syscfg_resume power: ipaq-micro-battery: use __maybe_unused to hide pm functions power: pm2301-charger: use __maybe_unused to hide pm functions mfd: ipaq-micro: use __maybe_unused to hide pm functions dma: sirf: use __maybe_unused to hide pm functions hw_random: exynos: use __maybe_unused to hide pm functions scsi: mvumi: use __maybe_unused to hide pm functions amd-xgbe: use __maybe_unused to hide pm functions wireless: cw1200: use __maybe_unused to hide pm functions_ input: spear-keyboard: use __maybe_unused to hide pm functions keyboard: snvs-pwrkey: use __maybe_unused to hide pm functions [media] omap3isp: use IS_ENABLED() to hide pm functions ASoC: rockchip: use __maybe_unused to hide st_irq_syscfg_resume drivers/char/hw_random/exynos-rng.c | 10 ++++------ drivers/dma/sirf-dma.c | 10 ++++------ drivers/input/keyboard/snvs_pwrkey.c | 4 ++-- drivers/input/keyboard/spear-keyboard.c | 6 ++---- drivers/irqchip/irq-st.c | 2 +- drivers/media/platform/omap3isp/isp.c | 13 +------------ drivers/mfd/ipaq-micro.c | 2 +- drivers/net/ethernet/amd/xgbe/xgbe-main.c | 6 ++---- drivers/net/wireless/st/cw1200/cw1200_spi.c | 9 ++------- drivers/net/wireless/st/cw1200/pm.h | 9 +++++++-- drivers/pinctrl/pinctrl-at91-pio4.c | 4 ++-- drivers/power/ipaq_micro_battery.c | 4 ++-- drivers/power/pm2301_charger.c | 22 ++++++---------------- drivers/scsi/mvumi.c | 4 ++-- sound/soc/rockchip/rockchip_spdif.c | 4 ++-- 15 files changed, 40 insertions(+), 69 deletions(-) -- 2.7.0 Cc: herbert(a)gondor.apana.org.au Cc: k.kozlowski(a)samsung.com Cc: dan.j.williams(a)intel.com Cc: vinod.koul(a)intel.com Cc: baohua(a)kernel.org Cc: dmitry.torokhov(a)gmail.com Cc: tglx(a)linutronix.de Cc: jason(a)lakedaemon.net Cc: marc.zyngier(a)arm.com Cc: laurent.pinchart(a)ideasonboard.com Cc: mchehab(a)osg.samsung.com Cc: lee.jones(a)linaro.org Cc: kvalo(a)codeaurora.org Cc: ludovic.desroches(a)atmel.com Cc: linus.walleij(a)linaro.org Cc: sre(a)kernel.org Cc: dbaryshkov(a)gmail.com Cc: JBottomley(a)odin.com Cc: martin.petersen(a)oracle.com Cc: broonie(a)kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: dmaengine(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-media(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-wireless(a)vger.kernel.org Cc: linux-gpio(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-rockchip(a)lists.infradead.org

2 2

[alsa-devel] [PATCH] ASoC: pcm3168a: remove unused variable
by Sudip Mukherjee 05 Mar '16

05 Mar '16

While building we were getting build warning about: sound/soc/codecs/pcm3168a.c:403:21: warning: variable 'channels' set but not used The variable channels were being assigned some value but that was never reused. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- Its upto you now. If you dont want to accept then I will add an extra From: in the commit. But since that will not be a part of my scripts so there will be chances of error. If you are worried about missing signoff, then I can assure you that my scripts will catch that before i can send it. sound/soc/codecs/pcm3168a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 0c7248a..992a77e 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -400,13 +400,12 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, struct pcm3168a_priv *pcm3168a = snd_soc_codec_get_drvdata(codec); bool tx, master_mode; u32 val, mask, shift, reg; - unsigned int rate, channels, fmt, ratio, max_ratio; + unsigned int rate, fmt, ratio, max_ratio; int i, min_frame_size; snd_pcm_format_t format; rate = params_rate(params); format = params_format(params); - channels = params_channels(params); ratio = pcm3168a->sysclk / rate; -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: rockchip: add bindings for spdif controller
by Sugar Zhang 05 Mar '16

05 Mar '16

this patch add compatible for rk3366/rk3368/rk3399 spdif, these three spdifs share the same type. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Documentation/devicetree/bindings/sound/rockchip-spdif.txt | 8 ++++++-- sound/soc/rockchip/rockchip_spdif.c | 13 ++++++++++--- 2 files changed, 16 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt index e64dbde..1104642 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt @@ -7,8 +7,12 @@ a fibre cable. Required properties: - compatible: should be one of the following: - - "rockchip,rk3288-spdif", "rockchip,rk3188-spdif" or - "rockchip,rk3066-spdif" + - "rockchip,rk3066-spdif" + - "rockchip,rk3188-spdif" + - "rockchip,rk3288-spdif" + - "rockchip,rk3366-spdif" + - "rockchip,rk3368-spdif" + - "rockchip,rk3399-spdif" - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the SPDIF interrupt. diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 5a806da..c589f6f 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -28,6 +28,7 @@ enum rk_spdif_type { RK_SPDIF_RK3066, RK_SPDIF_RK3188, RK_SPDIF_RK3288, + RK_SPDIF_RK3366, }; #define RK3288_GRF_SOC_CON2 0x24c @@ -45,11 +46,17 @@ struct rk_spdif_dev { static const struct of_device_id rk_spdif_match[] = { { .compatible = "rockchip,rk3066-spdif", - .data = (void *) RK_SPDIF_RK3066 }, + .data = (void *)RK_SPDIF_RK3066 }, { .compatible = "rockchip,rk3188-spdif", - .data = (void *) RK_SPDIF_RK3188 }, + .data = (void *)RK_SPDIF_RK3188 }, { .compatible = "rockchip,rk3288-spdif", - .data = (void *) RK_SPDIF_RK3288 }, + .data = (void *)RK_SPDIF_RK3288 }, + { .compatible = "rockchip,rk3366-spdif", + .data = (void *)RK_SPDIF_RK3366 }, + { .compatible = "rockchip,rk3368-spdif", + .data = (void *)RK_SPDIF_RK3366 }, + { .compatible = "rockchip,rk3399-spdif", + .data = (void *)RK_SPDIF_RK3366 }, {}, }; MODULE_DEVICE_TABLE(of, rk_spdif_match); -- 1.9.1

3 2

[alsa-devel] Applied "ASoC: rsnd: add CTU support" to the asoc tree
by Mark Brown 05 Mar '16

05 Mar '16

The patch ASoC: rsnd: add CTU support has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ec2ac01afe001360b911e27c915579fd003339e8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 25 Feb 2016 05:55:23 +0000 Subject: [PATCH] ASoC: rsnd: add CTU support This patch adds CTU (= Channel Transfer Unit) support on Renesas R-Car sound driver. It can Down/Up mixing and splitter. You need to check R-Car datasheet especially CTUn_CPMDR/CTUn_SV0xR/CTUn_SV1xR/CTUn_SV2xR/CTUn_SV3xR for setting parameter. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ctu.c | 234 ++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 197 insertions(+), 37 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index b326966ea407..9dcc1f9db026 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -12,8 +12,74 @@ #define CTU_NAME_SIZE 16 #define CTU_NAME "ctu" +/* + * User needs to setup CTU by amixer, and its settings are + * based on below registers + * + * CTUn_CPMDR : amixser set "CTU Pass" + * CTUn_SV0xR : amixser set "CTU SV0" + * CTUn_SV1xR : amixser set "CTU SV1" + * CTUn_SV2xR : amixser set "CTU SV2" + * CTUn_SV3xR : amixser set "CTU SV3" + * + * [CTU Pass] + * 0000: default + * 0001: Connect input data of channel 0 + * 0010: Connect input data of channel 1 + * 0011: Connect input data of channel 2 + * 0100: Connect input data of channel 3 + * 0101: Connect input data of channel 4 + * 0110: Connect input data of channel 5 + * 0111: Connect input data of channel 6 + * 1000: Connect input data of channel 7 + * 1001: Connect calculated data by scale values of matrix row 0 + * 1010: Connect calculated data by scale values of matrix row 1 + * 1011: Connect calculated data by scale values of matrix row 2 + * 1100: Connect calculated data by scale values of matrix row 3 + * + * [CTU SVx] + * [Output0] = [SV00, SV01, SV02, SV03, SV04, SV05, SV06, SV07] + * [Output1] = [SV10, SV11, SV12, SV13, SV14, SV15, SV16, SV17] + * [Output2] = [SV20, SV21, SV22, SV23, SV24, SV25, SV26, SV27] + * [Output3] = [SV30, SV31, SV32, SV33, SV34, SV35, SV36, SV37] + * [Output4] = [ 0, 0, 0, 0, 0, 0, 0, 0 ] + * [Output5] = [ 0, 0, 0, 0, 0, 0, 0, 0 ] + * [Output6] = [ 0, 0, 0, 0, 0, 0, 0, 0 ] + * [Output7] = [ 0, 0, 0, 0, 0, 0, 0, 0 ] + * + * [SVxx] + * Plus Minus + * value time dB value time dB + * ----------------------------------------------------------------------- + * H'7F_FFFF 2 6 H'80_0000 2 6 + * ... + * H'40_0000 1 0 H'C0_0000 1 0 + * ... + * H'00_0001 2.38 x 10^-7 -132 + * H'00_0000 0 Mute H'FF_FFFF 2.38 x 10^-7 -132 + * + * + * Ex) Input ch -> Output ch + * 1ch -> 0ch + * 0ch -> 1ch + * + * amixer set "CTU Reset" on + * amixer set "CTU Pass" 9,10 + * amixer set "CTU SV0" 0,4194304 + * amixer set "CTU SV1" 4194304,0 + * or + * amixer set "CTU Reset" on + * amixer set "CTU Pass" 2,1 + */ + struct rsnd_ctu { struct rsnd_mod mod; + struct rsnd_kctrl_cfg_m pass; + struct rsnd_kctrl_cfg_m sv0; + struct rsnd_kctrl_cfg_m sv1; + struct rsnd_kctrl_cfg_m sv2; + struct rsnd_kctrl_cfg_m sv3; + struct rsnd_kctrl_cfg_s reset; int channels; }; @@ -58,51 +124,91 @@ static int rsnd_ctu_probe_(struct rsnd_mod *mod, static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { + struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); + u32 cpmdr = 0; + u32 scmdr = 0; + int i; + + for (i = 0; i < RSND_MAX_CHANNELS; i++) { + u32 val = ctu->pass.val[i]; + + cpmdr |= val << (28 - (i * 4)); + + if ((val > 0x8) && (scmdr < (val - 0x8))) + scmdr = val - 0x8; + } + rsnd_mod_write(mod, CTU_CTUIR, 1); rsnd_mod_write(mod, CTU_ADINR, rsnd_runtime_channel_original(io)); - rsnd_mod_write(mod, CTU_CPMDR, 0); - rsnd_mod_write(mod, CTU_SCMDR, 0); - rsnd_mod_write(mod, CTU_SV00R, 0); - rsnd_mod_write(mod, CTU_SV01R, 0); - rsnd_mod_write(mod, CTU_SV02R, 0); - rsnd_mod_write(mod, CTU_SV03R, 0); - rsnd_mod_write(mod, CTU_SV04R, 0); - rsnd_mod_write(mod, CTU_SV05R, 0); - rsnd_mod_write(mod, CTU_SV06R, 0); - rsnd_mod_write(mod, CTU_SV07R, 0); - - rsnd_mod_write(mod, CTU_SV10R, 0); - rsnd_mod_write(mod, CTU_SV11R, 0); - rsnd_mod_write(mod, CTU_SV12R, 0); - rsnd_mod_write(mod, CTU_SV13R, 0); - rsnd_mod_write(mod, CTU_SV14R, 0); - rsnd_mod_write(mod, CTU_SV15R, 0); - rsnd_mod_write(mod, CTU_SV16R, 0); - rsnd_mod_write(mod, CTU_SV17R, 0); - - rsnd_mod_write(mod, CTU_SV20R, 0); - rsnd_mod_write(mod, CTU_SV21R, 0); - rsnd_mod_write(mod, CTU_SV22R, 0); - rsnd_mod_write(mod, CTU_SV23R, 0); - rsnd_mod_write(mod, CTU_SV24R, 0); - rsnd_mod_write(mod, CTU_SV25R, 0); - rsnd_mod_write(mod, CTU_SV26R, 0); - rsnd_mod_write(mod, CTU_SV27R, 0); - - rsnd_mod_write(mod, CTU_SV30R, 0); - rsnd_mod_write(mod, CTU_SV31R, 0); - rsnd_mod_write(mod, CTU_SV32R, 0); - rsnd_mod_write(mod, CTU_SV33R, 0); - rsnd_mod_write(mod, CTU_SV34R, 0); - rsnd_mod_write(mod, CTU_SV35R, 0); - rsnd_mod_write(mod, CTU_SV36R, 0); - rsnd_mod_write(mod, CTU_SV37R, 0); + rsnd_mod_write(mod, CTU_CPMDR, cpmdr); + + rsnd_mod_write(mod, CTU_SCMDR, scmdr); + + if (scmdr > 0) { + rsnd_mod_write(mod, CTU_SV00R, ctu->sv0.val[0]); + rsnd_mod_write(mod, CTU_SV01R, ctu->sv0.val[1]); + rsnd_mod_write(mod, CTU_SV02R, ctu->sv0.val[2]); + rsnd_mod_write(mod, CTU_SV03R, ctu->sv0.val[3]); + rsnd_mod_write(mod, CTU_SV04R, ctu->sv0.val[4]); + rsnd_mod_write(mod, CTU_SV05R, ctu->sv0.val[5]); + rsnd_mod_write(mod, CTU_SV06R, ctu->sv0.val[6]); + rsnd_mod_write(mod, CTU_SV07R, ctu->sv0.val[7]); + } + if (scmdr > 1) { + rsnd_mod_write(mod, CTU_SV10R, ctu->sv1.val[0]); + rsnd_mod_write(mod, CTU_SV11R, ctu->sv1.val[1]); + rsnd_mod_write(mod, CTU_SV12R, ctu->sv1.val[2]); + rsnd_mod_write(mod, CTU_SV13R, ctu->sv1.val[3]); + rsnd_mod_write(mod, CTU_SV14R, ctu->sv1.val[4]); + rsnd_mod_write(mod, CTU_SV15R, ctu->sv1.val[5]); + rsnd_mod_write(mod, CTU_SV16R, ctu->sv1.val[6]); + rsnd_mod_write(mod, CTU_SV17R, ctu->sv1.val[7]); + } + if (scmdr > 2) { + rsnd_mod_write(mod, CTU_SV20R, ctu->sv2.val[0]); + rsnd_mod_write(mod, CTU_SV21R, ctu->sv2.val[1]); + rsnd_mod_write(mod, CTU_SV22R, ctu->sv2.val[2]); + rsnd_mod_write(mod, CTU_SV23R, ctu->sv2.val[3]); + rsnd_mod_write(mod, CTU_SV24R, ctu->sv2.val[4]); + rsnd_mod_write(mod, CTU_SV25R, ctu->sv2.val[5]); + rsnd_mod_write(mod, CTU_SV26R, ctu->sv2.val[6]); + rsnd_mod_write(mod, CTU_SV27R, ctu->sv2.val[7]); + } + if (scmdr > 3) { + rsnd_mod_write(mod, CTU_SV30R, ctu->sv3.val[0]); + rsnd_mod_write(mod, CTU_SV31R, ctu->sv3.val[1]); + rsnd_mod_write(mod, CTU_SV32R, ctu->sv3.val[2]); + rsnd_mod_write(mod, CTU_SV33R, ctu->sv3.val[3]); + rsnd_mod_write(mod, CTU_SV34R, ctu->sv3.val[4]); + rsnd_mod_write(mod, CTU_SV35R, ctu->sv3.val[5]); + rsnd_mod_write(mod, CTU_SV36R, ctu->sv3.val[6]); + rsnd_mod_write(mod, CTU_SV37R, ctu->sv3.val[7]); + } rsnd_mod_write(mod, CTU_CTUIR, 0); } +static void rsnd_ctu_value_reset(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); + int i; + + if (!ctu->reset.val) + return; + + for (i = 0; i < RSND_MAX_CHANNELS; i++) { + ctu->pass.val[i] = 0; + ctu->sv0.val[i] = 0; + ctu->sv1.val[i] = 0; + ctu->sv2.val[i] = 0; + ctu->sv3.val[i] = 0; + } + ctu->reset.val = 0; +} + static int rsnd_ctu_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) @@ -164,12 +270,66 @@ static int rsnd_ctu_hw_params(struct rsnd_mod *mod, return 0; } +static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + struct snd_soc_pcm_runtime *rtd) +{ + struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); + int ret; + + /* CTU Pass */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU Pass", + NULL, + &ctu->pass, RSND_MAX_CHANNELS, + 0xC); + + /* ROW0 */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", + NULL, + &ctu->sv0, RSND_MAX_CHANNELS, + 0x00FFFFFF); + if (ret < 0) + return ret; + + /* ROW1 */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", + NULL, + &ctu->sv1, RSND_MAX_CHANNELS, + 0x00FFFFFF); + if (ret < 0) + return ret; + + /* ROW2 */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", + NULL, + &ctu->sv2, RSND_MAX_CHANNELS, + 0x00FFFFFF); + if (ret < 0) + return ret; + + /* ROW3 */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", + NULL, + &ctu->sv3, RSND_MAX_CHANNELS, + 0x00FFFFFF); + if (ret < 0) + return ret; + + /* Reset */ + ret = rsnd_kctrl_new_s(mod, io, rtd, "CTU Reset", + rsnd_ctu_value_reset, + &ctu->reset, 1); + + return ret; +} + static struct rsnd_mod_ops rsnd_ctu_ops = { .name = CTU_NAME, .probe = rsnd_ctu_probe_, .init = rsnd_ctu_init, .quit = rsnd_ctu_quit, .hw_params = rsnd_ctu_hw_params, + .pcm_new = rsnd_ctu_pcm_new, }; struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id) -- 2.7.0

1 0

[alsa-devel] [RFC v2 0/6] ASoC: Add mediatek HDMI codec support
by Philipp Zabel 05 Mar '16

05 Mar '16

Hi, this time we add an interface for Jyri's generic hdmi-codec driver [1] to the mediatek hdmi bridge driver. In place of a separate machine driver, the existing mt8173-rt5650-rt5676 driver is extended to link to the hdmi-codec instance provided by the hdmi bridge driver. The alsa jack support, previously implemented by the custom mediatek hdmi codec and machine drivers, is bolted on top of hdmi-codec using Russell's HDMI notifier prototype. It's all still a bit rough, but I could use some feedback at this point. Have we achieved consensus to go forward with the hdmi-codec approach? Is it ok to use an alsa jack for HDMI HPD status reporting? The mediatek drm patches apply on top of https://patchwork.kernel.org/patch/7949021/ ("drm/mediatek: Add HDMI support") the asoc patches depend on these two patches: https://patchwork.kernel.org/patch/7215121/ ("ALSA: pcm: add IEC958 channel status helper for hw_params") [1] https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") best regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the machine driver Philipp Zabel (5): drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver ASoC: mediatek: address dai link array entries by enum video: rmk's HDMI notification prototype drm/mediatek: hdmi: issue notifications ASoC: hdmi-codec: Use HDMI notifications to add jack support .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- drivers/gpu/drm/mediatek/Kconfig | 1 + drivers/gpu/drm/mediatek/mtk_cec.c | 5 + drivers/gpu/drm/mediatek/mtk_drm_hdmi_drv.c | 192 +++++++++++++++++++++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 26 +++ drivers/gpu/drm/mediatek/mtk_hdmi.h | 4 + drivers/video/Makefile | 2 +- drivers/video/hdmi-not.c | 61 +++++++ include/linux/hdmi-not.h | 39 +++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/hdmi-codec.c | 72 +++++++- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 66 ++++++- 12 files changed, 470 insertions(+), 9 deletions(-) create mode 100644 drivers/video/hdmi-not.c create mode 100644 include/linux/hdmi-not.h -- 2.6.2

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[alsa-devel] [PATCH v2 0/2] ASoC: ads117x: Add device tree support for TI ADS117x
by Florian Vaussard 05 Mar '16

05 Mar '16

Hello, This series adds device tree support for Texas Instruments ADS117x ADC chip. The first patch adds the binding documentation, while the second patch adds the compatible string into the driver. No further properties are necessary. Best regards, Florian --- Since v1: - Updated topic for the documentation patch - Removed wildcard in compatible string --- Florian Vaussard (2): ASoC: ads117x: Add bindings documentation for TI ADS117x ADC ASoC: ads117x: Add device tree compatible string Documentation/devicetree/bindings/sound/ti,ads117x.txt | 11 +++++++++++ sound/soc/codecs/ads117x.c | 12 ++++++++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,ads117x.txt -- 2.1.0

3 7

[alsa-devel] [PATCH] ASoC: dapm: Remove unneeded variable ret returning 0
by Rajan Vaja 05 Mar '16

05 Mar '16

This patch fixes coccicheck report: Unneeded variable: "ret". Return "0" on line 2817 Remove unneccesary variable ret created to return zero. Signed-off-by: Rajan Vaja <rajan.vaja(a)gmail.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5a2812f..8dc395f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2805,7 +2805,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret = 0; + int i; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { @@ -2814,7 +2814,7 @@ int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, } mutex_unlock(&dapm->card->dapm_mutex); - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_del_routes); -- 1.7.9.5

2 1

[alsa-devel] I created a wiki page about bug reporting, review requested
by Tanu Kaskinen 05 Mar '16

05 Mar '16

Hi, When talking with PulseAudio users, I sometimes need to tell people to file ALSA bugs, and it would be nice to have a web page that explains how to do that. (Also, I found that the alsa packages in OpenEmbedded point to a non-functional bug tracker, and I wanted to fix the packages, but I couldn't find a substitute link.) Therefore I created a wiki page: http://alsa-project.org/main/index.php/Bug_Tracking The Bug_Tracking wiki page points to another new page, explaining alsa- info.sh: http://alsa-project.org/main/index.php/AlsaInfo Are the ALSA maintainers happy with the contents? I'm not really an expert on how you prefer bugs to be reported. Also, the "Bug Tracker" link in the wiki sidebar points to https://bugtrack.alsa-project.org/alsa-bug which doesn't work. I think that link should be changed to point to the new wiki page, but I don't have edit rights for the sidebar. -- Tanu

2 2

[alsa-devel] [PATCH 0/6] ASoC: rsnd: add CTU support
by Kuninori Morimoto 05 Mar '16

05 Mar '16

Hi Mark These patches adds CTU (= Channel Transfer Unit) support to Renesas R-Car sound driver. It can exchange channel order ex) channel {0,1,2,3} -> channel {3,2,1,0} And it can add/reduce each channels sound ex) (input 0th ch x 0.5) + (input 2nd ch x 0.5) = output 1st ch You need to check R-Car datasheet for each detail settings, or check 6) patch Kuninori Morimoto (6): 1) ASoC: rsnd: don't call update callback if it was NULL 2) ASoC: rsrc-card: add convert channels support 3) ASoC: rsnd: tidyup rsnd_ssiu_init_gen2() 4) ASoC: rsnd: add CTU basic support 5) ASoC: rsnd: add rsnd_runtime_channel_xxx() 6) ASoC: rsnd: add CTU support Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 1 + sound/soc/sh/rcar/core.c | 71 ++++++++++++++---------- sound/soc/sh/rcar/ctu.c | 287 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++-------------- sound/soc/sh/rcar/dvc.c | 2 +- sound/soc/sh/rcar/mix.c | 2 +- sound/soc/sh/rcar/rsnd.h | 9 +++- sound/soc/sh/rcar/rsrc-card.c | 22 +++++--- sound/soc/sh/rcar/src.c | 2 +- sound/soc/sh/rcar/ssi.c | 15 +++--- sound/soc/sh/rcar/ssiu.c | 11 ++-- 10 files changed, 330 insertions(+), 92 deletions(-) Best regards --- Kuninori Morimoto

3 14

[alsa-devel] [PATCH] ASoC: wm8994: codec driver should manage its own MCLK
by Zidan Wang 05 Mar '16

05 Mar '16

codec driver should manage its own master clock, MCLK1 and MCLK2. When bias_level change from standby to prepare, enable codec MCLK. When bias_level change from prepare to standby, disable codec MCLK. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- drivers/mfd/wm8994-core.c | 7 +++++++ include/linux/mfd/wm8994/pdata.h | 3 +++ sound/soc/codecs/wm8994.c | 17 +++++++++++++++-- 3 files changed, 25 insertions(+), 2 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 7eec619..ace2cea 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -14,6 +14,7 @@ #include <linux/kernel.h> #include <linux/module.h> +#include <linux/clk.h> #include <linux/slab.h> #include <linux/i2c.h> #include <linux/err.h> @@ -270,6 +271,7 @@ static int wm8994_set_pdata_from_of(struct wm8994 *wm8994) { struct device_node *np = wm8994->dev->of_node; struct wm8994_pdata *pdata = &wm8994->pdata; + char tmp[8]; int i; if (!np) @@ -310,6 +312,11 @@ static int wm8994_set_pdata_from_of(struct wm8994 *wm8994) if (pdata->ldo[1].enable < 0) pdata->ldo[1].enable = 0; + for (i = 0; i < WM8994_NUM_MCLK; i++) { + sprintf(tmp, "MCLK%d", i + 1); + pdata->mclk[i] = devm_clk_get(wm8994->dev, tmp); + } + return 0; } #else diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 90c6052..657a828 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -18,6 +18,7 @@ #define WM8994_NUM_LDO 2 #define WM8994_NUM_GPIO 11 #define WM8994_NUM_AIF 3 +#define WM8994_NUM_MCLK 2 struct wm8994_ldo_pdata { /** GPIOs to enable regulator, 0 or less if not available */ @@ -233,6 +234,8 @@ struct wm8994_pdata { * GPIO for the IRQ pin if host only supports edge triggering */ int irq_gpio; + + struct clk *mclk[WM8994_NUM_MCLK]; }; #endif diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2ccbb32..7379eae 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> @@ -2474,6 +2475,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + struct wm8994_pdata *pdata = &control->pdata; + int i; wm_hubs_set_bias_level(codec, level); @@ -2495,8 +2498,13 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; } - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) { active_reference(codec); + + for (i = 0; i < WM8994_NUM_MCLK; i++) + if (!IS_ERR(pdata->mclk[i])) + clk_prepare_enable(pdata->mclk[i]); + } break; case SND_SOC_BIAS_STANDBY: @@ -2524,8 +2532,13 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_LINEOUT2_DISCH); } - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE) + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE) { + for (i = 0; i < WM8994_NUM_MCLK; i++) + if (!IS_ERR(pdata->mclk[i])) + clk_disable_unprepare(pdata->mclk[i]); + active_dereference(codec); + } /* MICBIAS into bypass mode on newer devices */ switch (control->type) { -- 1.9.1

3 2

[alsa-devel] [PATCH] ALSA: compress: fix more typos
by Vinod Koul 04 Mar '16

04 Mar '16

More inspection of code revealed few more typos so fix them as well Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c4b14b5f1b35..7d59dbda8897 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -703,7 +703,7 @@ static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) /* * We are called with lock held. So drop the lock while we wait for - * drain complete notfication from the driver + * drain complete notification from the driver * * It is expected that driver will notify the drain completion and then * stream will be moved to SETUP state, even if draining resulted in an @@ -758,7 +758,7 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; - /* you can signal next track isf this is intended to be a gapless stream + /* you can signal next track if this is intended to be a gapless stream * and current track metadata is set */ if (stream->metadata_set == false) -- 1.9.1

2 1

Re: [alsa-devel] [PATCH v3] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Nicolin Chen 04 Mar '16

04 Mar '16

On Fri, Mar 04, 2016 at 08:54:10AM +0000, Zidan Wang wrote: > On Friday, March 04, 2016 3:40 PM, Nicolin Chen wrote: > > > > > cpu-dai1 : The phandle of an CPU DAI controller for AIF1 > > > > cpu-dai2 : The phandle of an CPU DAI controller for AIF2 > > > > cpu-dai3 : The phandle of an CPU DAI controller for AIF3 > > > > > > > > > + data->card.num_links = DAI_LINK_NUM; > > > > > + data->card.dai_link = data->dai_link; > > > > > > > > Then, rather than hard coding the total DAI link number and creating > > > > useless dummy links, you may create the links dynamically according > > > > to the presence of the three properties above. > > > > > > I don't know how to using cpu-dai1/2/3. There don't have voice and bt > > controller on the IMX cpu, don't have such cpu nodes. > > > Do you means our hardware connection are wrong, and we must have voice > > and bt controller on the cpu? > > > > Okay. I was trying to give you an excuse to have all three aifs included, > > as I thought you might eventually need them to *connect* something via DT. > > If you think you don't have any idea to pre-allocate these links, you > > probably should ignore my suggestion here and just follow Mark's comments: > > I am just confused by our hardward connection. If dummy cpu dai is not ok, I think I should follow your suggestion. > > If we want to keep our hardware connection, which bt and voice directly connect to codec, and don't have bt and voice controller. > What's the best way? The only way to keep them while not having BT and voice controllers is just what I suggested you to make it through DT. Devicetree is a script to reflect your exact hardware connections. You can dis-link them for now in the DT as they are not "connected" in your current design and then link them whenever you get them connected -- I think there should be connections since you are using a validation board. You just don't have the extension board or something. So you may try to reach your hardware team, especially designer, for detail. The idea here is similar to the ASRC links in fsl-asoc-card driver: users can drop ASRC by merely removing the asrc phandle in the DT, depending on their board design or requirement. The drawback of this way is that you need to figure out an accurate DT binding: you might need to figure out a better name rather than "cpu-dai" because it sounds like WM8958 may not act as a Codec at all when it connects your BT route. That means WM8958 will become a cpu-dai somehow? You can send a patch with a decent DT binding. I believe Mark will give some better advice.

1 0

[alsa-devel] [PATCH 0/4] Fix headphone detection on Acer Chromebook 13
by Jonathan Tinkham 04 Mar '16

04 Mar '16

Headphone output works, but headphone detection does not. The problem is improper inversion of the jack gpio, and improper naming of the DAPM switch. With these changes, headphone insertion and removal is picked up by userspace, and the switch properly toggled. Jonathan Tinkham (4): soc/sound: tegra_max98090: do not invert headphone jack sound/soc: tegra_max98090: rename headphone jack DAPM sound/soc: tegra_max98090: document headphone jack rename soc/sound: tegra_max98090: update dts affected by rename .../devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 6 +++--- arch/arm/boot/dts/tegra124-nyan.dtsi | 4 ++-- arch/arm/boot/dts/tegra124-venice2.dts | 4 ++-- sound/soc/tegra/tegra_max98090.c | 10 +++++----- 4 files changed, 12 insertions(+), 12 deletions(-) -- 2.7.0

4 37

[alsa-devel] CM6206-LX usb sound card
by Nikita Orlov 04 Mar '16

04 Mar '16

Hello! First I sent these questions to alsa-user, but I think this mailing list is more proper place to ask such questions. I have cheap chinese usb audio card, based on CM6206-LX chip from CMEDIA. CM6206-LX is similar to CM6206, but not 7.1 channels output, it is 5.1 channels. I use ubuntu 15.* with kernel 4.2.0-30-generic. dmesg show me some warnings about sound card: [31010.394487] usb 1-1: new full-speed USB device number 24 using xhci_hcd [31010.523953] usb 1-1: New USB device found, idVendor=0d8c, idProduct=0102 [31010.523956] usb 1-1: New USB device strings: Mfr=0, Product=2, SerialNumber=0 [31010.523957] usb 1-1: Product: USB Sound Device [31010.526235] input: USB Sound Device as /devices/pci0000:00/0000:00:14.0/usb1/1-1/1-1:1.3/0003:0D8C:0102.0010/input/input33 [31010.578876] hid-generic 0003:0D8C:0102.0010: input,hidraw0: USB HID v1.00 Device [USB Sound Device ] on usb-0000:00:14.0-1/input3 [31010.643908] usb 1-1: current rate 30464 is different from the runtime rate 96000 [31010.646123] usb 1-1: Warning! Unlikely big volume range (=8065), cval->res is probably wrong. [31010.646126] usb 1-1: [9] FU [Mic Playback Volume] ch = 2, val = -6144/1921/1 [31010.646485] usb 1-1: Warning! Unlikely big volume range (=8065), cval->res is probably wrong. [31010.646486] usb 1-1: [11] FU [Line Playback Volume] ch = 2, val = -6144/1921/1 [31010.647884] usb 1-1: Warning! Unlikely big volume range (=6928), cval->res is probably wrong. [31010.647887] usb 1-1: [8] FU [Mic Capture Volume] ch = 2, val = -4096/2832/1 [31010.648291] usb 1-1: Warning! Unlikely big volume range (=6928), cval->res is probably wrong. [31010.648295] usb 1-1: [15] FU [Line Capture Volume] ch = 2, val = -4096/2832/1 [31010.648662] usb 1-1: Warning! Unlikely big volume range (=6928), cval->res is probably wrong. [31010.648664] usb 1-1: [2] FU [PCM Capture Volume] ch = 2, val = -4096/2832/1 [31010.648887] usbcore: registered new interface driver snd-usb-audio There are two types of errors, first that current rate 30464 is different from the runtime rate 96000. Unfortunately I have no idea about this warning. And warnings like Unlikely big volume range (=8065), cval->res is probably wrong. As I understand it is because auto value range for some mixers are not proper detected. I have test capturing with simple test app. It seems only one avalible sample rate is 48000 Hz. printf("Sample rate = %d\n", SampleRate); snd_pcm_hw_params_set_rate_near(Handle, params, &SampleRate, &dir); printf("Sample rate = %d\n", SampleRate); After calling these function SampleRate becomes 48000, even if before it was 44100. Maybe it is related to first error in dmesg. Anybody knows? But, according to cat /proc/asound/card2/stream0 USB Sound Device at usb-0000:00:14.0-1, full speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S16_LE Channels: 8 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 1 Altset 2 Format: S16_LE Channels: 2 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 1 Altset 3 Format: S16_LE Channels: 4 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 1 Altset 4 Format: S16_LE Channels: 6 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 1 Altset 5 Format: S16_LE Channels: 2 Endpoint: 6 OUT (ADAPTIVE) Rates: 96000 Capture: Status: Stop Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 44100, 48000 It is able to capture 44100. I have tried to capture via arecord: arecord -D hw:2,0 -f cd -c 2 > ./test2.wav and VLC said it is 44100, but I don`t have microphone at the moment to test is there proper sound recorded or not. I will try it later. I have found that init of sound card is related to snd_usb_audio.ko module and learn a bit how it works. There is a quirk that write internal registers. According to mailing list archives dated in 2009, values were found by usb traffic sniffing under Windows. I have found CM6206 datasheet and check description of internal registers. http://wenku.baidu.com/link?url=vkQLGhKUuvVmHC7vyobjKlF0yixpjn3ueAdD81OO6Ul… It is link to datasheet. Website is chinese, but datasheet itsel is in english. To see it after 3rd page click 下载文档到电脑，查找使用更方便. But there are no interesting things about registers and mostly values that are exist in current snd_usb_audio must work ok. Also in datasheet there some info about values of volume ranges that can be related to second dmesg warnings. I have uploaded images of related info from datasheet here https://yadi.sk/d/1kdCFs7Gpv8L8 I have try with mixer.c in sound/usb/ dir of kernel tree (4.2.0) and tried to add experimental code in volume_control_quirks function. But it seems function is not called for my device for some reason. It is more complicated that I think first. So another question is how to make quirk for proper mixer values? -- Nikita Orlov Skype: nik_stet QQ: 2717846083

1 0

[alsa-devel] compress offload patch
by Eric Laurent 04 Mar '16

04 Mar '16

Takashi, Vinod, Can you help with merging the attached patch for compress offload core upstream? Thanks. Eric.

3 4

[alsa-devel] [PATCH 1/2] ALSA: compress: Add SNDRV_PCM_STATE_PREPARED state explanation
by Vinod Koul 04 Mar '16

04 Mar '16

Stream states were explained in the code comments but SNDRV_PCM_STATE_PREPARED was missed so add it Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7fac3cae8abd..51a8eb495f54 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -74,6 +74,9 @@ struct snd_compr_file { * SNDRV_PCM_STATE_SETUP: When stream has been initialized. This is done by * calling SNDRV_COMPRESS_SET_PARAMS. running streams will come to this * state at stop by calling SNDRV_COMPRESS_STOP, or at end of drain. + * SNDRV_PCM_STATE_PREPARED: When a stream has been written to (for + * playback only). User after setting up stream writes the data buffer + * before starting the stream. * SNDRV_PCM_STATE_RUNNING: When stream has been started and is * decoding/encoding and rendering/capturing data. * SNDRV_PCM_STATE_DRAINING: When stream is draining current data. This is done -- 1.9.1

2 2

[alsa-devel] USB audio2.0 status interrupt endpoint
by Puneet Sharma 04 Mar '16

04 Mar '16

Hello all, I am using the uac2 kernel driver(drivers/usb/gadget/f_uac2.c) having kernel version 3.10.31 for an embedded board. The board is presented as a USB device to a Linux PC having USB host controller in it. I have added an interrupt endpoint in the uac2 kernel driver to support the status interrupt endpoint request as per usb audio 2.0 specification. For the interrupt transfer to happen, I prepared the interrupt packet structure as per audio 2.0 specification and queued the endpoint to transfer it to the host driver. The interrupt structure is prepared as follows: bInfo : 0 (Interface request) bAttribute : 0x1 (CUR request) wValue : 0x200 (VOLUME feature unit control selector) wIndex : 0x700 (FEATURE UNIT ID) As per my understanding, when the above interrupt endpoint is transferred to host in response to Interrupt IN request, the host should send me a GET CUR volume class-specific request to further qualify the cause of interrupt. However, I am not getting any class-specific request from host in response to the status interrupt transferred to host. My query is do i need to implement a mixer interface in the uac2 kernel driver to get the request or i am doing something wrong in the process. I am stuck onto this and cannot proceed further. Thanks Puneet

2 2

[alsa-devel] writing an alsa driver
by Michele Curti 04 Mar '16

04 Mar '16

Hi all, I'm going to *try* to "write" a driver for the sound card on my new laptop, an asus x205ta. To be fair I have almost no experience but I'm hoping that all the code is already present and it only needs some adaptations. The cpu is a intel bay trail z3735f, the sound card is a realtek rt5648 that has the same product Id of the rt5645 (0x6308). I see in kernel sources: - the codec driver sound/soc/codecs/rt5645.c - a baytrail board that uses the rt5640 sound/soc/intel/boards/byt-rt5640.c - a braswell board that uses the rt5645 sound/soc/intel/boards/cht_bsw_rt5645.c What do you think? Is it feasible, given a good amount of free-time? :p Thanks, Michele

10 46

[alsa-devel] [PATCH v2] ASoC: pcm3168a: remove unused variable
by Sudip Mukherjee 04 Mar '16

04 Mar '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> While building we were getting build warning about: sound/soc/codecs/pcm3168a.c:403:21: warning: variable 'channels' set but not used The variable channels were being assigned some value but that was never reused. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- v2: From: address changed sound/soc/codecs/pcm3168a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 0c7248a..992a77e 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -400,13 +400,12 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, struct pcm3168a_priv *pcm3168a = snd_soc_codec_get_drvdata(codec); bool tx, master_mode; u32 val, mask, shift, reg; - unsigned int rate, channels, fmt, ratio, max_ratio; + unsigned int rate, fmt, ratio, max_ratio; int i, min_frame_size; snd_pcm_format_t format; rate = params_rate(params); format = params_format(params); - channels = params_channels(params); ratio = pcm3168a->sysclk / rate; -- 1.9.1

1 0

[alsa-devel] [PATCH] Make alsa asound.h header more portable.
by Thomas Klausner 04 Mar '16

04 Mar '16

From: Thomas Klausner <wiz(a)NetBSD.org> This header is used as-is in the alsa-lib userland library, which is portable to other operating systems. For this reason, include linux/types.h only on Linux systems. Add sys/ioctl.h for _IOR/_IOW/etc. (works at least on *BSD and Solaris). Signed-off-by: Thomas Klausner <wiz(a)NetBSD.org> --- include/uapi/sound/asound.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index a82108e..3d581b5 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -23,7 +23,11 @@ #ifndef _UAPI__SOUND_ASOUND_H #define _UAPI__SOUND_ASOUND_H +#ifdef __linux__ #include <linux/types.h> +#else +#include <sys/ioctl.h> +#endif #ifndef __KERNEL__ #include <stdlib.h> -- 2.7.1

2 4

[alsa-devel] [PATCH 1/2] ALSA: hda - hdmi add wmb barrier for audio component
by libin.yang＠linux.intel.com 04 Mar '16

04 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> To make sure audio_ptr is set before intel_audio_codec_enable() or intel_audio_codec_disable() calling pin_eld_notify(), this patch adds wmb barrier to prevent optimizing. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index aa2d6a2..b5190e0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2939,6 +2939,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (codec_has_acomp(codec)) { codec->depop_delay = 0; spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; snd_hdac_i915_register_notifier(&spec->i915_audio_ops); } -- 1.9.1

2 1

Re: [alsa-devel] [PATCH v3] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Nicolin Chen 04 Mar '16

04 Mar '16

On Fri, Mar 04, 2016 at 07:15:44AM +0000, Zidan Wang wrote: > > cpu-dai1 : The phandle of an CPU DAI controller for AIF1 > > cpu-dai2 : The phandle of an CPU DAI controller for AIF2 > > cpu-dai3 : The phandle of an CPU DAI controller for AIF3 > > > > > + data->card.num_links = DAI_LINK_NUM; > > > + data->card.dai_link = data->dai_link; > > > > Then, rather than hard coding the total DAI link number and creating > > useless dummy links, you may create the links dynamically according to > > the presence of the three properties above. > > I don't know how to using cpu-dai1/2/3. There don't have voice and bt controller on the IMX cpu, don't have such cpu nodes. > Do you means our hardware connection are wrong, and we must have voice and bt controller on the cpu? Okay. I was trying to give you an excuse to have all three aifs included, as I thought you might eventually need them to *connect* something via DT. If you think you don't have any idea to pre-allocate these links, you probably should ignore my suggestion here and just follow Mark's comments: [ Why are you mapping in dummy DAIs? If these devices aren't connected then they're not connected and you shouldn't represent them. If they are connected to something then describe those connections, possibly in followup patches if you have other devices you need to support upstream first. ]

1 0

[alsa-devel] [PATCH] ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616
by Caesar Wang 04 Mar '16

04 Mar '16

This patch fixes that hasn't built the rt5616 driver with 'CONFIG_SND_SOC_RT5616=y' in .config. The tristate is the prompt on the 'make menuconfig', in other words, that can't show the prompt and select it if we don't say what's the tristate. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4383966..14eb5f3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -623,7 +623,7 @@ config SND_SOC_RT5514 tristate config SND_SOC_RT5616 - tristate + tristate "Realtek RT5616 CODEC" config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" -- 1.9.1

1 0

[alsa-devel] [PATCH] sound/usb: Use meaninful names for goto labels
by Shuah Khan 04 Mar '16

04 Mar '16

Fix to use meaningful names instead of numbered goto labels Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- sound/usb/media.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index b87d5b3..ab6a642 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -84,21 +84,21 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, ret = media_device_register_entity(mctl->media_dev, &mctl->media_entity); if (ret) - goto err1; + goto free_mctl; mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, MAJOR(pcm_dev->devt), MINOR(pcm_dev->devt)); if (!mctl->intf_devnode) { ret = -ENOMEM; - goto err2; + goto unregister_entity; } mctl->intf_link = media_create_intf_link(&mctl->media_entity, &mctl->intf_devnode->intf, MEDIA_LNK_FL_ENABLED); if (!mctl->intf_link) { ret = -ENOMEM; - goto err3; + goto devnode_remove; } /* create link between mixer and audio */ @@ -109,7 +109,7 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, &mctl->media_entity, 0, MEDIA_LNK_FL_ENABLED); if (ret) - goto err4; + goto remove_intf_link; break; } } @@ -117,13 +117,13 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, subs->media_ctl = mctl; return 0; -err4: +remove_intf_link: media_remove_intf_link(mctl->intf_link); -err3: +devnode_remove: media_devnode_remove(mctl->intf_devnode); -err2: +unregister_entity: media_device_unregister_entity(&mctl->media_entity); -err1: +free_mctl: kfree(mctl); return ret; } -- 2.5.0

2 1

[alsa-devel] [PATCH 2/2] ALSA: hda - hdmi defer to register acomp eld notifier
by libin.yang＠linux.intel.com 04 Mar '16

04 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> Defer to register acomp eld notifier until hdmi audio driver is fully ready. After registering eld notifier, gfx driver can use this callback function to notify audio driver the monitor connection event. However this action may happen when audio driver is adding the pins or doing other initialization. This is not always safe, however. For example, using per_pin->lock before the lock is initialized. Let's register the eld notifier after the initialization is done. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b5190e0..9bdb301 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2936,18 +2936,6 @@ static int patch_generic_hdmi(struct hda_codec *codec) is_broxton(codec)) codec->core.link_power_control = 1; - if (codec_has_acomp(codec)) { - codec->depop_delay = 0; - spec->i915_audio_ops.audio_ptr = codec; - /* intel_audio_codec_enable() or intel_audio_codec_disable() - * will call pin_eld_notify with using audio_ptr pointer - * We need make sure audio_ptr is really setup - */ - wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); - } - if (hdmi_parse_codec(codec) < 0) { if (spec->i915_bound) snd_hdac_i915_exit(&codec->bus->core); @@ -2969,6 +2957,18 @@ static int patch_generic_hdmi(struct hda_codec *codec) init_channel_allocations(); + if (codec_has_acomp(codec)) { + codec->depop_delay = 0; + spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); + spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + } + WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec)); return 0; } -- 1.9.1

1 0

[alsa-devel] Creative ca0132 on Gigabyte Z170X Gaming 7 motherboard: no sound
by Scott Doty 04 Mar '16

04 Mar '16

Hello, Didn't see a response on alsa-user, so thought I'd better roll up my sleeves and get to work. :) The ca0132 codec seems to do everything on my Gigabyte Z170X Gaming 7 motherboard but make sound. I can see the mixer settings, change them, even see the DSP settings -- but no sound out, either with HP or SPDIF. http://www.alsa-project.org/db/?f=6373811de293125e4e8c345c0c2d4a736c1fc2b8 I do see this in the startup messages: [ 9.314893] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 9.321116] snd_hda_intel 0000:00:1f.3: failed to add i915 component master (-19) I've tried all the fixes I could find reference to, as well as using iecset, but no joy. I also tried editing the driver to disable the "vga_switcheroo" flag, but that didn't work either. (Which is too bad, I thought that would be a quick fix that I could report. :/ ) If someone could point me in the right direction, I think I can investigate this and get it working. Any assistance greatly appreciated. Thanks, -Scott

2 2

[alsa-devel] [PATCH] Provide sequencer sound card number / PID via alsa-lib
by Martin Koegler 03 Mar '16

03 Mar '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This changes adds support for exposing the card number for kernel clients as well as the PID for user client. It supports kernels with and without the required support. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- include/seq.h | 2 ++ include/sound/asequencer.h | 6 ++++-- src/seq/seq.c | 26 ++++++++++++++++++++++++++ src/seq/seq_hw.c | 12 +++++++++++- 4 files changed, 43 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index 9576822..d05940e 100644 --- a/include/seq.h +++ b/include/seq.h @@ -143,6 +143,8 @@ snd_seq_client_type_t snd_seq_client_info_get_type(const snd_seq_client_info_t * const char *snd_seq_client_info_get_name(snd_seq_client_info_t *info); int snd_seq_client_info_get_broadcast_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..3ac70fd 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -24,7 +24,7 @@ /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +356,9 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card; /* RO: card number[kernel] */ + int pid; /* RO: pid[user] */ + char reserved[56]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..4405e68 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,6 +1522,32 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** + * \brief Get the sound card number. + * \param info client_info container + * \return card number or -1 if value is not available. + * + * Only available for SND_SEQ_KERNEL_CLIENT clients. + */ +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) +{ + assert(info); + return info->card; +} + +/** + * \brief Get the owning PID. + * \param info client_info container + * \return pid or -1 if value is not available. + * + * Only available for SND_SEQ_USER_CLIENT clients. + */ +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info) +{ + assert(info); + return info->pid; +} + +/** * \brief (DEPRECATED) Get the event filter bitmap of a client_info container * \param info client_info container * \return NULL if no event filter, or pointer to event filter bitmap diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..578ef12 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,10 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -368,6 +373,10 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -480,6 +489,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->version = ver; if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { -- 2.1.4

1 0

[alsa-devel] [PATCH v3] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 03 Mar '16

03 Mar '16

This is the initial imx-wm8958 device-tree-only machine driver working with fsl_sai driver. This sound card has three dai link, HIFI, VOICE and BT dai. HIFI dai link will support codec master and slave mode. VOICE an BT dai link have dummy cpu dai, and just support codec master mode. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v2->v3: 1. move codec mclk enable to codec driver 2. remove rate constraints 3. remove GPIO configurations, which can be configured in device tree .../devicetree/bindings/sound/imx-audio-wm8958.txt | 49 +++ arch/arm/configs/imx_v6_v7_defconfig | 1 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 327 +++++++++++++++++++++ 5 files changed, 393 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..7a2b8a9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,49 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - cpu-dai : The phandle of an CPU DAI controller + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + + - fsl,hifi-dai-master : If present, hifi dai works as master, and will + provide bit clock and frame clock. Otherwise, + hifi dai works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai = <&sai1>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; + + fsl,hifi-dai-master; +}; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..7ac5856 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -251,6 +251,7 @@ CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_PHYCORE_AC97=y CONFIG_SND_SOC_EUKREA_TLV320=y +CONFIG_SND_SOC_IMX_WM8958=y CONFIG_SND_SOC_IMX_WM8962=y CONFIG_SND_SOC_IMX_SGTL5000=y CONFIG_SND_SOC_IMX_SPDIF=y diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + 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..2a781c0 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-wm8958-objs := imx-wm8958.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 @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(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-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.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-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..01e7044 --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,327 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * 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/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link *dai_link; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_hifi_dai_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_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; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + if (!data->is_hifi_dai_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (!data->is_hifi_dai_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(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 = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == HIFI_DAI && !data->is_hifi_dai_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + .ops = &imx_voice_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + int ret, i; + + cpu_np = of_parse_phandle(np, "cpu-dai", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + return -EINVAL; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find cpu dai 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) + return -ENOMEM; + + data->dai_link = imx_wm8958_dai_link; + + /* + * AIF1 support codec master and slave mode + * AIF2 and AIF3 just support codec master mode + */ + if (of_property_read_bool(pdev->dev.of_node, "fsl,hifi-dai-master")) { + data->is_hifi_dai_master = true; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + data->is_hifi_dai_master = false; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(data->mclk[i])) + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->dai_link[HIFI_DAI].cpu_dai_name = dev_name(&cpu_pdev->dev); + data->dai_link[HIFI_DAI].platform_of_node = cpu_np; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = DAI_LINK_NUM; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + 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; + } + +fail: + of_node_put(cpu_np); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: cht_bsw_rt5645: Fix writing to string literal
by Carlo Caione 03 Mar '16

03 Mar '16

From: Carlo Caione <carlo(a)endlessm.com> We cannot use strcpy() to write to a const char * location. This is causing a 'BUG: unable to handle kernel paging request' error at boot when using the cht-bsw-rt5645 driver. With this patch we also fix a wrong indexing in the driver where the codec_name of the wrong dai_link is being overwritten. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 2d3afdd..a7b96a9 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -367,8 +367,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) + if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00")) + card->dai_link[i].codec_name = kstrdup(codec_name, GFP_KERNEL); + snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- 2.5.0

5 6

Re: [alsa-devel] Compute Stick baytrail audio problem
by Giacomo Comes 03 Mar '16

03 Mar '16

On Tue, Feb 23, 2016 at 12:32:23PM +0900, Toyo Abe wrote: > > Hi, > > I'd struggled a similar underrun problem on my baytrail device. The dmesg showed > the following warnings. > > [15356.757938] had: Driver detected 2 missed buffer done interrupt(s)!!!! > [15356.761594] had: Driver detected 1 missed buffer done interrupt(s)!!!! > [15356.774775] had: Unable to clear UNDERRUN bits > [15356.791312] had: Driver detected 1 missed buffer done interrupt(s)!!!! > [15356.808148] had: Unable to clear UNDERRUN bits > [15356.824692] had: Driver detected 2 missed buffer done interrupt(s)!!!! > > After some investigation, I found some problems on the driver and fixed a few of > them. I pushed the fix to my github. I hope it might help you. > https://raw.githubusercontent.com/toyoabe/baytrailaudio/fix-underrun/hdmi_a… > > Best Regards, > -toyo Hi, that was it!. I added your changes and now the audio works. I'm happy. I got finally every subsystem of the Compute Stick working. I have a minor question tought. Alsamixer says: This sound device does not have any controls. Is there a way to change the volume without enabling pulseaudio? Just for documentation, here is the small change that Toyo did to make the underrun disappear: diff -Nraub a/drivers/gpu/drm/i915/i915_irq.c b/drivers/gpu/drm/i915/i915_irq.c --- a/drivers/gpu/drm/i915/i915_irq.c 2016-02-23 08:03:42.406016576 -0400 +++ b/drivers/gpu/drm/i915/i915_irq.c 2016-02-23 08:03:58.714016576 -0400 @@ -2717,11 +2717,14 @@ int pipe = 1; spin_lock_irqsave(&dev_priv->irq_lock, irqflags); + i915_enable_lpe_pipestat(dev_priv, pipe); + imr = I915_READ(VLV_IMR); /* Audio is on Stream A */ imr &= ~I915_LPE_PIPE_A_INTERRUPT; I915_WRITE(VLV_IMR, imr); - i915_enable_lpe_pipestat(dev_priv, pipe); + I915_WRITE(VLV_IER, ~imr); + POSTING_READ(VLV_IER); spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); return 0; @@ -2769,7 +2772,10 @@ spin_lock_irqsave(&dev_priv->irq_lock, irqflags); imr = I915_READ(VLV_IMR); imr |= I915_LPE_PIPE_A_INTERRUPT; + I915_WRITE(VLV_IER, ~imr); I915_WRITE(VLV_IMR, imr); + POSTING_READ(VLV_IMR); + i915_disable_lpe_pipestat(dev_priv, pipe); spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); diff -Nraub a/sound/hdmi_audio/intel_mid_hdmi_audio.c b/sound/hdmi_audio/intel_mid_hdmi_audio.c --- a/sound/hdmi_audio/intel_mid_hdmi_audio.c 2016-02-23 08:03:42.414016576 -0400 +++ b/sound/hdmi_audio/intel_mid_hdmi_audio.c 2016-02-23 08:03:58.722016576 -0400 @@ -1128,7 +1128,6 @@ pr_debug("HDMI status =0x%x\n", hdmi_status); if (hdmi_status & AUD_CONFIG_MASK_UNDERRUN) { i++; - hdmi_status &= ~AUD_CONFIG_MASK_UNDERRUN; had_write_register(AUD_HDMI_STATUS_v2, hdmi_status); } else break; Thanks again. Giacomo > > Pierre-Louis Bossart wrote: > > On 02/19/2016 02:39 PM, Giacomo Comes wrote: > >> Hi, > >> I have recently bought a Intel Compute Stick and I am having trouble with= > > the audio. > >> Intel provides a custom Ubuntu release that support audio, but I want to = > > run the > >> linux distribution of my choice on it (openSUSE 13.2 for the moment). > >> > >> In order to have HDMI audio support it is necessary to compile a kernel w= > > ith this patch > >> (https://raw.githubusercontent.com/01org/baytrailaudio/master/hdmi_audio_= > > 20150319.patch) > >> provided by Intel for kernel 3.16.7 (https://01.org/ubuntu-hdmi) > >> > >> I compiled the openSUSE kernel using the default openSUSE .config plus th= > > e following: > >> CONFIG_SUPPORT_HDMI=3Dy > >> CONFIG_SND_SOC=3Dm > >> CONFIG_SND_SOC_INTEL_SST=3Dm > >> CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=3Dm > >> > >> I booted the new kernel and aplay -l gives this output: > >> **** List of PLAYBACK Hardware Devices **** > >> card 0: IntelHDMI [IntelHDMI], device 0: IntelHDMI [IntelHDMI] > >> Subdevices: 0/1 > >> Subdevice #0: subdevice #0 > >> > >> but when I run: > >> aplay /usr/share/sounds/alsa/test.wav > >> the audio plays for .2 seconds and then nothing more. > >> Sometime it will play later a little bit more and stop again. > >> At some point these messages appears: > >> underrun!!! (at least 29325.946 ms long) > >> underrun!!! (at least 1535.889 ms long) > >> underrun!!! (at least 3196.213 ms long) > >> > >> > >> I have looked around on google for a hint on how to fix this issue but I = > > found nothing. > >> Any idea where to look for a solution? > >> > >> Attached there are the dmesg and lsmod output for ubuntu > >> with the working audio and openSUSE 13.2 plus patch > >> without the working audio. > >> I can provide any other information you need. > > > > There are enough messages that make me believe something is very wrong > > from the start > > > > 19.664486] had: ******** HAD DRIVER loading.. Ver: 0.01.003 > > [ 19.664549] hdmi_audio_probe dma_mask: 0 > > [ 19.664937] card->dev is NULL!!!!! Should not be this case > > [ 19.668176] [drm] mid_hdmi_audio_register: Scheduling HDMI audio work qu= > > eue > > [ 19.670745] [drm:i915_had_wq] *ERROR* Checking for HDMI connection at bo= > > ot > > > > Some of us at Intel are trying to clean this up but it'll take time. > > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

3 4

[alsa-devel] feeding back pkgsrc changs
by Thomas Klausner 03 Mar '16

03 Mar '16

Hi! I've just updated the pkgsrc packages for alsa-lib, alsa-utils and alsa-plugins to 1.1.0 and noticed that there are quite a number of local patches we have. I'd like to get them integrated. When I visited the alsa-projects web page, I noticed that the link to the bug tracker is dead. What is the recommended method to provide the patches? Thanks, Thomas

3 29

[alsa-devel] [PATCH 1/5] Introduce bswap.h for portable definitions of byte swap macros.
by Thomas Klausner 03 Mar '16

03 Mar '16

From: Thomas Klausner <wiz(a)NetBSD.org> Signed-off-by: Thomas Klausner <wiz(a)NetBSD.org> --- include/Makefile.am | 2 +- include/bswap.h | 39 +++++++++++++++++++++++++++++++++++++++ src/pcm/pcm_adpcm.c | 2 +- src/pcm/pcm_alaw.c | 2 +- src/pcm/pcm_copy.c | 2 +- src/pcm/pcm_dmix_generic.c | 2 +- src/pcm/pcm_file.c | 2 +- src/pcm/pcm_iec958.c | 2 +- src/pcm/pcm_lfloat.c | 2 +- src/pcm/pcm_linear.c | 2 +- src/pcm/pcm_meter.c | 2 +- src/pcm/pcm_misc.c | 2 +- src/pcm/pcm_mulaw.c | 2 +- src/pcm/pcm_null.c | 2 +- src/pcm/pcm_rate.c | 2 +- src/pcm/pcm_rate_linear.c | 2 +- src/pcm/pcm_route.c | 2 +- src/pcm/pcm_softvol.c | 2 +- 18 files changed, 56 insertions(+), 17 deletions(-) create mode 100644 include/bswap.h diff --git a/include/Makefile.am b/include/Makefile.am index 8c83c11..31a3f74 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -59,7 +59,7 @@ alsainclude_HEADERS += alisp.h endif noinst_HEADERS = alsa sys.h search.h list.h aserver.h local.h alsa-symbols.h \ - asoundlib-head.h asoundlib-tail.h type_compat.h + asoundlib-head.h asoundlib-tail.h bswap.h type_compat.h DISTCLEANFILES = stamp-vh version.h alsa asoundlib.h diff --git a/include/bswap.h b/include/bswap.h new file mode 100644 index 0000000..68adce3 --- /dev/null +++ b/include/bswap.h @@ -0,0 +1,39 @@ +/* + * ALSA lib - compatibility header for providing byte swapping macros + * Copyright (c) 2016 by Thomas Klausner <wiz(a)NetBSD.org> + * + * + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * 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 Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#ifndef __BSWAP_H +#define __BSWAP_H + +#if defined(__FreeBSD__) || defined(__DragonFly__) || defined(__NetBSD__) +#include <sys/endian.h> +#define bswap_16 bswap16 +#define bswap_32 bswap32 +#define bswap_64 bswap64 +#elif defined (__sun) +#include <sys/byteorder.h> +#define bswap_16 BSWAP_16 +#define bswap_32 BSWAP_32 +#define bswap_64 BSWAP_64 +#else +#include <byteswap.h> +#endif + +#endif diff --git a/src/pcm/pcm_adpcm.c b/src/pcm/pcm_adpcm.c index 1a83c5a..1c88c83 100644 --- a/src/pcm/pcm_adpcm.c +++ b/src/pcm/pcm_adpcm.c @@ -56,7 +56,7 @@ IMA compatibility project proceedings, Vol 2, Issue 2, May 1992. come across a good description of XA yet. */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_alaw.c b/src/pcm/pcm_alaw.c index db759e3..fa58441 100644 --- a/src/pcm/pcm_alaw.c +++ b/src/pcm/pcm_alaw.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_copy.c b/src/pcm/pcm_copy.c index 66d3a47..0dbd5bd 100644 --- a/src/pcm/pcm_copy.c +++ b/src/pcm/pcm_copy.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_dmix_generic.c b/src/pcm/pcm_dmix_generic.c index 9e9d3c3..40c0874 100644 --- a/src/pcm/pcm_dmix_generic.c +++ b/src/pcm/pcm_dmix_generic.c @@ -125,7 +125,7 @@ static void mix_select_callbacks(snd_pcm_direct_t *dmix) (1ULL << SND_PCM_FORMAT_S24_LE) | (1ULL << SND_PCM_FORMAT_S24_3LE) | \ (1ULL << SND_PCM_FORMAT_U8)) -#include <byteswap.h> +#include "bswap.h" static void generic_mix_areas_16_native(unsigned int size, volatile signed short *dst, diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 9ab6964..92eb072 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <ctype.h> #include <string.h> #include "pcm_local.h" diff --git a/src/pcm/pcm_iec958.c b/src/pcm/pcm_iec958.c index 38c4ce7..86ac9cf 100644 --- a/src/pcm/pcm_iec958.c +++ b/src/pcm/pcm_iec958.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_lfloat.c b/src/pcm/pcm_lfloat.c index 2f3e578..22201f8 100644 --- a/src/pcm/pcm_lfloat.c +++ b/src/pcm/pcm_lfloat.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_linear.c b/src/pcm/pcm_linear.c index 9a92abd..50df779 100644 --- a/src/pcm/pcm_linear.c +++ b/src/pcm/pcm_linear.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index 034f582..1b0ccb4 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -27,7 +27,7 @@ */ -#include <byteswap.h> +#include "bswap.h" #include <time.h> #include <pthread.h> #include <dlfcn.h> diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index 5870f82..7d2b05d 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -23,7 +23,7 @@ #include <stdlib.h> #include <unistd.h> #include <string.h> -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" diff --git a/src/pcm/pcm_mulaw.c b/src/pcm/pcm_mulaw.c index 011b2a5..15dbdc1 100644 --- a/src/pcm/pcm_mulaw.c +++ b/src/pcm/pcm_mulaw.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index 5e63caa..685618e 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <limits.h> #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index c76db25..41bddac 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -28,7 +28,7 @@ * */ #include <inttypes.h> -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_rate.h" diff --git a/src/pcm/pcm_rate_linear.c b/src/pcm/pcm_rate_linear.c index 7481b38..70399e0 100644 --- a/src/pcm/pcm_rate_linear.c +++ b/src/pcm/pcm_rate_linear.c @@ -21,7 +21,7 @@ */ #include <inttypes.h> -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_rate.h" diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index 646517d..3611603 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <math.h> #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index c6cfd88..802aa4b 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <math.h> #include "pcm_local.h" #include "pcm_plugin.h" -- 2.7.1

1 4

[alsa-devel] [PATCH v2 0/7] ALSA: hda: Move chmap helpers to core
by Subhransu S. Prusty 03 Mar '16

03 Mar '16

The HDMI channel map helper APIs and control registration can be reused by ASoC skylake HDMI drivers. A common chmap object is created, channel map control callback handlers, helper APIs and data structures are moved to the core. Driver object store an instance of this chmap object to register chmap controls and program channel allocation for HDMI infoframe. Driver registers ops for specific operations. changes in v2: - Splitted patches for easier review - Access most of chmap helpers through ops, thus reducing number of exported APIs for chmap support Subhransu S. Prusty (7): ALSA: hda - Create common chmap object ALSA: hda - Register chmap obj as priv data instead of codec ALSA: hda - Use hdmi name space for CEA spk alloc structure ALSA: hda - Add hdmi chmap verb programming ops to chmap object ALSA: hda - Use ops instead of directly accessing chmap helpers ALSA: hda - Move chmap support helpers/ops to core ALSA: hda - Use snd_hdmi namespace prefix for printing ch alloc api include/sound/hdmi_chmap.h | 75 ++++ sound/hda/Makefile | 2 +- sound/hda/hdmi_chmap.c | 791 ++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_eld.c | 31 +- sound/pci/hda/patch_hdmi.c | 830 +++++---------------------------------------- 5 files changed, 957 insertions(+), 772 deletions(-) create mode 100644 include/sound/hdmi_chmap.h create mode 100644 sound/hda/hdmi_chmap.c -- 1.9.1

3 14

[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Fix mic issues on Acer Aspire E1-472
by Simon South 03 Mar '16

03 Mar '16

This patch applies the microphone-related fix created for the Acer Aspire E1-572 to the E1-472 as well, as it uses the same Realtek ALC282 CODEC and demonstrates the same issues. This patch allows an external, headset microphone to be used and limits the gain on the (quite noisy) internal microphone. Signed-off-by: Simon South <simon(a)simonsouth.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1527fc..5c672f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5224,6 +5224,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), + SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), -- 2.5.0

2 1

[alsa-devel] [PATCH 0/5] ASoC: pxa: Fix enum ctl accesses
by Takashi Iwai 02 Mar '16

02 Mar '16

Hi, here is a patchset to correct the wrong type accesses in pxa drivers for enum ctls. I didn't put Cc to stable for them, supposing these are only for 32bit systems. If not, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (5): ASoC: pxa: corgi: Fix enum ctl accesses in a wrong type ASoC: pxa: magician: Fix enum ctl accesses in a wrong type ASoC: pxa: poodle: Fix enum ctl accesses in a wrong type ASoC: pxa: spitz: Fix enum ctl accesses in a wrong type ASoC: pxa: tosa: Fix enum ctl accesses in a wrong type sound/soc/pxa/corgi.c | 12 ++++++------ sound/soc/pxa/magician.c | 6 +++--- sound/soc/pxa/poodle.c | 12 ++++++------ sound/soc/pxa/spitz.c | 12 ++++++------ sound/soc/pxa/tosa.c | 12 ++++++------ 5 files changed, 27 insertions(+), 27 deletions(-) -- 2.7.2

3 11

[alsa-devel] [PATCH v4 22/22] sound/usb: Use Media Controller API to share media resources
by Shuah Khan 02 Mar '16

02 Mar '16

Change ALSA driver to use Media Controller API to share media resources with DVB and V4L2 drivers on a AU0828 media device. Media Controller specific initialization is done after sound card is registered. ALSA creates Media interface and entity function graph nodes for Control, Mixer, PCM Playback, and PCM Capture devices. snd_usb_hw_params() will call Media Controller enable source handler interface to request the media resource. If resource request is granted, it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is returned. Media specific cleanup is done in usb_audio_disconnect(). Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- Changes since v3: - Fixed Kconfig to handle the following 1. CONFIG_MEDIA_SUPPORT and CONFIG_SND_USB_AUDIO are both modules CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER is selected 2. CONFIG_MEDIA_SUPPORT=y and CONFIG_SND_USB_AUDIO=m CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER is selected 3. CONFIG_MEDIA_SUPPORT=y and CONFIG_SND_USB_AUDIO=y CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER is selected 4. CONFIG_MEDIA_SUPPORT=m and CONFIG_SND_USB_AUDIO=y This is when we don't want CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER selected sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 +++ sound/usb/card.h | 3 + sound/usb/media.c | 318 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 72 +++++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 ++++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 11 files changed, 448 insertions(+), 5 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index a452ad7..d14bf41 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,6 +15,7 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE + select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -22,6 +23,9 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. +config SND_USB_AUDIO_USE_MEDIA_CONTROLLER + bool + config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 2d2d122..8dca3c4 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,6 +15,8 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o +snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o + snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 1f09d95..35fe256 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,6 +66,7 @@ #include "format.h" #include "power.h" #include "stream.h" +#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -561,6 +562,11 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; + if (quirk->media_device) { + /* don't want to fail when media_device_create() fails */ + media_device_create(chip, intf); + } + usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -617,6 +623,14 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } + /* + * Nice to check quirk && quirk->media_device + * need some special handlings. Doesn't look like + * we have access to quirk here + * Acceses mixer_list + */ + media_device_delete(chip); + /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 71778ca..34a0898 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,6 +105,8 @@ struct snd_usb_endpoint { struct list_head list; }; +struct media_ctl; + struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -156,6 +158,7 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ + struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c new file mode 100644 index 0000000..cff1459 --- /dev/null +++ b/sound/usb/media.c @@ -0,0 +1,318 @@ +/* + * media.c - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ + +#include <linux/init.h> +#include <linux/list.h> +#include <linux/mutex.h> +#include <linux/slab.h> +#include <linux/usb.h> + +#include <sound/pcm.h> +#include <sound/core.h> + +#include "usbaudio.h" +#include "card.h" +#include "mixer.h" +#include "media.h" + +static int media_enable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->enable_source) + return mctl->media_dev->enable_source(&mctl->media_entity, + &mctl->media_pipe); + return 0; +} + +static void media_disable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->disable_source) + mctl->media_dev->disable_source(&mctl->media_entity); +} + +int media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream) +{ + struct media_device *mdev; + struct media_ctl *mctl; + struct device *pcm_dev = &pcm->streams[stream].dev; + u32 intf_type; + int ret = 0; + u16 mixer_pad; + struct media_entity *entity; + + mdev = subs->stream->chip->media_dev; + if (!mdev) + return -ENODEV; + + if (subs->media_ctl) + return 0; + + /* allocate media_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; + mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; + mixer_pad = 1; + } else { + intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; + mctl->media_pad.flags = MEDIA_PAD_FL_SINK; + mixer_pad = 2; + } + mctl->media_entity.name = pcm->name; + media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) + goto err1; + + mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, + MAJOR(pcm_dev->devt), + MINOR(pcm_dev->devt)); + if (!mctl->intf_devnode) { + ret = -ENOMEM; + goto err2; + } + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &mctl->intf_devnode->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + ret = -ENOMEM; + goto err3; + } + + /* create link between mixer and audio */ + media_device_for_each_entity(entity, mdev) { + switch (entity->function) { + case MEDIA_ENT_F_AUDIO_MIXER: + ret = media_create_pad_link(entity, mixer_pad, + &mctl->media_entity, 0, + MEDIA_LNK_FL_ENABLED); + if (ret) + goto err4; + break; + } + } + + subs->media_ctl = mctl; + return 0; + +err4: + media_remove_intf_link(mctl->intf_link); +err3: + media_devnode_remove(mctl->intf_devnode); +err2: + media_device_unregister_entity(&mctl->media_entity); +err1: + kfree(mctl); + return ret; +} + +void media_stream_delete(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl && mctl->media_dev) { + struct media_device *mdev; + + mdev = subs->stream->chip->media_dev; + if (mdev && media_devnode_is_registered(&mdev->devnode)) { + media_devnode_remove(mctl->intf_devnode); + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + subs->media_ctl = NULL; + } +} + +int media_start_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + return media_enable_source(mctl); + return 0; +} + +void media_stop_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + media_disable_source(mctl); +} + +int media_mixer_init(struct snd_usb_audio *chip) +{ + struct device *ctl_dev = &chip->card->ctl_dev; + struct media_intf_devnode *ctl_intf; + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + struct media_mixer_ctl *mctl; + u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; + int ret; + + if (!mdev) + return -ENODEV; + + ctl_intf = chip->ctl_intf_media_devnode; + if (!ctl_intf) { + ctl_intf = media_devnode_create(mdev, intf_type, 0, + MAJOR(ctl_dev->devt), + MINOR(ctl_dev->devt)); + if (!ctl_intf) + return -ENOMEM; + chip->ctl_intf_media_devnode = ctl_intf; + } + + list_for_each_entry(mixer, &chip->mixer_list, list) { + + if (mixer->media_mixer_ctl) + continue; + + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; + mctl->media_entity.name = chip->card->mixername; + mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; + mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; + mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; + media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, + mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) { + kfree(mctl); + return ret; + } + + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &ctl_intf->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + kfree(mctl); + return -ENOMEM; + } + mctl->intf_devnode = ctl_intf; + mixer->media_mixer_ctl = mctl; + } + return 0; +} + +static void media_mixer_delete(struct snd_usb_audio *chip) +{ + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + + if (!mdev) + return; + + list_for_each_entry(mixer, &chip->mixer_list, list) { + struct media_mixer_ctl *mctl; + + mctl = mixer->media_mixer_ctl; + if (!mixer->media_mixer_ctl) + continue; + + if (media_devnode_is_registered(&mdev->devnode)) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + mixer->media_mixer_ctl = NULL; + } + if (media_devnode_is_registered(&mdev->devnode)) + media_devnode_remove(chip->ctl_intf_media_devnode); + chip->ctl_intf_media_devnode = NULL; +} + +int media_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) +{ + struct media_device *mdev; + struct usb_device *usbdev = interface_to_usbdev(iface); + int ret; + + mdev = media_device_get_devres(&usbdev->dev); + if (!mdev) + return -ENOMEM; + if (!mdev->dev) { + /* register media device */ + mdev->dev = &usbdev->dev; + if (usbdev->product) + strlcpy(mdev->model, usbdev->product, + sizeof(mdev->model)); + if (usbdev->serial) + strlcpy(mdev->serial, usbdev->serial, + sizeof(mdev->serial)); + strcpy(mdev->bus_info, usbdev->devpath); + mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); + media_device_init(mdev); + } + if (!media_devnode_is_registered(&mdev->devnode)) { + ret = media_device_register(mdev); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't register media device. Error: %d\n", + ret); + return ret; + } + } + + /* save media device - avoid lookups */ + chip->media_dev = mdev; + + /* Create media entities for mixer and control dev */ + ret = media_mixer_init(chip); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't create media mixer entities. Error: %d\n", + ret); + + /* clear saved media_dev */ + chip->media_dev = NULL; + + return ret; + } + return 0; +} + +void media_device_delete(struct snd_usb_audio *chip) +{ + struct media_device *mdev = chip->media_dev; + + media_mixer_delete(chip); + + if (mdev) { + if (media_devnode_is_registered(&mdev->devnode)) + media_device_unregister(mdev); + chip->media_dev = NULL; + } +} diff --git a/sound/usb/media.h b/sound/usb/media.h new file mode 100644 index 0000000..f66b4d7 --- /dev/null +++ b/sound/usb/media.h @@ -0,0 +1,72 @@ +/* + * media.h - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ +#ifndef __MEDIA_H + +#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER + +#include <media/media-device.h> +#include <media/media-entity.h> +#include <sound/asound.h> + +struct media_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad; + struct media_pipeline media_pipe; +}; + +/* + * One source pad each for SNDRV_PCM_STREAM_CAPTURE and + * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link + * to AUDIO Source +*/ +#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) + +struct media_mixer_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; + struct media_pipeline media_pipe; +}; + +int media_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface); +void media_device_delete(struct snd_usb_audio *chip); +int media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream); +void media_stream_delete(struct snd_usb_substream *subs); +int media_start_pipeline(struct snd_usb_substream *subs); +void media_stop_pipeline(struct snd_usb_substream *subs); +#else +static inline int media_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) + { return 0; } +static inline void media_device_delete(struct snd_usb_audio *chip) { } +static inline int media_stream_init(struct snd_usb_substream *subs, + struct snd_pcm *pcm, int stream) + { return 0; } +static inline void media_stream_delete(struct snd_usb_substream *subs) { } +static inline int media_start_pipeline(struct snd_usb_substream *subs) + { return 0; } +static inline void media_stop_pipeline(struct snd_usb_substream *subs) { } +#endif +#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 3417ef3..f378944 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,6 +3,8 @@ #include <sound/info.h> +struct media_mixer_ctl; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -22,6 +24,7 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; + struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9245f52..c3b8486 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,6 +35,7 @@ #include "pcm.h" #include "clock.h" #include "power.h" +#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -715,10 +716,14 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; + ret = media_start_pipeline(subs); + if (ret) + return ret; + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - return ret; + goto err_ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -732,22 +737,27 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - return -EINVAL; + ret = -EINVAL; + goto err_ret; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; + +err_ret: + media_stop_pipeline(subs); + return ret; } /* @@ -759,6 +769,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; + media_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1219,6 +1230,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; + int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1232,7 +1244,12 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - return setup_hw_info(runtime, subs); + ret = setup_hw_info(runtime, subs); + if (ret == 0) + ret = media_stream_init(subs, as->pcm, direction); + if (ret) + snd_usb_autosuspend(subs->stream->chip); + return ret; } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1241,6 +1258,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); + media_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c60a776..9d087b1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,6 +2886,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + .media_device = 1, \ } \ } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index c4dc577..6c7f5c1 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,6 +36,7 @@ #include "format.h" #include "clock.h" #include "stream.h" +#include "media.h" /* * free a substream @@ -52,6 +53,7 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); + media_stream_delete(subs); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b665d85..a161c7c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,6 +30,9 @@ * */ +struct media_device; +struct media_intf_devnode; + struct snd_usb_audio { int index; struct usb_device *dev; @@ -60,6 +63,8 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ + struct media_device *media_dev; + struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -110,6 +115,7 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; + bool media_device; const void *data; }; -- 2.5.0

3 4

[alsa-devel] [PATCH] ASoC: intel: remove unused variable
by Sudip Mukherjee 02 Mar '16

02 Mar '16

The variable cmd_id is only assigned some value and is never used. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/intel/atom/sst/sst_ipc.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3dc7358..8afa6fe 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -318,7 +318,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, union ipc_header_high msg_high; u32 msg_low; struct ipc_dsp_hdr *dsp_hdr; - unsigned int cmd_id; msg_high = msg->mrfld_header.p.header_high; msg_low = msg->mrfld_header.p.header_low_payload; @@ -357,7 +356,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, return; /* Copy command id so that we can use to put sst to reset */ dsp_hdr = (struct ipc_dsp_hdr *)data; - cmd_id = dsp_hdr->cmd_id; dev_dbg(sst_drv_ctx->dev, "cmd_id %d\n", dsp_hdr->cmd_id); if (sst_wake_up_block(sst_drv_ctx, msg_high.part.result, msg_high.part.drv_id, -- 1.9.1

4 13

[alsa-devel] [PATCH] ALSA: portman2x4: fix NULL pointer dereference
by Sudip Mukherjee 02 Mar '16

02 Mar '16

While registering pardev, the irq_func was also registered. As a result when we tried to probe for the card, an interrupt was generated and in the ISR we tried to dereference private_data. But private_data is still NULL as we have not yet done portman_create(). Lets probe for the device after card is created. Fixes: e6a1b7e88046 ("ALSA: portman2x4 - use new parport device model") Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/drivers/portman2x4.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 362533f..189e3e7 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -762,11 +762,6 @@ static int snd_portman_probe(struct platform_device *pdev) err = -EIO; goto free_pardev; } - err = portman_probe(p); - if (err) { - err = -EIO; - goto release_pardev; - } if ((err = portman_create(card, pardev, &pm)) < 0) { snd_printd("Cannot create main component\n"); @@ -774,6 +769,12 @@ static int snd_portman_probe(struct platform_device *pdev) } card->private_data = pm; card->private_free = snd_portman_card_private_free; + + err = portman_probe(p); + if (err) { + err = -EIO; + goto __err; + } if ((err = snd_portman_rawmidi_create(card)) < 0) { snd_printd("Creating Rawmidi component failed\n"); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: distinguish the different SSC
by Songjun Wu 02 Mar '16

02 Mar '16

Cpu_dai id always equals 0, can't distinguish the different SSC. Use platform_device id to record and distinguish the different SSC. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- drivers/misc/atmel-ssc.c | 1 + sound/soc/atmel/atmel_ssc_dai.c | 27 ++++++++++++++++++--------- 2 files changed, 19 insertions(+), 9 deletions(-) diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index e11a0bd..0516ecd 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -34,6 +34,7 @@ struct ssc_device *ssc_request(unsigned int ssc_num) if (ssc->pdev->dev.of_node) { if (of_alias_get_id(ssc->pdev->dev.of_node, "ssc") == ssc_num) { + ssc->pdev->id = ssc_num; ssc_valid = 1; break; } diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index ba8def5..2768970 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -285,7 +285,8 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; int ret; @@ -346,7 +347,8 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; @@ -392,7 +394,8 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; ssc_p->daifmt = fmt; return 0; @@ -404,7 +407,8 @@ static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { - struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; switch (div_id) { case ATMEL_SSC_CMR_DIV: @@ -445,7 +449,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int id = dai->id; + struct platform_device *pdev = to_platform_device(dai->dev); + int id = pdev->id; struct atmel_ssc_info *ssc_p = &ssc_info[id]; struct ssc_device *ssc = ssc_p->ssc; struct atmel_pcm_dma_params *dma_params; @@ -772,7 +777,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, static int atmel_ssc_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir; @@ -795,7 +801,8 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, static int atmel_ssc_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir; @@ -824,11 +831,12 @@ static int atmel_ssc_trigger(struct snd_pcm_substream *substream, static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) { struct atmel_ssc_info *ssc_p; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); if (!cpu_dai->active) return 0; - ssc_p = &ssc_info[cpu_dai->id]; + ssc_p = &ssc_info[pdev->id]; /* Save the status register before disabling transmit and receive */ ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); @@ -852,12 +860,13 @@ static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) { struct atmel_ssc_info *ssc_p; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); u32 cr; if (!cpu_dai->active) return 0; - ssc_p = &ssc_info[cpu_dai->id]; + ssc_p = &ssc_info[pdev->id]; /* restore SSC register settings */ ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); -- 1.7.9.5

4 4

[alsa-devel] [PATCH 1/2] ASoC: sh: rcar: core: don't open code of_device_get_match_data()
by Wolfram Sang 02 Mar '16

02 Mar '16

From: Wolfram Sang <wsa+renesas(a)sang-engineering.com> This change will also make Coverity happy by avoiding a theoretical NULL pointer dereference; yet another reason is to use the above helper function to tighten the code and make it more readable. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- Compile tested only. I am on the road and can't test the multimedia drivers because of no cables :( If someone could test it, that would be much appreciated. Or I'll do it next week. The pattern worked for other drivers I could actually test, though. 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 5227aad43e3854..3dae30710c34f2 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1119,7 +1119,6 @@ static int rsnd_probe(struct platform_device *pdev) struct rsnd_priv *priv; struct device *dev = &pdev->dev; struct rsnd_dai *rdai; - const struct of_device_id *of_id = of_match_device(rsnd_of_match, dev); int (*probe_func[])(struct rsnd_priv *priv) = { rsnd_gen_probe, rsnd_dma_probe, @@ -1145,7 +1144,7 @@ static int rsnd_probe(struct platform_device *pdev) } priv->pdev = pdev; - priv->flags = (unsigned long)of_id->data; + priv->flags = (unsigned long)of_device_get_match_data(dev); spin_lock_init(&priv->lock); /* -- 2.7.0

3 6

[alsa-devel] [PATCH v3 0/4] add IEC958 channel status control helper
by Arnaud Pouliquen 02 Mar '16

02 Mar '16

- Define helpers function to handle 'IEC958 Playback Default' control - add "pcm_new" ops for DAI initialisations that need pcm runtime context. This patchset is needed to be able to associate control to the PCM device. The iec control is linked to the DAI but also to the PCM device. Furthermore, some platforms like sti support both HDMI and SPDIF outputs in parallel. Logical way to differentiate them is to link them to pcm device index. V3: This patchset is extracted from following RFC http://permalink.gmane.org/gmane.linux.alsa.devel/149876 (sti: add audio interface to the hdmi driver) Notice that "ASoC: hdmi-codec: add IEC control" patch depends on integration of the hdmi-codec driver Patches update: - ALSA: pcm: add IEC958 channel status control helper - make mutex usage mandatory. - Set control index to pcm device id ( needed to be supported by iecset) - ASoC: core: add code to complete dai init after pcm creation - fix dai param of pcm_new ops for cpu dai. - ASoC: sti: use iec channel status control helper new patch: implementation of the helper for sti cpu-dai V2: - patch: ALSA: pcm: add IEC958 channel status control helper - Return 1 instead of 0 in snd_pcm_iec958_put - Add .access field in control structure - I have kept condition on mutex for flexibility (but could be cleaned to force user to use a mutex) V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Arnaud Pouliquen (4): ALSA: pcm: add IEC958 channel status control helper ASoC: core: add code to complete dai init after pcm creation ASoC: sti: use iec channel status control helper ASoC: hdmi-codec: add IEC control. include/sound/hdmi-codec.h | 1 + include/sound/pcm_iec958.h | 15 ++++++ include/sound/soc-dai.h | 7 +++ sound/core/pcm_iec958.c | 107 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdmi-codec.c | 59 ++++++++++++++++------ sound/soc/soc-core.c | 14 ++++++ sound/soc/sti/Kconfig | 1 + sound/soc/sti/uniperif_player.c | 79 ++++++++++++----------------- 8 files changed, 219 insertions(+), 62 deletions(-) -- 1.9.1

3 11

[alsa-devel] sound: uninterruptible hang in snd_seq_oss_writeq_sync
by Dmitry Vyukov 02 Mar '16

02 Mar '16

Hello, The following program creates an unkillable process: // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> #ifndef SYS_mmap #define SYS_mmap 9 #endif #ifndef SYS_syz_open_dev #define SYS_syz_open_dev 1000001 #endif #ifndef SYS_write #define SYS_write 1 #endif #ifndef SYS_read #define SYS_read 0 #endif #ifndef SYS_sync_file_range #define SYS_sync_file_range 277 #endif long r[61]; int main() { memset(r, -1, sizeof(r)); r[0] = syscall(SYS_mmap, 0x20000000ul, 0x5e000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); r[2] = syscall(SYS_open, "/dev/sequencer", 0x202ul, 0, 0, 0); *(uint8_t*)0x20057000 = (uint8_t)0x7; *(uint8_t*)0x20057001 = (uint8_t)0x6; *(uint8_t*)0x20057002 = (uint8_t)0x3; *(uint8_t*)0x20057003 = (uint8_t)0x99be; *(uint32_t*)0x2005700c = (uint32_t)0x4; *(uint8_t*)0x20057010 = (uint8_t)0x5; *(uint8_t*)0x20057011 = (uint8_t)0xffffffffffffffff; *(uint8_t*)0x20057012 = (uint8_t)0x100; *(uint8_t*)0x20057013 = (uint8_t)0x1; *(uint8_t*)0x20057024 = (uint8_t)0x8; *(uint8_t*)0x20057025 = (uint8_t)0xabe4; *(uint16_t*)0x20057026 = (uint16_t)0x3; *(uint64_t*)0x20057028 = (uint64_t)0x2005055c; *(uint8_t*)0x20057030 = (uint8_t)0xffffffffffffffe0; *(uint8_t*)0x20057031 = (uint8_t)0xffffffffffffffb8; *(uint8_t*)0x20057032 = (uint8_t)0x8; *(uint8_t*)0x20057033 = (uint8_t)0x7; *(uint32_t*)0x2005703c = (uint32_t)0x5; *(uint8_t*)0x20057040 = (uint8_t)0x5; *(uint8_t*)0x20057041 = (uint8_t)0x401; *(uint8_t*)0x20057042 = (uint8_t)0x8; *(uint8_t*)0x20057043 = (uint8_t)0x4; *(uint8_t*)0x2005704c = (uint8_t)0x0; *(uint8_t*)0x2005704d = (uint8_t)0x3; *(uint8_t*)0x2005704e = (uint8_t)0x200; *(uint8_t*)0x2005704f = (uint8_t)0xee; *(uint8_t*)0x20057050 = (uint8_t)0x2; *(uint8_t*)0x20057051 = (uint8_t)0x8; *(uint8_t*)0x20057052 = (uint8_t)0x800; *(uint8_t*)0x20057053 = (uint8_t)0xfffffffffffff000; *(uint64_t*)0x20057068 = (uint64_t)0x77359400; *(uint64_t*)0x20057070 = (uint64_t)0x0; *(uint8_t*)0x20057078 = (uint8_t)0x10000; *(uint8_t*)0x20057079 = (uint8_t)0x946; *(uint8_t*)0x2005707a = (uint8_t)0x1; *(uint8_t*)0x2005707b = (uint8_t)0x1ff; *(uint8_t*)0x2005708c = (uint8_t)0x3; *(uint32_t*)0x20057090 = (uint32_t)0x7; *(uint32_t*)0x20057094 = (uint32_t)0x1; *(uint8_t*)0x20057098 = (uint8_t)0x2; *(uint8_t*)0x20057099 = (uint8_t)0x7ff; *(uint8_t*)0x2005709a = (uint8_t)0x8; *(uint8_t*)0x2005709b = (uint8_t)0xdd8; *(uint64_t*)0x200570b0 = (uint64_t)0x1; *(uint64_t*)0x200570b8 = (uint64_t)0x0; *(uint8_t*)0x200570c0 = (uint8_t)0x9; *(uint8_t*)0x200570c1 = (uint8_t)0x3; *(uint8_t*)0x200570c2 = (uint8_t)0x7ff; *(uint8_t*)0x200570c3 = (uint8_t)0x8; *(uint32_t*)0x200570d4 = (uint32_t)0xec; *(uint64_t*)0x200570d8 = (uint64_t)0x20053f14; memcpy((void*)0x2005055c, "\xa4\x62\xd2\xb0\x88\xe0\x2b\xbd\xea\xa1\xa5\xf7\x7b\x89\xca" "\xb0\x43\xca\x19\x6d\x70\xb6\xa6\xcd\x30\xaa\x98\x07\xb3\xf3" "\xea\xf8\x44\x19\xc8\x83\xbc\xb7\x33\xd1\xd0\x5e\x8f\x34\x12" "\x82\xb4\xf8\xbd\x02\x63\x0b\xe7\x45\x23\x40\xa4\x51\x40\x23" "\x01\xcc\x57\x89\x40\x9c\xc1\x29\x36\xe2\xc0\x70\xe5\xda\xc9" "\x1f\x40\x25\xe0\x74\x98\x07\x75\xe9\x5a\xfc\x0e\x10\x3f\xd9" "\x4b\xc3\xc2\x48\xfa\x18\xb8\xda\x49\x3e\xeb\x69\x2c\x4b\xd2" "\xb7\xac\xc6\x5a\x1c\xfd", 111); memcpy((void*)0x20053f14, "\x1b\x4b\xa5\x94\xeb\xe1\xd9\x4b\x57\xf2\xeb\xab\xdd\xe5\xba" "\x24\xf2\xc8\x36\x90\x88\x21\x2c\xec\x8e\x5a\x0d\x24\x0f\x48" "\xcc\x0d\x3c\xab\x21\x8e\xfa\x8f\x0d\xcd\x8d\x14\x99\xc3\x64" "\xea\x7f\xf3\x88\xe9\xf7\x4f\x4c\x12\xe9\x0e\x23\xad\x73\x7d" "\xa3\x4c\x45\x3f\xb3\x48\xb7\x44\x89\x41\x20\x7b\x2c\xbd\xf2" "\x89\x16\x1f\xbd\x80\x24\x72\xa1\x8d\x4b\x13\x74\xb2\xf4\xdf" "\x49\xe0\x6c\x40\x0d\xf2\x7c\x8a\xbf\xf2\xb1\x62\xe0\x7e\x51" "\x9a\xd4\x60\x64\x49\x96\x4b\x83\x91\xe5\x51\x50\x94\x3c\x41" "\xf6\xd1\xef\x1f\x90\x5a\x48\xa5\xe2\xf8\xe3\xd9\xf9\x82\x5e" "\xa1\x5c\x85\x1c\xbe\x47\xa2\x29\xb7\x5a\xf4\x14\x24\x92\x11" "\x4b\x6b\x54\x95\x5e\x9f\x1b\xda\x02\xa3\x84\xb3\x2b\x07\x13" "\x18\x6b\x31\xb4\xc1\x04\x30\x5f\xa0\xa2\xe9\xe6\x50\x6f\xed" "\x15\x10\xf5\xda\xa1\xe5\x06\xaf\x28\xe0\x53\xea\x88\x49\xf8" "\x97\x52\x4b\xe8\xa2\xce\x9f\xc7\x47\x4f\xd6\xc0\xe2\xb7\xc1" "\x7f\xa9\x54\x3d\x11\xe4\x99\x28\x65\x39\xb1\xf3\xec\x48\x43" "\x58\x19\x12\xf6\x38\x5b\xee\xca\x33\x8e\x4d", 236); r[56] = syscall(SYS_write, r[2], 0x20057000ul, 0x9cul, 0, 0, 0); memcpy((void*)0x2005bfa7, "\x2f\x64\x65\x76\x2f\x76\x68\x63\x69\x00", 10); return 0; } The hang stack is: [<ffffffff85309f77>] snd_seq_oss_writeq_sync+0x327/0x790 sound/core/seq/oss/seq_oss_writeq.c:121 [<ffffffff852fa353>] snd_seq_oss_drain_write+0x113/0x160 sound/core/seq/oss/seq_oss_init.c:448 [<ffffffff852f8d04>] odev_release+0x44/0x80 sound/core/seq/oss/seq_oss.c:152 [<ffffffff817ccdc6>] __fput+0x236/0x780 fs/file_table.c:208 [<ffffffff817cd395>] ____fput+0x15/0x20 fs/file_table.c:244 [<ffffffff813b8db0>] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [<ffffffff81363ca0>] do_exit+0xaf0/0x2d20 kernel/exit.c:748 [<ffffffff81366048>] do_group_exit+0x108/0x330 kernel/exit.c:878 [< inline >] SYSC_exit_group kernel/exit.c:889 [<ffffffff8136628d>] SyS_exit_group+0x1d/0x20 kernel/exit.c:887 [<ffffffff866a3236>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 On commit fc77dbd34c5c99bce46d40a2491937c3bcbd10af (4.5-rc6).

2 7

[alsa-devel] [PATCH] ALSA: hda - hdmi defer to register acomp eld notifier
by libin.yang＠linux.intel.com 02 Mar '16

02 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> Defer to register acomp eld notifier until hdmi audio driver is fully ready. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index aa2d6a2..a3de50b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2936,13 +2936,6 @@ static int patch_generic_hdmi(struct hda_codec *codec) is_broxton(codec)) codec->core.link_power_control = 1; - if (codec_has_acomp(codec)) { - codec->depop_delay = 0; - spec->i915_audio_ops.audio_ptr = codec; - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); - } - if (hdmi_parse_codec(codec) < 0) { if (spec->i915_bound) snd_hdac_i915_exit(&codec->bus->core); @@ -2964,6 +2957,13 @@ static int patch_generic_hdmi(struct hda_codec *codec) init_channel_allocations(); + if (codec_has_acomp(codec)) { + codec->depop_delay = 0; + spec->i915_audio_ops.audio_ptr = codec; + spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + } + WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec)); return 0; } -- 1.9.1

3 2

[alsa-devel] [PATCH v2] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 02 Mar '16

02 Mar '16

This is the initial imx-wm8958 device-tree-only machine driver working with fsl_sai driver. This sound card has three dai link, HIFI, VOICE and BT dai. HIFI dai link will support codec master and slave mode. VOICE an BT dai link have dummy cpu dai, and just support codec master mode. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- .../devicetree/bindings/sound/imx-audio-wm8958.txt | 49 +++ arch/arm/configs/imx_v6_v7_defconfig | 1 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 432 +++++++++++++++++++++ 5 files changed, 498 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..50bad12 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,49 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - cpu-dai : The phandle of an CPU DAI controller + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + + - fsl,hifi-dai-master : If present, hifi dai works as master, and will + provide bit clock and frame clock. Otherwise, + hifi dai works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8960-audio"; + cpu-dai = <&sai1>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; + + fsl,hifi-dai-master; +}; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..7ac5856 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -251,6 +251,7 @@ CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_PHYCORE_AC97=y CONFIG_SND_SOC_EUKREA_TLV320=y +CONFIG_SND_SOC_IMX_WM8958=y CONFIG_SND_SOC_IMX_WM8962=y CONFIG_SND_SOC_IMX_SGTL5000=y CONFIG_SND_SOC_IMX_SPDIF=y diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + 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..2a781c0 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-wm8958-objs := imx-wm8958.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 @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(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-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.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-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..24eb66f --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,432 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * 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/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link *dai_link; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_hifi_dai_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static u32 imx_wm8958_rates[] = { 8000, 16000, 32000, 48000 }; + +static struct snd_pcm_hw_constraint_list imx_wm8958_rate_constraints = { + .count = ARRAY_SIZE(imx_wm8958_rates), + .list = imx_wm8958_rates, +}; + +static int imx_wm8958_hifi_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + + if (!data->is_hifi_dai_master) + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &imx_wm8958_rate_constraints); + + return ret; +} +static int imx_wm8958_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; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + /* + * Set GPIO1 pin function to reserve, so that DAC1 and ADC1 + * using shared LRCLK from DACLRCK1. + */ + snd_soc_update_bits(codec, WM8994_GPIO_1, 0x1f, 0x2); + + if (!data->is_hifi_dai_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (!data->is_hifi_dai_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } else if (id == VOICE_DAI) { + /* + * Set GPIO6 pin function to reserve, so that DAC2 and ADC2 + * using shared LRCLK from DACLRCK2. + */ + snd_soc_update_bits(codec, WM8994_GPIO_6, 0x1f, 0x2); + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(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 = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == HIFI_DAI && !data->is_hifi_dai_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static int imx_wm8958_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + int ret, i; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (card->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* need to enable mclk to write/read wm8958 register */ + for (i = 0; i < WM8958_MCLK_MAX; i++) { + if (!IS_ERR(data->mclk[i])) { + ret = clk_prepare_enable(data->mclk[i]); + if (ret) + dev_warn(card->dev, + "Failed to enable MCLK%d: %d\n", + i + 1, ret); + } + } + } + break; + default: + break; + } + + return 0; +} + +static int imx_wm8958_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + int i; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + for (i = 0; i < WM8958_MCLK_MAX; i++) { + if (!IS_ERR(data->mclk[i])) + clk_disable_unprepare(data->mclk[i]); + } + break; + default: + break; + } + + card->dapm.bias_level = level; + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, + .startup = imx_wm8958_hifi_startup, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + .ops = &imx_voice_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + int ret, i; + + cpu_np = of_parse_phandle(np, "cpu-dai", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + return -EINVAL; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find cpu dai 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) + return -ENOMEM; + + data->dai_link = imx_wm8958_dai_link; + + /* + * AIF1 support codec master and slave mode + * AIF2 and AIF3 just support codec master mode + */ + if (of_property_read_bool(pdev->dev.of_node, "fsl,hifi-dai-master")) { + data->is_hifi_dai_master = true; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + data->is_hifi_dai_master = false; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (IS_ERR(data->mclk[i])) { + ret = PTR_ERR(data->mclk[i]); + dev_err(&pdev->dev, "failed to get mclk%d clock: %d\n", + i + 1, ret); + } else { + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + } + + data->dai_link[HIFI_DAI].cpu_dai_name = dev_name(&cpu_pdev->dev); + data->dai_link[HIFI_DAI].platform_of_node = cpu_np; + + data->card.set_bias_level = imx_wm8958_set_bias_level; + data->card.set_bias_level_post = imx_wm8958_set_bias_level_post; + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = DAI_LINK_NUM; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + 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; + } + +fail: + of_node_put(cpu_np); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: mediatek: Add machine driver for rt5650 rt5514 codec
by PC Liao 02 Mar '16

02 Mar '16

From: Koro Chen <koro.chen(a)mediatek.com> This is the DPCM based machine driver with rt5650 and rt5514. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- .../bindings/sound/mt8173-rt5650-rt5514.txt | 15 ++ sound/soc/mediatek/Kconfig | 28 ++- sound/soc/mediatek/Makefile | 3 +- sound/soc/mediatek/mt8173-rt5650-rt5514.c | 258 ++++++++++++++++++++ 4 files changed, 294 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt create mode 100644 sound/soc/mediatek/mt8173-rt5650-rt5514.c diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt new file mode 100644 index 0000000..e8b3c80 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt @@ -0,0 +1,15 @@ +MT8173 with RT5650 RT5514 CODECS + +Required properties: +- compatible : "mediatek,mt8173-rt5650-rt5514" +- mediatek,audio-codec: the phandles of rt5650 and rt5514 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650-rt5514"; + mediatek,audio-codec = <&rt5650 &rt5514>; + mediatek,platform = <&afe>; + }; + diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 6c88087..f7e789e 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -18,14 +18,25 @@ config SND_SOC_MT8173_MAX98090 If unsure select "N". config SND_SOC_MT8173_RT5650 - tristate "ASoC Audio driver for MT8173 with RT5650 codec" - depends on SND_SOC_MEDIATEK && I2C - select SND_SOC_RT5645 - help - This adds ASoC driver for Mediatek MT8173 boards - with the RT5650 audio codec. - Select Y if you have such device. - If unsure select "N". + tristate "ASoC Audio driver for MT8173 with RT5650 codec" + depends on SND_SOC_MEDIATEK && I2C + select SND_SOC_RT5645 + help + This adds ASoC driver for Mediatek MT8173 boards + with the RT5650 audio codec. + Select Y if you have such device. + If unsure select "N". + +config SND_SOC_MT8173_RT5650_RT5514 + tristate "ASoC Audio driver for MT8173 with RT5650 RT5514 codecs" + depends on SND_SOC_MEDIATEK && I2C + select SND_SOC_RT5645 + select SND_SOC_RT5514 + help + This adds ASoC driver for Mediatek MT8173 boards + with the RT5650 and RT5514 codecs. + Select Y if you have such device. + If unsure select "N". config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" @@ -37,4 +48,3 @@ config SND_SOC_MT8173_RT5650_RT5676 with the RT5650 and RT5676 codecs. Select Y if you have such device. If unsure select "N". - diff --git a/sound/soc/mediatek/Makefile b/sound/soc/mediatek/Makefile index bcdcb06..d486860 100644 --- a/sound/soc/mediatek/Makefile +++ b/sound/soc/mediatek/Makefile @@ -2,5 +2,6 @@ obj-$(CONFIG_SND_SOC_MEDIATEK) += mtk-afe-pcm.o # Machine support obj-$(CONFIG_SND_SOC_MT8173_MAX98090) += mt8173-max98090.o -obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o obj-$(CONFIG_SND_SOC_MT8173_RT5650) += mt8173-rt5650.o +obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5514) += mt8173-rt5650-rt5514.o +obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173-rt5650-rt5514.c new file mode 100644 index 0000000..58e0836 --- /dev/null +++ b/sound/soc/mediatek/mt8173-rt5650-rt5514.c @@ -0,0 +1,258 @@ +/* + * mt8173-rt5650-rt5514.c -- MT8173 machine driver with RT5650/5514 codecs + * + * Copyright (c) 2016 MediaTek Inc. + * Author: Koro Chen <koro.chen(a)mediatek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * 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/gpio.h> +#include <linux/of_gpio.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5645.h" + +#define MCLK_FOR_CODECS 12288000 + +static const struct snd_soc_dapm_widget mt8173_rt5650_rt5514_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route mt8173_rt5650_rt5514_routes[] = { + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + {"Sub DMIC1L", NULL, "Int Mic"}, + {"Sub DMIC1R", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Headset Mic", NULL, "micbias1"}, + {"Headset Mic", NULL, "micbias2"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, +}; + +static const struct snd_kcontrol_new mt8173_rt5650_rt5514_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + + /* pll from mclk 12.288M */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, + params_rate(params) * 512); + if (ret) + return ret; + + /* sysclk from pll */ + ret = snd_soc_dai_set_sysclk(codec_dai, 1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + return 0; +} + +static struct snd_soc_ops mt8173_rt5650_rt5514_ops = { + .hw_params = mt8173_rt5650_rt5514_hw_params, +}; + +static struct snd_soc_jack mt8173_rt5650_rt5514_jack; + +static int mt8173_rt5650_rt5514_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + int ret; + + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* enable jack detection */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &mt8173_rt5650_rt5514_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "Can't new Headset Jack %d\n", ret); + return ret; + } + + return rt5645_set_jack_detect(codec, + &mt8173_rt5650_rt5514_jack, + &mt8173_rt5650_rt5514_jack, + &mt8173_rt5650_rt5514_jack); +} + +static struct snd_soc_dai_link_component mt8173_rt5650_rt5514_codecs[] = { + { + .dai_name = "rt5645-aif1", + }, + { + .dai_name = "rt5514-aif1", + }, +}; + +enum { + DAI_LINK_PLAYBACK, + DAI_LINK_CAPTURE, + DAI_LINK_CODEC_I2S, +}; + +/* Digital audio interface glue - connects codec <---> CPU */ +static struct snd_soc_dai_link mt8173_rt5650_rt5514_dais[] = { + /* Front End DAI links */ + [DAI_LINK_PLAYBACK] = { + .name = "rt5650_rt5514 Playback", + .stream_name = "rt5650_rt5514 Playback", + .cpu_dai_name = "DL1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, + [DAI_LINK_CAPTURE] = { + .name = "rt5650_rt5514 Capture", + .stream_name = "rt5650_rt5514 Capture", + .cpu_dai_name = "VUL", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_capture = 1, + }, + /* Back End DAI links */ + [DAI_LINK_CODEC_I2S] = { + .name = "Codec", + .cpu_dai_name = "I2S", + .no_pcm = 1, + .codecs = mt8173_rt5650_rt5514_codecs, + .num_codecs = 2, + .init = mt8173_rt5650_rt5514_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &mt8173_rt5650_rt5514_ops, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +static struct snd_soc_codec_conf mt8173_rt5650_rt5514_codec_conf[] = { + { + .name_prefix = "Sub", + }, +}; + +static struct snd_soc_card mt8173_rt5650_rt5514_card = { + .name = "mtk-rt5650-rt5514", + .owner = THIS_MODULE, + .dai_link = mt8173_rt5650_rt5514_dais, + .num_links = ARRAY_SIZE(mt8173_rt5650_rt5514_dais), + .codec_conf = mt8173_rt5650_rt5514_codec_conf, + .num_configs = ARRAY_SIZE(mt8173_rt5650_rt5514_codec_conf), + .controls = mt8173_rt5650_rt5514_controls, + .num_controls = ARRAY_SIZE(mt8173_rt5650_rt5514_controls), + .dapm_widgets = mt8173_rt5650_rt5514_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8173_rt5650_rt5514_widgets), + .dapm_routes = mt8173_rt5650_rt5514_routes, + .num_dapm_routes = ARRAY_SIZE(mt8173_rt5650_rt5514_routes), +}; + +static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mt8173_rt5650_rt5514_card; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_rt5514_dais[i].platform_name) + continue; + mt8173_rt5650_rt5514_dais[i].platform_of_node = platform_node; + } + + mt8173_rt5650_rt5514_codecs[0].of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); + if (!mt8173_rt5650_rt5514_codecs[0].of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + mt8173_rt5650_rt5514_codecs[1].of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); + if (!mt8173_rt5650_rt5514_codecs[1].of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + mt8173_rt5650_rt5514_codec_conf[0].of_node = + mt8173_rt5650_rt5514_codecs[1].of_node; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + return ret; +} + +static const struct of_device_id mt8173_rt5650_rt5514_dt_match[] = { + { .compatible = "mediatek,mt8173-rt5650-rt5514", }, + { } +}; +MODULE_DEVICE_TABLE(of, mt8173_rt5650_rt5514_dt_match); + +static struct platform_driver mt8173_rt5650_rt5514_driver = { + .driver = { + .name = "mtk-rt5650-rt5514", + .of_match_table = mt8173_rt5650_rt5514_dt_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, + .probe = mt8173_rt5650_rt5514_dev_probe, +}; + +module_platform_driver(mt8173_rt5650_rt5514_driver); + +/* Module information */ +MODULE_DESCRIPTION("MT8173 RT5650 and RT5514 SoC machine driver"); +MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:mtk-rt5650-rt5514"); + -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Update the IVI feedback path
by Vinod Koul 02 Mar '16

02 Mar '16

This patch series fixes the blob query direction and then uses that in IVI feedback path in nau88125_ssm4567 machine. Jeeja KP (3): ASoC: Intel: Skylake: correct direction in blob query ASoC: Intel: Skylake: Add IVI feedback path in nau88125_ssm4567 machine ASoC: Intel: Skylake: remove IVI codec loop in skl_nau88l25_ssm4567 machine sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 21 +++------------------ sound/soc/intel/skylake/skl-topology.c | 15 ++++++++------- 2 files changed, 11 insertions(+), 25 deletions(-) -- 1.9.1

2 6

Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Liam Girdwood 02 Mar '16

02 Mar '16

On Tue, 2016-02-16 at 21:47 +0530, Jeeja KP wrote: > We have no mechanism to configure hw_params for DAIs which are hostless. > Like a loop or DAIs connected to internal sinks/sources. > Fwiw, a few years back Peter, Sebastien and I were considering a similar mechanism for the ABE on OMAP4. We never got around to implementing or upstreaming though.... One comment below. > Problem Statement: > During a BE-BE loop of DSP or a SRC->SINK path, we cannot program > hw_params. There are no hw_params from user as these are hostless. > This scenario can happens in below 2 cases: > > Case 1: BE-BE loop aka hostless stream through SoC DSP > Hostless stream can be defined by BE to BE connection and in this case > the path is enabled or disabled by the state of the DAPM graph. This > usually means there is a mixer control that can be used to connect or > disconnect the path between both DAIs. > Example: > |--------------/----------------| > | | > | | > codec out modem_in > | | > | | > ssp0_TX ssp1_RX > | | > | | > AIFI_Playback dummy_capture > > In the above case (SSP0,codec playback,SSp1 dai needs to be configured > and turned on with the selected hw_params). > > Case 2: SoC Internal Sink or Source > Example: > WOV SINK (DAPM Sink) > | > | > | > Switch > | > | > w1 > | > dmic rx (BE dai widget) > | > Capture(codec dai widget) > | > DMIC AIF > | > DMic > | > Soc DMic (MIC Widget) (SRC) > > In the above case BE (dmic RX) dai hw_params needs to applied. > > Approach: > 1. Allow params to be defined for other BE dai link also. this > currently applicable only for codec_codec link. > > 2. when params is defined for BE dai link, create params kcontrols > and associate it with the dai widgets. this will allow params to be > selected via Kcontrol for each dai defined in the dai link. > The only problem we have with using a kcontrol to define params is that we already have an existing kernel ioctl and public C API to define SW/HW params. This would add another mechanism, unless each new kcontrol allows modification of a single param ? Liam > 3. Before powering on the dai widget and if the dai is not powered > with the stream event, then dai_ops(hw_params) for the dai widget will > be called.This will apply the selected hw_params. > > 4. with the above, fix_up can be removed and can use .params to > select the parameter. > > Core Changes: > 1. params definition needs to changed, currently this is defined to used > only for codec codec link. define new flag to identifying codec > codec link. > > 2. if params is defined for any BE dai link, then create kcontrol for > each dai defined in the dai_link and associated the kcontrol to dai > widget. > > 3. In the powering sequence check if the dai widget and if the event is > not a stream event, SND_SOC_DAPM_STREAM_NOP , then call dai ops with > the selected params. This will apply the hw_params for dai. > > 1. params definition: > - Add new flag in struct snd_soc_dai_link. > - Based on this flag create, the codec codec link instead of > params > struct snd_soc_dai_link { > const struct snd_soc_pcm_stream *params; > unsigned int num_params; > > + /* flag to create codec codec based on the flag*/ > + unsigned int codec_link:1; > + > unsigned int dai_fmt; /* format to set on init */ > > enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ > > - creation of dai link based on the codec_link flag > static int soc_probe_link_dais(struct snd_soc_card *card, > int num, int order) > } > } else { > > - if (!dai_link->params) { > + if (!dai_link->codec_link) { > /* create the pcm */ > ret = soc_new_pcm(rtd, num); > if (ret < 0) { > > 2. when params is defined for BE dai link, create params kcontrols and > associate it with the dai widgets. this will allow params to be selected > via Kcontrol for each dai defined in the dai link. > a. Move the param control creation from snd_soc_dapm_new_pcm and add new > API to created kcontrols with the params. > +snd_soc_dapm_create_param_control(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params) > { > /* create kcontrol based on the params */ > } > > - modify the snd_soc_dapm_new_pcm to create the param control > 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) > { > > template.kcontrol_news = snd_soc_dapm_create_param_control(card, > params, num_params); > 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; > + goto outfree_link_name; > } > > > b. add new API snd_soc_dapm_new_dai_params() to create kcontrol for dai and associated it to dai. > +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params, > + struct snd_soc_dapm_widget *dai_w) > +{ > + struct snd_kcontrol_new *kcontrol_news; > + > + kcontrol_news = snd_soc_dapm_create_param_control(card, params, > + num_params); > + if (!kcontrol_news) { > + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", > + dai_w->name); > + return -EINVAL; > + } > + > + dai_w->num_kcontrols = 1; > + dai_w->kcontrol_news = kcontrol_news; > + > + dai_w->params = params; > + dai_w->num_params = num_params; > + > + return 0; > + > +} > + > +++ b/include/sound/soc-dapm.h > +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params, > + struct snd_soc_dapm_widget *dai_w); > > > c. if the be dai link has .params defined, call new API snd_soc_dapm_new_dai_params > > +static int soc_link_be_params_widget(struct snd_soc_card *card, > + struct snd_soc_dai *dai, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params) > +{ > + int ret; > + struct snd_soc_dapm_widget *w; > + > + if (!strcmp(dai->name, "snd-soc-dummy")) > + return 0; > + > + if (dai->playback_widget) { > + w = dai->playback_widget; > + ret = snd_soc_dapm_new_dai_params(card, params, > + num_params, w); > + > + if (ret != 0) { > + dev_err(card->dev, > + "ASoC: Can't create params widget=%s %d\n", > + w->name, ret); > + return ret; > + } > + } > + > + if (dai->capture_widget) { > + w = dai->capture_widget; > + ret = snd_soc_dapm_new_dai_params(card, params, > + num_params, w); > + > + if (ret != 0) { > + dev_err(card->dev, > + "ASoC: Can't create params widget=%s %d\n", > + w->name, ret); > + return ret; > + } > + } > + > + return 0; > +} > + > +static int soc_link_be_params_dai_widgets(struct snd_soc_card *card, > + struct snd_soc_dai_link *dai_link, > + struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + int ret, i; > + > + for (i = 0; i < rtd->num_codecs; i++) { > + struct snd_soc_dai *codec = rtd->codec_dais[i]; > + > + ret = soc_link_be_params_widget(card, codec, > + dai_link->params, > + dai_link->num_params); > + > + if (!ret) > + return ret; > + } > + > + return soc_link_be_params_widget(card, cpu_dai, dai_link->params, > + dai_link->num_params); > +} > + > > @@ -1651,6 +1714,13 @@ static int soc_probe_link_dais(struct snd_soc_card *card, > dai_link->stream_name, ret); > return ret; > } > + if (dai_link->params && dai_link->no_pcm) { > + /* create dai link controls */ > + ret = soc_link_be_params_dai_widgets(card, > + dai_link, rtd); > + if (ret) > + return ret; > + } > } else { > INIT_DELAYED_WORK(&rtd->delayed_work, > codec2codec_close_delayed_work); > > > 3. Applying the hw_params for dai widget > > --- a/sound/soc/soc-dapm.c > +++ b/sound/soc/soc-dapm.c > @@ -1398,9 +1398,64 @@ static void dapm_seq_check_event(struct snd_soc_card * > card, > } > } > > +static int snd_soc_dai_params(struct snd_soc_dapm_widget *w, int event) > +{ > + struct snd_soc_dai *dai; > + const struct snd_soc_pcm_stream *config = w->params + w->params_select; > + struct snd_pcm_substream substream; > + struct snd_pcm_hw_params *params = NULL; > + u64 fmt; > + int ret = 0; > + > + if (WARN_ON(!config)) > + return -EINVAL; > + > + dai = w->priv; > + > + if (config->formats) { > + fmt = ffs(config->formats) - 1; > + } else { > + dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n", > + config->formats); > + fmt = 0; > + } > + > + /* Currently very limited parameter selection */ > + params = kzalloc(sizeof(*params), GFP_KERNEL); > + if (!params) { > + ret = -ENOMEM; > + goto out; > + } > + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); > + > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = > + config->rate_min; > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = > + config->rate_max; > + > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min > + = config->channels_min; > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max > + = config->channels_max; > + > + memset(&substream, 0, sizeof(substream)); > + if (w->id == snd_soc_dapm_dai_in) > + substream.stream = SNDRV_PCM_STREAM_CAPTURE; > + else > + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; > + > + ret = soc_dai_hw_params(&substream, params, dai); > + if (ret < 0) > + goto out; > +out: > + kfree(params); > + return ret; > +} > + > /* Apply the coalesced changes from a DAPM sequence */ > static void dapm_seq_run_coalesced(struct snd_soc_card *card, > - struct list_head *pending) > + struct list_head *pending, > + int event) > { > struct snd_soc_dapm_context *dapm; > struct snd_soc_dapm_widget *w; > @@ -1413,6 +1468,14 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, > dapm = w->dapm; > > list_for_each_entry(w, pending, power_list) { > + if ((event == SND_SOC_DAPM_STREAM_NOP) && > + ((w->id == snd_soc_dapm_dai_out) || > + (w->id == snd_soc_dapm_dai_in))) { > + snd_soc_dai_params(w, SND_SOC_DAPM_PRE_PMU); > + } > + } > + > + list_for_each_entry(w, pending, power_list) { > WARN_ON(reg != w->reg || dapm != w->dapm); > w->power = w->new_power; > > @@ -1447,6 +1510,13 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, > dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMU); > dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMD); > } > + list_for_each_entry(w, pending, power_list) { > + if ((event == SND_SOC_DAPM_STREAM_NOP) && > + ((w->id == snd_soc_dapm_dai_out) || > + (w->id == snd_soc_dapm_dai_in))) { > + snd_soc_dai_params(w, SND_SOC_DAPM_POST_PMD); > + } > + } > } > > /* Apply a DAPM power sequence. > @@ -1482,7 +1552,7 @@ static void dapm_seq_run(struct snd_soc_card *card, > if (sort[w->id] != cur_sort || w->reg != cur_reg || > w->dapm != cur_dapm || w->subseq != cur_subseq) { > if (!list_empty(&pending)) > - dapm_seq_run_coalesced(card, &pending); > + dapm_seq_run_coalesced(card, &pending, event); > > if (cur_dapm && cur_dapm->seq_notifier) { > for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) > @@ -1545,7 +1615,7 @@ static void dapm_seq_run(struct snd_soc_card *card, > } > > if (!list_empty(&pending)) > - dapm_seq_run_coalesced(card, &pending); > + dapm_seq_run_coalesced(card, &pending, event); > > if (cur_dapm && cur_dapm->seq_notifier) { > for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) > -- > 1.7.9.5 > > Comments welcome, based on discussion we will send patches. > > --Jeeja

3 2

Re: [alsa-devel] [PATCH v2] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Nicolin Chen 02 Mar '16

02 Mar '16

It looks like you replied me off the maillist. So I just add the maillist and related TO/CCs back. On Tue, Mar 01, 2016 at 05:43:22AM +0000, Zidan Wang wrote: > On Fri, Feb 26, 2016 at 11:42:42AM +0800, Zidan Wang wrote: > > This is the initial imx-wm8958 device-tree-only machine driver working > > with fsl_sai driver. This sound card has three dai link, HIFI, VOICE > > and BT dai. HIFI dai link will support codec master and slave mode. > > VOICE an BT dai link have dummy cpu dai, and just support codec master > > mode. > > Why VOICE and BT are using dummy cpu dai? Where can I find the schematics > for wm8958 from NXP website? > [Wang Zidan-B50113] I have attached the schematics for wm8958. In our > release, we can't support VOICE and BT, and just using AIF1. But in > order to upstream, I add the voice and bt dai. The way to support the other two AIFs are not very graceful -- looks like you will have to hard code the CPU DAIs in the driver. I personally suggest you to use fsl-asoc-card. Since the Codec IC provides three DAIs. And in the schematics, you do have three DAI links. It sounds plausible to me that we might treat these three DAI links as three different cards. Then use different compatible names: "imx-audio-wm8958-aif1" and so on. And each one can also has ASRC back-end support. If you really have big trouble to integrate it into fsl-asoc-card (not 100% sure if the 3 cards methodology is totally feasible), I will be okay with a dedicated driver for wm8958. But you will need to refine the DT binding to support three CPU DAIs in my opinion. > > +static u32 imx_wm8958_rates[] = { 8000, 16000, 32000, 48000 }; > > + > > +static struct snd_pcm_hw_constraint_list imx_wm8958_rate_constraints = { > > + .count = ARRAY_SIZE(imx_wm8958_rates), > > + .list = imx_wm8958_rates, > > +}; > > Where are these rate constraints coming from? > [Wang Zidan-B50113] these rates are restricted by the SAI MCLK. If the SAI MCLK runs at a different rate, it would be invalid. So there should be at least two constraints depending on the MCLK rate. And I think it might be better to have this in the SAI driver as a refinement since the current driver only errors out for those unsupported rates. > > + if (id == HIFI_DAI) { > > + /* > > + * Set GPIO1 pin function to reserve, so that DAC1 and ADC1 > > + * using shared LRCLK from DACLRCK1. > > + */ > > + snd_soc_update_bits(codec, WM8994_GPIO_1, 0x1f, 0x2); > > I don't see any reverse operation against this GPIO configuration here. > So is it really an operation that has to be done every hw_params()? > > > + } else if (id == VOICE_DAI) { > > + /* > > + * Set GPIO6 pin function to reserve, so that DAC2 and ADC2 > > + * using shared LRCLK from DACLRCK2. > > + */ > > + snd_soc_update_bits(codec, WM8994_GPIO_6, 0x1f, 0x2); > > Same here. Actually I just took a look at the DT binding doc of WM8994. It supports GPIO configurations in the codec DT nodes. So you may not need to do these over here at all. > > +static int imx_wm8958_set_bias_level_post(struct snd_soc_card *card, > > + struct snd_soc_dapm_context *dapm, > > + enum snd_soc_bias_level level) { > > + switch (level) { > > + case SND_SOC_BIAS_OFF: > > + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) > > + for (i = 0; i < WM8958_MCLK_MAX; i++) { > > + if (!IS_ERR(data->mclk[i])) > > + clk_disable_unprepare(data->mclk[i]); > > The set_bias() funcs are merely en/disabling mclk. > Would be possible to do it in the codec driver? > [Wang Zidan-B50113] It can be moved to codec driver, > but the wm8994 driver can support wm8958 wm8994 wm1811 codec, > it's hard for me to do like that, and I also can't test them. Sending a patch to the maillist is a way to let people be aware of the change and then test it. The driver maintainer will review it and give you feedback. If it really causes compatible issues, you can then limit the code for wm8958 only. > > +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { > > + [BT_DAI] = { > > + .name = "Bluetooth", > > + .stream_name = "Bluetooth", > > + .cpu_dai_name = "snd-soc-dummy-dai", > > + .codec_name = "wm8994-codec", > > + .codec_dai_name = "wm8994-aif3", > > + .platform_name = "snd-soc-dummy", > > + .ignore_pmdown_time = 1, > > + }, > > A little curious...how is the BT gonna work for you? > [Wang Zidan-B50113] I have not test the BT dai. The BT is directly > connect to AIF3, when AIF3 receive date from BT, it will route the > date to AIF1 or AIF2. And the date from AIF1 and AIF2 can also route > to AIF3 to BT. So we just need to set the amixer setting to open the > audio route. Sounds interesting. So the Codec totally becomes a router. If this design can not be supported by using three cards via the fsl-asoc-card, you may continue this driver as I can see the situation here. But I encourage you to try that first so as to avoid duplicated code/driver.

2 2

[alsa-devel] [PATCH] Add a quirk for Plantronics DA45
by Dennis Kadioglu 01 Mar '16

01 Mar '16

Plantronics DA45 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x4" and "cannot get freq at ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu <denk(a)post.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2585c17..5b03296 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1123,6 +1123,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ ---

2 1

[alsa-devel] [PATCH] Add a quirk for Plantronics DA45
by denk 01 Mar '16

01 Mar '16

Plantronics DA45 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x4" and "cannot get freq at ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and avoids those error messages. --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2585c17..5b03296 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1123,6 +1123,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ --- Signed-off-by: Dennis Kadioglu <denk(a)post.com>

4 5

[alsa-devel] [PATCH 1/1] alsabat: truncate sample frames for faster FFT analysis
by han.lu＠intel.com 01 Mar '16

01 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Truncate the sample frames to powers of 2, since the FFTW algorithm runs especially fast in this case, and other sizes may be computed by means of a slow, general-purpose algorithm. In my test environment applying the patch, a sound clip of 33072 frames is cut off to 32768 frames before analysis, and the time cost is reduced from 6.128s to 0.224s. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/analyze.c b/bat/analyze.c index 60e2d1c..5cfdac3 100644 --- a/bat/analyze.c +++ b/bat/analyze.c @@ -256,6 +256,20 @@ static int reorder_data(struct bat *bat) return 0; } +/* truncate sample frames for faster FFT analysis process */ +static int truncate_frames(struct bat *bat) +{ + int shift = SHIFT_MAX; + + for (; shift > SHIFT_MIN; shift--) + if (bat->frames & (1 << shift)) { + bat->frames = 1 << shift; + return 0; + } + + return -EINVAL; +} + int analyze_capture(struct bat *bat) { int err = 0; @@ -263,6 +277,13 @@ int analyze_capture(struct bat *bat) int c; struct analyze a; + err = truncate_frames(bat); + if (err < 0) { + fprintf(bat->err, _("Invalid frame number for analysis: %d\n"), + bat->frames); + return err; + } + fprintf(bat->log, _("\nBAT analysis: signal has %d frames at %d Hz,"), bat->frames, bat->rate); fprintf(bat->log, _(" %d channels, %d bytes per sample.\n"), diff --git a/bat/common.h b/bat/common.h index b8f6670..ff03fc1 100644 --- a/bat/common.h +++ b/bat/common.h @@ -68,6 +68,12 @@ #define FOUND_DC (1<<1) #define FOUND_WRONG_PEAK (1<<0) +/* Truncate sample frames to (1 << N), for faster FFT analysis process. The + * valid range of N is (SHIFT_MIN, SHIFT_MAX). When N increases, the analysis + * will be more time-consuming, and the result will be more accurate. */ +#define SHIFT_MAX (sizeof(int) * 8 - 2) +#define SHIFT_MIN 8 + struct wav_header { unsigned int magic; /* 'RIFF' */ unsigned int length; /* file len */ -- 2.5.0

2 1

[alsa-devel] [PATCH] ALSA: hda - hdmi_find_pcm_slot return value bug fix
by libin.yang＠linux.intel.com 01 Mar '16

01 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> hdmi_find_pcm_slot return -EBUSY when not no pcm slot found, not -ENODEV. So the caller should compare with -EBUSY. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d77509a..aa2d6a2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1758,7 +1758,7 @@ static void hdmi_attach_hda_pcm(struct hdmi_spec *spec, if (per_pin->pcm) return; idx = hdmi_find_pcm_slot(spec, per_pin); - if (idx == -ENODEV) + if (idx == -EBUSY) return; per_pin->pcm_idx = idx; per_pin->pcm = get_hdmi_pcm(spec, idx); -- 1.9.1

2 1

[alsa-devel] [PATCH] teac ud-501/ud-503 usb delay
by Guillaume Fougnies 01 Mar '16

01 Mar '16

Hello, This is a small patch against sound/usb/quirks.c to correct "Teac Corp." products delay. Tested on Teac UD-501 and UD-503 products, could not harm other Teac products. It corrects the "clock source 41 is not valid, cannot use" error for both devices. And more critically on the new UD-503, a complete freeze of the teac usb interface when switching between different rate/format. (Freeze only solved by a power switch of the device...) The patch matches the problem of the "Playback Design" products. I just refactored a bit the code to include Teac products properly. Regards, Guillaume --- sound/usb/quirks.c | 58 +++++++++++++++++++++++++++++++----------------------- 1 file changed, 33 insertions(+), 25 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 23ea6d8..f3faaef 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1202,40 +1202,48 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) void snd_usb_set_interface_quirk(struct usb_device *dev) { /* - * "Playback Design" products need a 50ms delay after setting the + * Products needing a 50ms delay after setting the * USB interface. */ - if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) - mdelay(50); + switch (le16_to_cpu(dev->descriptor.idVendor)) { + case 0x23ba: /* Playback Design */ + case 0x0644: /* TEAC Corp. */ + mdelay(50); + } } void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) { - /* - * "Playback Design" products need a 20ms delay after each - * class compliant request - */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); - - /* Marantz/Denon devices with USB DAC functionality need a delay - * after each class compliant request - */ - if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), - le16_to_cpu(dev->descriptor.idProduct))) - && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + if ((requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { + switch (le16_to_cpu(dev->descriptor.idVendor)) { + /* + * "Playback Design"/"TEAC Corp." products need a 20ms delay after each + * class compliant request + */ + case 0x23ba: /* "Playback Design" */ + case 0x0644: /* TEAC Corp. */ + mdelay(20); + return; + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + case 0x1686: + /* restrict to 0x00dd */ + if (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) + mdelay(1); + return; + } - /* Zoom R16/24 needs a tiny delay here, otherwise requests like - * get/set frequency return as failed despite actually succeeding. - */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && - (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(1); + /* Marantz/Denon devices with USB DAC functionality need a delay + * after each class compliant request + */ + if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct)))) + mdelay(20); + } } /* -- 2.6.4

4 5

[alsa-devel] Applied "ASoC: intel: remove unused variable" to the asoc tree
by Mark Brown 01 Mar '16

01 Mar '16

The patch ASoC: intel: remove unused variable has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From dbc77eed6553362e08c1874ff30df640c96eca92 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee <sudipm.mukherjee(a)gmail.com> Date: Mon, 29 Feb 2016 22:53:46 +0530 Subject: [PATCH] ASoC: intel: remove unused variable The variable cmd_id is only assigned some value and is never used. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_ipc.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3dc7358828b3..8afa6fe7b0b0 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -318,7 +318,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, union ipc_header_high msg_high; u32 msg_low; struct ipc_dsp_hdr *dsp_hdr; - unsigned int cmd_id; msg_high = msg->mrfld_header.p.header_high; msg_low = msg->mrfld_header.p.header_low_payload; @@ -357,7 +356,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, return; /* Copy command id so that we can use to put sst to reset */ dsp_hdr = (struct ipc_dsp_hdr *)data; - cmd_id = dsp_hdr->cmd_id; dev_dbg(sst_drv_ctx->dev, "cmd_id %d\n", dsp_hdr->cmd_id); if (sst_wake_up_block(sst_drv_ctx, msg_high.part.result, msg_high.part.drv_id, -- 2.7.0

1 0

[alsa-devel] [PATCH] ASoC: dapm: Fix ctl value accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

snd_soc_dapm_dai_link_get() and _put() access the associated ctl values as value.integer.value[]. However, this is an enum ctl, and it has to be accessed via value.enumerated.item[]. The former is long while the latter is unsigned int, so they don't align. Fixes: c66150824b8a ('ASoC: dapm: add code to configure dai link parameters') Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d3707987900..581175a51ecf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3573,7 +3573,7 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = w->params_select; + ucontrol->value.enumerated.item[0] = w->params_select; return 0; } @@ -3587,13 +3587,13 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, if (w->power) return -EBUSY; - if (ucontrol->value.integer.value[0] == w->params_select) + if (ucontrol->value.enumerated.item[0] == w->params_select) return 0; - if (ucontrol->value.integer.value[0] >= w->num_params) + if (ucontrol->value.enumerated.item[0] >= w->num_params) return -EINVAL; - w->params_select = ucontrol->value.integer.value[0]; + w->params_select = ucontrol->value.enumerated.item[0]; return 0; } -- 2.7.2

2 1

[alsa-devel] [PATCH 0/2] ASoC: omap: Fix enum ctl accesses
by Takashi Iwai 01 Mar '16

01 Mar '16

Hi, here is a patchset to correct the wrong type accesses in omap drivers for enum ctls. I didn't put Cc to stable for them, supposing these are only for 32bit systems. If not, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (2): ASoC: omap: n810: Fix enum ctl accesses in a wrong type ASoC: omap: rx51: Fix enum ctl accesses in a wrong type sound/soc/omap/n810.c | 18 +++++++++--------- sound/soc/omap/rx51.c | 18 +++++++++--------- 2 files changed, 18 insertions(+), 18 deletions(-) -- 2.7.2

3 6

[alsa-devel] [PATCH] ASoC: intel: mfld: Fix enum ctl accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

"Playback Switch" and "Lineout Mux" ctls in medfld machine driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I didn't put Cc to stable, as I assume medfiled is 32bit only. If it also supports 64bit, put Cc to stable. sound/soc/intel/boards/mfld_machine.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c index 49c09a0add79..34f46c72a0e2 100644 --- a/sound/soc/intel/boards/mfld_machine.c +++ b/sound/soc/intel/boards/mfld_machine.c @@ -94,7 +94,7 @@ static const struct soc_enum lo_enum = static int headset_get_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = hs_switch; + ucontrol->value.enumerated.item[0] = hs_switch; return 0; } @@ -104,12 +104,12 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_context *dapm = &card->dapm; - if (ucontrol->value.integer.value[0] == hs_switch) + if (ucontrol->value.enumerated.item[0] == hs_switch) return 0; snd_soc_dapm_mutex_lock(dapm); - if (ucontrol->value.integer.value[0]) { + if (ucontrol->value.enumerated.item[0]) { pr_debug("hs_set HS path\n"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); @@ -123,7 +123,7 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, snd_soc_dapm_mutex_unlock(dapm); - hs_switch = ucontrol->value.integer.value[0]; + hs_switch = ucontrol->value.enumerated.item[0]; return 0; } @@ -148,7 +148,7 @@ static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm) static int lo_get_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = lo_dac; + ucontrol->value.enumerated.item[0] = lo_dac; return 0; } @@ -158,7 +158,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_context *dapm = &card->dapm; - if (ucontrol->value.integer.value[0] == lo_dac) + if (ucontrol->value.enumerated.item[0] == lo_dac) return 0; snd_soc_dapm_mutex_lock(dapm); @@ -168,7 +168,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, */ lo_enable_out_pins(dapm); - switch (ucontrol->value.integer.value[0]) { + switch (ucontrol->value.enumerated.item[0]) { case 0: pr_debug("set vibra path\n"); snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT"); @@ -202,7 +202,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, snd_soc_dapm_mutex_unlock(dapm); - lo_dac = ucontrol->value.integer.value[0]; + lo_dac = ucontrol->value.enumerated.item[0]; return 0; } -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: cs42l51: Fix enum ctl accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

"PCM channel mixer" ctl in cs42l51 codec driver is enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- like other patchsets, I didn't put Cc to stable. If this driver is used for 64bit archs, feel free to put Cc to stable. sound/soc/codecs/cs42l51.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index b3951524339f..35488f14e237 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -60,15 +60,15 @@ static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol, switch (value) { default: case 0: - ucontrol->value.integer.value[0] = 0; + ucontrol->value.enumerated.item[0] = 0; break; /* same value : (L+R)/2 and (R+L)/2 */ case 1: case 2: - ucontrol->value.integer.value[0] = 1; + ucontrol->value.enumerated.item[0] = 1; break; case 3: - ucontrol->value.integer.value[0] = 2; + ucontrol->value.enumerated.item[0] = 2; break; } @@ -85,7 +85,7 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); unsigned char val; - switch (ucontrol->value.integer.value[0]) { + switch (ucontrol->value.enumerated.item[0]) { default: case 0: val = CHAN_MIX_NORMAL; -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: da732x: Fix enum ctl accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

"DAC1 High Pass Filter Mode" & co in da732x codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- like other patchsets, I didn't put Cc to stable. If this driver is used for 64bit archs, feel free to put Cc to stable. sound/soc/codecs/da732x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 1d5a89c5164b..461506a4ca6a 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -334,7 +334,7 @@ static int da732x_hpf_set(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; unsigned int reg = enum_ctrl->reg; - unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.enumerated.item[0]; unsigned int bits; switch (sel) { @@ -368,13 +368,13 @@ static int da732x_hpf_get(struct snd_kcontrol *kcontrol, switch (val) { case DA732X_HPF_VOICE_EN: - ucontrol->value.integer.value[0] = DA732X_HPF_VOICE; + ucontrol->value.enumerated.item[0] = DA732X_HPF_VOICE; break; case DA732X_HPF_MUSIC_EN: - ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC; + ucontrol->value.enumerated.item[0] = DA732X_HPF_MUSIC; break; default: - ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED; + ucontrol->value.enumerated.item[0] = DA732X_HPF_DISABLED; break; } -- 2.7.2

2 1

[alsa-devel] [PATCH 0/5] ASoC: Fix enum ctl accesses in various codec drivers
by Takashi Iwai 01 Mar '16

01 Mar '16

Hi, here are the rest of buggy drivers to be fixed for the wrong ctl access types to enum ctls. Again, put Cc to stable appropriately if needed. thanks, Takashi === Takashi Iwai (5): ASoC: ab8500: Fix enum ctl accesses in a wrong type ASoC: max98088: Fix enum ctl accesses in a wrong type ASoC: max98095: Fix enum ctl accesses in a wrong type ASoC: tlv320dac33: Fix enum ctl accesses in a wrong type ASoC: wl1273: Fix enum ctl accesses in a wrong type sound/soc/codecs/ab8500-codec.c | 8 ++++---- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/codecs/tlv320dac33.c | 9 ++++----- sound/soc/codecs/wl1273.c | 13 ++++++------- 5 files changed, 17 insertions(+), 19 deletions(-) -- 2.7.2

2 10

[alsa-devel] [PATCH v2 0/6] ALSA core X32 ABI fixes
by Takashi Iwai 29 Feb '16

29 Feb '16

Hi, this is the revised patchset for fixing the incompatible ioctls in ALSA core with X32 ABI. Too bad that we have to deal in this way, but it's life... Takashi === Takashi Iwai (6): ALSA: ctl: Fix ioctls for X32 ABI ALSA: pcm: Fix ioctls for X32 ABI ALSA: rawmidi: Use comapt_put_timespec() ALSA: rawmidi: Fix ioctls X32 ABI ALSA: timer: Fix broken compat timer user status ioctl ALSA: timer: Fix ioctls for X32 ABI sound/core/control_compat.c | 90 ++++++++++++++++++---- sound/core/pcm_compat.c | 177 +++++++++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 56 +++++++++++++- sound/core/timer_compat.c | 18 ++++- 4 files changed, 320 insertions(+), 21 deletions(-) -- 2.7.2

2 8

[alsa-devel] [PATCH] ALSA: seq: Remove unimplemented ioctls
by Takashi Iwai 29 Feb '16

29 Feb '16

SNDRV_SEQ_IOCTL_{GET|SET}_QUEUE_OWNER and *_{GET|SET}_QUEUE_SYNC ioctls have been never implemented. Get rid of the definitions from uapi header file. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/asequencer.h | 6 ------ 1 file changed, 6 deletions(-) diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 5a5fa4956ebd..af96f2044f91 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -594,14 +594,8 @@ struct snd_seq_query_subs { #define SNDRV_SEQ_IOCTL_GET_QUEUE_STATUS _IOWR('S', 0x40, struct snd_seq_queue_status) #define SNDRV_SEQ_IOCTL_GET_QUEUE_TEMPO _IOWR('S', 0x41, struct snd_seq_queue_tempo) #define SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO _IOW ('S', 0x42, struct snd_seq_queue_tempo) -#define SNDRV_SEQ_IOCTL_GET_QUEUE_OWNER _IOWR('S', 0x43, struct snd_seq_queue_owner) -#define SNDRV_SEQ_IOCTL_SET_QUEUE_OWNER _IOW ('S', 0x44, struct snd_seq_queue_owner) #define SNDRV_SEQ_IOCTL_GET_QUEUE_TIMER _IOWR('S', 0x45, struct snd_seq_queue_timer) #define SNDRV_SEQ_IOCTL_SET_QUEUE_TIMER _IOW ('S', 0x46, struct snd_seq_queue_timer) -/* XXX -#define SNDRV_SEQ_IOCTL_GET_QUEUE_SYNC _IOWR('S', 0x53, struct snd_seq_queue_sync) -#define SNDRV_SEQ_IOCTL_SET_QUEUE_SYNC _IOW ('S', 0x54, struct snd_seq_queue_sync) -*/ #define SNDRV_SEQ_IOCTL_GET_QUEUE_CLIENT _IOWR('S', 0x49, struct snd_seq_queue_client) #define SNDRV_SEQ_IOCTL_SET_QUEUE_CLIENT _IOW ('S', 0x4a, struct snd_seq_queue_client) #define SNDRV_SEQ_IOCTL_GET_CLIENT_POOL _IOWR('S', 0x4b, struct snd_seq_client_pool) -- 2.7.2

1 0

[alsa-devel] [PATCH] ASoC: qcom: add dependency on HAS_DMA
by Sudip Mukherjee 29 Feb '16

29 Feb '16

m32r allmodconfig build is failing with the error: ERROR: "bad_dma_ops" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! ERROR: "dma_common_mmap" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! lpass-platform uses DMA so add HAS_DMA in the Kconfig. Also add the same for SND_SOC_LPASS_IPQ806X and SND_SOC_LPASS_APQ8016 as they will also select SND_SOC_LPASS_PLATFORM. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- m32r allmodconfig build for next-20160229 is at: https://travis-ci.org/sudipm-mukherjee/parport/jobs/112534168 sound/soc/qcom/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 3cc252e..1e7a783 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -11,15 +11,18 @@ config SND_SOC_LPASS_CPU config SND_SOC_LPASS_PLATFORM tristate + depends on HAS_DMA select REGMAP_MMIO config SND_SOC_LPASS_IPQ806X tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_LPASS_APQ8016 tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM -- 1.9.1

1 1

[alsa-devel] [PATCH] ALSA: mts64: fix NULL pointer dereference
by Sudip Mukherjee 29 Feb '16

29 Feb '16

While registering pardev, the irq_func was also registered. As a result when we tried to probe for the card, an interrupt was generated and in the ISR we tried to dereference private_data. But private_data is still NULL as we have not yet done snd_mts64_create(). Lets probe for the card after mts64 is created. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Fixes: 94a573500d48 ("ALSA: mts64: use new parport device model") Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/drivers/mts64.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index c76bd87..fd4d18d 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -964,11 +964,6 @@ static int snd_mts64_probe(struct platform_device *pdev) err = -EIO; goto free_pardev; } - err = mts64_probe(p); - if (err) { - err = -EIO; - goto release_pardev; - } if ((err = snd_mts64_create(card, pardev, &mts)) < 0) { snd_printd("Cannot create main component\n"); @@ -976,6 +971,12 @@ static int snd_mts64_probe(struct platform_device *pdev) } card->private_data = mts; card->private_free = snd_mts64_card_private_free; + + err = mts64_probe(p); + if (err) { + err = -EIO; + goto __err; + } if ((err = snd_mts64_rawmidi_create(card)) < 0) { snd_printd("Creating Rawmidi component failed\n"); -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: hdsp: Fix wrong boolean ctl value accesses
by Takashi Iwai 29 Feb '16

29 Feb '16

snd-hdsp driver accesses enum item values (int) instead of boolean values (long) wrongly for some ctl elements. This patch fixes them. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme9652/hdsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 2875b4f6d8c9..7c8941b8b2de 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -2879,7 +2879,7 @@ static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); + ucontrol->value.integer.value[0] = hdsp_dds_offset(hdsp); return 0; } @@ -2891,7 +2891,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; - val = ucontrol->value.enumerated.item[0]; + val = ucontrol->value.integer.value[0]; spin_lock_irq(&hdsp->lock); if (val != hdsp_dds_offset(hdsp)) change = (hdsp_set_dds_offset(hdsp, val) == 0) ? 1 : 0; -- 2.7.2

1 0

[alsa-devel] [PATCH 1/2] ALSA: hdspm: Fix wrong boolean ctl value accesses
by Takashi Iwai 29 Feb '16

29 Feb '16

snd-hdspm driver accesses enum item values (int) instead of boolean values (long) wrongly for some ctl elements. This patch fixes them. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme9652/hdspm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8bc8016c173d..b047b1ba48fd 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2261,7 +2261,7 @@ static int snd_hdspm_put_system_sample_rate(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - hdspm_set_dds_value(hdspm, ucontrol->value.enumerated.item[0]); + hdspm_set_dds_value(hdspm, ucontrol->value.integer.value[0]); return 0; } @@ -4449,7 +4449,7 @@ static int snd_hdspm_get_tco_word_term(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = hdspm->tco->term; + ucontrol->value.integer.value[0] = hdspm->tco->term; return 0; } @@ -4460,8 +4460,8 @@ static int snd_hdspm_put_tco_word_term(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - if (hdspm->tco->term != ucontrol->value.enumerated.item[0]) { - hdspm->tco->term = ucontrol->value.enumerated.item[0]; + if (hdspm->tco->term != ucontrol->value.integer.value[0]) { + hdspm->tco->term = ucontrol->value.integer.value[0]; hdspm_tco_write(hdspm); -- 2.7.2

1 1

[alsa-devel] [PATCH] ALSA: ctl: Fix incompatible ioctls for X32 ABI
by Takashi Iwai 29 Feb '16

29 Feb '16

The X32 ABI takes the same alignment like x86-64, and this may result in the incompatible struct size from ia32. Unfortunately, we hit this in some control ABI: struct snd_ctl_elem_value differs between them due to the position of 64bit variable array. This ends up with the unknown ioctl (ENOTTY) error. The fix is to add the compat entries for the new aligned struct. Reported-and-tested-by: Steven Newbury <steve(a)snewbury.org.uk> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control_compat.c | 90 +++++++++++++++++++++++++++++++++++++-------- 1 file changed, 74 insertions(+), 16 deletions(-) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index b9c0910fb8c4..dff3671217c8 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -170,6 +170,19 @@ struct snd_ctl_elem_value32 { unsigned char reserved[128]; }; +#ifdef CONFIG_X86_X32 +/* x32 has a different alignment for 64bit values from ia32 */ +struct snd_ctl_elem_value_x32 { + struct snd_ctl_elem_id id; + unsigned int indirect; /* bit-field causes misalignment */ + union { + s32 integer[128]; + unsigned char data[512]; + s64 integer64[64]; + } value; + unsigned char reserved[128]; +}; +#endif /* CONFIG_X86_X32 */ /* get the value type and count of the control */ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, @@ -219,9 +232,11 @@ static int get_elem_size(int type, int count) static int copy_ctl_value_from_user(struct snd_card *card, struct snd_ctl_elem_value *data, - struct snd_ctl_elem_value32 __user *data32, + void __user *userdata, + void __user *valuep, int *typep, int *countp) { + struct snd_ctl_elem_value32 __user *data32 = userdata; int i, type, size; int uninitialized_var(count); unsigned int indirect; @@ -239,8 +254,9 @@ static int copy_ctl_value_from_user(struct snd_card *card, if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || type == SNDRV_CTL_ELEM_TYPE_INTEGER) { for (i = 0; i < count; i++) { + s32 __user *intp = valuep; int val; - if (get_user(val, &data32->value.integer[i])) + if (get_user(val, &intp[i])) return -EFAULT; data->value.integer.value[i] = val; } @@ -250,8 +266,7 @@ static int copy_ctl_value_from_user(struct snd_card *card, dev_err(card->dev, "snd_ioctl32_ctl_elem_value: unknown type %d\n", type); return -EINVAL; } - if (copy_from_user(data->value.bytes.data, - data32->value.data, size)) + if (copy_from_user(data->value.bytes.data, valuep, size)) return -EFAULT; } @@ -261,7 +276,8 @@ static int copy_ctl_value_from_user(struct snd_card *card, } /* restore the value to 32bit */ -static int copy_ctl_value_to_user(struct snd_ctl_elem_value32 __user *data32, +static int copy_ctl_value_to_user(void __user *userdata, + void __user *valuep, struct snd_ctl_elem_value *data, int type, int count) { @@ -270,22 +286,22 @@ static int copy_ctl_value_to_user(struct snd_ctl_elem_value32 __user *data32, if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || type == SNDRV_CTL_ELEM_TYPE_INTEGER) { for (i = 0; i < count; i++) { + s32 __user *intp = valuep; int val; val = data->value.integer.value[i]; - if (put_user(val, &data32->value.integer[i])) + if (put_user(val, &intp[i])) return -EFAULT; } } else { size = get_elem_size(type, count); - if (copy_to_user(data32->value.data, - data->value.bytes.data, size)) + if (copy_to_user(valuep, data->value.bytes.data, size)) return -EFAULT; } return 0; } -static int snd_ctl_elem_read_user_compat(struct snd_card *card, - struct snd_ctl_elem_value32 __user *data32) +static int ctl_elem_read_user(struct snd_card *card, + void __user *userdata, void __user *valuep) { struct snd_ctl_elem_value *data; int err, type, count; @@ -294,7 +310,9 @@ static int snd_ctl_elem_read_user_compat(struct snd_card *card, if (data == NULL) return -ENOMEM; - if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0) + err = copy_ctl_value_from_user(card, data, userdata, valuep, + &type, &count); + if (err < 0) goto error; snd_power_lock(card); @@ -303,14 +321,15 @@ static int snd_ctl_elem_read_user_compat(struct snd_card *card, err = snd_ctl_elem_read(card, data); snd_power_unlock(card); if (err >= 0) - err = copy_ctl_value_to_user(data32, data, type, count); + err = copy_ctl_value_to_user(userdata, valuep, data, + type, count); error: kfree(data); return err; } -static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, - struct snd_ctl_elem_value32 __user *data32) +static int ctl_elem_write_user(struct snd_ctl_file *file, + void __user *userdata, void __user *valuep) { struct snd_ctl_elem_value *data; struct snd_card *card = file->card; @@ -320,7 +339,9 @@ static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, if (data == NULL) return -ENOMEM; - if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0) + err = copy_ctl_value_from_user(card, data, userdata, valuep, + &type, &count); + if (err < 0) goto error; snd_power_lock(card); @@ -329,12 +350,39 @@ static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, err = snd_ctl_elem_write(card, file, data); snd_power_unlock(card); if (err >= 0) - err = copy_ctl_value_to_user(data32, data, type, count); + err = copy_ctl_value_to_user(userdata, valuep, data, + type, count); error: kfree(data); return err; } +static int snd_ctl_elem_read_user_compat(struct snd_card *card, + struct snd_ctl_elem_value32 __user *data32) +{ + return ctl_elem_read_user(card, data32, &data32->value); +} + +static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, + struct snd_ctl_elem_value32 __user *data32) +{ + return ctl_elem_write_user(file, data32, &data32->value); +} + +#ifdef CONFIG_X86_X32 +static int snd_ctl_elem_read_user_x32_compat(struct snd_card *card, + struct snd_ctl_elem_value_x32 __user *data32) +{ + return ctl_elem_read_user(card, data32, &data32->value); +} + +static int snd_ctl_elem_write_user_x32_compat(struct snd_ctl_file *file, + struct snd_ctl_elem_value_x32 __user *data32) +{ + return ctl_elem_write_user(file, data32, &data32->value); +} +#endif /* CONFIG_X86_X32 */ + /* add or replace a user control */ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, struct snd_ctl_elem_info32 __user *data32, @@ -391,6 +439,10 @@ enum { SNDRV_CTL_IOCTL_ELEM_INFO32 = _IOWR('U', 0x11, struct snd_ctl_elem_info32), SNDRV_CTL_IOCTL_ELEM_READ32 = _IOWR('U', 0x12, struct snd_ctl_elem_value32), SNDRV_CTL_IOCTL_ELEM_WRITE32 = _IOWR('U', 0x13, struct snd_ctl_elem_value32), +#ifdef CONFIG_X86_X32 + SNDRV_CTL_IOCTL_ELEM_READ_X32 = _IOWR('U', 0x12, struct snd_ctl_elem_value_x32), + SNDRV_CTL_IOCTL_ELEM_WRITE_X32 = _IOWR('U', 0x13, struct snd_ctl_elem_value_x32), +#endif SNDRV_CTL_IOCTL_ELEM_ADD32 = _IOWR('U', 0x17, struct snd_ctl_elem_info32), SNDRV_CTL_IOCTL_ELEM_REPLACE32 = _IOWR('U', 0x18, struct snd_ctl_elem_info32), }; @@ -427,6 +479,12 @@ static inline long snd_ctl_ioctl_compat(struct file *file, unsigned int cmd, uns return snd_ctl_elem_read_user_compat(ctl->card, argp); case SNDRV_CTL_IOCTL_ELEM_WRITE32: return snd_ctl_elem_write_user_compat(ctl, argp); +#ifdef CONFIG_X86_X32 + case SNDRV_CTL_IOCTL_ELEM_READ_X32: + return snd_ctl_elem_read_user_x32_compat(ctl->card, argp); + case SNDRV_CTL_IOCTL_ELEM_WRITE_X32: + return snd_ctl_elem_write_user_x32_compat(ctl, argp); +#endif case SNDRV_CTL_IOCTL_ELEM_ADD32: return snd_ctl_elem_add_compat(ctl, argp, 0); case SNDRV_CTL_IOCTL_ELEM_REPLACE32: -- 2.7.2

1 1

[alsa-devel] ERROR: USB Sound card with unsafe characters
by Brian Jones 29 Feb '16

29 Feb '16

Newby alert - but I don't know how to get this bug to the developers I have been trying to work on a fix for Bug 1498129 - https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1498129 on Ubuntu My USB soundcard is an Asus Xonar U7 Echelon Edition <https://www.asus.com/Sound-Cards/Xonar_U7_Echelon_Edition/> which identifies itself to alsa as "Xonar U7 Echelon Ed." The card is supported by alsa and works well in analog mode but I cannot get digital output via alsa. It has iec958 as device 1 rather than device 0 and I can access digital output by referring to directly to hw:x,1 but cannot access digital output via alsa. Attempts to change the iec958 device in USB-Audio.conf fail because of the period char and apparently my level of OS (Mint 17.3 based on Ubuntu 14.04, kernel 3.19.0-32) does not have alsa supporting USB vendorid:productid naming so I don't have a workaround. If a patch exists, advice on applying/rebuilding alsa would be gratefully appreciated.

2 2

[alsa-devel] [PATCH] ASoC: nau8825: reduce stanby power consumption
by John Hsu 28 Feb '16

28 Feb '16

Fix unnecessary widget power up when standby; And decrease internal clock frequency to save power. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 49 +++++++++++++++++++++++++++++++++------------- 1 file changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 52882e4..f9752cc 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -262,6 +262,29 @@ static int nau8825_playback_event(struct snd_soc_dapm_widget *w, return 0; } +static int nau8825_dacr_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + break; + case SND_SOC_DAPM_POST_PMD: + if (!nau8825->irq) + regmap_update_bits(nau8825->regmap, + NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + static const char * const nau8825_adc_decimation[] = { "32", "64", "128", "256" @@ -337,16 +360,17 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { 0), /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), - SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC_E("DDACR", NULL, SND_SOC_NOPM, 0, 0, + nau8825_dacr_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACL_SFT, 0), SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), @@ -878,13 +902,6 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* The interrupt clock is gated by x1[10:8], - * one of them needs to be enabled all the time for - * interrupts to happen. - */ - snd_soc_dapm_force_enable_pin(dapm, "DDACR"); - snd_soc_dapm_sync(dapm); - /* Unmask interruptions. Handler uses dapm object so we can enable * interruptions only after dapm is fully initialized. */ @@ -1075,6 +1092,12 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_DCO_EN); regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; @@ -1309,9 +1332,7 @@ static int nau8825_setup_irq(struct nau8825 *nau8825) /* Enable internal VCO needed for interruptions */ nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - /* Enable DDACR needed for interrupts - * It is the same as force_enable_pin("DDACR") we do later - */ + /* Enable DDACR needed for interrupts */ regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); -- 2.6.4

2 5

[alsa-devel] [PATCH] ALSA: dice: drop duplex streams synchronization to transfer own time stamps
by Takashi Sakamoto 28 Feb '16

28 Feb '16

This commit drops implementation of duplex streams synchronization from ALSA dice driver, due to a reason of hardware design. This patch allows dice-based units to generate sounds correctly when isochronous packet streaming starts at first time. In IEC 61883-6:2005, CIP packetization layer for AM824 data format utilizes the value of SYT field in CIP header of received packet for a reference to phase lock loop. Figure 3 in clause 4.3 describes it. The value is an offset from cycle_time field of every cycle start packet from cycle master on IEEE 1394 bus. The time calculated with these two fields is called as 'presentation timestamp' which represents the time to play data included in the packet. Although, this idea includes some problems due to accuracy of timekeep in cycle master, accuracy of transmission of cycle start packet on the bus with the other units, accuracy of sampling clock in data transmitter side and accuracy of replay in data receiver side. In most case, these accuracies somewhat worse because there's no such ideal hardwares in this world. For the issues, ASICs for Dice include Jitter Elimination Technologies (JET) PLL. The PLL can handle several sources of clock and compensate it with high-precision internal clock source. The sequence of value in syt field of received AMDTP packets is one of the sources, therefore transmitters on IEEE 1394 bus should transfer it. On the other hand, current ALSA dice driver is programmed with a mode of duplex streams with synchronization. In this mode, the driver outputs packets after some incoming packets are handled, to re-use the value of SYT field in incoming packets to the value for outgoing packets. This mode is enabled when source signal of sampling clock is set to internal, and this is a major use case. Thus, in most cases, the unit receives no packets during a short time after packet streaming starts. As long as I experienced, this causes the units to generate no sounds at first time to receive packets. This issue occurs only with Dice II. I guess this is due to a quirk of the PLL. In short, the PLL cannot generate firm signals to ADCs/DACs or the other ICs when no packets are received in the beginning of packet streaming. While, on second time or later, the unit generates sound correctly. I guess that starting packet streaming at first time sets the PLL correctly. Well, still based on my hypothesis and no way to prove it, this commit drops duplex streams synchronization from this driver. At least, the PLL requires the sequence of value in SYT field of received AMDTP packets as one of source of clock signals with internal clock source. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 42 ++------------------------------------- 1 file changed, 2 insertions(+), 40 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index a64b3cc..df035b1 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -191,53 +191,17 @@ end: return err; } -static int get_sync_mode(struct snd_dice *dice, enum cip_flags *sync_mode) -{ - u32 source; - int err; - - err = snd_dice_transaction_get_clock_source(dice, &source); - if (err < 0) - goto end; - - switch (source) { - /* So-called 'SYT Match' modes, sync_to_syt value of packets received */ - case CLOCK_SOURCE_ARX4: /* in 4th stream */ - case CLOCK_SOURCE_ARX3: /* in 3rd stream */ - case CLOCK_SOURCE_ARX2: /* in 2nd stream */ - err = -ENOSYS; - break; - case CLOCK_SOURCE_ARX1: /* in 1st stream, which this driver uses */ - *sync_mode = 0; - break; - default: - *sync_mode = CIP_SYNC_TO_DEVICE; - break; - } -end: - return err; -} - int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) { struct amdtp_stream *master, *slave; unsigned int curr_rate; - enum cip_flags sync_mode; int err = 0; if (dice->substreams_counter == 0) goto end; - err = get_sync_mode(dice, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &dice->tx_stream; - slave = &dice->rx_stream; - } else { - master = &dice->rx_stream; - slave = &dice->tx_stream; - } + master = &dice->rx_stream; + slave = &dice->tx_stream; /* Some packet queueing errors. */ if (amdtp_streaming_error(master) || amdtp_streaming_error(slave)) @@ -261,8 +225,6 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) stop_stream(dice, slave); snd_dice_transaction_clear_enable(dice); - amdtp_stream_set_sync(sync_mode, master, slave); - err = ensure_phase_lock(dice); if (err < 0) { dev_err(&dice->unit->device, -- 2.5.0

2 1

[alsa-devel] [PATCH 0/2] clk: imx6: add kpp clock for i.MX6UL
by Lothar Waßmann 28 Feb '16

28 Feb '16

This patchset adds the clock which is necessary to operate the KPP unit on i.MX6UL. The first patch removes bogus whitespace before TABs in indentation. The second patch adds the clock definition.

3 4

[alsa-devel] [PATCH 1/4] ASoC: wm9713: add binding for WM9713 codec
by Robert Jarzmik 27 Feb '16

27 Feb '16

This adds a binding for the Wolfson WM9713 audio codec. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Documentation/devicetree/bindings/sound/wm9713.txt | 14 ++++++++++++++ 1 file changed, 14 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm9713.txt diff --git a/Documentation/devicetree/bindings/sound/wm9713.txt b/Documentation/devicetree/bindings/sound/wm9713.txt new file mode 100644 index 000000000000..761001bc3201 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm9713.txt @@ -0,0 +1,14 @@ +WM9713 audio CODEC + +This devices supports I2C. + +Required properties: + - compatible : + "wlf,wm9713" + +Example: + wm9713: wm9713@0 { + compatible = "wlf,wm9713"; + #sound-dai-cells = <1>; + status = "okay"; + }; -- 2.1.4

3 16

[alsa-devel] [PATCH] ASoC: rt5640: add master clock handling for rt5640
by Sugar Zhang 27 Feb '16

27 Feb '16

enable/disable master clock when codec is active or not. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Documentation/devicetree/bindings/sound/rt5640.txt | 3 +++ sound/soc/codecs/rt5640.c | 31 ++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2 ++ 3 files changed, 36 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index 9e62f6e..57fe646 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -12,6 +12,9 @@ Required properties: Optional properties: +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + - realtek,in1-differential - realtek,in2-differential - realtek,in3-differential diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 11d032c..6cd84fb 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1949,7 +1949,33 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, static int rt5640_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* + * SND_SOC_BIAS_PREPARE is called while preparing for a + * transition to ON or away from ON. If current bias_level + * is SND_SOC_BIAS_ON, then it is preparing for a transition + * away from ON. Disable the clock in that case, otherwise + * enable it. + */ + if (IS_ERR(rt5640->mclk)) + break; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { + clk_disable_unprepare(rt5640->mclk); + } else { + ret = clk_prepare_enable(rt5640->mclk); + if (ret) + return ret; + } + break; + case SND_SOC_BIAS_STANDBY: if (SND_SOC_BIAS_OFF == snd_soc_codec_get_bias_level(codec)) { snd_soc_update_bits(codec, RT5640_PWR_ANLG1, @@ -2088,6 +2114,11 @@ static int rt5640_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* Check if MCLK provided */ + rt5640->mclk = devm_clk_get(codec->dev, "mclk"); + if (PTR_ERR(rt5640->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + rt5640->codec = codec; snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 83a7150..1761c3a9 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -12,6 +12,7 @@ #ifndef _RT5640_H #define _RT5640_H +#include <linux/clk.h> #include <sound/rt5640.h> /* Info */ @@ -2097,6 +2098,7 @@ struct rt5640_priv { struct snd_soc_codec *codec; struct rt5640_platform_data pdata; struct regmap *regmap; + struct clk *mclk; int sysclk; int sysclk_src; -- 1.9.1

5 9

[alsa-devel] [PATCH] ALSA: hda - Autosuspend controller after probe even if codecs are already suspended
by ville.syrjala＠linux.intel.com 26 Feb '16

26 Feb '16

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> azx_probe_continue() uses pm_runtime_put_noidle() to drop the rpm usage_count, which means that if it's the last reference the autosuspend of the controller won't actually happen. So if the codecs autosuspend before the azx_probe_continue() drops the last reference we'll fail to autosuspend the controller. This does happen in practice, but not every time. As can be seen in [1] the controller autosuspend attempt fails due to the usage_count when suspending the codecs. A bit later we see the the contoller usage_count dropping to zero without further attempts at autosuspend. Fix the problem by using pm_runtime_put_autosuspend() instead, which will kick off the autosuspend of the controller even if the codecs are already asleep. As can be seen in [2] the controller autosuspend still fails while suspending the codecs, but later on we see another autosuspend attempt after dropping the usage_count to 0. I was also a bit worried that there might still be a race between the controller autosuspend and the rest of the code in azx_probe_continue(). So I also tried replacing the the put_noidle() with put_sync_suspend(). No explosions occurred, so I'm somewhat satisfied that there are no serious problems in this area. [1] kworker/1:2-122 [001] .... 63.661310: __pm_runtime_suspend: hdaudioC0D0 usage_count 0 kworker/1:2-122 [001] d..2 63.661316: rpm_suspend: hdaudioC0D0 flags-d cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:2-122 [001] d..1 63.661317: rpm_check_suspend_allowed: hdaudioC0D0 retval 0 kworker/1:2-122 [001] d..2 63.661332: rpm_return_int: rpm_suspend+0x406/0x5e8:hdaudioC0D0 ret=0 kworker/1:1-72 [001] d..2 63.661543: rpm_suspend: hdaudioC0D0 flags-a cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:1-72 [001] d..1 63.661544: rpm_check_suspend_allowed: hdaudioC0D0 retval 0 kworker/1:1-72 [001] .... 63.661545: hda_codec_runtime_suspend: hdaudioC0D0 suspend kworker/1:1-72 [001] d..2 63.661614: rpm_idle: 0000:00:03.0 flags-1 cnt-1 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:1-72 [001] d..1 63.661615: rpm_check_suspend_allowed: 0000:00:03.0 usage_count 1 kworker/1:1-72 [001] d..1 63.661615: rpm_check_suspend_allowed: 0000:00:03.0 retval -11 kworker/1:1-72 [001] d..2 63.661616: rpm_return_int: rpm_idle+0x249/0x487:0000:00:03.0 ret=-11 kworker/1:1-72 [001] d..2 63.661616: rpm_return_int: rpm_suspend+0x406/0x5e8:hdaudioC0D0 ret=0 kworker/1:2-122 [001] d..2 63.664834: rpm_idle: hdaudioC0D0 flags-8 cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:2-122 [001] d..1 63.664835: rpm_check_suspend_allowed: hdaudioC0D0 retval 1 kworker/1:2-122 [001] d..2 63.664836: rpm_return_int: rpm_idle+0x249/0x487:hdaudioC0D0 ret=-11 kworker/1:2-122 [001] d..2 63.664841: rpm_idle: hdaudioC0D0 flags-8 cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:2-122 [001] d..1 63.664841: rpm_check_suspend_allowed: hdaudioC0D0 retval 1 kworker/1:2-122 [001] d..2 63.664841: rpm_return_int: rpm_idle+0x249/0x487:hdaudioC0D0 ret=-11 kworker/1:2-122 [001] .... 63.664842: azx_probe_continue: 0000:00:03.0 usage_count=0 [2] kworker/0:0-4 [000] .... 50.354567: __pm_runtime_suspend: hdaudioC0D0 usage_count 0 kworker/0:0-4 [000] d..2 50.354574: rpm_suspend: hdaudioC0D0 flags-d cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.354575: rpm_check_suspend_allowed: hdaudioC0D0 retval 0 kworker/0:0-4 [000] d..2 50.354589: rpm_return_int: rpm_suspend+0x406/0x5e8:hdaudioC0D0 ret=0 kworker/0:2-135 [000] d..2 50.354809: rpm_suspend: hdaudioC0D0 flags-a cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:2-135 [000] d..1 50.354810: rpm_check_suspend_allowed: hdaudioC0D0 retval 0 kworker/0:2-135 [000] .... 50.354816: hda_codec_runtime_suspend: hdaudioC0D0 suspend kworker/0:2-135 [000] d..2 50.354908: rpm_idle: 0000:00:03.0 flags-1 cnt-1 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:2-135 [000] d..1 50.354909: rpm_check_suspend_allowed: 0000:00:03.0 usage_count 1 kworker/0:2-135 [000] d..1 50.354909: rpm_check_suspend_allowed: 0000:00:03.0 retval -11 kworker/0:2-135 [000] d..2 50.354909: rpm_return_int: rpm_idle+0x249/0x487:0000:00:03.0 ret=-11 kworker/0:2-135 [000] d..2 50.354910: rpm_return_int: rpm_suspend+0x406/0x5e8:hdaudioC0D0 ret=0 kworker/0:0-4 [000] d..2 50.373791: rpm_idle: hdaudioC0D0 flags-8 cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.373792: rpm_check_suspend_allowed: hdaudioC0D0 retval 1 kworker/0:0-4 [000] d..2 50.373793: rpm_return_int: rpm_idle+0x249/0x487:hdaudioC0D0 ret=-11 kworker/0:0-4 [000] d..2 50.373797: rpm_idle: hdaudioC0D0 flags-8 cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.373798: rpm_check_suspend_allowed: hdaudioC0D0 retval 1 kworker/0:0-4 [000] d..2 50.373798: rpm_return_int: rpm_idle+0x249/0x487:hdaudioC0D0 ret=-11 kworker/0:0-4 [000] .... 50.373799: __pm_runtime_suspend: 0000:00:03.0 usage_count 0 kworker/0:0-4 [000] d..2 50.373800: rpm_suspend: 0000:00:03.0 flags-d cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.373800: rpm_check_suspend_allowed: 0000:00:03.0 retval 0 kworker/0:0-4 [000] d..2 50.373803: rpm_return_int: rpm_suspend+0x406/0x5e8:0000:00:03.0 ret=0 kworker/0:0-4 [000] d..2 50.385164: rpm_suspend: 0000:00:03.0 flags-a cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.385165: rpm_check_suspend_allowed: 0000:00:03.0 retval 0 kworker/0:0-4 [000] .... 50.385174: azx_runtime_suspend: 0000:00:03.0 azx suspend releaseing power well kworker/0:0-4 [000] .... 50.385179: azx_runtime_suspend: 0000:00:03.0 azx suspend kworker/0:0-4 [000] d..2 50.386872: rpm_return_int: rpm_suspend+0x406/0x5e8:0000:00:03.0 ret=0 Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e5240cb3749f..2624cfe98884 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2145,7 +2145,7 @@ static int azx_probe_continue(struct azx *chip) azx_add_card_list(chip); snd_hda_set_power_save(&chip->bus, power_save * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) - pm_runtime_put_noidle(&pci->dev); + pm_runtime_put_autosuspend(&pci->dev); out_free: if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL -- 2.4.10

2 1

[alsa-devel] [PATCH] ASoC: Extend FLL function
by John Hsu 26 Feb '16

26 Feb '16

Extend FFL clock source from MCLK/BCLK/FS; And modify some FLL parameter setting. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 100 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/nau8825.h | 14 ++++++- 2 files changed, 86 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 7fc7b4e..16da8ef 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -926,7 +926,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, struct nau8825_fll *fll_param) { regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, NAU8825_FLL_RATIO_MASK, fll_param->ratio); /* FLL 16-bit fractional input */ @@ -939,10 +940,20 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, - NAU8825_FLL_FILTER_SW_MASK, 0x0000); - /* FLL sigma delta modulator enable */ + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FILTER_SW_MASK, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FILTER_SW_REF); + /* Disable free-running mode */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + NAU8825_DCO_EN, 0); + /* FLL sigma delta modulator enable */ + if (fll_param->fll_frac) + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + else + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN_MASK, 0); } /* freq_out must be 256*Fs in order to achieve the best performance */ @@ -970,6 +981,37 @@ static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, return 0; } +static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) +{ + int ret = 0; + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + goto done; + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + goto done; + } + } + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + goto done; + } + } + +done: + return ret; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -981,29 +1023,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); - - /* We selected MCLK source but the clock itself managed externally */ - if (!nau8825->mclk) - break; - - if (!nau8825->mclk_freq) { - ret = clk_prepare_enable(nau8825->mclk); - if (ret) { - dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); - return ret; - } - } - - if (nau8825->mclk_freq != freq) { - nau8825->mclk_freq = freq; - - freq = clk_round_rate(nau8825->mclk, freq); - ret = clk_set_rate(nau8825->mclk, freq); - if (ret) { - dev_err(nau8825->dev, "Unable to set mclk rate\n"); - return ret; - } - } + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; break; case NAU8825_CLK_INTERNAL: @@ -1018,6 +1040,30 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, } break; + case NAU8825_CLK_FLL_MCLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_MCLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_BLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_BLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_FS: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_FS); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; default: dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); return -EINVAL; diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb..e8f7ca5 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -112,12 +112,21 @@ /* FLL3 (0x06) */ #define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) +#define NAU8825_FLL_CLK_SRC_SFT 10 +#define NAU8825_FLL_CLK_SRC_MASK (0x3 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_MCLK (0 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_BLK (0x2 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) /* FLL5 (0x08) */ -#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) +#define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8825_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 13) +#define NAU8825_FLL_FILTER_SW_N2 (0x1 << 13) +#define NAU8825_FLL_FILTER_SW_REF (0x0 << 13) /* FLL6 (0x9) */ #define NAU8825_DCO_EN_MASK (0x1 << 15) @@ -306,6 +315,9 @@ enum { NAU8825_CLK_MCLK = 0, NAU8825_CLK_INTERNAL, + NAU8825_CLK_FLL_MCLK, + NAU8825_CLK_FLL_BLK, + NAU8825_CLK_FLL_FS, }; struct nau8825 { -- 1.9.1

3 7

[alsa-devel] [PATCH RESEND] ASoC: pxa: remove unused variable
by Arnd Bergmann 26 Feb '16

26 Feb '16

As pointed out by Zhangfei Gao, the sspa_div variable in brownstone_wm8994_hw_params() is completely unused, so as a cleanup following a prior patch, this removes both the variable and the division. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- Submitted originally on Nov 23 2015, this apparently got lost, resending it now. sound/soc/pxa/brownstone.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 416ea646c3b1..ec522e94b0e2 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -52,7 +52,6 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int freq_out, sspa_mclk, sysclk; - int sspa_div; if (params_rate(params) > 11025) { freq_out = params_rate(params) * 512; @@ -63,7 +62,6 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, sysclk = params_rate(params) * 512; sspa_mclk = params_rate(params) * 64; } - sspa_div = freq_out / sspa_mclk; snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk); -- 2.7.0

2 2

Re: [alsa-devel] Fwd: Lenovo Y700-17ISK notebook - multiple issues
by Takashi Iwai 26 Feb '16

26 Feb '16

On Wed, 24 Feb 2016 21:20:06 +0100, Aljosa wrote: > > OS: Ubuntu 16.04 > I tested first the original Ubuntu kernel and now I'm running > 4.5-rc5-wily. > I didn't test any model option because I don't know how to do it (I'm > not an expert). > > Please find attached the 'alsa-info.sh output' file. Please don't top-post. Also, again, don't drop Cc to ML. Judging from alsa-info.sh output, there is no pin declared for the bass output by BIOS. So you'd need to identify the pin by yourself, e.g. via hdajackretask, and set it up manually. Once when you figuring out which pin corresponds to the bass output, let us know. We can add it statically in the driver fixup. thanks, Takashi > > Thanks, ciao :) > Aljosa > > > > On 2016-02-24 20:29, Takashi Iwai wrote: > > On Wed, 24 Feb 2016 19:00:36 +0100, > > Aljosa wrote: > >> > >> Hi Takashi, > >> thank you very much for your reply. > >> Notebook subwoofer doesn't work at all (it does not produce any > >> sound). > > > > OK, could you give more details? Which kernel did you test? Did you > > test any model option? > > > > Also give alsa-info.sh output (run it with --no-upload option and > > attach the output). This contains the h/w details. > > > > Last but not least, don't drop Cc to ML. > > > > > > thanks, > > > > Takashi > > > >> Regards, Aljosa > >> > >> > >> > >> On 2016-02-24 16:35, Takashi Iwai wrote: > >> > On Tue, 23 Feb 2016 20:37:21 +0100, > >> > Aljosa wrote: > >> >> > >> >> Hello, > >> >> advised by one of the kernel developers - I'm reporting you the > >> >> problem > >> >> nr.2 from the below link. > >> > > >> > Could you describe what are the exact issues, instead of pointing some > >> > URL? > >> > > >> > > >> > thanks, > >> > > >> > Takashi > >> > upload=true&script=true&cardinfo= > !!################################ > !!ALSA Information Script v 0.4.64 > !!################################ > > !!Script ran on: Wed Feb 24 20:16:58 UTC 2016 > > > !!Linux Distribution > !!------------------ > > Ubuntu Xenial Xerus (development branch) \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu Xenial Xerus (development branch)" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 16.04" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" UBUNTU_CODENAME=xenial > > > !!DMI Information > !!--------------- > > Manufacturer: LENOVO > Product Name: 80Q0 > Product Version: Lenovo ideapad Y700-17ISK > Firmware Version: CDCN26WW > > > !!Kernel Information > !!------------------ > > Kernel release: 4.5.0-040500rc5-generic > Operating System: GNU/Linux > Architecture: x86_64 > Processor: x86_64 > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: k4.5.0-040500rc5-generic > Library version: 1.1.0 > Utilities version: 1.1.0 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [PCH ]: HDA-Intel - HDA Intel PCH > HDA Intel PCH at 0x94320000 irq 132 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1f.3 Audio device: Intel Corporation Sunrise Point-H HD Audio (rev 31) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!------------------------------------------------------- > > 00:1f.3 0403: 8086:a170 (rev 31) > Subsystem: 17aa:3802 > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd_pcsp: index=-2 > snd_usb_audio: index=-2 > snd_atiixp_modem: index=-2 > snd_intel8x0m: index=-2 > snd_via82xx_modem: index=-2 > snd_atiixp_modem: index=-2 > snd_intel8x0m: index=-2 > snd_via82xx_modem: index=-2 > snd_usb_audio: index=-2 > snd_usb_caiaq: index=-2 > snd_usb_ua101: index=-2 > snd_usb_us122l: index=-2 > snd_usb_usx2y: index=-2 > snd_cmipci: mpu_port=0x330 fm_port=0x388 > snd_pcsp: index=-2 > snd_usb_audio: index=-2 > > > !!Loaded sound module options > !!--------------------------- > > !!Module: snd_hda_intel > align_buffer_size : -1 > bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > enable_msi : -1 > id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > power_save : 0 > power_save_controller : Y > probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > single_cmd : N > snoop : -1 > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: Realtek ALC233 > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x10ec0235 > Subsystem Id: 0x17aa3829 > Revision Id: 0x100002 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > State of AFG node 0x01: > Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS > Power: setting=D0, actual=D0 > GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out > Control: name="Speaker Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x50 0x50] > Converter: stream=1, channel=0 > PCM: > rates [0x60]: 44100 48000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out > Control: name="Headphone Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Device: name="ALC233 Analog", type="Audio", device=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x00 0x00] > Converter: stream=1, channel=0 > PCM: > rates [0x60]: 44100 48000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Control: name="Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Capture Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Device: name="ALC233 Analog", type="Audio", device=0 > Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 > Amp-In vals: [0x30 0x30] > Converter: stream=1, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 > Amp-In vals: [0x97 0x97] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 1 > 0x22 > Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Control: name="Mic Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=1, ofs=0 > Control: name="Mic Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=1, ofs=0 > Control: name="Beep Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=4, ofs=0 > Control: name="Beep Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=4, ofs=0 > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 5 > 0x18 0x19 0x1a 0x1b 0x1d > Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x03 0x0b > Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] > Connection: 1 > 0x0d > Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In > Control: name="Internal Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00000020: IN > Pin Default 0x90a60140: [Fixed] Mic at Int N/A > Conn = Digital, Color = Unknown > DefAssociation = 0x4, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > Control: name="Speaker Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00010014: OUT EAPD Detect > EAPD 0x2: EAPD > Pin Default 0x90170110: [Fixed] Speaker at Int N/A > Conn = Analog, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Connection: 1 > 0x0c > Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80] > Pincap 0x00000010: OUT > Pin Default 0x40000000: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x0, Sequence = 0x0 > Pin-ctls: 0x00: > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 1 > 0x0f > Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00003724: IN Detect > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In > Control: name="Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00003724: IN Detect > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x04a11030: [Jack] Mic at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=02, enabled=1 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00003724: IN Detect > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0001373c: IN OUT HP EAPD Detect > Vref caps: HIZ 50 GRD 80 100 > EAPD 0x2: EAPD > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 2 > 0x0c* 0x0d > Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > Pincap 0x00000020: IN > Pin Default 0x40f79a2d: [N/A] Other at Ext N/A > Conn = Analog, Color = Pink > DefAssociation = 0x2, Sequence = 0xd > Pin-ctls: 0x20: IN > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 1 > 0x06 > Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=76 > Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > Control: name="Headphone Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0001001c: OUT HP EAPD Detect > EAPD 0x2: EAPD > Pin Default 0x04211020: [Jack] HP Out at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=01, enabled=1 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 2 > 0x0c 0x0d* > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 6 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x00 0x00] > Connection: 7 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12 > Codec: Intel Skylake HDMI > Address: 2 > AFG Function Id: 0x1 (unsol 0) > Vendor Id: 0x80862809 > Subsystem Id: 0x80860101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > State of AFG node 0x01: > Power states: D0 D3 CLKSTOP EPSS > Power: setting=D0, actual=D0, Clock-stop-OK > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital > Converter: stream=0, channel=0 > Digital: Enabled KAE > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1a]: 16 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital > Converter: stream=0, channel=0 > Digital: Enabled KAE > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1a]: 16 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital > Converter: stream=0, channel=0 > Digital: Enabled KAE > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1a]: 16 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Control: name="IEC958 Playback Con Mask", index=0, device=0 > Control: name="IEC958 Playback Pro Mask", index=0, device=0 > Control: name="IEC958 Playback Default", index=0, device=0 > Control: name="IEC958 Playback Switch", index=0, device=0 > Control: name="ELD", index=0, device=3 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0b000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Devices: 0 > Connection: 0 > In-driver Connection: 3 > 0x02 0x03 0x04 > Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Control: name="IEC958 Playback Con Mask", index=1, device=0 > Control: name="IEC958 Playback Pro Mask", index=1, device=0 > Control: name="IEC958 Playback Default", index=1, device=0 > Control: name="IEC958 Playback Switch", index=1, device=0 > Control: name="ELD", index=0, device=7 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0b000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Devices: 0 > Connection: 0 > In-driver Connection: 3 > 0x02 0x03 0x04 > Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Control: name="IEC958 Playback Con Mask", index=2, device=0 > Control: name="IEC958 Playback Pro Mask", index=2, device=0 > Control: name="IEC958 Playback Default", index=2, device=0 > Control: name="IEC958 Playback Switch", index=2, device=0 > Control: name="ELD", index=0, device=8 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0b000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Devices: 0 > Connection: 0 > In-driver Connection: 3 > 0x02 0x03 0x04 > Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono > --endcollapse-- > > > !!ALSA Device nodes > !!----------------- > > crw-rw----+ 1 root audio 116, 2 Feb 24 21:02 /dev/snd/controlC0 > crw-rw----+ 1 root audio 116, 8 Feb 24 21:02 /dev/snd/hwC0D0 > crw-rw----+ 1 root audio 116, 9 Feb 24 21:02 /dev/snd/hwC0D2 > crw-rw----+ 1 root audio 116, 4 Feb 24 21:02 /dev/snd/pcmC0D0c > crw-rw----+ 1 root audio 116, 3 Feb 24 21:02 /dev/snd/pcmC0D0p > crw-rw----+ 1 root audio 116, 5 Feb 24 21:02 /dev/snd/pcmC0D3p > crw-rw----+ 1 root audio 116, 6 Feb 24 21:02 /dev/snd/pcmC0D7p > crw-rw----+ 1 root audio 116, 7 Feb 24 21:02 /dev/snd/pcmC0D8p > crw-rw----+ 1 root audio 116, 1 Feb 24 21:02 /dev/snd/seq > crw-rw----+ 1 root audio 116, 33 Feb 24 21:02 /dev/snd/timer > > /dev/snd/by-path: > total 0 > drwxr-xr-x 2 root root 60 Feb 24 21:02 . > drwxr-xr-x 3 root root 260 Feb 24 21:02 .. > lrwxrwxrwx 1 root root 12 Feb 24 21:02 pci-0000:00:1f.3 -> ../controlC0 > > > !!Aplay/Arecord output > !!-------------------- > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: PCH [HDA Intel PCH], device 0: ALC233 Analog [ALC233 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > ARECORD > > **** List of CAPTURE Hardware Devices **** > card 0: PCH [HDA Intel PCH], device 0: ALC233 Analog [ALC233 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > !!Amixer output > !!------------- > > !!-------Mixer controls for card 0 [PCH] > > Card hw:0 'PCH'/'HDA Intel PCH at 0x94320000 irq 132' > Mixer name : 'Realtek ALC233' > Components : 'HDA:10ec0235,17aa3829,00100002 HDA:80862809,80860101,00100000' > Controls : 44 > Simple ctrls : 14 > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > Playback channels: Mono > Limits: Playback 0 - 87 > Mono: Playback 80 [92%] [-5.25dB] [on] > Simple mixer control 'Headphone',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 87 > Mono: > Front Left: Playback 0 [0%] [-65.25dB] [off] > Front Right: Playback 0 [0%] [-65.25dB] [off] > Simple mixer control 'Speaker',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 87 > Mono: > Front Left: Playback 87 [100%] [0.00dB] [on] > Front Right: Playback 87 [100%] [0.00dB] [on] > Simple mixer control 'PCM',0 > Capabilities: pvolume > Playback channels: Front Left - Front Right > Limits: Playback 0 - 255 > Mono: > Front Left: Playback 255 [100%] [0.00dB] > Front Right: Playback 255 [100%] [0.00dB] > Simple mixer control 'Mic',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 0 [0%] [-34.50dB] [off] > Front Right: Playback 0 [0%] [-34.50dB] [off] > Simple mixer control 'Mic Boost',0 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'IEC958',0 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'IEC958',1 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [on] > Simple mixer control 'IEC958',2 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [on] > Simple mixer control 'Beep',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 0 [0%] [-34.50dB] [off] > Front Right: Playback 0 [0%] [-34.50dB] [off] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 63 > Front Left: Capture 48 [76%] [18.75dB] [on] > Front Right: Capture 48 [76%] [18.75dB] [on] > Simple mixer control 'Auto-Mute Mode',0 > Capabilities: enum > Items: 'Disabled' 'Enabled' > Item0: 'Enabled' > Simple mixer control 'Internal Mic Boost',0 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'Loopback Mixing',0 > Capabilities: enum > Items: 'Disabled' 'Enabled' > Item0: 'Enabled' > > > !!Alsactl output > !!-------------- > > --startcollapse-- > state.PCH { > control.1 { > iface MIXER > name 'Headphone Playback Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 87' > dbmin -6525 > dbmax 0 > dbvalue.0 -6525 > dbvalue.1 -6525 > } > } > control.2 { > iface MIXER > name 'Headphone Playback Switch' > value.0 false > value.1 false > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.3 { > iface MIXER > name 'Speaker Playback Volume' > value.0 87 > value.1 87 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 87' > dbmin -6525 > dbmax 0 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.4 { > iface MIXER > name 'Speaker Playback Switch' > value.0 true > value.1 true > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.5 { > iface MIXER > name 'Loopback Mixing' > value Enabled > comment { > access 'read write' > type ENUMERATED > count 1 > item.0 Disabled > item.1 Enabled > } > } > control.6 { > iface MIXER > name 'Mic Playback Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 31' > dbmin -3450 > dbmax 1200 > dbvalue.0 -3450 > dbvalue.1 -3450 > } > } > control.7 { > iface MIXER > name 'Mic Playback Switch' > value.0 false > value.1 false > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.8 { > iface MIXER > name 'Auto-Mute Mode' > value Enabled > comment { > access 'read write' > type ENUMERATED > count 1 > item.0 Disabled > item.1 Enabled > } > } > control.9 { > iface MIXER > name 'Capture Volume' > value.0 48 > value.1 48 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 63' > dbmin -1725 > dbmax 3000 > dbvalue.0 1875 > dbvalue.1 1875 > } > } > control.10 { > iface MIXER > name 'Capture Switch' > value.0 true > value.1 true > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.11 { > iface MIXER > name 'Mic Boost Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.12 { > iface MIXER > name 'Internal Mic Boost Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.13 { > iface MIXER > name 'Master Playback Volume' > value 80 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 87' > dbmin -6525 > dbmax 0 > dbvalue.0 -525 > } > } > control.14 { > iface MIXER > name 'Master Playback Switch' > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.15 { > iface CARD > name 'Mic Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.16 { > iface CARD > name 'Internal Mic Phantom Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.17 { > iface CARD > name 'Headphone Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.18 { > iface CARD > name 'Speaker Phantom Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.19 { > iface MIXER > name 'Beep Playback Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 31' > dbmin -3450 > dbmax 1200 > dbvalue.0 -3450 > dbvalue.1 -3450 > } > } > control.20 { > iface MIXER > name 'Beep Playback Switch' > value.0 false > value.1 false > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.21 { > iface PCM > name 'Playback Channel Map' > value.0 0 > value.1 0 > comment { > access read > type INTEGER > count 2 > range '0 - 36' > } > } > control.22 { > iface PCM > name 'Capture Channel Map' > value.0 0 > value.1 0 > comment { > access read > type INTEGER > count 2 > range '0 - 36' > } > } > control.23 { > iface CARD > name 'HDMI/DP,pcm=3 Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.24 { > iface MIXER > name 'IEC958 Playback Con Mask' > value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.25 { > iface MIXER > name 'IEC958 Playback Pro Mask' > value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.26 { > iface MIXER > name 'IEC958 Playback Default' > value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access 'read write' > type IEC958 > count 1 > } > } > control.27 { > iface MIXER > name 'IEC958 Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.28 { > iface PCM > device 3 > name ELD > value '' > comment { > access 'read volatile' > type BYTES > count 0 > } > } > control.29 { > iface CARD > name 'HDMI/DP,pcm=7 Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.30 { > iface MIXER > name 'IEC958 Playback Con Mask' > index 1 > value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.31 { > iface MIXER > name 'IEC958 Playback Pro Mask' > index 1 > value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.32 { > iface MIXER > name 'IEC958 Playback Default' > index 1 > value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access 'read write' > type IEC958 > count 1 > } > } > control.33 { > iface MIXER > name 'IEC958 Playback Switch' > index 1 > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.34 { > iface PCM > device 7 > name ELD > value '' > comment { > access 'read volatile' > type BYTES > count 0 > } > } > control.35 { > iface CARD > name 'HDMI/DP,pcm=8 Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.36 { > iface MIXER > name 'IEC958 Playback Con Mask' > index 2 > value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.37 { > iface MIXER > name 'IEC958 Playback Pro Mask' > index 2 > value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.38 { > iface MIXER > name 'IEC958 Playback Default' > index 2 > value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access 'read write' > type IEC958 > count 1 > } > } > control.39 { > iface MIXER > name 'IEC958 Playback Switch' > index 2 > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.40 { > iface PCM > device 8 > name ELD > value '' > comment { > access 'read volatile' > type BYTES > count 0 > } > } > control.41 { > iface PCM > device 3 > name 'Playback Channel Map' > value.0 0 > value.1 0 > value.2 0 > value.3 0 > value.4 0 > value.5 0 > value.6 0 > value.7 0 > comment { > access 'read write' > type INTEGER > count 8 > range '0 - 36' > } > } > control.42 { > iface PCM > device 7 > name 'Playback Channel Map' > value.0 0 > value.1 0 > value.2 0 > value.3 0 > value.4 0 > value.5 0 > value.6 0 > value.7 0 > comment { > access 'read write' > type INTEGER > count 8 > range '0 - 36' > } > } > control.43 { > iface PCM > device 8 > name 'Playback Channel Map' > value.0 0 > value.1 0 > value.2 0 > value.3 0 > value.4 0 > value.5 0 > value.6 0 > value.7 0 > comment { > access 'read write' > type INTEGER > count 8 > range '0 - 36' > } > } > control.44 { > iface MIXER > name 'PCM Playback Volume' > value.0 255 > value.1 255 > comment { > access 'read write user' > type INTEGER > count 2 > range '0 - 255' > tlv '0000000100000008ffffec1400000014' > dbmin -5100 > dbmax 0 > dbvalue.0 0 > dbvalue.1 0 > } > } > } > --endcollapse-- > > > !!All Loaded Modules > !!------------------ > > Module > drbg > ansi_cprng > ctr > ccm > cmac > rfcomm > bnep > bbswitch > arc4 > binfmt_misc > nls_iso8859_1 > snd_hda_codec_hdmi > snd_hda_codec_realtek > snd_hda_codec_generic > intel_rapl > x86_pkg_temp_thermal > intel_powerclamp > coretemp > iwlmvm > kvm_intel > mac80211 > kvm > irqbypass > crct10dif_pclmul > snd_hda_intel > crc32_pclmul > uvcvideo > snd_hda_codec > ghash_clmulni_intel > videobuf2_vmalloc > videobuf2_memops > snd_hda_core > aesni_intel > videobuf2_v4l2 > snd_hwdep > aes_x86_64 > videobuf2_core > lrw > iwlwifi > gf128mul > glue_helper > ablk_helper > snd_pcm > videodev > cryptd > snd_seq_midi > media > snd_seq_midi_event > cfg80211 > input_leds > snd_rawmidi > joydev > snd_seq > serio_raw > snd_seq_device > snd_timer > btusb > btrtl > snd > soundcore > mei_me > shpchp > mei > ideapad_laptop > sparse_keymap > hci_uart > btbcm > btqca > btintel > bluetooth > intel_lpss_acpi > wmi > intel_lpss > mac_hid > tpm_crb > acpi_pad > ip6t_REJECT > nf_reject_ipv6 > nf_log_ipv6 > xt_hl > ip6t_rt > nf_conntrack_ipv6 > nf_defrag_ipv6 > ipt_REJECT > nf_reject_ipv4 > nf_log_ipv4 > nf_log_common > xt_LOG > xt_limit > xt_tcpudp > xt_addrtype > nf_conntrack_ipv4 > nf_defrag_ipv4 > xt_conntrack > ip6table_filter > ip6_tables > nf_conntrack_netbios_ns > nf_conntrack_broadcast > nf_nat_ftp > nf_nat > nf_conntrack_ftp > nf_conntrack > iptable_filter > parport_pc > ip_tables > ppdev > x_tables > lp > parport > autofs4 > hid_generic > usbhid > i915 > i2c_algo_bit > drm_kms_helper > psmouse > syscopyarea > sysfillrect > sysimgblt > fb_sys_fops > r8169 > drm > sdhci_pci > ahci > mii > sdhci > libahci > i2c_hid > pinctrl_sunrisepoint > video > hid > pinctrl_intel > fjes > > > !!Sysfs Files > !!----------- > > /sys/class/sound/hwC0D0/init_pin_configs: > 0x12 0x90a60140 > 0x14 0x90170110 > 0x17 0x40000000 > 0x18 0x411111f0 > 0x19 0x04a11030 > 0x1a 0x411111f0 > 0x1b 0x411111f0 > 0x1d 0x40f79a2d > 0x1e 0x411111f0 > 0x21 0x04211020 > > /sys/class/sound/hwC0D0/driver_pin_configs: > > /sys/class/sound/hwC0D0/user_pin_configs: > > /sys/class/sound/hwC0D0/init_verbs: > > /sys/class/sound/hwC0D0/hints: > > /sys/class/sound/hwC0D2/init_pin_configs: > 0x05 0x18560010 > 0x06 0x18560010 > 0x07 0x18560010 > > /sys/class/sound/hwC0D2/driver_pin_configs: > > /sys/class/sound/hwC0D2/user_pin_configs: > > /sys/class/sound/hwC0D2/init_verbs: > > /sys/class/sound/hwC0D2/hints: > > > !!ALSA/HDA dmesg > !!-------------- > > [ 3.118546] iwlwifi 0000:08:00.0: loaded firmware version 16.242414.0 op_mode iwlmvm > [ 3.119214] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) > [ 3.119989] uvcvideo 1-6:1.0: Entity type for entity Realtek Extended Controls Unit was not initialized! > -- > [ 3.159964] Bluetooth: hci0: Failed to send firmware data (-38) > [ 3.161319] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC233: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker > [ 3.161321] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > [ 3.161323] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) > [ 3.161324] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 > [ 3.161325] snd_hda_codec_realtek hdaudioC0D0: inputs: > [ 3.161326] snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 > [ 3.161327] snd_hda_codec_realtek hdaudioC0D0: Internal Mic=0x12 > [ 3.170162] nvidia_uvm: Loaded the UVM driver, major device number 245 > [ 3.172370] Adding 32529404k swap on /dev/sda3. Priority:-1 extents:1 across:32529404k SSFS > [ 3.181718] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input12 > [ 3.181763] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input13 > [ 3.181802] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input14 > [ 3.181846] input: HDA Intel PCH HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input15 > [ 3.181905] input: HDA Intel PCH HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input16 > [ 3.259924] audit: type=1400 audit(1456344123.829:2): apparmor="STATUS" operation="profile_load" name="/usr/lib/lightdm/lightdm-guest-session" pid=676 comm="apparmor_parser" > >

1 0

[alsa-devel] USB Sound card with unsafe characters
by Brian Jones 26 Feb '16

26 Feb '16

I have been trying to work on a fix for Bug 1498129 - https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1498129 on Ubuntu My USB soundcard is an Asus Xonar U7 Echelon Edition <https://www.asus.com/Sound-Cards/Xonar_U7_Echelon_Edition/> which identifies itself to alsa as "Xonar U7 Echelon Ed." The card is supported by alsa and works well in analog mode but I cannot get digital output via alsa or pulseaudio. It has iec958 as device 1 rather than device 0 and I can access digital output by referring to directly to hw:x,1 but cannot access digital output via alsa. The fix I believe is to add "Xonar U7 Echelon Ed." 1 to the USB-Audio.pcm.iec958_device section of USB-Audio.conf. This does not work. As I suspected, and now have confirmed by Raymond on the bug report, this is because the alsa name for this card contains the period character '.' Is there a workaround for this or do I need to await an alsa code change?

1 0

[alsa-devel] [RFC: PATCH] ALSA: hda - clear IRQ statu twice on some Intel platforms
by libin.yang＠linux.intel.com 26 Feb '16

26 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> On some Intel platforms, we found the interrupt issue in the below scenario: 1. driver is in irq handler 2. there is another interrupt from HW after interrupt status is cleared and before exiting from interrupt handler 3. exit from the current irq handling After exiting the irq handler, it should raise another interrupt for driver to handle the new interrupt. But actually, it failed to raise the interrupt and driver will never have chance to clear the interrupt status. The patch clears the interrupt status again just before exiting for interrupt handler. This can reduce the contest dramatically. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/hda_controller.c | 3 +++ sound/pci/hda/hda_controller.h | 3 +++ sound/pci/hda/hda_intel.c | 40 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 46 insertions(+) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 37cf9ce..2ca95c7 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -963,6 +963,9 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } + if (chip->post_irq) + chip->post_irq(chip); + spin_unlock(&bus->reg_lock); return IRQ_HANDLED; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index ec63bbf..ce59997 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -146,6 +146,9 @@ struct azx { const struct firmware *fw; #endif + /* callback at the end of interrupt handler */ + void (*post_irq)(struct azx *); + /* flags */ int bdl_pos_adj; int poll_count; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4045dca..6a47d88 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1663,6 +1663,43 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, return 0; } +/* HSW, BDW, SKL and BXT need do post_irq() */ +#define INTEL_IRQ_POST(chip) \ + (((chip)->pci->vendor == PCI_VENDOR_ID_INTEL) && \ + (((chip)->pci->device == 0x0a0c) || \ + ((chip)->pci->device == 0x0c0c) || \ + ((chip)->pci->device == 0x0d0c) || \ + ((chip)->pci->device == 0x160c) || \ + ((chip)->pci->device == 0xa170) || \ + ((chip)->pci->device == 0x9d70) || \ + ((chip)->pci->device == 0x5a98))) + + +/* on some intel platforms, if there occurs an interrupt + * when irq is being handled, interrupt signal will not be raised + * after the irq handler returns. And the interrupt status may never + * be cleared. + * So let's clear all the interrupt status before return from irq handler. + * This can help to reduce the contest between the irq handler and the signal. + */ +static void intel_post_irq(struct azx *chip) +{ + struct hdac_bus *bus = azx_bus(chip); + struct hdac_stream *azx_dev; + + list_for_each_entry(azx_dev, &bus->stream_list, list) + snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); + + /* clear STATESTS */ + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + + /* clear rirb status */ + snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK); + + /* clear int status */ + snd_hdac_chip_writel(bus, INTSTS, AZX_INT_CTRL_EN | AZX_INT_ALL_STREAM); +} + static int azx_first_init(struct azx *chip) { int dev = chip->dev_index; @@ -1717,6 +1754,9 @@ static int azx_first_init(struct azx *chip) if (chip->pci->vendor == PCI_VENDOR_ID_AMD) dma_bits = 40; + if (INTEL_IRQ_POST(chip)) + chip->post_irq = intel_post_irq; + /* disable SB600 64bit support for safety */ if (chip->pci->vendor == PCI_VENDOR_ID_ATI) { struct pci_dev *p_smbus; -- 1.9.1

3 7

[alsa-devel] snd-hda-intel runtime PM fail after module reload
by Ville Syrjälä 26 Feb '16

26 Feb '16

Hi, My investigation into some sporadic i915 runtime PM failures seem to point the finger at snd-hda-intel. I just tried to play around unloding and reloading snd-hda-intel and sometimes I get snd-hda-intel loaded with runtime PM supposedly enabled, but actually the device won't runtime suspend. At which point frobbing with power/control is enough to kick it back into submission. # for i in `seq 1 3`; do sleep 1 ; grep . /sys/bus/pci/devices/0000\:00\:03.0/power/* ; done /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:117045 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:222657 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:117045 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:223666 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:117045 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:224674 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 # rmmod snd-hda-intel # modprobe snd-hda-intel # for i in `seq 1 3`; do sleep 1 ; grep . /sys/bus/pci/devices/0000\:00\:03.0/power/* ; done /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:121454 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:active /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:228560 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:122463 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:active /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:228560 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:123472 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:active /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:228560 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 # echo on > /sys/bus/pci/devices/0000:00:03.0/power/control # echo auto > /sys/bus/pci/devices/0000:00:03.0/power/control # for i in `seq 1 3`; do sleep 1 ; grep . /sys/bus/pci/devices/0000\:00\:03.0/power/* ; done /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:141459 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:231699 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:141459 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:232706 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:141459 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:233713 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 -- Ville Syrjälä Intel OTC

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[alsa-devel] [PATCH 0/4 RESEND] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 26 Feb '16

26 Feb '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Jose Abreu (4): [adv7511] Add audio support [adv7511] Update defconfigs so that audio is enabled [dwc] Add I2S HDMI audio support using custom platform driver [dwc] Update defconfigs so that I2S is enabled arch/arc/boot/dts/axs10x_mb.dtsi | 20 +- arch/arc/configs/axs101_defconfig | 4 + arch/arc/configs/axs103_defconfig | 6 + arch/arc/configs/axs103_smp_defconfig | 6 + drivers/gpu/drm/i2c/Kconfig | 8 + drivers/gpu/drm/i2c/Makefile | 2 + drivers/gpu/drm/i2c/adv7511.c | 1024 --------------------------------- drivers/gpu/drm/i2c/adv7511.h | 41 ++ drivers/gpu/drm/i2c/adv7511_audio.c | 310 ++++++++++ drivers/gpu/drm/i2c/adv7511_core.c | 1005 ++++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/Makefile | 4 +- sound/soc/dwc/designware_i2s.c | 139 ++++- sound/soc/dwc/designware_pcm.c | 264 +++++++++ sound/soc/dwc/designware_pcm.h | 21 + 16 files changed, 1813 insertions(+), 1043 deletions(-) delete mode 100644 drivers/gpu/drm/i2c/adv7511.c create mode 100644 drivers/gpu/drm/i2c/adv7511_audio.c create mode 100644 drivers/gpu/drm/i2c/adv7511_core.c create mode 100644 sound/soc/dwc/designware_pcm.c create mode 100644 sound/soc/dwc/designware_pcm.h -- 1.9.1

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[alsa-devel] [PATCH 0/15 v2][RFC] cleanup regmap write functions
by Kuninori Morimoto 26 Feb '16

26 Feb '16

Hi Mark Current regmap has many similar functions regmap_update_bits() regmap_update_bits_async() regmap_update_bits_check() regmap_update_bits_check_async() But difference is very few. And I would like to have _force_ feature on it too. So, these patches add new regmap_update_bits_base() which has _check, _async, and _force option. Above functions are now defined as macro. I used [RFC], because regmap has big effects. 1), 6), 9) adds new regmap_xxx_update_bits_base() functions and current each functions are merged into it by macro. Please review carefully especially 7), 10). I think using ~0 mask is correct for regmap_field_write() <-> regmap_field_update_bits_base() 1) - 5) for regmap_xxx() 6) - 8) for regmap_field_xxx() 9) - 11) for regmap_fields_xxx() 12) - 13) adds _force_ macro 14) remove unused regmap_write_bits() which was one of _force_ function 15) rsnd driver uses _force_ function for register access BTW, I noticed #if - #else - #endif on ${LINUX}/include/linux/regmap.h are strange. Many functions/struct/macro are defined under #ifdef CONFIG_REGMAP, but few are defined under #else. It can be trouble ? Do we really need this #if ? Kuninori Morimoto (15): 1) regmap: add regmap_update_bits_base() 2) regmap: merge regmap_update_bits() into macro 3) regmap: merge regmap_update_bits_async() into macro 4) regmap: merge regmap_update_bits_check() into macro 5) regmap: merge regmap_update_bits_check_async() into macro 6) regmap: add regmap_field_update_bits_base() 7) regmap: merge regmap_field_write() into macro 8) regmap: merge regmap_field_update_bits() into macro 9) regmap: add regmap_fields_update_bits_base() 10) regmap: merge regmap_fields_write() into macro 11) regmap: merge regmap_fields_update_bits() into macro 12) regmap: add regmap_field_force_xxx() macros 13) regmap: add regmap_fields_force_xxx() macros 14) regmpa: remove regmap_write_bits() 15) ASoC: rsnd: rsnd_write() / rsnd_bset() uses regmap _force_ function drivers/base/regmap/regmap.c | 217 ++++++++++++++++++++++++----------------------------------------------------------------------------------------------------------- include/linux/regmap.h | 94 ++++++++++++++++++++++++++++----------------------------- sound/soc/sh/rcar/gen.c | 21 ++----------- sound/soc/sh/rcar/rsnd.h | 2 -- 4 files changed, 87 insertions(+), 247 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/6][RFC] cleanup regmap write functions
by Kuninori Morimoto 26 Feb '16

26 Feb '16

Hi Mark Current regmap has many similar functions regmap_update_bits() regmap_update_bits_async() regmap_update_bits_check() regmap_update_bits_check_async() But difference is very few. And I would like to have _force_ feature on it. So, these patches add new regmap_raw_update_bits() which has _check, _async, _force option. Above functions are now defined as macro. 4), 5) adds _force_ feature. We can easily add _check, _async. I used [RFC], because regmap has many effect. BTW, I noticed #if - #else - #endif on ${LINUX}/include/linux/regmap.h are strange. Many functions/struct/macro are defined under #ifdef CONFIG_REGMAP, but few are defined under #else. It can be trouble ? Do we really need this #if ? Kuninori Morimoto (7): 1) regmap: add regmap_raw_update_bits() and merge all regmap_update_bits_xxx() 2) regmap: regmap_field_xxx() uses regmap_raw_update_bits() 3) regmap: regmap_fields_xxx() uses regmap_raw_update_bits() 4) regmap: add regmap_field_force_xxx() functions 5) regmap: add regmap_fields_force_xxx() functions 6) regmpa: remove regmap_write_bits() 7) ASoC: rsnd: rsnd_write() / rsnd_bset() uses regmap _force_ function drivers/base/regmap/regmap.c | 191 +++++++++++++++++++++++++++++------------------------------------------------------------------------------------------------------ include/linux/regmap.h | 102 ++++++++++++++++++++++++++++++---------------------------------------- sound/soc/sh/rcar/gen.c | 21 ++------------- sound/soc/sh/rcar/rsnd.h | 2 -- 4 files changed, 88 insertions(+), 228 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: mediatek: Add I03/I04 widgets and corresponding routes
by PC Liao 26 Feb '16

26 Feb '16

From: Koro Chen <koro.chen(a)mediatek.com> Add these widgets to allow another path from I2S input. Change-Id: Ib6804bbbde388830d8359245fe13bbb0f2bb36ba Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 689c51f..5de878f 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -40,6 +40,7 @@ #define AFE_CONN1 0x0024 #define AFE_CONN2 0x0028 +#define AFE_CONN3 0x002c #define AFE_CONN7 0x0460 #define AFE_CONN8 0x0464 #define AFE_HDMI_CONN0 0x0390 @@ -63,6 +64,7 @@ #define AFE_HDMI_OUT_CUR 0x0378 #define AFE_HDMI_OUT_END 0x037c +#define AFE_ADDA_TOP_CON0 0x0120 #define AFE_ADDA2_TOP_CON0 0x0600 #define AFE_HDMI_OUT_CON0 0x0370 @@ -259,6 +261,7 @@ static int mtk_afe_set_i2s(struct mtk_afe *afe, unsigned int rate) return -EINVAL; /* from external ADC */ + regmap_update_bits(afe->regmap, AFE_ADDA_TOP_CON0, 0x1, 0x1); regmap_update_bits(afe->regmap, AFE_ADDA2_TOP_CON0, 0x1, 0x1); /* set input */ @@ -903,15 +906,19 @@ static const struct snd_kcontrol_new mtk_afe_o04_mix[] = { }; static const struct snd_kcontrol_new mtk_afe_o09_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I03 Switch", AFE_CONN3, 0, 1, 0), SOC_DAPM_SINGLE_AUTODISABLE("I17 Switch", AFE_CONN7, 30, 1, 0), }; static const struct snd_kcontrol_new mtk_afe_o10_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I04 Switch", AFE_CONN3, 3, 1, 0), SOC_DAPM_SINGLE_AUTODISABLE("I18 Switch", AFE_CONN8, 0, 1, 0), }; static const struct snd_soc_dapm_widget mtk_afe_pcm_widgets[] = { /* inter-connections */ + SND_SOC_DAPM_MIXER("I03", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I04", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("I05", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("I06", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("I17", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -934,12 +941,16 @@ static const struct snd_soc_dapm_route mtk_afe_pcm_routes[] = { {"I2S Playback", NULL, "O04"}, {"VUL", NULL, "O09"}, {"VUL", NULL, "O10"}, + {"I03", NULL, "I2S Capture"}, + {"I04", NULL, "I2S Capture"}, {"I17", NULL, "I2S Capture"}, {"I18", NULL, "I2S Capture"}, { "O03", "I05 Switch", "I05" }, { "O04", "I06 Switch", "I06" }, { "O09", "I17 Switch", "I17" }, + { "O09", "I03 Switch", "I03" }, { "O10", "I18 Switch", "I18" }, + { "O10", "I04 Switch", "I04" }, }; static const struct snd_soc_dapm_route mtk_afe_hdmi_routes[] = { -- 1.7.9.5

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[alsa-devel] [patch 2/2] ASoC: max9867: silence and array overflow warning
by Dan Carpenter 26 Feb '16

26 Feb '16

Smatch complains that we might reach the end of this loop without finding what we're looking for leading to a buffer overflow. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 4fb6fc7..9fb03c6 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -156,6 +156,8 @@ static inline int get_ni_value(int mclk, int rate) if (ni_div[i].mclk >= mclk) break; } + if (i == ARRAY_SIZE(ni_div)) + return -EINVAL; switch (rate) { case 8000:

3 2

[alsa-devel] [PATCH] sound: hdmi: avoid dereferencing uninitialized 'jack' pointer
by Arnd Bergmann 26 Feb '16

26 Feb '16

When CONFIG_SND_JACK is disabled, the hdmi driver tries to create a new output jack structure and dereferences it even though the pointer is never assigned: sound/pci/hda/patch_hdmi.c: In function 'generic_hdmi_build_jack': sound/pci/hda/patch_hdmi.c:2534:30: error: 'jack' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/pci/hda/patch_hdmi.c:2526:19: note: 'jack' was declared here This uses a compile-time check to avoid calling a nonworking snd_jack_new() function, and sets an explicit NULL pointer instead. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 788d441a164c ("ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling") --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f4443b5fbf6e..c0872ad80aa0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2526,6 +2526,11 @@ static int add_hdmi_jack_kctl(struct hda_codec *codec, struct snd_jack *jack; int err; + if (!IS_ENABLED(CONFIG_SND_JACK)) { + spec->pcm_rec[pcm_idx].jack = NULL; + return 0; + } + err = snd_jack_new(codec->card, name, SND_JACK_AVOUT, &jack, true, false); if (err < 0) -- 2.7.0

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: improve headphone pop when system resumes from S3
by John Lin 26 Feb '16

26 Feb '16

When playing audio through headphone, headphone makes pop noise if system resumes from S3 to S0. We modify the sequence of writing register for avoiding pop sound. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 93e8c90..d57119d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3029,13 +3029,18 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, RT5645_PWR_BG | RT5645_PWR_VREF2, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2); + mdelay(10); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_FV1 | RT5645_PWR_FV2, RT5645_PWR_FV1 | RT5645_PWR_FV2); - if (rt5645->en_button_func && - snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) - queue_delayed_work(system_power_efficient_wq, - &rt5645->jack_detect_work, msecs_to_jiffies(0)); + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140); + msleep(40); + if (rt5645->en_button_func) + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, + msecs_to_jiffies(0)); + } break; case SND_SOC_BIAS_OFF: -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: pxa: remove incorrect do_div() call
by Arnd Bergmann 26 Feb '16

26 Feb '16

The new optimized do_div implementation (now in asm-generic/next) exposes a glitch in the brownstone audio driver by producing a compile-time warning: sound/soc/pxa/brownstone.c: In function 'brownstone_wm8994_hw_params': sound/soc/pxa/brownstone.c:67:85: warning: comparison of distinct pointer types lacks a cast sound/soc/pxa/brownstone.c:67:10125: warning: right shift count >= width of type [-Wshift-count-overflow] sound/soc/pxa/brownstone.c:67:10254: warning: passing argument 1 of '__div64_32' from incompatible pointer type [-Wincompatible-pointer-types] The driver just divides two plain integer values, so it should not use do_div to start with, but has apparently done so ever since the code was first merged. This replaces do_div with a simple division operator. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- I think this one is harmless, so we only need it in 4.5 to avoid the warning, but not backported to earlier kernels. Found today on ARM allmodconfig diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 6147e86e9b0f..416ea646c3b1 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -63,8 +63,7 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, sysclk = params_rate(params) * 512; sspa_mclk = params_rate(params) * 64; } - sspa_div = freq_out; - do_div(sspa_div, sspa_mclk); + sspa_div = freq_out / sspa_mclk; snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);

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[alsa-devel] [PATCH] ASoC: mediatek: remove soft reset and add second I2S clock
by PC Liao 26 Feb '16

26 Feb '16

To improve I2S flow, this patch removes soft reset and adds second I2S clock to use. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 689c51f..3456bfa 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -283,20 +283,13 @@ static void mtk_afe_set_i2s_enable(struct mtk_afe *afe, bool enable) regmap_read(afe->regmap, AFE_I2S_CON2, &val); if (!!(val & AFE_I2S_CON2_EN) == enable) - return; /* must skip soft reset */ - - /* I2S soft reset begin */ - regmap_update_bits(afe->regmap, AUDIO_TOP_CON1, 0x4, 0x4); + return; /* input */ regmap_update_bits(afe->regmap, AFE_I2S_CON2, 0x1, enable); /* output */ regmap_update_bits(afe->regmap, AFE_I2S_CON1, 0x1, enable); - - /* I2S soft reset end */ - udelay(1); - regmap_update_bits(afe->regmap, AUDIO_TOP_CON1, 0x4, 0); } static int mtk_afe_dais_enable_clks(struct mtk_afe *afe, @@ -365,6 +358,7 @@ static int mtk_afe_i2s_startup(struct snd_pcm_substream *substream, return 0; mtk_afe_dais_enable_clks(afe, afe->clocks[MTK_CLK_I2S1_M], NULL); + mtk_afe_dais_enable_clks(afe, afe->clocks[MTK_CLK_I2S2_M], NULL); regmap_update_bits(afe->regmap, AUDIO_TOP_CON0, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M, 0); return 0; @@ -384,6 +378,7 @@ static void mtk_afe_i2s_shutdown(struct snd_pcm_substream *substream, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M); mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S1_M], NULL); + mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S2_M], NULL); } static int mtk_afe_i2s_prepare(struct snd_pcm_substream *substream, @@ -397,6 +392,9 @@ static int mtk_afe_i2s_prepare(struct snd_pcm_substream *substream, mtk_afe_dais_set_clks(afe, afe->clocks[MTK_CLK_I2S1_M], runtime->rate * 256, NULL, 0); + mtk_afe_dais_set_clks(afe, + afe->clocks[MTK_CLK_I2S2_M], runtime->rate * 256, + NULL, 0); /* config I2S */ ret = mtk_afe_set_i2s(afe, substream->runtime->rate); if (ret) -- 1.7.9.5

2 1

[alsa-devel] [RESEND PATCH] ASoC: cs4271: add regulator consumer support
by pascal.huerst＠gmail.com 26 Feb '16

26 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> The cs4271 has three power domains: vd, vl and va. Enable them all, as long as the codec is in use. While at it, factored out the reset code into its own function. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- Documentation/devicetree/bindings/sound/cs4271.txt | 7 +++ sound/soc/codecs/cs4271.c | 69 +++++++++++++++++++--- 2 files changed, 68 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index e2cd1d7..6e699ce 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -33,12 +33,19 @@ Optional properties: Note that this is not needed in case the clocks are stable throughout the entire runtime of the codec. + - vd-supply: Digital power + - vl-supply: Logic power + - va-supply: Analog Power + Examples: codec_i2c: cs4271@10 { compatible = "cirrus,cs4271"; reg = <0x10>; reset-gpio = <&gpio 23 0>; + vd-supply = <&vdd_3v3_reg>; + vl-supply = <&vdd_3v3_reg>; + va-supply = <&vdd_3v3_reg>; }; codec_spi: cs4271@0 { diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e770ee6..0c0010b 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/tlv.h> @@ -157,6 +158,10 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) return reg == CS4271_CHIPID; } +static const char * const supply_names[] = { + "vd", "vl", "va" +}; + struct cs4271_private { unsigned int mclk; bool master; @@ -170,6 +175,7 @@ struct cs4271_private { int gpio_disable; /* enable soft reset workaround */ bool enable_soft_reset; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = { @@ -487,6 +493,20 @@ static struct snd_soc_dai_driver cs4271_dai = { .symmetric_rates = 1, }; +static int cs4271_reset(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(cs4271->gpio_nreset)) { + gpio_set_value(cs4271->gpio_nreset, 0); + mdelay(1); + gpio_set_value(cs4271->gpio_nreset, 1); + mdelay(1); + } + + return 0; +} + #ifdef CONFIG_PM static int cs4271_soc_suspend(struct snd_soc_codec *codec) { @@ -499,6 +519,9 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec) if (ret < 0) return ret; + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; } @@ -507,6 +530,16 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) int ret; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + + /* Do a proper reset after power up */ + cs4271_reset(codec); + /* Restore codec state */ ret = regcache_sync(cs4271->regmap); if (ret < 0) @@ -553,19 +586,24 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec) } #endif + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + if (cs4271plat) { amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_is_valid(cs4271->gpio_nreset)) { - /* Reset codec */ - gpio_direction_output(cs4271->gpio_nreset, 0); - mdelay(1); - gpio_set_value(cs4271->gpio_nreset, 1); - /* Give the codec time to wake up */ - mdelay(1); - } + /* Reset codec */ + cs4271_reset(codec); + + ret = regcache_sync(cs4271->regmap); + if (ret < 0) + return ret; ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, @@ -595,6 +633,9 @@ static int cs4271_codec_remove(struct snd_soc_codec *codec) /* Set codec to the reset state */ gpio_set_value(cs4271->gpio_nreset, 0); + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; }; @@ -617,6 +658,7 @@ static int cs4271_common_probe(struct device *dev, { struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; + int i, ret; cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); if (!cs4271) @@ -638,6 +680,17 @@ static int cs4271_common_probe(struct device *dev, return ret; } + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + cs4271->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + *c = cs4271; return 0; } -- 2.4.3

2 1

[alsa-devel] [PATCH 0/1] ALSA: hda - Use acpi_dev_present()
by Lukas Wunner 25 Feb '16

25 Feb '16

Hi Takashi, the acpi_dev_present() API has now landed in Linus' tree. Thus, after Linus' tree gets merged back into yours, it would be possible to use the API in the Thinkpad hda drivers as per the following patch. I've also pushed it to GitHub in case anyone prefers perusing it in a browser: https://github.com/l1k/linux/commit/a1473d726b57eaf97c4de8812c5967603068e261 An ack for this patch was kindly provided by Hui Wang with: Message-ID: <5653C291.9090607(a)canonical.com> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100962.h… Thanks, Lukas Lukas Wunner (1): ALSA: hda - Use acpi_dev_present() sound/pci/hda/thinkpad_helper.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) -- 2.1.0

3 8

Re: [alsa-devel] [PATCH] ASoC: nau8825: Modify power management and add interrupt wakeup
by John Hsu 25 Feb '16

25 Feb '16

Hi Mark, This patch has cancel and only submit enhance codec suspend and resume patch. Please review the following link for the patch. Many thanks. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-January/102924.ht… [alsa-devel] [PATCH] ASoC: nau8825: fix interrupt fails and unstable after resume On 1/8/2016 9:14 PM, Mark Brown wrote: > On Fri, Jan 08, 2016 at 03:11:22PM +0800, John Hsu wrote: > >> 1. Enhance codec suspend and resume sequence 2. Add interrupt wakeup >> function >> > > If this is two or more separate changes you should be sending two or more separate patches. You also need a better changelog which describes what the patch is supposed to do, in what way is suspend and resume enhanced for example? > > >> +/** >> + * nau8825_init_wakeup - set wakeup capability for codec >> + * >> + * @codec: codec device component >> + * >> + * After this function done, codec can support system wakeup by button. >> + */ >> +int nau8825_init_wakeup(struct snd_soc_codec *codec) { >> + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); >> + >> + device_init_wakeup(nau8825->dev, true); >> + >> + return 0; >> +} >> +EXPORT_SYMBOL_GPL(nau8825_init_wakeup); >> > > You need a bit more explanation as to what's going on here. Why is this not something the driver just does or if this should be done elsewhere why do we need the wrapper function? > > >> +int nau8825_irq_wakeup(struct snd_soc_codec *codec, int on) { >> + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); >> > > Same thing here, this looks like an unclear interface. >

2 2

[alsa-devel] Headset doesn't work for HP EliteBook 755 G2
by Derick Eddington 25 Feb '16

25 Feb '16

Hi, first time here, following suggestion [1] to report non-working default configuration, which I found how to make work. I checked using the latest stable kernel 4.4.2 and my headset jack doesn't work by default but my fix does work, on Scientific Linux 7.2 using the 4.4.2 kernel from the ELRepo repository. I also checked using Ubuntu 16.04 alpha with its 4.4.0 kernel and the problem and fix are the same. My laptop is an HP EliteBook 755 G2 from mid 2015 [2] that has a combined headset (headphone + mic) jack that doesn't work by default. The speakers work fine by default and they mute when a headset is plugged in, but the headphones don't work. The snd-hda-intel module handles my device. The codec is Realtek ALC3228. I have the latest BIOS update from HP. I got it working by passing model=hp,headset-mic to snd-hda-intel. I found this fix in an alsa-user message about a different HP laptop with the same codec [3], but I had to modify the message's fix by only including the module option for model= and omitting the other options and aliases that seem to be specific to Arch Linux. I tried using only "hp" or only "headset-mic" as the value for model= but neither worked. Attached are my outputs of alsa-info.sh from before and after I did model=hp,headset-mic. Also, there's another problem after enabling model=hp,headset-mic, which I found how to fix: There is a very annoying squeaky background noise in the headphones that's independent of adjusting the volume level but it does stop when the headphones are muted. I tried a different headset and it's the same. I found how to suppress this by using the white_noise_fix.py script from [4] as described at [5,6], which apparently does some ioctl verbs to adjust the amp gain. I thought this might be something that should be fixed/adjusted in the kernel module(s) that handle this device, so that the white_noise_fix.py isn't needed. If so, maybe it's worth mentioning that when my laptop resumes from sleep, the background noise comes back, so I have to run white_noise_fix.py again. I'm willing to help investigate my hardware more or try other configurations or distros if requested. If it's worth mentioning, I found someone else with the same laptop who has the same non-working headset problem, see [7]. I'd like to help get this fixed in Ubuntu 16.04 before its release. If the proper fix is in the kernel ALSA, I'm willing to go through Ubuntu's process for asking them to incorporate it in 16.04. Apologies if I've misdirected this report. Thanks for your work on ALSA! -- Derick [1] linux-doc/sound/alsa/ALSA-Configuration.txt [2] Product number L9H51UT#ABA [3] https://sourceforge.net/p/alsa/mailman/message/33087916/ [4] http://xps13-9333.appspot.com/root/usr/local/bin/white_noise_fix.py [5] http://xps13-9333.appspot.com/#background_noise [6] http://askubuntu.com/questions/524344/noise-from-headphones [7] http://askubuntu.com/questions/714920/audio-card-alc3228-issue

2 3

[alsa-devel] [PATCH] hda-emu: add acpi_dev_present
by David Henningsson 25 Feb '16

25 Feb '16

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/acpi/acpi.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/acpi/acpi.h b/include/acpi/acpi.h index d7d6cb9..9fd9c86 100644 --- a/include/acpi/acpi.h +++ b/include/acpi/acpi.h @@ -33,4 +33,6 @@ acpi_get_devices(const char *HID, return user_function(NULL, 0, context, NULL); } +static inline bool acpi_dev_present(const char *hid) { return true; } + #endif /* __ACPI_H__ */ -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Fixup speaker pass-through control for nid 0x14 on ALC225
by David Henningsson 25 Feb '16

25 Feb '16

On one of the machines we enable, we found that the actual speaker volume did not always correspond to the volume set in alsamixer. This patch fixes that problem. This patch was orginally written by Kailang @ Realtek, I've rebased it to fit sound git master. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1549660 Co-Authored-By: Kailang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 72fa58d..7fded69 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3801,6 +3801,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, static void alc_headset_mode_default(struct hda_codec *codec) { + static struct coef_fw coef0225[] = { + UPDATE_COEF(0x45, 0x3f<<10, 0x34<<10), + {} + }; static struct coef_fw coef0255[] = { WRITE_COEF(0x45, 0xc089), WRITE_COEF(0x45, 0xc489), @@ -3842,6 +3846,9 @@ static void alc_headset_mode_default(struct hda_codec *codec) }; switch (codec->core.vendor_id) { + case 0x10ec0225: + alc_process_coef_fw(codec, coef0225); + break; case 0x10ec0255: case 0x10ec0256: alc_process_coef_fw(codec, coef0255); @@ -4750,6 +4757,7 @@ enum { ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, + ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5375,6 +5383,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable pass-through path for FRONT 14h */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x36 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5646,10 +5665,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x21, 0x03211020} static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701a0}), - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701b0}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fixup speaker pass-through control for nid 0x14 on ALC225
by David Henningsson 25 Feb '16

25 Feb '16

On one of the machines we enable, we found that the actual speaker volume did not always correspond to the volume set in alsamixer. This patch fixes that problem. This patch was orginally written by Kailang @ Realtek, I've rebased it to fit sound git master. BugLink: https://bugs.launchpad.net/bugs/1549660 Co-Authored-By: Kailang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index efd4980..ab353b0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3801,6 +3801,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, static void alc_headset_mode_default(struct hda_codec *codec) { + static struct coef_fw coef0225[] = { + UPDATE_COEF(0x45, 0x3f<<10, 0x34<<10), + {} + }; static struct coef_fw coef0255[] = { WRITE_COEF(0x45, 0xc089), WRITE_COEF(0x45, 0xc489), @@ -3842,6 +3846,9 @@ static void alc_headset_mode_default(struct hda_codec *codec) }; switch (codec->core.vendor_id) { + case 0x10ec0225: + alc_process_coef_fw(codec, coef0225); + break; case 0x10ec0255: case 0x10ec0256: alc_process_coef_fw(codec, coef0255); @@ -4749,6 +4756,7 @@ enum { ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, + ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5368,6 +5376,17 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_lenovo_line2_mic_hotkey, }, + [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable pass-through path for FRONT 14h */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x36 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5638,10 +5657,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x21, 0x03211020} static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701a0}), - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701b0}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, -- 1.9.1

2 2

[alsa-devel] [patch 1/2] ASoC: max9867: signedness bug in max9867_dai_hw_params()
by Dan Carpenter 25 Feb '16

25 Feb '16

"value" needs to be signed for the error handling to work. Fixes: 805d132dcbbd ('ASoC: Add max9867 codec driver') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index ab7a94d..4fb6fc7 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -180,7 +180,8 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); - unsigned int ni_h, ni_l, value; + unsigned int ni_h, ni_l; + int value; value = get_ni_value(max9867->sysclk, params_rate(params)); if (value < 0)

2 1

[alsa-devel] [PATCH] ALSA: hda - Fixing background noise on Dell Inspiron 3162
by Kai-Heng Feng 25 Feb '16

25 Feb '16

After login to the desktop on Dell Inspiron 3162, there's a very loud background noise comes from the builtin speaker. The noise does not go away even if the speaker is muted. The noise disappears after using the aamix fixup. Codec: Realtek ALC3234 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0255 Subsystem Id: 0x10280725 Revision Id: 0x100002 No Modem Function Group found BugLink: http://bugs.launchpad.net/bugs/1549620 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index efd4980..72fa58d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4749,6 +4749,7 @@ enum { ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, + ALC255_FIXUP_DELL_SPK_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5368,6 +5369,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_lenovo_line2_mic_hotkey, }, + [ALC255_FIXUP_DELL_SPK_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5410,6 +5417,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.5.0

2 1

[alsa-devel] [PATCH 0/2 v2] ALSA: oxfw: implement retries for scs1x at transaction failure
by Takashi Sakamoto 25 Feb '16

25 Feb '16

Hi, This patchset updates my former post below. Changes are just adding my sign (I forgot it...) and improvements of commit messages. [alsa-devel] [PATCH 0/2] ALSA: oxfw: implement retries for scs1x at transaction failure http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104579.h… In thread of the post, I got some comments about work to extend ALSA rawmidi core. I've investigated it in a few days ago and realized that it may not so light work, mainly because of a lack of unified state management[0]. In this development cycle, I'd like to give it up and use my time for the other work within my plan for the cycle. Please just apply these two patches. [0] In short, the state management is understandable for developers who can understand it, and I'm not. Regards Takashi Sakamoto (2): ALSA: oxfw: retry MIDI transferring for scs1x at transaction failure ALSA: oxfw: discontinue MIDI substream for scs1x at transaction failure sound/firewire/oxfw/oxfw-scs1x.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) -- 2.5.0

2 4

[alsa-devel] [PATCH] ASoC: arizona: Make logging of FLL calculations clearer
by Richard Fitzgerald 25 Feb '16

25 Feb '16

The debug logging of FLL calculations was confusing. Values were printed in hex without indicating this by a leading 0x, and despite these normally being required in decimal. Also where the register value isn't the actual value (it s a power-of-two or 0 means 1, 1 means 2, ...) it was unclear whether the actual or register value was shown. This patch changes the log print so that all mathematical values are shown in decimal, all register values are shown in hex with a 0x prefix, and where the register value isn't the actual integer value the decimal integer value is shown in () after the hex register value. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 913cfa8..92d22a0 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2149,11 +2149,12 @@ static int arizona_calc_fll(struct arizona_fll *fll, return -EINVAL; } - arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", + arizona_fll_dbg(fll, "N=%d THETA=%d LAMBDA=%d\n", cfg->n, cfg->theta, cfg->lambda); - arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", - cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); - arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain); + arizona_fll_dbg(fll, "FRATIO=0x%x(%d) OUTDIV=%d REFCLK_DIV=0x%x(%d)\n", + cfg->fratio, ratio, cfg->outdiv, + cfg->refdiv, 1 << cfg->refdiv); + arizona_fll_dbg(fll, "GAIN=0x%x(%d)\n", cfg->gain, 1 << cfg->gain); return 0; -- 1.9.1

2 4

Re: [alsa-devel] Fwd: Lenovo Y700-17ISK notebook - multiple issues
by Takashi Iwai 24 Feb '16

24 Feb '16

On Wed, 24 Feb 2016 19:00:36 +0100, Aljosa wrote: > > Hi Takashi, > thank you very much for your reply. > Notebook subwoofer doesn't work at all (it does not produce any sound). OK, could you give more details? Which kernel did you test? Did you test any model option? Also give alsa-info.sh output (run it with --no-upload option and attach the output). This contains the h/w details. Last but not least, don't drop Cc to ML. thanks, Takashi > Regards, Aljosa > > > > On 2016-02-24 16:35, Takashi Iwai wrote: > > On Tue, 23 Feb 2016 20:37:21 +0100, > > Aljosa wrote: > >> > >> Hello, > >> advised by one of the kernel developers - I'm reporting you the > >> problem > >> nr.2 from the below link. > > > > Could you describe what are the exact issues, instead of pointing some > > URL? > > > > > > thanks, > > > > Takashi >

1 0

[alsa-devel] Codec "virtual multiplexer" representation
by Petr Kulhavy 24 Feb '16

24 Feb '16

Hi, I'm writing an ASoC driver for the TAS5721 amplifier. This 2.1 amplifier provides a complex configuration of the audio channels (3), PWMs(4) and output(4) mappings. However only few (about 6) configurations really make sense. I'm trying to model the output configurations in ALSA. In order to avoid a complicated set of switches and multiplexers I would like to add just one multiplexer like control which selects one of these 6 output configurations and a callback function which sets the proper registers accordingly (there are 5 affected registers). Is there a way to create such a "virtual multiplexer" in ALSA? Thanks Petr

2 2

[alsa-devel] Fwd: Lenovo Y700-17ISK notebook - multiple issues
by Aljosa 24 Feb '16

24 Feb '16

Hello, advised by one of the kernel developers - I'm reporting you the problem nr.2 from the below link. Best regards, Aljosa -------- Original Message -------- Subject: Lenovo Y700-17ISK notebook - multiple issues Date: 2016-02-21 11:18 From: Aljosa <aljosa(a)cromalternativemoney.org> To: jkosina(a)suse.com Reply-To: aljosa(a)cromalternativemoney.org Hi, please excuse me if I contact you in this way - I found your email by googling "Lenovo" inside the Linux kernel mailing list archive. Few days ago I purchased a new Lenovo Y700-17ISK notebook, and after installing "current Ubuntu 16.04 daily build" I'm facing some issues which I think are kernel related: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1547934 Consequently, I would like to inform kernel developers about these problems. If you are not the right person to contact on this subject, please route this e-mail to whomever can help me or let me know who to contact - thank you very much in advance. Best regards, Aljosa PS. I have just tried freshly released kernel v4.5-rc5-wily (http://kernel.ubuntu.com/~kernel-ppa/mainline/?C=N;O=D), and with it my wireless connection now works, but there are still plenty problematic lines inside DMESG output.

2 1

[alsa-devel] [BUG] [REGRESSION] [BISECTED] -rc1 breaks audio over HDMI for i915
by Martin Kepplinger 24 Feb '16

24 Feb '16

The following change: 788d441 ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling breaks audio over HDMI on my snd_hda_intel laptop. It is the first bad commit. This was merged for -rc1 and isn't fixed until now, so I got nervous. There are no errors in the log that stand out. I'm no sound or alsa developer, and just reverting it isn't easy since there is stuff depending on this. I'm happy to test possible fixes and hope v4.5 not to ship with such a big regression ;)

4 23

[alsa-devel] ASoC: simultaneous start of input/output channels
by Jonah Petri 24 Feb '16

24 Feb '16

Hello! Our application is very low-latency full duplex audio on a Freescale SoC. ALSA is configured with 2 64-frame buffers, with DSP happening on a realtime user thread. With small buffer sizes, we're obviously very tight on timing, and I've run into an issue which seems architectural in nature, and seemingly is worthy of consulting the list. What we're seeing is a large temporal latency between the DMA completion IRQ for the audio input DMA and audio output DMA. This delay is on the order of 200 microseconds, but also somewhat variable (ok, "large" is relative here, but it matters to us!). In the ideal case, these two interrupts would be perfectly aligned in time, allowing us to run DSP for the full buffer's duration. However, since the interrupts are offset by about 20% of a buffer, we effectively lose that time for our DSP. And, since the timing is variable, and we're in a realtime application, this produces scared engineers. Working backwards from the hardware drivers, I see that snd_soc_dai_ops's trigger callback is used to start and stop the DMA. The fundamental "start" action comes down to a single register write which can easily start both input and output streams in the same write. However, it seems that the snd_soc_dai_ops does not provide for simultaneous start of both input and output streams. Tracing back through the kernel interface and into user space, it seems that starting both the input and output streams necessitates two separate calls from userspace, which could explain the delay, and some of the variability in timing. My question is: * Did I miss some API or technique which could allow simultaneous start of full duplex streams? * If not, I would certainly appreciate advice on where we could fit this into the existing architecture! We would be glad to do the kernel work to make this change, as this capability seems generally useful! Thanks much for any help, Jonah

2 2

[alsa-devel] [asoc:topic/intel 18/22] sound/soc/intel/boards/skl_nau88l25_max98357a.c:25:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory
by kbuild test robot 24 Feb '16

24 Feb '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 8be3838af0e992846d4725fae152b818560c1068 commit: 0ab338ff33762d50e5005d8d28b30b25f23aa934 [18/22] ASoC: Intel: boards: Enable HDMI and DP on SKL nau88l25_max98357 machine config: i386-allmodconfig (attached as .config) reproduce: git checkout 0ab338ff33762d50e5005d8d28b30b25f23aa934 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): >> sound/soc/intel/boards/skl_nau88l25_max98357a.c:25:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory compilation terminated. vim +25 sound/soc/intel/boards/skl_nau88l25_max98357a.c 19 #include <sound/core.h> 20 #include <sound/jack.h> 21 #include <sound/pcm.h> 22 #include <sound/pcm_params.h> 23 #include <sound/soc.h> 24 #include "../../codecs/nau8825.h" > 25 #include "../../codecs/hdac_hdmi.h" 26 27 #define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" 28 #define SKL_MAXIM_CODEC_DAI "HiFi" --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:topic/intel 16/22] sound/soc/intel/boards/skl_rt286.c:29:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory
by kbuild test robot 24 Feb '16

24 Feb '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 8be3838af0e992846d4725fae152b818560c1068 commit: 23905cd1ca48cbc852a670a3dbbfc5e7d694580f [16/22] ASoC: Intel: boards: Enable HDMI and DP on skl_rt286 machine config: i386-allmodconfig (attached as .config) reproduce: git checkout 23905cd1ca48cbc852a670a3dbbfc5e7d694580f # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): >> sound/soc/intel/boards/skl_rt286.c:29:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory compilation terminated. vim +29 sound/soc/intel/boards/skl_rt286.c 23 #include <sound/core.h> 24 #include <sound/pcm.h> 25 #include <sound/soc.h> 26 #include <sound/jack.h> 27 #include <sound/pcm_params.h> 28 #include "../../codecs/rt286.h" > 29 #include "../../codecs/hdac_hdmi.h" 30 31 static struct snd_soc_jack skylake_headset; 32 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [asoc:topic/intel 17/22] sound/soc/intel/boards/skl_nau88l25_ssm4567.c:29:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory
by kbuild test robot 24 Feb '16

24 Feb '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 8be3838af0e992846d4725fae152b818560c1068 commit: bc5f6ac9a66f90f8e16e568ebd463cbe44f46b81 [17/22] ASoC: Intel: boards: Enable HDMI and DP on nau88l2 machine config: i386-allmodconfig (attached as .config) reproduce: git checkout bc5f6ac9a66f90f8e16e568ebd463cbe44f46b81 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): >> sound/soc/intel/boards/skl_nau88l25_ssm4567.c:29:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory compilation terminated. vim +29 sound/soc/intel/boards/skl_nau88l25_ssm4567.c 23 #include <sound/core.h> 24 #include <sound/pcm.h> 25 #include <sound/soc.h> 26 #include <sound/jack.h> 27 #include <sound/pcm_params.h> 28 #include "../../codecs/nau8825.h" > 29 #include "../../codecs/hdac_hdmi.h" 30 31 #define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" 32 #define SKL_SSM_CODEC_DAI "ssm4567-hifi" --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 1/3] ASoC: rt298: enable IRQ for jack detection
by Bard Liao 24 Feb '16

24 Feb '16

There are some registers needed for enabling rt298 IRQ missed on current driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 6 ++++++ sound/soc/codecs/rt298.h | 8 ++++++++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 30c6de6..e67ef5f 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1225,6 +1225,12 @@ static int rt298_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt298->regmap, RT298_WIND_FILTER_CTRL, 0x0082, 0x0082); regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); + + regmap_write(rt298->regmap, RT298_UNSOLICITED_INLINE_CMD, 0x81); + regmap_write(rt298->regmap, RT298_UNSOLICITED_HP_OUT, 0x82); + regmap_write(rt298->regmap, RT298_UNSOLICITED_MIC1, 0x84); + regmap_update_bits(rt298->regmap, RT298_IRQ_FLAG_CTRL, 0x2, 0x2); + rt298->is_hp_in = -1; if (rt298->i2c->irq) { diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index 31da162..d66f884 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -34,6 +34,7 @@ #define RT298_HP_OUT 0x21 #define RT298_MIXER_IN1 0x22 #define RT298_MIXER_IN2 0x23 +#define RT298_INLINE_CMD 0x55 #define RT298_SET_PIN_SFT 6 #define RT298_SET_PIN_ENABLE 0x40 @@ -124,6 +125,12 @@ VERB_CMD(AC_VERB_SET_COEF_INDEX, RT298_VENDOR_REGISTERS, 0) #define RT298_PROC_COEF\ VERB_CMD(AC_VERB_SET_PROC_COEF, RT298_VENDOR_REGISTERS, 0) +#define RT298_UNSOLICITED_INLINE_CMD\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_INLINE_CMD, 0) +#define RT298_UNSOLICITED_HP_OUT\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_HP_OUT, 0) +#define RT298_UNSOLICITED_MIC1\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_MIC1, 0) /* Index registers */ #define RT298_A_BIAS_CTRL1 0x01 @@ -148,6 +155,7 @@ #define RT298_DEPOP_CTRL2 0x67 #define RT298_DEPOP_CTRL3 0x68 #define RT298_DEPOP_CTRL4 0x69 +#define RT298_IRQ_FLAG_CTRL 0x7c /* SPDIF (0x06) */ #define RT298_SPDIF_SEL_SFT 0 -- 1.8.1.1.439.g50a6b54

2 7

[alsa-devel] [PATCH 0/7] ASoC: Intel: boards: Enabled HDMI/DP and update WOV on SKL machines
by Vinod Koul 24 Feb '16

24 Feb '16

This series adds the multiple port support for HDMI and DP on respective SKL machine drivers. Also the WoV sink is removed as that needs to be associated with PCM device and not a DAPM Sink Jeeja KP (5): ASoC: Intel: boards: Enable HDMI and DP on skl_rt286 machine ASoC: Intel: boards: Enable HDMI and DP on nau88l2 machine ASoC: Intel: boards: Remove WoV sink in skl_rt286 ASoC: Intel: boards: Remove WoV sink in nau88125_ssm4567 machine ASoC: Intel: boards: Remove WoV sink in nau88125_max98357a machine Sathya Prakash M R (1): ASoC: Intel: boards: Add IV feedback route for nau88l25_ssm4567 machine Sathyanarayana Nujella (1): ASoC: Intel: boards: Enable HDMI and DP on SKL nau88l25_max98357 machine sound/soc/intel/Kconfig | 3 + sound/soc/intel/boards/skl_nau88l25_max98357a.c | 122 +++++++++++++++++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 145 +++++++++++++++++++++--- sound/soc/intel/boards/skl_rt286.c | 113 ++++++++++++++++-- 4 files changed, 339 insertions(+), 44 deletions(-) -- 1.9.1

2 14

[alsa-devel] [PATCH RESEND 1/2] ASoC: rt298: enable IRQ for jack detection
by Bard Liao 24 Feb '16

24 Feb '16

There are some registers needed for enabling rt298 IRQ missed on current driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 6 ++++++ sound/soc/codecs/rt298.h | 8 ++++++++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 30c6de6..e67ef5f 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1225,6 +1225,12 @@ static int rt298_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt298->regmap, RT298_WIND_FILTER_CTRL, 0x0082, 0x0082); regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); + + regmap_write(rt298->regmap, RT298_UNSOLICITED_INLINE_CMD, 0x81); + regmap_write(rt298->regmap, RT298_UNSOLICITED_HP_OUT, 0x82); + regmap_write(rt298->regmap, RT298_UNSOLICITED_MIC1, 0x84); + regmap_update_bits(rt298->regmap, RT298_IRQ_FLAG_CTRL, 0x2, 0x2); + rt298->is_hp_in = -1; if (rt298->i2c->irq) { diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index 31da162..d66f884 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -34,6 +34,7 @@ #define RT298_HP_OUT 0x21 #define RT298_MIXER_IN1 0x22 #define RT298_MIXER_IN2 0x23 +#define RT298_INLINE_CMD 0x55 #define RT298_SET_PIN_SFT 6 #define RT298_SET_PIN_ENABLE 0x40 @@ -124,6 +125,12 @@ VERB_CMD(AC_VERB_SET_COEF_INDEX, RT298_VENDOR_REGISTERS, 0) #define RT298_PROC_COEF\ VERB_CMD(AC_VERB_SET_PROC_COEF, RT298_VENDOR_REGISTERS, 0) +#define RT298_UNSOLICITED_INLINE_CMD\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_INLINE_CMD, 0) +#define RT298_UNSOLICITED_HP_OUT\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_HP_OUT, 0) +#define RT298_UNSOLICITED_MIC1\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_MIC1, 0) /* Index registers */ #define RT298_A_BIAS_CTRL1 0x01 @@ -148,6 +155,7 @@ #define RT298_DEPOP_CTRL2 0x67 #define RT298_DEPOP_CTRL3 0x68 #define RT298_DEPOP_CTRL4 0x69 +#define RT298_IRQ_FLAG_CTRL 0x7c /* SPDIF (0x06) */ #define RT298_SPDIF_SEL_SFT 0 -- 1.8.1.1.439.g50a6b54

1 1

[alsa-devel] [PATCH 2/2] ASoC: max9867: fix type of variable containing error codes
by Andrzej Hajda 24 Feb '16

24 Feb '16

value variable can contain error values and is compared with zero. Its type must be signed. The problem has been detected using coccinelle script scripts/coccinelle/tests/unsigned_lesser_than_zero.cocci Signed-off-by: Andrzej Hajda <a.hajda(a)samsung.com> --- sound/soc/codecs/max9867.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index ab7a94d..4fb6fc7 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -180,7 +180,8 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); - unsigned int ni_h, ni_l, value; + unsigned int ni_h, ni_l; + int value; value = get_ni_value(max9867->sysclk, params_rate(params)); if (value < 0) -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: max9867: Use params_width instead of open-coded
by Axel Lin 24 Feb '16

24 Feb '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max9867.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index ab7a94d..e3ee391 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -206,10 +206,9 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream, unsigned long int bclk_rate, pclk_bclk_ratio; int bclk_value; - bclk_rate = params_rate(params) * 2 * - snd_pcm_format_width(params_format(params)); + bclk_rate = params_rate(params) * 2 * params_width(params); pclk_bclk_ratio = max9867->pclk/bclk_rate; - switch (snd_pcm_format_width(params_format(params))) { + switch (params_width(params)) { case 8: case 16: switch (pclk_bclk_ratio) { -- 2.1.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: adau17x1: Minor fixes and DT support
by Andreas Irestål 24 Feb '16

24 Feb '16

This set of patches fixes some minor typos and adds DT support for the adau17x1 codec drivers. I've only been able to test the 1761 part on real hardware since I don't have any adau1781 hardware at the moment. Currently, quite a lot of options are provided through the platform_data struct, so it made sense parsing the DT and create a platform_data struct from these parameters not to break anything. I've peeked at a few other drivers and the TI tlv320aic31xx driver works this way as well. Andreas Irestål (4): ASoC: adau17x1: Fix incorrect BCLK ratio definitions ASoc: adau17x1: Correct typos in file headers ASoC: adau17x1: Add basic DT support for adau17x1 ASoC: adau17x1: Add option to provide platform data via DT .../devicetree/bindings/sound/adi,adau17x1.txt | 54 +++++++++ include/dt-bindings/sound/adau17x1.h | 14 +++ include/linux/platform_data/adau17x1.h | 2 +- sound/soc/codecs/adau1761-i2c.c | 14 ++- sound/soc/codecs/adau1761-spi.c | 14 ++- sound/soc/codecs/adau1761.c | 133 ++++++++++++++++++++- sound/soc/codecs/adau1781-i2c.c | 10 ++ sound/soc/codecs/adau1781-spi.c | 10 ++ sound/soc/codecs/adau1781.c | 59 ++++++++- sound/soc/codecs/adau17x1.h | 6 +- 10 files changed, 308 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau17x1.txt create mode 100644 include/dt-bindings/sound/adau17x1.h -- 2.1.4

3 8

[alsa-devel] [PATCH v2 0/5] Minor fixes and DT support
by Andreas Irestål 24 Feb '16

24 Feb '16

This set of patches fixes some minor typos and adds DT support for the adau17x1 codec drivers. I've only been able to test the 1761 part on real hardware since I don't have any adau1781 hardware at the moment. Currently, quite a lot of options are provided through the platform_data struct, so it made sense parsing the DT and create a platform_data struct from these parameters not to break anything. Changes since v1: - Added vendor prefix "adi," to DT properties - DT parsing moved to main probe function Andreas Irestål (5): ASoC: adau17x1: Fix incorrect BCLK ratio definitions ASoC: adau17x1: Correct typos in file headers ASoC: adau17x1: Add adau17x1 bindings document ASoC: adau17x1: Add basic DT support for adau17x1 ASoC: adau17x1: Support platform data via DT .../devicetree/bindings/sound/adi,adau17x1.txt | 55 +++++++++ include/dt-bindings/sound/adau17x1.h | 14 +++ include/linux/platform_data/adau17x1.h | 2 +- sound/soc/codecs/adau1761-i2c.c | 14 ++- sound/soc/codecs/adau1761-spi.c | 14 ++- sound/soc/codecs/adau1761.c | 129 ++++++++++++++++++++- sound/soc/codecs/adau1781-i2c.c | 10 ++ sound/soc/codecs/adau1781-spi.c | 10 ++ sound/soc/codecs/adau1781.c | 50 +++++++- sound/soc/codecs/adau17x1.h | 6 +- 10 files changed, 296 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau17x1.txt create mode 100644 include/dt-bindings/sound/adau17x1.h -- 2.1.4

4 12

[alsa-devel] [PATCH v8 1/2] ASoC: hdac_hdmi: Fix to wait for D3 before powering off codec
by Subhransu S. Prusty 24 Feb '16

24 Feb '16

Powering off codec immediately after sending D3 verb may not set the node to D3 state. So wait for a confirmation response before shutting down codec. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 2445b43..aa953a5 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1486,6 +1486,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; + unsigned long timeout; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1495,10 +1496,19 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) return 0; /* Power down afg */ - if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) + if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) { snd_hdac_codec_write(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + /* Wait till power state is set to D3 */ + timeout = jiffies + msecs_to_jiffies(1000); + while (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3) + && time_before(jiffies, timeout)) { + + msleep(50); + } + } + err = snd_hdac_display_power(bus, false); if (err < 0) { dev_err(bus->dev, "Cannot turn on display power on i915\n"); -- 1.9.1

3 4

[alsa-devel] [PATCH v9 1/2] ASoC: hdac_hdmi: Fix to wait for D3 before powering off codec
by Subhransu S. Prusty 24 Feb '16

24 Feb '16

Powering off codec immediately after sending D3 verb may not set the node to D3 state. So wait for a confirmation response before shutting down codec. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 2445b43..aa953a5 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1486,6 +1486,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; + unsigned long timeout; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1495,10 +1496,19 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) return 0; /* Power down afg */ - if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) + if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) { snd_hdac_codec_write(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + /* Wait till power state is set to D3 */ + timeout = jiffies + msecs_to_jiffies(1000); + while (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3) + && time_before(jiffies, timeout)) { + + msleep(50); + } + } + err = snd_hdac_display_power(bus, false); if (err < 0) { dev_err(bus->dev, "Cannot turn on display power on i915\n"); -- 1.9.1

2 2

[alsa-devel] [PATCH v2] ASoC: wm9713: fix regmap free path
by Robert Jarzmik 24 Feb '16

24 Feb '16

In the conversion to regmap, I assumed that the devm_() variant could be used in the soc probe function. As a mater of fact with the current code the regmap is freed twice because of the devm_() call: (mutex_lock) from [<c01f6624>] (debugfs_remove_recursive+0x50/0x1d0) (debugfs_remove_recursive) from [<c02bf800>] (regmap_debugfs_exit+0x1c/0xd4) (regmap_debugfs_exit) from [<c02ba1f8>] (regmap_exit+0x28/0xc8) (regmap_exit) from [<c02aa258>] (release_nodes+0x18c/0x204) (release_nodes) from [<c02a278c>] (device_release+0x18/0x90) (device_release) from [<c0239030>] (kobject_release+0x90/0x1bc) (kobject_release) from [<c0395c94>] (wm9713_soc_remove+0x1c/0x24) (wm9713_soc_remove) from [<c0384884>] (soc_remove_component+0x50/0x7c) (soc_remove_component) from [<c0386c28>] (soc_remove_dai_links+0x118/0x228) (soc_remove_dai_links) from [<c038721c>] (snd_soc_register_card+0x4e4/0xdd4) (snd_soc_register_card) from [<c0393c54>] (devm_snd_soc_register_card+0x34/0x70) Fix this by replacing the devm_regmap initialization code with the non devm_() variant. Fixes: 700dadfefc3d ASoC: wm9713: convert to regmap Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: changed the fix into removing the devm_() variant --- sound/soc/codecs/wm9713.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4e522302bb8a..edd270f6d8e5 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1212,7 +1212,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); - regmap = devm_regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); + regmap = regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); if (IS_ERR(regmap)) { snd_soc_free_ac97_codec(wm9713->ac97); return PTR_ERR(regmap); -- 2.1.4

4 3

[alsa-devel] [PATCH] ASoC: rockchip: i2s: add default values for registers
by Sugar Zhang 24 Feb '16

24 Feb '16

this patch add default values for registers according description from TRM. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 6561c4c..a0b0942 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -440,11 +440,21 @@ static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) } } +static const struct reg_default rockchip_i2s_reg_defaults[] = { + {0x00, 0x0000000f}, + {0x04, 0x0000000f}, + {0x08, 0x00071f1f}, + {0x10, 0x001f0000}, + {0x14, 0x01f00000}, +}; + static const struct regmap_config rockchip_i2s_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, .max_register = I2S_RXDR, + .reg_defaults = rockchip_i2s_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rockchip_i2s_reg_defaults), .writeable_reg = rockchip_i2s_wr_reg, .readable_reg = rockchip_i2s_rd_reg, .volatile_reg = rockchip_i2s_volatile_reg, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: compress: Add DSP private metadata interface
by Tim Sheridan 23 Feb '16

23 Feb '16

Provides an area to communicate with a DSP to implement support for weird and wonderful features of DSP hardware that aren't supported in the compress offload API. Signed-off-by: Tim Sheridan <tim.sheridan(a)imgtec.com> --- include/uapi/sound/compress_offload.h | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index e00d8cb..65ac445 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 3) /** * struct snd_compressed_buffer - compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -150,6 +150,8 @@ struct snd_compr_metadata { * SNDRV_COMPRESS_SET_PARAMS: Set codec and stream parameters * Note: only codec params can be changed runtime and stream params cant be * SNDRV_COMPRESS_GET_PARAMS: Query codec params + * SNDRV_COMPRESS_SET_DSP_PRIV: Set private DSP metadata + * SNDRV_COMPRESS_GET_DSP_PRIV: Get private DSP metadata * SNDRV_COMPRESS_TSTAMP: get the current timestamp value * SNDRV_COMPRESS_AVAIL: get the current buffer avail value. * This also queries the tstamp properties @@ -171,6 +173,8 @@ struct snd_compr_metadata { struct snd_compr_metadata) #define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ struct snd_compr_metadata) +#define SNDRV_COMPRESS_SET_DSP_PRIV _IOW('C', 0x16, struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_DSP_PRIV _IOWR('C', 0x17, struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) -- 1.7.1

4 6

[alsa-devel] [RFC PATCH] ALSA: hda - hdmi eld control created based on pcm
by libin.yang＠linux.intel.com 23 Feb '16

23 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> eld control is created based on pcm now. When monitor is connected, eld control will be bound to pin automatically. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 74 ++++++++++++++++++++++++++++++---------------- 1 file changed, 48 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 541986f..d77509a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -86,7 +86,6 @@ struct hdmi_spec_per_pin { struct hdmi_eld sink_eld; struct mutex lock; struct delayed_work work; - struct snd_kcontrol *eld_ctl; struct hdmi_pcm *pcm; /* pointer to spec->pcm_rec[n] dynamically*/ int pcm_idx; /* which pcm is attached. -1 means no pcm is attached */ int repoll_count; @@ -136,6 +135,7 @@ struct hdmi_ops { struct hdmi_pcm { struct hda_pcm *pcm; struct snd_jack *jack; + struct snd_kcontrol *eld_ctl; }; struct hdmi_spec { @@ -471,17 +471,22 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; - int pin_idx; + int pcm_idx; uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; - pin_idx = kcontrol->private_value; - per_pin = get_pin(spec, pin_idx); + pcm_idx = kcontrol->private_value; + mutex_lock(&spec->pcm_lock); + per_pin = pcm_idx_to_pin(spec, pcm_idx); + if (!per_pin) { + /* no pin is bound to the pcm */ + uinfo->count = 0; + mutex_unlock(&spec->pcm_lock); + return 0; + } eld = &per_pin->sink_eld; - - mutex_lock(&per_pin->lock); uinfo->count = eld->eld_valid ? eld->eld_size : 0; - mutex_unlock(&per_pin->lock); + mutex_unlock(&spec->pcm_lock); return 0; } @@ -493,16 +498,23 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; - int pin_idx; + int pcm_idx; - pin_idx = kcontrol->private_value; - per_pin = get_pin(spec, pin_idx); + pcm_idx = kcontrol->private_value; + mutex_lock(&spec->pcm_lock); + per_pin = pcm_idx_to_pin(spec, pcm_idx); + if (!per_pin) { + /* no pin is bound to the pcm */ + memset(ucontrol->value.bytes.data, 0, + ARRAY_SIZE(ucontrol->value.bytes.data)); + mutex_unlock(&spec->pcm_lock); + return 0; + } eld = &per_pin->sink_eld; - mutex_lock(&per_pin->lock); if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || eld->eld_size > ELD_MAX_SIZE) { - mutex_unlock(&per_pin->lock); + mutex_unlock(&spec->pcm_lock); snd_BUG(); return -EINVAL; } @@ -512,7 +524,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, if (eld->eld_valid) memcpy(ucontrol->value.bytes.data, eld->eld_buffer, eld->eld_size); - mutex_unlock(&per_pin->lock); + mutex_unlock(&spec->pcm_lock); return 0; } @@ -525,7 +537,7 @@ static struct snd_kcontrol_new eld_bytes_ctl = { .get = hdmi_eld_ctl_get, }; -static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, +static int hdmi_create_eld_ctl(struct hda_codec *codec, int pcm_idx, int device) { struct snd_kcontrol *kctl; @@ -535,14 +547,17 @@ static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, kctl = snd_ctl_new1(&eld_bytes_ctl, codec); if (!kctl) return -ENOMEM; - kctl->private_value = pin_idx; + kctl->private_value = pcm_idx; kctl->id.device = device; - err = snd_hda_ctl_add(codec, get_pin(spec, pin_idx)->pin_nid, kctl); + /* no pin nid is associated with the kctl now + * tbd: associate pin nid to eld ctl later + */ + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; - get_pin(spec, pin_idx)->eld_ctl = kctl; + get_hdmi_pcm(spec, pcm_idx)->eld_ctl = kctl; return 0; } @@ -1841,7 +1856,10 @@ static void update_eld(struct hda_codec *codec, struct hdmi_spec *spec = codec->spec; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; + int pcm_idx = -1; + /* for monitor disconnection, save pcm_idx firstly */ + pcm_idx = per_pin->pcm_idx; if (spec->dyn_pcm_assign) { if (eld->eld_valid) { hdmi_attach_hda_pcm(spec, per_pin); @@ -1851,6 +1869,11 @@ static void update_eld(struct hda_codec *codec, hdmi_detach_hda_pcm(spec, per_pin); } } + /* if pcm_idx == -1, it means this is in monitor connection event + * we can get the correct pcm_idx now. + */ + if (pcm_idx == -1) + pcm_idx = per_pin->pcm_idx; if (eld->eld_valid) snd_hdmi_show_eld(codec, &eld->info); @@ -1884,11 +1907,11 @@ static void update_eld(struct hda_codec *codec, hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } - if (eld_changed) + if (eld_changed && pcm_idx >= 0) snd_ctl_notify(codec->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, - &per_pin->eld_ctl->id); + &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id); } /* update ELD and jack state via HD-audio verbs */ @@ -2629,17 +2652,16 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pcm_idx); - } - - for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); /* add control for ELD Bytes */ - err = hdmi_create_eld_ctl(codec, pin_idx, - get_pcm_rec(spec, pin_idx)->device); - + err = hdmi_create_eld_ctl(codec, pcm_idx, + get_pcm_rec(spec, pcm_idx)->device); if (err < 0) return err; + } + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hdmi_present_sense(per_pin, 0); } -- 1.9.1

3 3

[alsa-devel] [alsa-lib][PATCH 0/6] control: add APIs for control element set
by Takashi Sakamoto 23 Feb '16

23 Feb '16

This patchset adds some APIs for control element set, according to design of ALSA control core. Additionally, for the APIs, some API documentation is improved. In ALSA control core, each sound card can have control elements. The elements are managed according to below design. * element set * A set of elements with the same attribute (i.e. name, get/put operations). Some element sets can be added to a sound card. [0] * element * An element can be identified by userspace applications. * channel * An element has channels which can be changed by userspace applications or hardware device itself. * value * Each channel has its own value. The place to kept the value depends on design of each device driver (i.e. in hardware register or kernel space). There're two ways to identify the element; by identification number(numid), or by a combination of name of the element set and index from the first element in the element set [1]. Both of device drivers in kernel land and ALSA control applications in user land can add the element set. When userspace applications add them, ALSA control core works as Inter Process Communication mechanism between event-listening applications and usual control applications. Such element is referred as 'user-defined control elements' in alsa-lib. Typically, device drivers in kernel land adds one element set for a single element. So users and userspace developers are hard to realize that an existence of the element set. There are more items to describe the design. But here I focus on what relates to this patchset. Well, current alsa-lib implementation includes small bugs, some misleadings, a lack of feature related to user-defined control elements. This patchset is my attempt to improves them. [0] In kernel, the element set is represented by 'struct snd_kcontrol'. Some developers call it as 'kernel control'. The others call it as 'control group'. Here, I suggest 'element set', because from userspace the 'struct snd_kcontrol' is not visible and any elements are just visible. Furthermore, I'd like to use unique name against Linux control groups. [1] Actually, when identifying an element according to name/index pair, the other members of 'struct snd_ctl_elem_id' such as iface, device and subdevice are required. They're required to seek element set, then index is used to find an element. Regards Takashi Sakamoto (6): control: rename variables so that it represents the number of channels which an element has control: improve API documentation for functions to add elements control: improve API documentation for functions to set/get channel values for an element control: rename variables so that it represents a channel in element control: add a function to add an element of bytes type control: add functions to add element set include/control.h | 55 +++-- src/control/control.c | 589 ++++++++++++++++++++++++++++++++++++++------------ 2 files changed, 487 insertions(+), 157 deletions(-) -- 2.5.0

3 10

[alsa-devel] Compute Stick baytrail audio problem
by Giacomo Comes 23 Feb '16

23 Feb '16

Hi, I have recently bought a Intel Compute Stick and I am having trouble with the audio. Intel provides a custom Ubuntu release that support audio, but I want to run the linux distribution of my choice on it (openSUSE 13.2 for the moment). In order to have HDMI audio support it is necessary to compile a kernel with this patch (https://raw.githubusercontent.com/01org/baytrailaudio/master/hdmi_audio_201…) provided by Intel for kernel 3.16.7 (https://01.org/ubuntu-hdmi) I compiled the openSUSE kernel using the default openSUSE .config plus the following: CONFIG_SUPPORT_HDMI=y CONFIG_SND_SOC=m CONFIG_SND_SOC_INTEL_SST=m CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=m I booted the new kernel and aplay -l gives this output: **** List of PLAYBACK Hardware Devices **** card 0: IntelHDMI [IntelHDMI], device 0: IntelHDMI [IntelHDMI] Subdevices: 0/1 Subdevice #0: subdevice #0 but when I run: aplay /usr/share/sounds/alsa/test.wav the audio plays for .2 seconds and then nothing more. Sometime it will play later a little bit more and stop again. At some point these messages appears: underrun!!! (at least 29325.946 ms long) underrun!!! (at least 1535.889 ms long) underrun!!! (at least 3196.213 ms long) I have looked around on google for a hint on how to fix this issue but I found nothing. Any idea where to look for a solution? Attached there are the dmesg and lsmod output for ubuntu with the working audio and openSUSE 13.2 plus patch without the working audio. I can provide any other information you need. Thanks for any help. Giacomo

2 2

[alsa-devel] [PATCH v1 0/2] patches to add support for rockchip rt5640 card
by Sugar Zhang 23 Feb '16

23 Feb '16

add default values for i2s registers from TRM. rt5640: add master clock handing. Sugar Zhang (2): ASoC: rockchip: i2s: add default values for registers ASoC: rt5640: add master clock handling for rt5640 Documentation/devicetree/bindings/sound/rt5640.txt | 3 +++ sound/soc/codecs/rt5640.c | 31 ++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2 ++ sound/soc/rockchip/rockchip_i2s.c | 10 +++++++ 4 files changed, 46 insertions(+) -- 1.9.1

3 6

[alsa-devel] Compress offload: DSP buffer timestamp passing
by Tim Sheridan 22 Feb '16

22 Feb '16

Hi, I've been working on the Imagination Technologies AXD (audio DSP) compress offload driver. One of the features of the DSP is scheduled playback of audio using timestamps (that are obtained by an application from userspace). A current problem with the compress offload API is that there's no "blessed" way to get these timestamps from tinycompress through to our compress offload driver. Currently, I've added a SNDRV_COMPRESS_ENCODER_PTS value to the sndrv_compress_encoder enum, and exposed this with an API in tinycompress, handling this in my driver to pass it to the DSP. Does this sound like a reasonable approach to go about adding compress offload API support for this? Or is there some other part of the ALSA API which would be more appropriate to use instead of this? Cheers, Tim -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

2 2

[alsa-devel] [PATCH] ALSA: hda - Apply clock gate workaround to Skylake, too
by Takashi Iwai 22 Feb '16

22 Feb '16

Some Skylake machines show the codec probe errors in certain situations, e.g. HP Z240 desktop fails to probe the onboard Realtek codec at reloading the snd-hda-intel module like: snd_hda_intel 0000:00:1f.3: spurious response 0x200:0x2, last cmd=0x000000 snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to polling mode: lastcmd=0x000f0000 snd_hda_intel 0000:00:1f.3: No response from codec, disabling MSI: last cmd=0x000f0000 snd_hda_intel 0000:00:1f.3: Codec #0 probe error; disabling it... hdaudio hdaudioC0D2: no AFG or MFG node found snd_hda_intel 0000:00:1f.3: no codecs initialized Also, HP G470 G3 suffers from the similar problem, as reported in bugzilla below. On this machine, the codec probe error appears even at a fresh boot. As Libin suggested, the same workaround used for Broxton in the commit [6639484ddaf6: ALSA: hda - disable dynamic clock gating on Broxton before reset] can be applied for Skylake in order to fix this problem. The Intel HW team also confirmed that this is needed for SKL. This patch makes the workaround applied to both SKL and BXT platforms. The referred macros are moved and one superfluous macro (IS_BROXTON()) is another one (IS_BXT()) as well. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=112731 Suggested-by: Libin Yang <libin.yang(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> # v4.4+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ce6b97f31390..e5240cb3749f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -363,7 +363,10 @@ enum { ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c)) -#define IS_BROXTON(pci) ((pci)->device == 0x5a98) +#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) +#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) +#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) +#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -540,13 +543,13 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, true); - if (IS_BROXTON(pci)) { + if (IS_SKL_PLUS(pci)) { pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val); val = val & ~INTEL_HDA_CGCTL_MISCBDCGE; pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); } azx_init_chip(chip, full_reset); - if (IS_BROXTON(pci)) { + if (IS_SKL_PLUS(pci)) { pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val); val = val | INTEL_HDA_CGCTL_MISCBDCGE; pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); @@ -555,7 +558,7 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) snd_hdac_set_codec_wakeup(bus, false); /* reduce dma latency to avoid noise */ - if (IS_BROXTON(pci)) + if (IS_BXT(pci)) bxt_reduce_dma_latency(chip); } @@ -977,11 +980,6 @@ static int azx_resume(struct device *dev) /* put codec down to D3 at hibernation for Intel SKL+; * otherwise BIOS may still access the codec and screw up the driver */ -#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) -#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) -#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) -#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) - static int azx_freeze_noirq(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); -- 2.7.1

1 0

[alsa-devel] [PATCH] ALSA: mts64: use new parport device model
by Sudip Mukherjee 22 Feb '16

22 Feb '16

Modify mts64 driver to use the new parallel port device model. The advantage of using the device model is that the driver gets binded to the hardware, we get the feature of hotplug, we can bind/unbind the driver at runtime. The changes are in the way the driver gets registered with the parallel port subsystem and the temporary device to probe mts64 card is removed and mts64_probe() is used in the probe callback. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- exactly same changes as done in portman2x4. BTW, there is one portman2x4 available on ebay. I was bidding for it but someone is more desperate than I am and outbidding me everytime :( sound/drivers/mts64.c | 95 +++++++++++++++++++++++---------------------------- 1 file changed, 43 insertions(+), 52 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 2a008a9..c76bd87 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -65,8 +65,6 @@ struct mts64 { struct snd_card *card; struct snd_rawmidi *rmidi; struct pardevice *pardev; - int pardev_claimed; - int open_count; int current_midi_output_port; int current_midi_input_port; @@ -850,30 +848,6 @@ __out: spin_unlock(&mts->lock); } -static int snd_mts64_probe_port(struct parport *p) -{ - struct pardevice *pardev; - int res; - - pardev = parport_register_device(p, DRIVER_NAME, - NULL, NULL, NULL, - 0, NULL); - if (!pardev) - return -EIO; - - if (parport_claim(pardev)) { - parport_unregister_device(pardev); - return -EIO; - } - - res = mts64_probe(p); - - parport_release(pardev); - parport_unregister_device(pardev); - - return res; -} - static void snd_mts64_attach(struct parport *p) { struct platform_device *device; @@ -907,10 +881,20 @@ static void snd_mts64_detach(struct parport *p) /* nothing to do here */ } +static int snd_mts64_dev_probe(struct pardevice *pardev) +{ + if (strcmp(pardev->name, DRIVER_NAME)) + return -ENODEV; + + return 0; +} + static struct parport_driver mts64_parport_driver = { - .name = "mts64", - .attach = snd_mts64_attach, - .detach = snd_mts64_detach + .name = "mts64", + .probe = snd_mts64_dev_probe, + .match_port = snd_mts64_attach, + .detach = snd_mts64_detach, + .devmodel = true, }; /********************************************************************* @@ -922,8 +906,7 @@ static void snd_mts64_card_private_free(struct snd_card *card) struct pardevice *pardev = mts->pardev; if (pardev) { - if (mts->pardev_claimed) - parport_release(pardev); + parport_release(pardev); parport_unregister_device(pardev); } @@ -938,6 +921,12 @@ static int snd_mts64_probe(struct platform_device *pdev) struct snd_card *card = NULL; struct mts64 *mts = NULL; int err; + struct pardev_cb mts64_cb = { + .preempt = NULL, + .wakeup = NULL, + .irq_func = snd_mts64_interrupt, /* ISR */ + .flags = PARPORT_DEV_EXCL, /* flags */ + }; p = platform_get_drvdata(pdev); platform_set_drvdata(pdev, NULL); @@ -946,8 +935,6 @@ static int snd_mts64_probe(struct platform_device *pdev) return -ENODEV; if (!enable[dev]) return -ENOENT; - if ((err = snd_mts64_probe_port(p)) < 0) - return err; err = snd_card_new(&pdev->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -960,23 +947,32 @@ static int snd_mts64_probe(struct platform_device *pdev) sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); - pardev = parport_register_device(p, /* port */ - DRIVER_NAME, /* name */ - NULL, /* preempt */ - NULL, /* wakeup */ - snd_mts64_interrupt, /* ISR */ - PARPORT_DEV_EXCL, /* flags */ - (void *)card); /* private */ - if (pardev == NULL) { + mts64_cb.private = card; /* private */ + pardev = parport_register_dev_model(p, /* port */ + DRIVER_NAME, /* name */ + &mts64_cb, /* callbacks */ + pdev->id); /* device number */ + if (!pardev) { snd_printd("Cannot register pardevice\n"); err = -EIO; goto __err; } + /* claim parport */ + if (parport_claim(pardev)) { + snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); + err = -EIO; + goto free_pardev; + } + err = mts64_probe(p); + if (err) { + err = -EIO; + goto release_pardev; + } + if ((err = snd_mts64_create(card, pardev, &mts)) < 0) { snd_printd("Cannot create main component\n"); - parport_unregister_device(pardev); - goto __err; + goto release_pardev; } card->private_data = mts; card->private_free = snd_mts64_card_private_free; @@ -986,14 +982,6 @@ static int snd_mts64_probe(struct platform_device *pdev) goto __err; } - /* claim parport */ - if (parport_claim(pardev)) { - snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); - err = -EIO; - goto __err; - } - mts->pardev_claimed = 1; - /* init device */ if ((err = mts64_device_init(p)) < 0) goto __err; @@ -1009,6 +997,10 @@ static int snd_mts64_probe(struct platform_device *pdev) snd_printk(KERN_INFO "ESI Miditerminal 4140 on 0x%lx\n", p->base); return 0; +release_pardev: + parport_release(pardev); +free_pardev: + parport_unregister_device(pardev); __err: snd_card_free(card); return err; @@ -1024,7 +1016,6 @@ static int snd_mts64_remove(struct platform_device *pdev) return 0; } - static struct platform_driver snd_mts64_driver = { .probe = snd_mts64_probe, .remove = snd_mts64_remove, -- 1.9.1

2 1

[alsa-devel] [PATCH] control: keep consistent usage of assertion
by Takashi Sakamoto 22 Feb '16

22 Feb '16

In most codes for control APIs, assertions are used to check given arguments from applications. If the arguments are not expected, the application is forced to abort. When NDEBUG is defined in the beginning of the codes, no checks are performed. Although, in snd_ctl_elem_set_bytes(), assertion is used with condition statement. There's an intention to check the arguments regardless of NDEBUG. However, this is not the same fashion in the codes. This commit applies the same fashion to the function, to keep consistent usage of assertion. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 328920d..70d168d 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2627,10 +2627,7 @@ void snd_ctl_elem_value_set_byte(snd_ctl_elem_value_t *obj, unsigned int idx, un void snd_ctl_elem_set_bytes(snd_ctl_elem_value_t *obj, void *data, size_t size) { assert(obj); - if (size >= ARRAY_SIZE(obj->value.bytes.data)) { - assert(0); - return; - } + assert(size < ARRAY_SIZE(obj->value.bytes.data)); memcpy(obj->value.bytes.data, data, size); } -- 2.5.0

2 1

[alsa-devel] [alsa-lib][PATCH] control: correct assertion in snd_ctl_elem_set_bytes()
by Takashi Sakamoto 22 Feb '16

22 Feb '16

The assert(3) aborts running process when expression arguments is false (zero). Although, the snd_ctl_elem_set_bytes() has return statement after assert(3). It has meaningless. This commit corrects the statements. Fixes: 7893ea238d5a('Added mode argument to open functions where it was missing. First part of CTL documentation') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 328920d..70d168d 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2627,10 +2627,7 @@ void snd_ctl_elem_value_set_byte(snd_ctl_elem_value_t *obj, unsigned int idx, un void snd_ctl_elem_set_bytes(snd_ctl_elem_value_t *obj, void *data, size_t size) { assert(obj); - if (size >= ARRAY_SIZE(obj->value.bytes.data)) { - assert(0); - return; - } + assert(size < ARRAY_SIZE(obj->value.bytes.data)); memcpy(obj->value.bytes.data, data, size); } -- 2.5.0

2 5

[alsa-devel] [PATCH] ASoC: mediatek: Add machine driver for ALC5650 codec
by PC Liao 22 Feb '16

22 Feb '16

From: Koro Chen <koro.chen(a)mediatek.com> This is the DPCM based machine driver with rt5650 Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 15 ++ sound/soc/mediatek/Kconfig | 10 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8173-rt5650.c | 236 ++++++++++++++++++++ 4 files changed, 262 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650.txt create mode 100644 sound/soc/mediatek/mt8173-rt5650.c diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt new file mode 100644 index 0000000..fe5a5ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -0,0 +1,15 @@ +MT8173 with RT5650 CODECS + +Required properties: +- compatible : "mediatek,mt8173-rt5650" +- mediatek,audio-codec: the phandles of rt5650 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650"; + mediatek,audio-codec = <&rt5650>; + mediatek,platform = <&afe>; + }; + diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 9769676..6c88087 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -17,6 +17,16 @@ config SND_SOC_MT8173_MAX98090 Select Y if you have such device. If unsure select "N". +config SND_SOC_MT8173_RT5650 + tristate "ASoC Audio driver for MT8173 with RT5650 codec" + depends on SND_SOC_MEDIATEK && I2C + select SND_SOC_RT5645 + help + This adds ASoC driver for Mediatek MT8173 boards + with the RT5650 audio codec. + Select Y if you have such device. + If unsure select "N". + config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" depends on SND_SOC_MEDIATEK && I2C diff --git a/sound/soc/mediatek/Makefile b/sound/soc/mediatek/Makefile index 75effbe..bcdcb06 100644 --- a/sound/soc/mediatek/Makefile +++ b/sound/soc/mediatek/Makefile @@ -3,3 +3,4 @@ obj-$(CONFIG_SND_SOC_MEDIATEK) += mtk-afe-pcm.o # Machine support obj-$(CONFIG_SND_SOC_MT8173_MAX98090) += mt8173-max98090.o obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o +obj-$(CONFIG_SND_SOC_MT8173_RT5650) += mt8173-rt5650.o diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c new file mode 100644 index 0000000..bb09bb1 --- /dev/null +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -0,0 +1,236 @@ +/* + * mt8173-rt5650.c -- MT8173 machine driver with RT5650 codecs + * + * Copyright (c) 2016 MediaTek Inc. + * Author: Koro Chen <koro.chen(a)mediatek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * 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/gpio.h> +#include <linux/of_gpio.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5645.h" + +#define MCLK_FOR_CODECS 12288000 + +static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Headset Mic", NULL, "micbias1"}, + {"Headset Mic", NULL, "micbias2"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, +}; + +static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + + /* pll from mclk 12.288M */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, + params_rate(params) * 512); + if (ret) + return ret; + + /* sysclk from pll */ + ret = snd_soc_dai_set_sysclk(codec_dai, 1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + return 0; +} + +static struct snd_soc_ops mt8173_rt5650_ops = { + .hw_params = mt8173_rt5650_hw_params, +}; + +static struct snd_soc_jack mt8173_rt5650_jack; + +static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + int ret; + + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + /* enable jack detection */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &mt8173_rt5650_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "Can't new Headset Jack %d\n", ret); + return ret; + } + + return rt5645_set_jack_detect(codec, + &mt8173_rt5650_jack, + &mt8173_rt5650_jack, + &mt8173_rt5650_jack); +} + +static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { + { + .dai_name = "rt5645-aif1", + }, +}; + +enum { + DAI_LINK_PLAYBACK, + DAI_LINK_CAPTURE, + DAI_LINK_CODEC_I2S, +}; + +/* Digital audio interface glue - connects codec <---> CPU */ +static struct snd_soc_dai_link mt8173_rt5650_dais[] = { + /* Front End DAI links */ + [DAI_LINK_PLAYBACK] = { + .name = "rt5650 Playback", + .stream_name = "rt5650 Playback", + .cpu_dai_name = "DL1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, + [DAI_LINK_CAPTURE] = { + .name = "rt5650 Capture", + .stream_name = "rt5650 Capture", + .cpu_dai_name = "VUL", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_capture = 1, + }, + /* Back End DAI links */ + [DAI_LINK_CODEC_I2S] = { + .name = "Codec", + .cpu_dai_name = "I2S", + .no_pcm = 1, + .codecs = mt8173_rt5650_codecs, + .num_codecs = 1, + .init = mt8173_rt5650_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &mt8173_rt5650_ops, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +static struct snd_soc_card mt8173_rt5650_card = { + .name = "mtk-rt5650", + .owner = THIS_MODULE, + .dai_link = mt8173_rt5650_dais, + .num_links = ARRAY_SIZE(mt8173_rt5650_dais), + .controls = mt8173_rt5650_controls, + .num_controls = ARRAY_SIZE(mt8173_rt5650_controls), + .dapm_widgets = mt8173_rt5650_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8173_rt5650_widgets), + .dapm_routes = mt8173_rt5650_routes, + .num_dapm_routes = ARRAY_SIZE(mt8173_rt5650_routes), +}; + +static int mt8173_rt5650_dev_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mt8173_rt5650_card; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_dais[i].platform_name) + continue; + mt8173_rt5650_dais[i].platform_of_node = platform_node; + } + + mt8173_rt5650_codecs[0].of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); + if (!mt8173_rt5650_codecs[0].of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + return ret; +} + +static const struct of_device_id mt8173_rt5650_dt_match[] = { + { .compatible = "mediatek,mt8173-rt5650", }, + { } +}; +MODULE_DEVICE_TABLE(of, mt8173_rt5650_dt_match); + +static struct platform_driver mt8173_rt5650_driver = { + .driver = { + .name = "mtk-rt5650", + .of_match_table = mt8173_rt5650_dt_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, + .probe = mt8173_rt5650_dev_probe, +}; + +module_platform_driver(mt8173_rt5650_driver); + +/* Module information */ +MODULE_DESCRIPTION("MT8173 RT5650 SoC machine driver"); +MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:mtk-rt5650"); + -- 1.7.9.5

2 3

[alsa-devel] [PATCH 0/3] ALSA: hda: Move chmap helpers to core
by Subhransu S. Prusty 22 Feb '16

22 Feb '16

The HDMI channel map helper APIs and control registration can be reused by ASoC skylake HDMI drivers. A common chmap object is created, channel map control callback handlers, helper APIs and data structures are moved to the core. Driver object store an instance of this chmap object to register chmap controls and program channel allocation for HDMI infoframe. Driver registers ops for specific operations. Subhransu S. Prusty (3): ALSA: hda - Create common chmap object ALSA: hda - Move chmap support helpers/structures to core ALSA: hda - Use the helpers from HDA core include/sound/hdaudio.h | 1 + include/sound/hdmi_chmap.h | 84 +++++ sound/hda/Makefile | 2 +- sound/hda/hdmi_chmap.c | 763 +++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_eld.c | 31 +- sound/pci/hda/patch_hdmi.c | 794 ++++----------------------------------------- 6 files changed, 923 insertions(+), 752 deletions(-) create mode 100644 include/sound/hdmi_chmap.h create mode 100644 sound/hda/hdmi_chmap.c -- 1.9.1

2 7

[alsa-devel] [PATCH] ASoC: rockchip: add bindings for rk3399 i2s
by jianqun.xu 22 Feb '16

22 Feb '16

From: Jianqun Xu <jay.xu(a)rock-chips.com> Add devicetree bindings for i2s controller found on rk3399 processors from rockchip. It's helpful to add full set of compatible strings for serials of Rockchip SoCs (rk3066, rk3188, rk3288, rk3399). Acked-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Jianqun Xu <jay.xu(a)rock-chips.com> --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 1 + sound/soc/rockchip/rockchip_i2s.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index b7f3a93..6e86d8a 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -9,6 +9,7 @@ Required properties: - "rockchip,rk3066-i2s": for rk3066 - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 + - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the I2S interrupt. diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 6561c4c..531a6d2 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -575,6 +575,9 @@ static int rockchip_i2s_remove(struct platform_device *pdev) static const struct of_device_id rockchip_i2s_match[] = { { .compatible = "rockchip,rk3066-i2s", }, + { .compatible = "rockchip,rk3188-i2s", }, + { .compatible = "rockchip,rk3288-i2s", }, + { .compatible = "rockchip,rk3399-i2s", }, {}, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: topology: fix build warning
by Sudip Mukherjee 22 Feb '16

22 Feb '16

We are getting build warning about: sound/soc/soc-topology.c:249:31: warning: 'get_dobj_type' defined but not used commit 64527e8a3529 has removed the only caller of get_dobj_type() but missed to remove the function which is now unused. And after removing get_dobj_type() we do not have any use of get_dobj_mixer_type(). Fixes: 64527e8a3529 ("ASoC: topology: Add FE DAIs dynamically") CC: Mengdong Lin <mengdong.lin(a)linux.intel.com> Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/soc-topology.c | 45 --------------------------------------------- 1 file changed, 45 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0eb01e8..07f61a6 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -223,51 +223,6 @@ static int get_widget_id(int tplg_type) return -EINVAL; } -static enum snd_soc_dobj_type get_dobj_mixer_type( - struct snd_soc_tplg_ctl_hdr *control_hdr) -{ - if (control_hdr == NULL) - return SND_SOC_DOBJ_NONE; - - switch (control_hdr->ops.info) { - case SND_SOC_TPLG_CTL_VOLSW: - case SND_SOC_TPLG_CTL_VOLSW_SX: - case SND_SOC_TPLG_CTL_VOLSW_XR_SX: - case SND_SOC_TPLG_CTL_RANGE: - case SND_SOC_TPLG_CTL_STROBE: - return SND_SOC_DOBJ_MIXER; - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_CTL_ENUM_VALUE: - return SND_SOC_DOBJ_ENUM; - case SND_SOC_TPLG_CTL_BYTES: - return SND_SOC_DOBJ_BYTES; - default: - return SND_SOC_DOBJ_NONE; - } -} - -static enum snd_soc_dobj_type get_dobj_type(struct snd_soc_tplg_hdr *hdr, - struct snd_soc_tplg_ctl_hdr *control_hdr) -{ - switch (hdr->type) { - case SND_SOC_TPLG_TYPE_MIXER: - return get_dobj_mixer_type(control_hdr); - case SND_SOC_TPLG_TYPE_DAPM_GRAPH: - case SND_SOC_TPLG_TYPE_MANIFEST: - return SND_SOC_DOBJ_NONE; - case SND_SOC_TPLG_TYPE_DAPM_WIDGET: - return SND_SOC_DOBJ_WIDGET; - case SND_SOC_TPLG_TYPE_DAI_LINK: - return SND_SOC_DOBJ_DAI_LINK; - case SND_SOC_TPLG_TYPE_PCM: - return SND_SOC_DOBJ_PCM; - case SND_SOC_TPLG_TYPE_CODEC_LINK: - return SND_SOC_DOBJ_CODEC_LINK; - default: - return SND_SOC_DOBJ_NONE; - } -} - static inline void soc_bind_err(struct soc_tplg *tplg, struct snd_soc_tplg_ctl_hdr *hdr, int index) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fix memory leak
by Sudip Mukherjee 22 Feb '16

22 Feb '16

If dai_link is already bound then we just returned and leaked rtd and rtd->codec_dais which were allocated by soc_new_pcm_runtime(). We do not need this newly allocated rtd to check if dai_link is already binded. Lets check first if it is already binded before allocating this memory. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 790ee2b..d2e62b15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -986,16 +986,16 @@ static int soc_bind_dai_link(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); - rtd = soc_new_pcm_runtime(card, dai_link); - if (!rtd) - return -ENOMEM; - if (soc_is_dai_link_bound(card, dai_link)) { dev_dbg(card->dev, "ASoC: dai link %s already bound\n", dai_link->name); return 0; } + rtd = soc_new_pcm_runtime(card, dai_link); + if (!rtd) + return -ENOMEM; + cpu_dai_component.name = dai_link->cpu_name; cpu_dai_component.of_node = dai_link->cpu_of_node; cpu_dai_component.dai_name = dai_link->cpu_dai_name; -- 1.9.1

2 1

[alsa-devel] [PATCH v2]ASoC: topology: Fix setting of stream rates, rate_min and rate_max
by mengdong.lin＠linux.intel.com 22 Feb '16

22 Feb '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Directly set a stream's rates, rate_min and rate_max from the topology info. Also define set_stream_info to wrap setting of the stream info. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0eb01e8..726cea7 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1575,6 +1575,18 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } +static void set_stream_info(struct snd_soc_pcm_stream *stream, + struct snd_soc_tplg_stream_caps *caps) +{ + stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + stream->channels_min = caps->channels_min; + stream->channels_max = caps->channels_max; + stream->rates = caps->rates; + stream->rate_min = caps->rate_min; + stream->rate_max = caps->rate_max; + stream->formats = caps->formats; +} + static int soc_tplg_dai_create(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *pcm) { @@ -1593,25 +1605,13 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - - stream->stream_name = kstrdup(caps->name, GFP_KERNEL); - stream->channels_min = caps->channels_min; - stream->channels_max = caps->channels_max; - stream->rates = snd_pcm_rate_range_to_bits(caps->rate_min, - caps->rate_max); - stream->formats = caps->formats; + set_stream_info(stream, caps); } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - - stream->stream_name = kstrdup(caps->name, GFP_KERNEL); - stream->channels_min = caps->channels_min; - stream->channels_max = caps->channels_max; - stream->rates = snd_pcm_rate_range_to_bits(caps->rate_min, - caps->rate_max); - stream->formats = caps->formats; + set_stream_info(stream, caps); } /* pass control to component driver for optional further init */ -- 2.5.0

2 1

[alsa-devel] [RFC 0/4] Add support for DAI link addition dynamically
by Vaibhav Agarwal 22 Feb '16

22 Feb '16

Following patches are based on for-next branch from broonie tree. First 2 patches include changes in existing soc, core framework to prepare for adding support for dynamic DAI link addition/ removal Patch 3 & 4 contains actual changes to enable dynamic DAI link support NOTE: Currently, the code is tested on Pandaboad ES revB3 for playback usecase. Vaibhav Agarwal (4): ASoc: Use ref_count for soc DAI & component alsa: add locked variant for snd_ctl_remove_id ASoC: Enable dynamic DAIlink insertion & removal ASoC: Change soc-card codec_conf array to a list include/sound/control.h | 1 + include/sound/soc-dai.h | 1 + include/sound/soc-dapm.h | 7 +- include/sound/soc-dpcm.h | 1 + include/sound/soc.h | 18 ++- sound/core/control.c | 23 +++ sound/soc/Kconfig | 4 + sound/soc/soc-core.c | 359 ++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/soc-dapm.c | 105 +++++++++++--- sound/soc/soc-pcm.c | 25 ++++ 10 files changed, 501 insertions(+), 43 deletions(-) -- 2.1.4

5 22

Re: [alsa-devel] [PATCH] ASoC: topology: Fix setting of stream rates, rate_min and rate_max
by Mark Brown 22 Feb '16

22 Feb '16

On Tue, Feb 16, 2016 at 02:27:32PM +0800, kbuild test robot wrote: > sound/soc/soc-topology.c: In function 'set_stream_info': > >> sound/soc/soc-topology.c:1588:1: warning: no return statement in function returning non-void [-Wreturn-type] > } > ^ This is a real bug that 0day has identified.

2 1

[alsa-devel] [PATCH v7 0/2] ASoC: Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 22 Feb '16

22 Feb '16

From: Marcus Cooper <codekipper(a)gmail.com> For now just the SPDIF transmitter has been tested on a Mele A2000(A10) and a Itead Ibox(A20). In order for this patch set to be functional we require an audio clock patch which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK --- Changes since v6: - removed legacy dapm components that are no longer needed. - changed dt-bindings patch subject styling to match that of the sound subsystem Changes since v5: - fixed warning that had sneaked in. Changes since v4: - corrected indentation issues - removed old macros - removed unnecessary enabling of apb clock Changes since v3: - removed code required for spdif capture - added disable/enable of apb clk to suspend/resume. - removed unnecessary dt checks in probe - fixed error conditions in probe/resume - modified pm_runtime functionality and confirmed that driver worked when PM was disabled. - removed .owner from platform driver - renamed bindings file Changes since v2: - removed sunxi-machine-spdif and replaced with simple audio card - removed untested compatibilities from device tree documentation - added pm_runtime and remove shutdown - removed pll2 as it's the parent of the spdif clock - rename clk to spdif_clk - removed enabling of mclk output - removed interrupts status being cleared as it's not being used - use default fifo settings for now - fixed alignments for wrapped lines Changes since v1: - Moved sunxi-machine-spdif.c to separate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into separate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. Marcus Cooper (2): ASoC: sunxi :add sun4i SPDIF dt bindings ASoC: sunxi: Add support for the SPDIF block .../bindings/sound/sunxi,sun4i-spdif.txt | 39 ++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 542 +++++++++++++++++++++ 4 files changed, 590 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c -- 2.7.1

3 4

[alsa-devel] [PATCH 1/3] ASoC: Intel: Skylake: fix pointer scaling
by Vinod Koul 22 Feb '16

22 Feb '16

From: Alan <gnomes(a)lxorguk.ukuu.org.uk> skl_tplg_tlv_control_set does pointer maths on data but forgets that data is not uint8_t so the maths is already scaled in the pointer type. Signed-off-by: Alan Cox <alan(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ba0d02d1613e..22a9e6fe1c05 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1111,7 +1111,7 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, return -EFAULT; } else { if (copy_from_user(ac->params, - data + 2 * sizeof(u32), size)) + data + 2, size)) return -EFAULT; } -- 1.9.1

2 9

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: remove bogus comparison of an array with NULL
by Vinod Koul 22 Feb '16

22 Feb '16

From: Alan Cox <alan(a)linux.intel.com> dfw_ac->params is an array not a pointer. It will never be NULL. The check on ac->max appears sufficient. Signed-off-by: Alan Cox <alan(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 22a9e6fe1c05..99f50a034e1d 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1657,8 +1657,7 @@ static int skl_init_algo_data(struct device *dev, struct soc_bytes_ext *be, if (!ac->params) return -ENOMEM; - if (dfw_ac->params) - memcpy(ac->params, dfw_ac->params, ac->max); + memcpy(ac->params, dfw_ac->params, ac->max); } be->dobj.private = ac; -- 1.9.1

1 0

[alsa-devel] Applied "regmap: Add explict native endian flag to DT bindings" to the asoc tree
by Mark Brown 22 Feb '16

22 Feb '16

The patch regmap: Add explict native endian flag to DT bindings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a06c488da0b0c1eebf710017675f00b9d5f9cf42 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Tue, 26 Jan 2016 17:59:30 +0000 Subject: [PATCH] regmap: Add explict native endian flag to DT bindings Currently the binding document says that if no endianness is configured we use native endian but this is not in fact true for all binding types and we do have some devices that really want native endianness such as Broadcom MIPS SoCs where switching the endianness of the CPU also switches the endianness of external IPs. Provide an explicit option for this. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/regmap/regmap.txt | 11 +++++++---- drivers/base/regmap/regmap.c | 2 ++ 2 files changed, 9 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/regmap/regmap.txt b/Documentation/devicetree/bindings/regmap/regmap.txt index b494f8b..e98a9652 100644 --- a/Documentation/devicetree/bindings/regmap/regmap.txt +++ b/Documentation/devicetree/bindings/regmap/regmap.txt @@ -5,15 +5,18 @@ Index Device Endianness properties --------------------------------------------------- 1 BE 'big-endian' 2 LE 'little-endian' +3 Native 'native-endian' For one device driver, which will run in different scenarios above on different SoCs using the devicetree, we need one way to simplify this. -Required properties: -- {big,little}-endian: these are boolean properties, if absent - meaning that the CPU and the Device are in the same endianness mode, - these properties are for register values and all the buffers only. +Optional properties: +- {big,little,native}-endian: these are boolean properties, if absent + then the implementation will choose a default based on the device + being controlled. These properties are for register values and all + the buffers only. Native endian means that the CPU and device have + the same endianness. Examples: Scenario 1 : CPU in LE mode & device in LE mode. diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index ee54e84..3432634 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -557,6 +557,8 @@ enum regmap_endian regmap_get_val_endian(struct device *dev, endian = REGMAP_ENDIAN_BIG; else if (of_property_read_bool(np, "little-endian")) endian = REGMAP_ENDIAN_LITTLE; + else if (of_property_read_bool(np, "native-endian")) + endian = REGMAP_ENDIAN_NATIVE; /* If the endianness was specified in DT, use that */ if (endian != REGMAP_ENDIAN_DEFAULT) -- 2.7.0

1 0

[alsa-devel] Applied "MIPS: dt: Explicitly specify native endian behaviour for syscon" to the asoc tree
by Mark Brown 22 Feb '16

22 Feb '16

The patch MIPS: dt: Explicitly specify native endian behaviour for syscon has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 25d6463e48ea41149c88dc40648a888699e77158 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Tue, 26 Jan 2016 18:08:06 +0000 Subject: [PATCH] MIPS: dt: Explicitly specify native endian behaviour for syscon On many MIPS systems the endianness of IP blocks is kept the same as that of the CPU by the hardware. This includes the system controllers on these systems which are controlled via syscon which uses the regmap API which used readl() and writel() to interact with the hardware, meaning that all writes are converted to little endian when writing to the hardware. This caused a bad interaction with the regmap core in big endian mode since it was not aware of the byte swapping and so ended up performing little endian writes. Unfortunately when this issue was noticed it was addressed by updating the DT for the affected devices to specify them as little endian. This happened to work since it resulted in two endianness swaps which cancelled each other out and gave little endian behaviour but meant that the DT was clearly not accurately describing the hardware. The intention of commit 29bb45f25ff305 (regmap-mmio: Use native endianness for read/write) was to fix this by making regmap default to native endianness but this breaks most other MMIO users where the hardware has a fixed endianness and the implementation uses the __raw accessors which are not intended to be used outside of architecture code. Instead use the newly added native-endian DT property to say exactly what we want for these systems. Fixes: 29bb45f25ff305 (regmap-mmio: Use native endianness for read/write) Reported-by: Johannes Berg <johannes(a)sipsolutions.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> Acked-by: Ralf Baechle <ralf(a)linux-mips.org> --- arch/mips/boot/dts/brcm/bcm6328.dtsi | 1 + arch/mips/boot/dts/brcm/bcm6368.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7125.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7346.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7358.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7360.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7362.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7420.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7425.dtsi | 1 + arch/mips/boot/dts/brcm/bcm7435.dtsi | 1 + 10 files changed, 10 insertions(+), 1 deletion(-) diff --git a/arch/mips/boot/dts/brcm/bcm6328.dtsi b/arch/mips/boot/dts/brcm/bcm6328.dtsi index 459b9b2..9d19236 100644 --- a/arch/mips/boot/dts/brcm/bcm6328.dtsi +++ b/arch/mips/boot/dts/brcm/bcm6328.dtsi @@ -74,6 +74,7 @@ timer: timer@10000040 { compatible = "syscon"; reg = <0x10000040 0x2c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm6368.dtsi b/arch/mips/boot/dts/brcm/bcm6368.dtsi index 9c8d3fe2..1f6b9b5 100644 --- a/arch/mips/boot/dts/brcm/bcm6368.dtsi +++ b/arch/mips/boot/dts/brcm/bcm6368.dtsi @@ -54,7 +54,7 @@ periph_cntl: syscon@10000000 { compatible = "syscon"; reg = <0x10000000 0x14>; - little-endian; + native-endian; }; reboot: syscon-reboot@10000008 { diff --git a/arch/mips/boot/dts/brcm/bcm7125.dtsi b/arch/mips/boot/dts/brcm/bcm7125.dtsi index 4fc7ece..3ae1605 100644 --- a/arch/mips/boot/dts/brcm/bcm7125.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7125.dtsi @@ -98,6 +98,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7125-sun-top-ctrl", "syscon"; reg = <0x404000 0x60c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7346.dtsi b/arch/mips/boot/dts/brcm/bcm7346.dtsi index a3039bb..be79919 100644 --- a/arch/mips/boot/dts/brcm/bcm7346.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7346.dtsi @@ -118,6 +118,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7346-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7358.dtsi b/arch/mips/boot/dts/brcm/bcm7358.dtsi index 4274ff4..060805b 100644 --- a/arch/mips/boot/dts/brcm/bcm7358.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7358.dtsi @@ -112,6 +112,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7358-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7360.dtsi b/arch/mips/boot/dts/brcm/bcm7360.dtsi index 0dcc9163..bcdb09b 100644 --- a/arch/mips/boot/dts/brcm/bcm7360.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7360.dtsi @@ -112,6 +112,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7360-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7362.dtsi b/arch/mips/boot/dts/brcm/bcm7362.dtsi index 2f3f9fc..d3b1b76 100644 --- a/arch/mips/boot/dts/brcm/bcm7362.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7362.dtsi @@ -118,6 +118,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7362-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7420.dtsi b/arch/mips/boot/dts/brcm/bcm7420.dtsi index bee221b..3302a1b 100644 --- a/arch/mips/boot/dts/brcm/bcm7420.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7420.dtsi @@ -99,6 +99,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7420-sun-top-ctrl", "syscon"; reg = <0x404000 0x60c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7425.dtsi b/arch/mips/boot/dts/brcm/bcm7425.dtsi index 571f30f..15b27aa 100644 --- a/arch/mips/boot/dts/brcm/bcm7425.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7425.dtsi @@ -100,6 +100,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7425-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { diff --git a/arch/mips/boot/dts/brcm/bcm7435.dtsi b/arch/mips/boot/dts/brcm/bcm7435.dtsi index 614ee21..adb33e3 100644 --- a/arch/mips/boot/dts/brcm/bcm7435.dtsi +++ b/arch/mips/boot/dts/brcm/bcm7435.dtsi @@ -114,6 +114,7 @@ sun_top_ctrl: syscon@404000 { compatible = "brcm,bcm7425-sun-top-ctrl", "syscon"; reg = <0x404000 0x51c>; + native-endian; }; reboot { -- 2.7.0

1 0

[alsa-devel] Applied "regmap: mmio: Convert to regmap_bus and fix accessor usage" to the asoc tree
by Mark Brown 22 Feb '16

22 Feb '16

The patch regmap: mmio: Convert to regmap_bus and fix accessor usage has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 922a9f936e40001f9b921379aab90047d5990923 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Wed, 27 Jan 2016 04:50:07 +0000 Subject: [PATCH] regmap: mmio: Convert to regmap_bus and fix accessor usage Currently regmap-mmio uses the __raw accessors to read and write from memory. This is not safe as these interact poorly with spinlocks and are not guaranteed to generate emulated instructions on at least ARM where regmap is commonly used. The APIs that are provided all provide some byte swapping so this is difficult to do with the current regmap-mmio implementation which attempts to use the regmap core byte swapping. We can fix this by modernising the MMIO implementation to use reg_read() and reg_write() operations which were added after the API was implemented and pass simple unsigned integers through to the bus, making use of the formatting provided by the I/O accessors using a similar pattern to that used by the core. This will be less efficient for block I/O operations since we now enable and disable any required clocks per register but it is not clear that any users of regmap-mmio actually use block I/O and there is room to optimise later. This removes support for big endian I/O on 64 bit registers since no I/O accessors are provided, no current users were found and support can be added easily once they are available. In addition make the default endianness little endian. This was the behaviour prior to 29bb45f25ff305 (regmap-mmio: Use native endianness for read/write) and is the behaviour desired by most existing users, the users have been audited and those that need native endianness converted to request it explicitly. Previously native was documented as the default but due to the byte swapping in the accessors this was not correctly implemented. Fixes: 29bb45f25ff305 (regmap-mmio: Use native endianness for read/write) Reported-by: Johannes Berg <johannes(a)sipsolutions.net> Tested-by: Johannes Berg <johannes(a)sipsolutions.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regmap-mmio.c | 259 ++++++++++++++++++++------------------ 1 file changed, 139 insertions(+), 120 deletions(-) diff --git a/drivers/base/regmap/regmap-mmio.c b/drivers/base/regmap/regmap-mmio.c index 8812bfb..7526906 100644 --- a/drivers/base/regmap/regmap-mmio.c +++ b/drivers/base/regmap/regmap-mmio.c @@ -25,26 +25,14 @@ struct regmap_mmio_context { void __iomem *regs; - unsigned reg_bytes; unsigned val_bytes; - unsigned pad_bytes; struct clk *clk; -}; -static inline void regmap_mmio_regsize_check(size_t reg_size) -{ - switch (reg_size) { - case 1: - case 2: - case 4: -#ifdef CONFIG_64BIT - case 8: -#endif - break; - default: - BUG(); - } -} + void (*reg_write)(struct regmap_mmio_context *ctx, + unsigned int reg, unsigned int val); + unsigned int (*reg_read)(struct regmap_mmio_context *ctx, + unsigned int reg); +}; static int regmap_mmio_regbits_check(size_t reg_bits) { @@ -88,72 +76,62 @@ static int regmap_mmio_get_min_stride(size_t val_bits) return min_stride; } -static inline void regmap_mmio_count_check(size_t count, u32 offset) +static void regmap_mmio_write8(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) +{ + writeb(val, ctx->regs + reg); +} + +static void regmap_mmio_write16le(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) { - BUG_ON(count <= offset); + writew(val, ctx->regs + reg); } -static inline unsigned int -regmap_mmio_get_offset(const void *reg, size_t reg_size) +static void regmap_mmio_write16be(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) { - switch (reg_size) { - case 1: - return *(u8 *)reg; - case 2: - return *(u16 *)reg; - case 4: - return *(u32 *)reg; + iowrite16be(val, ctx->regs + reg); +} + +static void regmap_mmio_write32le(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) +{ + writel(val, ctx->regs + reg); +} + +static void regmap_mmio_write32be(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) +{ + iowrite32be(val, ctx->regs + reg); +} + #ifdef CONFIG_64BIT - case 8: - return *(u64 *)reg; -#endif - default: - BUG(); - } +static void regmap_mmio_write64le(struct regmap_mmio_context *ctx, + unsigned int reg, + unsigned int val) +{ + writeq(val, ctx->regs + reg); } +#endif -static int regmap_mmio_gather_write(void *context, - const void *reg, size_t reg_size, - const void *val, size_t val_size) +static int regmap_mmio_write(void *context, unsigned int reg, unsigned int val) { struct regmap_mmio_context *ctx = context; - unsigned int offset; int ret; - regmap_mmio_regsize_check(reg_size); - if (!IS_ERR(ctx->clk)) { ret = clk_enable(ctx->clk); if (ret < 0) return ret; } - offset = regmap_mmio_get_offset(reg, reg_size); - - while (val_size) { - switch (ctx->val_bytes) { - case 1: - __raw_writeb(*(u8 *)val, ctx->regs + offset); - break; - case 2: - __raw_writew(*(u16 *)val, ctx->regs + offset); - break; - case 4: - __raw_writel(*(u32 *)val, ctx->regs + offset); - break; -#ifdef CONFIG_64BIT - case 8: - __raw_writeq(*(u64 *)val, ctx->regs + offset); - break; -#endif - default: - /* Should be caught by regmap_mmio_check_config */ - BUG(); - } - val_size -= ctx->val_bytes; - val += ctx->val_bytes; - offset += ctx->val_bytes; - } + ctx->reg_write(ctx, reg, val); if (!IS_ERR(ctx->clk)) clk_disable(ctx->clk); @@ -161,59 +139,56 @@ static int regmap_mmio_gather_write(void *context, return 0; } -static int regmap_mmio_write(void *context, const void *data, size_t count) +static unsigned int regmap_mmio_read8(struct regmap_mmio_context *ctx, + unsigned int reg) { - struct regmap_mmio_context *ctx = context; - unsigned int offset = ctx->reg_bytes + ctx->pad_bytes; + return readb(ctx->regs + reg); +} + +static unsigned int regmap_mmio_read16le(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return readw(ctx->regs + reg); +} + +static unsigned int regmap_mmio_read16be(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return ioread16be(ctx->regs + reg); +} + +static unsigned int regmap_mmio_read32le(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return readl(ctx->regs + reg); +} - regmap_mmio_count_check(count, offset); +static unsigned int regmap_mmio_read32be(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return ioread32be(ctx->regs + reg); +} - return regmap_mmio_gather_write(context, data, ctx->reg_bytes, - data + offset, count - offset); +#ifdef CONFIG_64BIT +static unsigned int regmap_mmio_read64le(struct regmap_mmio_context *ctx, + unsigned int reg) +{ + return readq(ctx->regs + reg); } +#endif -static int regmap_mmio_read(void *context, - const void *reg, size_t reg_size, - void *val, size_t val_size) +static int regmap_mmio_read(void *context, unsigned int reg, unsigned int *val) { struct regmap_mmio_context *ctx = context; - unsigned int offset; int ret; - regmap_mmio_regsize_check(reg_size); - if (!IS_ERR(ctx->clk)) { ret = clk_enable(ctx->clk); if (ret < 0) return ret; } - offset = regmap_mmio_get_offset(reg, reg_size); - - while (val_size) { - switch (ctx->val_bytes) { - case 1: - *(u8 *)val = __raw_readb(ctx->regs + offset); - break; - case 2: - *(u16 *)val = __raw_readw(ctx->regs + offset); - break; - case 4: - *(u32 *)val = __raw_readl(ctx->regs + offset); - break; -#ifdef CONFIG_64BIT - case 8: - *(u64 *)val = __raw_readq(ctx->regs + offset); - break; -#endif - default: - /* Should be caught by regmap_mmio_check_config */ - BUG(); - } - val_size -= ctx->val_bytes; - val += ctx->val_bytes; - offset += ctx->val_bytes; - } + *val = ctx->reg_read(ctx, reg); if (!IS_ERR(ctx->clk)) clk_disable(ctx->clk); @@ -232,14 +207,11 @@ static void regmap_mmio_free_context(void *context) kfree(context); } -static struct regmap_bus regmap_mmio = { +static const struct regmap_bus regmap_mmio = { .fast_io = true, - .write = regmap_mmio_write, - .gather_write = regmap_mmio_gather_write, - .read = regmap_mmio_read, + .reg_write = regmap_mmio_write, + .reg_read = regmap_mmio_read, .free_context = regmap_mmio_free_context, - .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, - .val_format_endian_default = REGMAP_ENDIAN_NATIVE, }; static struct regmap_mmio_context *regmap_mmio_gen_context(struct device *dev, @@ -265,24 +237,71 @@ static struct regmap_mmio_context *regmap_mmio_gen_context(struct device *dev, if (config->reg_stride < min_stride) return ERR_PTR(-EINVAL); - switch (config->reg_format_endian) { - case REGMAP_ENDIAN_DEFAULT: - case REGMAP_ENDIAN_NATIVE: - break; - default: - return ERR_PTR(-EINVAL); - } - ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return ERR_PTR(-ENOMEM); ctx->regs = regs; ctx->val_bytes = config->val_bits / 8; - ctx->reg_bytes = config->reg_bits / 8; - ctx->pad_bytes = config->pad_bits / 8; ctx->clk = ERR_PTR(-ENODEV); + switch (config->reg_format_endian) { + case REGMAP_ENDIAN_DEFAULT: + case REGMAP_ENDIAN_LITTLE: +#ifdef __LITTLE_ENDIAN + case REGMAP_ENDIAN_NATIVE: +#endif + switch (config->val_bits) { + case 8: + ctx->reg_read = regmap_mmio_read8; + ctx->reg_write = regmap_mmio_write8; + break; + case 16: + ctx->reg_read = regmap_mmio_read16le; + ctx->reg_write = regmap_mmio_write16le; + break; + case 32: + ctx->reg_read = regmap_mmio_read32le; + ctx->reg_write = regmap_mmio_write32le; + break; +#ifdef CONFIG_64BIT + case 64: + ctx->reg_read = regmap_mmio_read64le; + ctx->reg_write = regmap_mmio_write64le; + break; +#endif + default: + ret = -EINVAL; + goto err_free; + } + break; + case REGMAP_ENDIAN_BIG: +#ifdef __BIG_ENDIAN + case REGMAP_ENDIAN_NATIVE: +#endif + switch (config->val_bits) { + case 8: + ctx->reg_read = regmap_mmio_read8; + ctx->reg_write = regmap_mmio_write8; + break; + case 16: + ctx->reg_read = regmap_mmio_read16be; + ctx->reg_write = regmap_mmio_write16be; + break; + case 32: + ctx->reg_read = regmap_mmio_read32be; + ctx->reg_write = regmap_mmio_write32be; + break; + default: + ret = -EINVAL; + goto err_free; + } + break; + default: + ret = -EINVAL; + goto err_free; + } + if (clk_id == NULL) return ctx; -- 2.7.0

1 0

[alsa-devel] [PATCH 1/2] regmap: Return an error if a caller attempts to do an unsupported raw read
by Mark Brown 22 Feb '16

22 Feb '16

regmaps without raw I/O access can't implement raw I/O operations, return an error if someone tries to do that rather than crashing. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regmap.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 343263449aff..e2f68807d970 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -2255,6 +2255,9 @@ static int _regmap_raw_read(struct regmap *map, unsigned int reg, void *val, WARN_ON(!map->bus); + if (!map->bus || !map->bus->read) + return -EINVAL; + range = _regmap_range_lookup(map, reg); if (range) { ret = _regmap_select_page(map, &reg, range, -- 2.7.0.rc3

3 7

[alsa-devel] Applied "regmap: cache: Fall back to register by register read for cache defaults" to the asoc tree
by Mark Brown 22 Feb '16

22 Feb '16

The patch regmap: cache: Fall back to register by register read for cache defaults has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3245d460a1eb55b5c3ca31dde7b5c5ac71546edf Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Tue, 2 Feb 2016 10:16:51 -0200 Subject: [PATCH] regmap: cache: Fall back to register by register read for cache defaults If we are unable to read the cache defaults for a regmap then fall back on attempting to read them word by word. This is going to be painfully slow for large regmaps but might be adequate for smaller ones. Signed-off-by: Mark Brown <broonie(a)kernel.org> [maciej: Use cache_bypass around read and skipping of unreadable regs] Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> Tested-by: Fabio Estevam <fabio.estevam(a)nxp.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regcache.c | 41 ++++++++++++++++++++++++++++++----------- 1 file changed, 30 insertions(+), 11 deletions(-) diff --git a/drivers/base/regmap/regcache.c b/drivers/base/regmap/regcache.c index 348be3a..cccceb5 100644 --- a/drivers/base/regmap/regcache.c +++ b/drivers/base/regmap/regcache.c @@ -30,7 +30,7 @@ static int regcache_hw_init(struct regmap *map) int i, j; int ret; int count; - unsigned int val; + unsigned int reg, val; void *tmp_buf; if (!map->num_reg_defaults_raw) @@ -67,27 +67,46 @@ static int regcache_hw_init(struct regmap *map) ret = regmap_raw_read(map, 0, tmp_buf, map->num_reg_defaults_raw); map->cache_bypass = cache_bypass; - if (ret < 0) - goto err_cache_free; - - map->reg_defaults_raw = tmp_buf; - map->cache_free = 1; + if (ret == 0) { + map->reg_defaults_raw = tmp_buf; + map->cache_free = 1; + } else { + kfree(tmp_buf); + } } /* fill the reg_defaults */ for (i = 0, j = 0; i < map->num_reg_defaults_raw; i++) { - if (regmap_volatile(map, i * map->reg_stride)) + reg = i * map->reg_stride; + + if (!regmap_readable(map, reg)) continue; - val = regcache_get_val(map, map->reg_defaults_raw, i); - map->reg_defaults[j].reg = i * map->reg_stride; + + if (regmap_volatile(map, reg)) + continue; + + if (map->reg_defaults_raw) { + val = regcache_get_val(map, map->reg_defaults_raw, i); + } else { + bool cache_bypass = map->cache_bypass; + + map->cache_bypass = true; + ret = regmap_read(map, reg, &val); + map->cache_bypass = cache_bypass; + if (ret != 0) { + dev_err(map->dev, "Failed to read %d: %d\n", + reg, ret); + goto err_free; + } + } + + map->reg_defaults[j].reg = reg; map->reg_defaults[j].def = val; j++; } return 0; -err_cache_free: - kfree(tmp_buf); err_free: kfree(map->reg_defaults); -- 2.7.0

1 0

[alsa-devel] [PATCH 3/3] ASoC: fsl_ssi: remove register defaults
by Maciej S. Szmigiero 22 Feb '16

22 Feb '16

There is no guarantee that on fsl_ssi module load SSI registers will have their power-on-reset values. In fact, if the driver is reloaded the values in registers will be whatever they were set to previously. This fixes hard lockup on fsl_ssi module reload, at least in AC'97 mode. Fixes: 05cf237972fe ("ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast") Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 40dfd8a36484..f6ef7d64acb3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -112,20 +112,6 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val tx; }; -static const struct reg_default fsl_ssi_reg_defaults[] = { - {CCSR_SSI_SCR, 0x00000000}, - {CCSR_SSI_SIER, 0x00003003}, - {CCSR_SSI_STCR, 0x00000200}, - {CCSR_SSI_SRCR, 0x00000200}, - {CCSR_SSI_STCCR, 0x00040000}, - {CCSR_SSI_SRCCR, 0x00040000}, - {CCSR_SSI_SACNT, 0x00000000}, - {CCSR_SSI_STMSK, 0x00000000}, - {CCSR_SSI_SRMSK, 0x00000000}, - {CCSR_SSI_SACCEN, 0x00000000}, - {CCSR_SSI_SACCDIS, 0x00000000}, -}; - static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -190,8 +176,6 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .reg_defaults = fsl_ssi_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg,

4 27

[alsa-devel] [PATCH v2] ASoC: wm9713: fix regmap free path
by Robert Jarzmik 22 Feb '16

22 Feb '16

In the conversion to regmap, I assumed that the 2 following functions was working symetrically: - snd_soc_codec_init_regmap() - snd_soc_codec_exit_regmap(codec) As a mater of fact with the current code the regmap is freed twice because of the devm_() call: (mutex_lock) from (debugfs_remove_recursive+0x50/0x1d0) (debugfs_remove_recursive) from (regmap_debugfs_exit+0x1c/0xd4) (regmap_debugfs_exit) from (regmap_exit+0x28/0xc8) (regmap_exit) from (release_nodes+0x18c/0x204) (release_nodes) from (device_release+0x18/0x90) (device_release) from (kobject_release+0x90/0x1bc) (kobject_release) from (wm9713_soc_remove+0x1c/0x24) (wm9713_soc_remove) from (soc_remove_component+0x50/0x7c) (soc_remove_component) from (soc_remove_dai_links+0x118/0x228) (soc_remove_dai_links) from (snd_soc_register_card+0x4e4/0xdd4) (snd_soc_register_card) from (devm_snd_soc_register_card+0x34/0x70) Fix this by removing the doubled regmap free. Fixes: 700dadfefc3d ASoC: wm9713: convert to regmap Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: fix the diff second hunk, god knows where it came from. --- sound/soc/codecs/wm9713.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 79e143625ac3..dd31a5207527 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1230,7 +1230,6 @@ static int wm9713_soc_remove(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9713->ac97); return 0; } -- 2.1.4

3 3

[alsa-devel] [alsa-lib][PATCH] control: initialize element channel values according to identification number at adding a new element
by Takashi Sakamoto 21 Feb '16

21 Feb '16

In ALSA control core, there're two ways to identify elements. One is by numid. Another is by name/index. The former way is lighter than the latter way. In control API, after adding elements by snd_ctl_elem_add_integer() and snd_ctl_elem_add_integer64(), these functions initialize value of each channel. When executing this operation, used ID information is still given by applications. Although, in this time, the element can be identified according to numid. This commit uses the numid instead of name/index pair. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index a44ae96..8a5d530 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -292,7 +292,7 @@ int snd_ctl_elem_add_integer(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, if (err < 0) return err; snd_ctl_elem_value_alloca(&val); - val->id = *id; + val->id = info->id; for (i = 0; i < count; i++) val->value.integer.value[i] = min; err = ctl->ops->element_write(ctl, val); @@ -330,7 +330,7 @@ int snd_ctl_elem_add_integer64(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, if (err < 0) return err; snd_ctl_elem_value_alloca(&val); - val->id = *id; + val->id = info->id; for (i = 0; i < count; i++) val->value.integer64.value[i] = min; err = ctl->ops->element_write(ctl, val); -- 2.5.0

2 1

[alsa-devel] [alsa-lib][PATCH] control: fulfill a description about element type with integer64
by Takashi Sakamoto 21 Feb '16

21 Feb '16

In UAPI of asound.h, there're 6 types of control element; boolean, integer, enumerated, bytes, iec958 and integer64 (The none type seems to be added to start the value at non-zero). Although, current description in alsa-lib has a lack of integer64 type. This commit fulfil the type. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/control.c b/src/control/control.c index 70d168d..a44ae96 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -35,7 +35,7 @@ also interface notifying about control and structure changes. \section control_general_overview General overview -The primitive controls can be integer, boolean, enumerators, bytes +The primitive controls can be integer, inter64, boolean, enumerators, bytes and IEC958 structure. */ -- 2.5.0

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm_adsp: Avoid narrow race condition on compr pointer
by Charles Keepax 20 Feb '16

20 Feb '16

There is a very small window between then wm_adsp_compr_free gets call and when the DSP is actually powered down. If we get an IRQ from the DSP in this window then the wm_adsp_compr pointer will be NULL. This patch adds a check for this into the IRQ handler to avoid any issues when this happens. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ed90e12..00a7d68 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2857,7 +2857,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) goto out; } - if (compr->stream) + if (compr && compr->stream) snd_compr_fragment_elapsed(compr->stream); out: -- 2.1.4

2 7

[alsa-devel] [PATCH] ASoC: max98926: Constify max98926_reg and max98926_regmap
by Axel Lin 20 Feb '16

20 Feb '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98926.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 5245e10..8d14ada 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -40,7 +40,7 @@ static const char *const max98926_hpf_cutoff_txt[] = { "200Hz", "400Hz", "800Hz", }; -static struct reg_default max98926_reg[] = { +static const struct reg_default max98926_reg[] = { { 0x0B, 0x00 }, /* IRQ Enable0 */ { 0x0C, 0x00 }, /* IRQ Enable1 */ { 0x0D, 0x00 }, /* IRQ Enable2 */ @@ -506,7 +506,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98926 = { .num_dapm_widgets = ARRAY_SIZE(max98926_dapm_widgets), }; -static struct regmap_config max98926_regmap = { +static const struct regmap_config max98926_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = MAX98926_VERSION, -- 2.1.4

3 2

[alsa-devel] [PATCH 1/3] ASoC: max9867: Add CONFIG_SND_SOC_MAX9867 to build
by Axel Lin 20 Feb '16

20 Feb '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Makefile | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 621c0a3..185a712 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -283,6 +283,7 @@ obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o +obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o -- 2.1.4

3 6

[alsa-devel] [PATCH] ASoC: wm9713: fix regmap free path
by Robert Jarzmik 20 Feb '16

20 Feb '16

In the conversion to regmap, I assumed that the 2 following functions was working symetrically: - snd_soc_codec_init_regmap() - snd_soc_codec_exit_regmap(codec) As a mater of fact with the current code the regmap is freed twice because of the devm_() call: (mutex_lock) from (debugfs_remove_recursive+0x50/0x1d0) (debugfs_remove_recursive) from (regmap_debugfs_exit+0x1c/0xd4) (regmap_debugfs_exit) from (regmap_exit+0x28/0xc8) (regmap_exit) from (release_nodes+0x18c/0x204) (release_nodes) from (device_release+0x18/0x90) (device_release) from (kobject_release+0x90/0x1bc) (kobject_release) from (wm9713_soc_remove+0x1c/0x24) (wm9713_soc_remove) from (soc_remove_component+0x50/0x7c) (soc_remove_component) from (soc_remove_dai_links+0x118/0x228) (soc_remove_dai_links) from (snd_soc_register_card+0x4e4/0xdd4) (snd_soc_register_card) from (devm_snd_soc_register_card+0x34/0x70) Fix this by removing the doubled regmap free. Fixes: 700dadfefc3d ASoC: wm9713: convert to regmap Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/codecs/wm9713.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 79e143625ac3..0242b7dcc1bb 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1230,7 +1230,6 @@ static int wm9713_soc_remove(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9713->ac97); return 0; } @@ -1272,3 +1271,2 @@ static int wm9713_remove(struct platform_device *pdev) return 0; } -- 2.1.4

2 1

[alsa-devel] [PATCH 0/11] ASoC: rsnd: cleanup SSI multi mode and sequence
by Kuninori Morimoto 20 Feb '16

20 Feb '16

Hi Mark Current rsnd driver is supporting many features, and now we are confirming these register settings sequence. These patches fixup not good sequence (and bug) Kuninori Morimoto (11): ASoC: rsnd: rename RSND_DVC_CHANNELS to RSND_MAX_CHANNELS ASoC: rsnd: setup SRC_ROUTE_MODE0 before SRC_SRCIR ASoC: rsnd: indicates status failed SSI ASoC: rsnd: SSI function parameter uses "mod" instead of "ssi" ASoC: rsnd: fixup forever loop bug on SSI ASoC: rsnd: add missing .irq callback for DMA ASoC: rsnd: tidyup SSI init/start sequence ASoC: rsnd: Parent SSI attach is not needed if not clock master ASoC: rsnd: move rsnd_ssi_irq() position ASoC: rsnd: judge multi SSI in runtime ASoC: rsnd: judge work SSI in runtime sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/sh/rcar/dvc.c | 6 ++--- sound/soc/sh/rcar/rsnd.h | 6 ++--- sound/soc/sh/rcar/src.c | 3 ++- sound/soc/sh/rcar/ssi.c | 166 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------------------------------------- sound/soc/sh/rcar/ssiu.c | 6 ++--- 6 files changed, 126 insertions(+), 65 deletions(-)

2 22

[alsa-devel] [PATCH v5 00/14] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 20 Feb '16

20 Feb '16

This patch series adds DP audio and hotplug notification support. In skylake platform the DP is on a different port which is not enabled by default. A vendor widget is programmed to enable playback on all ports. This also enables all the converters. Codec Dais, widgets, graph are now created dynamically based on the nodes enumerated, thus removing the hardcoding. Mux controls are used to establish routing stream to a specific port. For DP audio support, infoframe needs to be different. Based on ELD queried we either pack HDMI or DP infoframe. Also with this series, Jack notification support is added and Jack event is reported to userspace as is done in case of legacy driver. Few fixes are added to check for possible memory leak, reconfiguring register during resume from D3 etc. The 8th patch is adding a small macro for getting connection type in drm header. It is reviewed by drm folks, CCed them. Please merge it through sound trees. Changes in v5: - Added more comments and reworded some commit messages - Jack reporting support to user space added - Fix to check for possible memory leak in hw_params - Fix not to fail in dai startup to make userland APIs happy Cc: Jani Nikula <jani.nikula(a)intel.com> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Daniel Vetter <daniel.vetter(a)intel.com> Jeeja KP (2): ASoC: hdac_hdmi: Add jack reporting ASoC: hdac_hdmi: Add PM support Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep codec power active during enumeration. Subhransu S. Prusty (11): ASoC: hdac_hdmi: Add hotplug notification and read eld ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi: create dais based on number of cvts ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Enable playback on all enumerated ports drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio ASoC: hdac_hdmi: Fix possible memory leak in hw_params ASoC: hdac_hdmi: Don't fail in dai startup to make userland happy ASoC: hdac_hdmi: Fix to reconfigure registers in runtime resume include/drm/drm_edid.h | 12 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 1197 +++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/hdac_hdmi.h | 6 + 4 files changed, 1094 insertions(+), 122 deletions(-) create mode 100644 sound/soc/codecs/hdac_hdmi.h -- 1.9.1

4 25

[alsa-devel] [PATCH v1 00/14] ASoC: qcom: Add capture support.
by Srinivas Kandagatla 20 Feb '16

20 Feb '16

LPASS IP on QCOM SOC supports both Playback and capture via I2S, but this feature is missing in existing code. This patchset aims at adding capture support to lpass IP. First few patches in this series does cleanup the driver to make easy to add capture support. Most of these patches are acked by Kenneth. These patches are tested on DB410C with Headset Mic, secondary mic. Thanks, srini Changes since RFC: - made patches bisectable, reported by 0day testing - merged two patches as suggested by Kenneth - used corect device for snd_dma alloc/free apis, spotted by Kenneth - renamed LPAIF_I2SCTL_MICMODE_QUAD02 to LPAIF_I2SCTL_MICMODE_QUAD23, spotted by Kenneth. - Fixed minor issues suggested by Kenneth. Srinivas Kandagatla (14): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: apq8016: add wrdma support ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma register definitions ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: add mic support ASoC: qcom: apq8016-sbc: add mic support sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +++-- sound/soc/qcom/lpass-cpu.c | 146 +++++++++++++++++------ sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 116 +++++++++++++------ sound/soc/qcom/lpass-platform.c | 243 +++++++++++++++++++++++---------------- sound/soc/qcom/lpass.h | 10 +- 7 files changed, 390 insertions(+), 177 deletions(-) -- 1.9.1

3 26

[alsa-devel] [PATCH v3 00/14] ASoC: qcom: Add capture support.
by Srinivas Kandagatla 20 Feb '16

20 Feb '16

LPASS IP on QCOM SOC supports both Playback and capture via I2S, but this feature is missing in existing code. This patchset aims at adding capture support to lpass IP. First few patches in this series does cleanup the driver to make easy to add capture support. All these patches are acked by Kenneth. These patches are tested on DB410C with Headset Mic, secondary mic. Thanks, srini Changes since v1/v2: - Fixed patch version number. - renamed rdma_port to dma_port in patch 3, as suggested by Kenneth. - fixed correct device for snd_dma_alloc/free as suggested by Kenneth in patch 13. Changes since RFC: - made patches bisectable, reported by 0day testing - merged two patches as suggested by Kenneth - used corect device for snd_dma alloc/free apis, spotted by Kenneth - renamed LPAIF_I2SCTL_MICMODE_QUAD02 to LPAIF_I2SCTL_MICMODE_QUAD23, spotted by Kenneth. - Fixed minor issues suggested by Kenneth. Srinivas Kandagatla (14): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: apq8016: add wrdma support ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma register definitions ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: add mic support ASoC: qcom: apq8016-sbc: add mic support sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +++-- sound/soc/qcom/lpass-cpu.c | 146 ++++++++++++++++++----- sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 116 +++++++++++++------ sound/soc/qcom/lpass-platform.c | 244 +++++++++++++++++++++++---------------- sound/soc/qcom/lpass.h | 10 +- 7 files changed, 391 insertions(+), 177 deletions(-) -- 1.9.1

2 17

[alsa-devel] [PATCH RFC 00/15] ASoC: qcom: Add capture support.
by Srinivas Kandagatla 20 Feb '16

20 Feb '16

LPASS IP on QCOM SOC supports both Playback and capture via I2S, but this feature is missing in existing code. This patchset aims at adding capture support to lpass IP. First few patches in this series does cleanup the driver to make easy to add capture support. These patches are tested on DB410C with Headset Mic. Thanks, srini Srinivas Kandagatla (15): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma dma channel start ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: apq8016: add wrdma support ASoC: qcom: add wrdma register definations ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: apq8016-sbc: add mic support ASoC: qcom: add mic support sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +++-- sound/soc/qcom/lpass-cpu.c | 146 +++++++++++++++++------ sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 118 +++++++++++++------ sound/soc/qcom/lpass-platform.c | 243 +++++++++++++++++++++++---------------- sound/soc/qcom/lpass.h | 10 +- 7 files changed, 392 insertions(+), 177 deletions(-) -- 1.9.1

3 45

[alsa-devel] [PATCH] ASoC: cht_bsw_rt5645: Enable jack detection
by Carlo Caione 20 Feb '16

20 Feb '16

From: Carlo Caione <carlo(a)endlessm.com> Add missing DAPM pins and enable jack detection on those pins for Cherrytrail and Braswell. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index e6cf800..c7af271 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -147,6 +147,17 @@ static const struct snd_kcontrol_new cht_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Ext Spk"), }; +static struct snd_soc_jack_pin cht_bsw_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -202,9 +213,9 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) else jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; - ret = snd_soc_card_jack_new(runtime->card, "Headset Jack", + ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, &ctx->jack, - NULL, 0); + cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); if (ret) { dev_err(runtime->dev, "Headset jack creation failed %d\n", ret); return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH v4 0/2] ASoC: Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 20 Feb '16

20 Feb '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver. For now just the SPDIF transmitter has been tested on a Mele A2000(A10) and a Itead Ibox(A20). In order for this patch set to be functional we require an audio clock patch which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK --- Changes since v3: - removed code required for spdif capture - added disable/enable of apb clk to suspend/resume. - removed unnecessary dt checks in probe - fixed error conditions in probe/resume - modified pm_runtime functionality(confirmed driver worked with PM disabled). - removed .owner from platform driver - renamed bindings file Changes since v2: - removed sunxi-machine-spdif and replaced with simple audio card - removed untested compatibilities from device tree documentation - added pm_runtime and remove shutdown - removed pll2 as it's the parent of the spdif clock - rename clk to spdif_clk - removed enabling of mclk output - removed interrupts status being cleared as it's not being used - use default fifo settings for now - fixed alignments for wrapped lines Changes since v1: - Moved sunxi-machine-spdif.c to separate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into separate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. Marcus Cooper (2): dt-bindings: sunxi :add sun4i SPDIF transceiver ASoC: sunxi: Add support for the SPDIF block .../bindings/sound/sunxi,sun4i-spdif.txt | 39 ++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 561 +++++++++++++++++++++ 4 files changed, 609 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c -- 2.7.0

4 5

[alsa-devel] [PATCH] SoC: cht_bsw_rt5645: Fix writing to string literal
by Carlo Caione 20 Feb '16

20 Feb '16

From: Carlo Caione <carlo(a)endlessm.com> We cannot use strcpy() to write to a const char * location. This is causing a 'BUG: unable to handle kernel paging request' error at boot when using the cht-bsw-rt5645 driver. With this patch we also fix a wrong indexing in the driver where the codec_name of the wrong dai_link is being overwritten. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/atom/sst-atom-controls.h | 1 + sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index e011311..6f88d1c 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -30,6 +30,7 @@ enum { MERR_DPCM_AUDIO = 0, MERR_DPCM_DEEP_BUFFER, MERR_DPCM_COMPR, + MERR_DPCM_BE, }; /* define a bit for each mixer input */ diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index e6cf800..89b4c032 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -282,7 +282,7 @@ static struct snd_soc_dai_link cht_dailink[] = { }, /* CODEC<->CODEC link */ /* back ends */ - { + [MERR_DPCM_BE] = { .name = "SSP2-Codec", .be_id = 1, .cpu_dai_name = "ssp2-port", @@ -357,7 +357,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + card->dai_link[MERR_DPCM_BE].codec_name = kstrdup(codec_name, GFP_KERNEL); snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- 1.9.1

3 3

[alsa-devel] [PATCH] fireworks: serialize transactions to update connections at bus reset
by Takashi Sakamoto 20 Feb '16

20 Feb '16

In IEC 61883-1, at bus-reset, applications can continue isochronous streaming by updating connections. In ALSA fireworks driver, the operation is executed in 'update' handler for bus driver. The connection resources are also changed in process contexts of PCM/MIDI applications. Therefore, bus-reset handling has race condition against connection. Current ALSA fireworks driver has a bug for the condition. This commit fixes the bug, by expand critical section with mutex. As a result, connection updating operation in bus-reset handler and connection changing operation in process context are serialized. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks.c | 3 +++ sound/firewire/fireworks/fireworks_stream.c | 6 ++---- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index d5b19bc..8f27b67 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -301,7 +301,10 @@ static void efw_update(struct fw_unit *unit) struct snd_efw *efw = dev_get_drvdata(&unit->device); snd_efw_transaction_bus_reset(efw->unit); + + mutex_lock(&efw->mutex); snd_efw_stream_update_duplex(efw); + mutex_unlock(&efw->mutex); } static void efw_remove(struct fw_unit *unit) diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 968a40a..425db8d 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -313,12 +313,10 @@ void snd_efw_stream_stop_duplex(struct snd_efw *efw) void snd_efw_stream_update_duplex(struct snd_efw *efw) { - if ((cmp_connection_update(&efw->out_conn) < 0) || - (cmp_connection_update(&efw->in_conn) < 0)) { - mutex_lock(&efw->mutex); + if (cmp_connection_update(&efw->out_conn) < 0 || + cmp_connection_update(&efw->in_conn) < 0) { stop_stream(efw, &efw->rx_stream); stop_stream(efw, &efw->tx_stream); - mutex_unlock(&efw->mutex); } else { amdtp_stream_update(&efw->rx_stream); amdtp_stream_update(&efw->tx_stream); -- 2.5.0

2 2

[alsa-devel] [PATCH 0/4] ALSA: bebob: give up updating streams at bus reset
by Takashi Sakamoto 20 Feb '16

20 Feb '16

Hi, This patchset is to simplify codes of ALSA bebob driver by giving up updating streams at bus reset. DM1000/DM1100/DM1500 chipsets have a quirk to transfer packets with discontinuous value in 'dbc' field of CIP header. This is against packet streaming protocol (IEC 61883-1/6). In the specification, units should continue packet streaming during 1 seconds after bus reset. During this period, connections are updated. Else, the streaming stops. The discontinuity causes firewire-lib module (packet streaming layer) to stop streaming immediately. This is done far earlier than executing bus-reset handler. Therefore, updating streams at the bus-reset handler has no meaning. This patchset will have little impacts for userspace applications because most users won't update IEEE 1394 bus when playbacking/recording. And PCM substream recovery is still available. Regards Takashi Sakamoto (4): ALSA: bebob: simplify bus-reset handling ALSA: bebob: move mutex from function callee to callers ALSA: bebob: change type of substream counter from atomic_t to unsigned int ALSA: bebob: give up updating streams at bus reset handler sound/firewire/bebob/bebob.c | 21 ++++++++++---- sound/firewire/bebob/bebob.h | 5 +--- sound/firewire/bebob/bebob_midi.c | 16 +++++++--- sound/firewire/bebob/bebob_pcm.c | 28 +++++++++++++----- sound/firewire/bebob/bebob_stream.c | 58 +++---------------------------------- 5 files changed, 53 insertions(+), 75 deletions(-) -- 2.5.0

2 5

Re: [alsa-devel] Issues with ALSA:ice1724 sound cards on powerpc machines
by Julian Margetson 20 Feb '16

20 Feb '16

On 2/17/2016 2:29 PM, Julian Margetson wrote: > Hi > > Having an issue issue with ice1724 sound cards on powerpc motherboards. > Motherboards in question Acube Sam460ex . > The card appears as a valid device but there is no sound output on the > majority of boots of the computer. > Logs attached. > On the first set boot audio playback produced sound ok. > On the second set boot no sound was produce with audio playback > > > Regards > Julian Margetson > > > Issue also with A-Eon Cyrus powerpc motherboard .

1 0

[alsa-devel] [PATCH v6 0/2] ASoC: Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 19 Feb '16

19 Feb '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver. For now just the SPDIF transmitter has been tested on a Mele A2000(A10) and a Itead Ibox(A20). In order for this patch set to be functional we require an audio clock patch which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK --- Changes since v5: - fixed warning that had sneaked in.' Changes since v4: - corrected indentation issues - removed old macros - removed unnecessary enabling of apb clock Changes since v3: - removed code required for spdif capture - added disable/enable of apb clk to suspend/resume. - removed unnecessary dt checks in probe - fixed error conditions in probe/resume - modified pm_runtime functionality and confirmed that driver worked when PM was disabled. - removed .owner from platform driver - renamed bindings file Changes since v2: - removed sunxi-machine-spdif and replaced with simple audio card - removed untested compatibilities from device tree documentation - added pm_runtime and remove shutdown - removed pll2 as it's the parent of the spdif clock - rename clk to spdif_clk - removed enabling of mclk output - removed interrupts status being cleared as it's not being used - use default fifo settings for now - fixed alignments for wrapped lines Changes since v1: - Moved sunxi-machine-spdif.c to separate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into separate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. Marcus Cooper (2): dt-bindings: sunxi :add sun4i SPDIF transceiver ASoC: sunxi: Add support for the SPDIF block .../bindings/sound/sunxi,sun4i-spdif.txt | 39 ++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 550 +++++++++++++++++++++ 4 files changed, 598 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c -- 2.7.1

3 6

[alsa-devel] [PATCH 0/2] ALSA: oxfw: implement retries for scs1x at transaction failure
by Takashi Sakamoto 19 Feb '16

19 Feb '16

Hi, Current ALSA oxfw driver has a TODO to retry IEEE 1394 asynchronous transactions when transaction failure occurs. This patchset is for it. Additionally, this patchset classifies reason of the failure and stop the retries to prevent from endless loop of scheduling work. Regards Takashi Sakamoto (2): ALSA: oxfw: retry MIDI transferring for scs1x at transaction failure ALSA: oxfw: discontinue MIDI substream for scs1x at transaction failure sound/firewire/oxfw/oxfw-scs1x.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) -- 2.5.0

2 11

[alsa-devel] [PATCH v7 0/9] ASoC: hdac_hdmi: Update to HDMI driver
by Subhransu S. Prusty 19 Feb '16

19 Feb '16

This is remaining patches of v6 and appended platform changes for enumerating HDMI codec Jeeja KP (1): ASoC: hdac_hdmi: Add PM support Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep codec power active during enumeration. Subhransu S. Prusty (6): ASoC: hdac_hdmi: Fix possible memory leak in hw_params ASoC: hdac_hdmi: Don't fail in dai startup to make userland happy ASoC: hdac_hdmi: Fix to reconfigure registers in runtime resume ASoC: hdac_hdmi: Fix to wait for D3 before powering off codec ASoC: Intel: Skylake: Fix possible memory corruption in codec dai dma params ASoC: Intel: Skylake: Add HDMI FE and BE cpu dais Vinod Koul (1): ASoC: Intel: Skylake: Add i915 enabling in skl probe sound/soc/codecs/hdac_hdmi.c | 178 +++++++++++++++++++++++++++++++------- sound/soc/intel/Kconfig | 1 + sound/soc/intel/skylake/skl-pcm.c | 84 ++++++++++++++++-- sound/soc/intel/skylake/skl.c | 51 +++++++++++ 4 files changed, 275 insertions(+), 39 deletions(-) -- 1.9.1

2 16

[alsa-devel] [PATCH 0/4] Add I2S/ADV7511 audio support for ARC AXS10x boards.
by Carlos Palminha 19 Feb '16

19 Feb '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Jose Abreu (4): drm/i2c/adv7511: Add audio support ARC: axs10x: Update defconfigs so that audio is enabled ASoC: dwc: Add I2S HDMI audio support using custom platform ARC: axs10x: Update defconfigs so that I2S is enabled arch/arc/boot/dts/axs10x_mb.dtsi | 20 +- arch/arc/configs/axs101_defconfig | 4 + arch/arc/configs/axs103_defconfig | 6 + arch/arc/configs/axs103_smp_defconfig | 6 + drivers/gpu/drm/i2c/Kconfig | 8 + drivers/gpu/drm/i2c/Makefile | 2 + drivers/gpu/drm/i2c/adv7511.c | 1024 --------------------------------- drivers/gpu/drm/i2c/adv7511.h | 41 ++ drivers/gpu/drm/i2c/adv7511_audio.c | 310 ++++++++++ drivers/gpu/drm/i2c/adv7511_core.c | 1005 ++++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/Makefile | 4 +- sound/soc/dwc/designware_i2s.c | 139 ++++- sound/soc/dwc/designware_pcm.c | 264 +++++++++ sound/soc/dwc/designware_pcm.h | 21 + 16 files changed, 1813 insertions(+), 1043 deletions(-) delete mode 100644 drivers/gpu/drm/i2c/adv7511.c create mode 100644 drivers/gpu/drm/i2c/adv7511_audio.c create mode 100644 drivers/gpu/drm/i2c/adv7511_core.c create mode 100644 sound/soc/dwc/designware_pcm.c create mode 100644 sound/soc/dwc/designware_pcm.h -- 2.5.0

1 3

[alsa-devel] [RFC v2 0/6] sti: add audio interface to the hdmi driver
by Arnaud Pouliquen 19 Feb '16

19 Feb '16

V2: - patch: ALSA: pcm: add IEC958 channel status control helper - Return 1 instead of 0 in snd_pcm_iec958_put - Add .access field in control structure - I have kept condition on mutex for flexibility (but could be cleaned to force user to use a mutex) - patch: video: hdmi: add helper function for N and CTS - Duplicate function to have a separte treatment for coherent and non-coherent clcocks - Add ratio fiel for alternate CTS value - Clock frequency in Hz for TMDS and audio clocks - Add information concerning clocks and CTS calculation. V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Arnaud Pouliquen (6): video: hdmi: add helper function for N and CTS ALSA: pcm: add IEC958 channel status control helper ASoC: core: add code to complete dai init after pcm creation drm: sti: Add ASoC generic hdmi codec support. ASoc: hdmi-codec: add IEC control. ARM: DT: b2120: add audio HDMI dai link in audio card arch/arm/boot/dts/stihxxx-b2120.dtsi | 15 ++- drivers/gpu/drm/sti/Kconfig | 1 + drivers/gpu/drm/sti/sti_hdmi.c | 248 ++++++++++++++++++++++++++++++++--- drivers/gpu/drm/sti/sti_hdmi.h | 10 ++ drivers/video/hdmi.c | 222 +++++++++++++++++++++++++++++++ include/linux/hdmi.h | 22 ++++ include/sound/hdmi-codec.h | 1 + include/sound/pcm_iec958.h | 16 +++ include/sound/soc-dai.h | 7 + sound/core/pcm_iec958.c | 99 ++++++++++++++ sound/soc/codecs/hdmi-codec.c | 59 +++++++-- sound/soc/soc-core.c | 14 ++ 12 files changed, 682 insertions(+), 32 deletions(-) -- 1.9.1

4 17

[alsa-devel] [PATCH] ALSA: hda - hdmi get jack from hda_jack_tbl when not dyn_pcm_assign
by libin.yang＠linux.intel.com 19 Feb '16

19 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> On Intel platform, if !dyn_pcm_assign, spec->pcm_rec[].jack is not NULL even after snd_hda_jack_tbl_clear() is called to free snd_jack. This may cause access invalid memory when calling snd_jack_report. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 30 ++++++++++++++++++++++++++---- 1 file changed, 26 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f4443b5..541986f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1956,6 +1956,29 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, return ret; } +static struct snd_jack *pin_idx_to_jack(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + struct hdmi_spec *spec = codec->spec; + struct snd_jack *jack = NULL; + struct hda_jack_tbl *jack_tbl; + + /* if !dyn_pcm_assign, get jack from hda_jack_tbl + * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not + * NULL even after snd_hda_jack_tbl_clear() is called to + * free snd_jack. This may cause access invalid memory + * when calling snd_jack_report + */ + if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) + jack = spec->pcm_rec[per_pin->pcm_idx].jack; + else if (!spec->dyn_pcm_assign) { + jack_tbl = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + if (jack_tbl) + jack = jack_tbl->jack; + } + return jack; +} + /* update ELD and jack state via audio component */ static void sync_eld_via_acomp(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) @@ -1989,11 +2012,10 @@ static void sync_eld_via_acomp(struct hda_codec *codec, /* pcm_idx >=0 before update_eld() means it is in monitor * disconnected event. Jack must be fetched before update_eld() */ - if (per_pin->pcm_idx >= 0) - jack = spec->pcm_rec[per_pin->pcm_idx].jack; + jack = pin_idx_to_jack(codec, per_pin); update_eld(codec, per_pin, eld); - if (jack == NULL && per_pin->pcm_idx >= 0) - jack = spec->pcm_rec[per_pin->pcm_idx].jack; + if (jack == NULL) + jack = pin_idx_to_jack(codec, per_pin); if (jack == NULL) goto unlock; snd_jack_report(jack, -- 1.9.1

2 1

Re: [alsa-devel] [pulseaudio-discuss] [PATCH 0/1] alsa: Make profiles' availability dynamic
by Raymond Yau 19 Feb '16

19 Feb '16

2016-2-18 下午5:38於 "Bob Ham" <bob.ham(a)collabora.com>寫道： > > Make ALSA profiles stick around and update their availability if the > HDMI ELD changes. This alters the presence of card profiles quite > dramatically. Specifically, the existing autogeneration of profiles > generates a large set of input/output mapping combinations and all of > these combinations are reported by the API, marked as unavailable. To > give an idea of the consequences, here is what pavucontrol looks like > with this patch: > > https://people.collabora.com/~rah/pavucontrol-unavailable-profiles.png > > Similarly, here is the output of "pactl list cards": > > https://people.collabora.com/~rah/pactl-list-cards-output.txt > AFAIK, 1) mono is not supported by hda-intel 2) those desktop with three jacks at rear panel need retasking input jacks to output jacks by changing control channel mode from 2ch to 6ch, you may need to add hook for alsa devices to change the channel mode in alsa HDA-Intel.conf 3) even HDMI codec support 8 channels, it still require your hdmi monitor support multi channels

1 0

[alsa-devel] [PATCH] ASoC: samsung: Use IRQ safe spin lock calls
by Charles Keepax 19 Feb '16

19 Feb '16

Lockdep warns of a potential lock inversion, i2s->lock is held numerous times whilst we are under the substream lock (snd_pcm_stream_lock). If we use the IRQ unsafe spin lock calls, you can also end up locking snd_pcm_stream_lock whilst under i2s->lock (if an IRQ happens whilst we are holding i2s->lock). This could result in deadlock. [ 18.147001] CPU0 CPU1 [ 18.151509] ---- ---- [ 18.156022] lock(&(&pri_dai->spinlock)->rlock); [ 18.160701] local_irq_disable(); [ 18.166622] lock(&(&substream->self_group.lock)->rlock); [ 18.174595] lock(&(&pri_dai->spinlock)->rlock); [ 18.181806] <Interrupt> [ 18.184408] lock(&(&substream->self_group.lock)->rlock); [ 18.190045] [ 18.190045] *** DEADLOCK *** This patch changes to using the irq safe spinlock calls, to avoid this issue. Fixes: ce8bcdbb61d9 ("ASoC: samsung: i2s: Protect more registers with a spinlock") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/i2s.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 84d9e77..70a2559 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -481,10 +481,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, unsigned int cdcon_mask = 1 << i2s_regs->cdclkcon_off; unsigned int rsrc_mask = 1 << i2s_regs->rclksrc_off; u32 mod, mask, val = 0; + unsigned long flags; - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); switch (clk_id) { case SAMSUNG_I2S_OPCLK: @@ -575,11 +576,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, return -EINVAL; } - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); mod = (mod & ~mask) | val; writel(mod, i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); return 0; } @@ -590,6 +591,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, struct i2s_dai *i2s = to_info(dai); int lrp_shift, sdf_shift, sdf_mask, lrp_rlow, mod_slave; u32 mod, tmp = 0; + unsigned long flags; lrp_shift = i2s->variant_regs->lrp_off; sdf_shift = i2s->variant_regs->sdf_off; @@ -649,7 +651,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); /* * Don't change the I2S mode if any controller is active on this @@ -657,7 +659,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, */ if (any_active(i2s) && ((mod & (sdf_mask | lrp_rlow | mod_slave)) != tmp)) { - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; @@ -666,7 +668,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, mod &= ~(sdf_mask | lrp_rlow | mod_slave); mod |= tmp; writel(mod, i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); return 0; } @@ -676,6 +678,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, { struct i2s_dai *i2s = to_info(dai); u32 mod, mask = 0, val = 0; + unsigned long flags; if (!is_secondary(i2s)) mask |= (MOD_DC2_EN | MOD_DC1_EN); @@ -744,11 +747,11 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); mod = readl(i2s->addr + I2SMOD); mod = (mod & ~mask) | val; writel(mod, i2s->addr + I2SMOD); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); -- 2.1.4

4 3

[alsa-devel] Closing the pcm streams
by Andy Ng 19 Feb '16

19 Feb '16

Hello, I am working on a system that a DSP provides the audio algorithms and in case where the DSP crashes, I would like to shutdown nicely any opened PCM streams. The DSP has an RPC protocol to the external world. I use a small watchdog on ARM to monitor the "health" of the DSP. When the DSP goes down, I reset it. However, I would like to close down smoothly the PCM streams and inform the user that there was an error while playing. Is there any way to inform the ALSA kernel framework in the kernel that a specific PCM stream gone bad? Where is the best point to reset ALSA kernel framework? Thank you Andy

2 1

[alsa-devel] [PATCH] Changed the if-else case to switch case
by anish kumar 18 Feb '16

18 Feb '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/codecs/max9867.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index b9d989f..5a8481b 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -212,16 +212,22 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream, switch (snd_pcm_format_width(params_format(params))) { case 8: case 16: - if (pclk_bclk_ratio == 2) + switch (pclk_bclk_ratio) { + case 2: bclk_value = MAX9867_IFC1B_PCLK_2; - else if (pclk_bclk_ratio == 4) + break; + case 4: bclk_value = MAX9867_IFC1B_PCLK_4; - else if (pclk_bclk_ratio == 8) + break; + case 8: bclk_value = MAX9867_IFC1B_PCLK_8; - else if (pclk_bclk_ratio == 16) + break; + case 16: bclk_value = MAX9867_IFC1B_PCLK_16; - else { - dev_err(codec->dev, "unsupported sampling rate\n"); + break; + default: + dev_err(codec->dev, + "unsupported sampling rate\n"); return -EINVAL; } break; -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: Add max9867 codec driver
by anish kumar 18 Feb '16

18 Feb '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max9867.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max9867.c | 537 +++++++++++++++++++++ sound/soc/codecs/max9867.h | 83 ++++ 5 files changed, 642 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max9867.txt create mode 100755 sound/soc/codecs/max9867.c create mode 100755 sound/soc/codecs/max9867.h diff --git a/Documentation/devicetree/bindings/sound/max9867.txt b/Documentation/devicetree/bindings/sound/max9867.txt new file mode 100644 index 0000000..394cd4e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9867.txt @@ -0,0 +1,17 @@ +max9867 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max9867" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max9867: max9867@0x18 { + compatible = "maxim,max9867"; + reg = <0x18>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c3178e3..913c83c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C select SND_SOC_MAX9850 if I2C @@ -490,6 +491,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX9867 + tristate + config SND_SOC_MAX98925 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 84843f3..f39dee7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -70,6 +70,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max9850-objs := max9850.o diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c new file mode 100755 index 0000000..b9d989f --- /dev/null +++ b/sound/soc/codecs/max9867.c @@ -0,0 +1,537 @@ +/* + * max9867.c -- max9867 ALSA SoC Audio driver + * + * Copyright 2013-15 Maxim Integrated Products + * + * 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/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max9867.h" + +static const char *const max9867_spmode[] = { + "Stereo Diff", "Mono Diff", + "Stereo Cap", "Mono Cap", + "Stereo Single", "Mono Single", + "Stereo Single Fast", "Mono Single Fast" +}; +static const char *const max9867_sidetone_text[] = { + "None", "Left", "Right", "LeftRight", "LeftRightDiv2", +}; +static const char *const max9867_filter_text[] = {"IIR", "FIR"}; + +static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, + max9867_filter_text); +static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, + max9867_spmode); +static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, + max9867_sidetone_text); +static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); +static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); +static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); +static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); +static const unsigned int max98088_micboost_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + +static const struct snd_kcontrol_new max9867_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 63, 1), + SOC_DOUBLE_R_TLV("Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, + 0, 15, 1, max9860_capture_tlv), + SOC_DOUBLE_R_TLV("Mic Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 31, 1, max9860_mic_tlv), + SOC_DOUBLE_R_TLV("Mic Boost Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 3, 0, max98088_micboost_tlv), + SOC_ENUM("Digital Sidetone Src", max9867_sidetone), + SOC_SINGLE("Sidetone Volume", MAX9867_DACGAIN, 0, 31, 1), + SOC_SINGLE("DAC Volume", MAX9867_DACLEVEL, 4, 3, 0), + SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), + SOC_SINGLE_TLV("ADC Left Volume", MAX9867_ADCLEVEL, + 4, 15, 1, max9860_adc_left_tlv), + SOC_SINGLE_TLV("ADC Right Volume", MAX9867_ADCLEVEL, + 0, 15, 1, max9860_adc_right_tlv), + SOC_ENUM("Speaker Mode", max9867_spkmode), + SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), + SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_ENUM("DSP Filter", max9867_filter), +}; + +static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; + +static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, + MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, + max9867_mux); + +static const struct snd_kcontrol_new max9867_dapm_mux_controls = + SOC_DAPM_ENUM("Route", max9867_mux_enum); + +static const struct snd_kcontrol_new max9867_left_dapm_control = + SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); +static const struct snd_kcontrol_new max9867_right_dapm_control = + SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); +static const struct snd_kcontrol_new max9867_line_dapm_control = + SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); + +static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), + SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HPOUT"), + + SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), + SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &max9867_dapm_mux_controls), + + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, + &max9867_left_dapm_control), + SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, + &max9867_right_dapm_control), + SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, + &max9867_line_dapm_control), + SND_SOC_DAPM_INPUT("LINE_IN"), +}; + +static const struct snd_soc_dapm_route max9867_audio_map[] = { + {"Left DAC", NULL, "DAI_OUT"}, + {"Right DAC", NULL, "DAI_OUT"}, + {"Output Mixer", NULL, "Left DAC"}, + {"Output Mixer", NULL, "Right DAC"}, + {"HPOUT", NULL, "Output Mixer"}, + + {"Left ADC", NULL, "DAI_IN"}, + {"Right ADC", NULL, "DAI_IN"}, + {"Input Mixer", NULL, "Left ADC"}, + {"Input Mixer", NULL, "Right ADC"}, + {"Input Mux", "Line", "Input Mixer"}, + {"Input Mux", "Mic", "Input Mixer"}, + {"Input Mux", "Mic_Line", "Input Mixer"}, + {"Right Line", "Switch", "Input Mux"}, + {"Left Line", "Switch", "Input Mux"}, + {"LINE_IN", NULL, "Left Line"}, + {"LINE_IN", NULL, "Right Line"}, +}; + +enum rates { + pcm_rate_8, pcm_rate_16, pcm_rate_24, + pcm_rate_32, pcm_rate_44, + pcm_rate_48, max_pcm_rate, +}; + +struct ni_div_rates { + u32 mclk; + u16 ni[max_pcm_rate]; +} ni_div[] = { + {11289600, {0x116A, 0x22D4, 0x343F, 0x45A9, 0x6000, 0x687D} }, + {12000000, {0x1062, 0x20C5, 0x3127, 0x4189, 0x5A51, 0x624E} }, + {12288000, {0x1000, 0x2000, 0x3000, 0x4000, 0x5833, 0x6000} }, + {13000000, {0x0F20, 0x1E3F, 0x2D5F, 0x3C7F, 0x535F, 0x5ABE} }, + {19200000, {0x0A3D, 0x147B, 0x1EB8, 0x28F6, 0x3873, 0x3D71} }, + {24000000, {0x1062, 0x20C5, 0x1893, 0x4189, 0x5A51, 0x624E} }, + {26000000, {0x0F20, 0x1E3F, 0x16AF, 0x3C7F, 0x535F, 0x5ABE} }, + {27000000, {0x0E90, 0x1D21, 0x15D8, 0x3A41, 0x5048, 0x5762} }, +}; + +static inline int get_ni_value(int mclk, int rate) +{ + int i, ret = 0; + + /* find the closest rate index*/ + for (i = 0; i < ARRAY_SIZE(ni_div); i++) { + if (ni_div[i].mclk >= mclk) + break; + } + + switch (rate) { + case 8000: + return ni_div[i].ni[pcm_rate_8]; + case 16000: + return ni_div[i].ni[pcm_rate_16]; + case 32000: + return ni_div[i].ni[pcm_rate_32]; + case 44100: + return ni_div[i].ni[pcm_rate_44]; + case 48000: + return ni_div[i].ni[pcm_rate_48]; + default: + pr_err("%s wrong rate %d\n", __func__, rate); + ret = -EINVAL; + } + return ret; +} + +static int max9867_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + unsigned int ni_h, ni_l, value; + + value = get_ni_value(max9867->sysclk, params_rate(params)); + if (value < 0) + return value; + + ni_h = (0xFF00 & value) >> 8; + ni_l = 0x00FF & value; + /* set up the ni value */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_NI_HIGH_MASK, ni_h); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_NI_LOW_MASK, ni_l); + if (!max9867->master) { + /* + * digital pll locks on to any externally supplied LRCLK signal + * and also enable rapid lock mode. + */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_PLL, MAX9867_PLL); + } else { + unsigned long int bclk_rate, pclk_bclk_ratio; + int bclk_value; + + bclk_rate = params_rate(params) * 2 * + snd_pcm_format_width(params_format(params)); + pclk_bclk_ratio = max9867->pclk/bclk_rate; + switch (snd_pcm_format_width(params_format(params))) { + case 8: + case 16: + if (pclk_bclk_ratio == 2) + bclk_value = MAX9867_IFC1B_PCLK_2; + else if (pclk_bclk_ratio == 4) + bclk_value = MAX9867_IFC1B_PCLK_4; + else if (pclk_bclk_ratio == 8) + bclk_value = MAX9867_IFC1B_PCLK_8; + else if (pclk_bclk_ratio == 16) + bclk_value = MAX9867_IFC1B_PCLK_16; + else { + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + break; + case 24: + bclk_value = MAX9867_IFC1B_24BIT; + break; + case 32: + bclk_value = MAX9867_IFC1B_32BIT; + break; + default: + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + regmap_update_bits(max9867->regmap, MAX9867_IFC1B, + MAX9867_IFC1B_BCLK_MASK, bclk_value); + } + return 0; +} + +static int max9867_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN_MASK, MAX9867_SHTDOWN_MASK); + return 0; +} + +static int max9867_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + if (mute) + regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); + else + regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + MAX9867_DAC_MUTE_MASK, 0); + return 0; +} + +static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + int value = 0; + + /* Set the prescaler based on the master clock frequency*/ + if (freq >= 10000000 && freq <= 20000000) { + value |= MAX9867_PSCLK_10_20; + max9867->pclk = freq; + } else if (freq >= 20000000 && freq <= 40000000) { + value |= MAX9867_PSCLK_20_40; + max9867->pclk = freq/2; + } else if (freq >= 40000000 && freq <= 60000000) { + value |= MAX9867_PSCLK_40_60; + max9867->pclk = freq/4; + } else { + pr_err("bad clock frequency %d", freq); + return -EINVAL; + } + value = value << MAX9867_PSCLK_SHIFT; + max9867->sysclk = freq; + /* exact integer mode is not supported */ + value &= ~MAX9867_FREQ_MASK; + regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, + MAX9867_PSCLK_MASK, value); + return 0; +} + +static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + u8 iface1A = 0, iface1B = 0; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + max9867->master = 1; + iface1A |= MAX9867_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + max9867->master = 0; + iface1A &= ~MAX9867_MASTER; + break; + default: + return -EINVAL; + } + + /* for i2s compatible mode */ + iface1A |= MAX9867_I2S_DLY; + /* SDOUT goes to hiz state after all data is transferred */ + iface1A |= MAX9867_SDOUT_HIZ; + + /* Clock inversion bits, BCI and WCI */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface1A |= MAX9867_WCI_MODE | MAX9867_BCI_MODE; + break; + case SND_SOC_DAIFMT_IB_NF: + iface1A |= MAX9867_BCI_MODE; + break; + case SND_SOC_DAIFMT_NB_IF: + iface1A |= MAX9867_WCI_MODE; + break; + default: + return -EINVAL; + } + + ret = regmap_write(max9867->regmap, MAX9867_IFC1A, iface1A); + ret = regmap_write(max9867->regmap, MAX9867_IFC1B, iface1B); + return 0; +} + +static struct snd_soc_dai_ops max9867_dai_ops = { + .set_fmt = max9867_dai_set_fmt, + .set_sysclk = max9867_set_dai_sysclk, + .prepare = max9867_prepare, + .digital_mute = max9867_mute, + .hw_params = max9867_dai_hw_params, +}; + +#define MAX9867_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define MAX9867_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_driver max9867_dai[] = { + { + .name = "max9867-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9867_RATES, + .formats = MAX9867_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9867_RATES, + .formats = MAX9867_FORMATS, + }, + .ops = &max9867_dai_ops, + } +}; + +#ifdef CONFIG_PM_SLEEP +static int max9867_suspend(struct device *dev) +{ + struct max9867_priv *max9867 = dev_get_drvdata(dev); + + /* Drop down to power saving mode when system is suspended */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN_MASK, ~MAX9867_SHTDOWN_MASK); + return 0; +} + +static int max9867_resume(struct device *dev) +{ + struct max9867_priv *max9867 = dev_get_drvdata(dev); + + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_SHTDOWN_MASK, MAX9867_SHTDOWN_MASK); + return 0; +} +#endif + +static int max9867_probe(struct snd_soc_codec *codec) +{ + struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "max98090_probe\n"); + max9867->codec = codec; + return 0; +} + +static struct snd_soc_codec_driver max9867_codec = { + .probe = max9867_probe, + .controls = max9867_snd_controls, + .num_controls = ARRAY_SIZE(max9867_snd_controls), + .dapm_routes = max9867_audio_map, + .num_dapm_routes = ARRAY_SIZE(max9867_audio_map), + .dapm_widgets = max9867_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9867_dapm_widgets), +}; + +static bool max9867_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9867_STATUS: + case MAX9867_JACKSTATUS: + case MAX9867_AUXHIGH: + case MAX9867_AUXLOW: + return true; + default: + return false; + } +} + +static struct reg_default max9867_reg[] = { + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x0B, 0x00 }, + { 0x0C, 0x00 }, + { 0x0D, 0x00 }, + { 0x0E, 0x00 }, + { 0x0F, 0x00 }, + { 0x10, 0x00 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, +}; + +static struct regmap_config max9867_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX9867_REVISION, + .reg_defaults = max9867_reg, + .num_reg_defaults = ARRAY_SIZE(max9867_reg), + .volatile_reg = max9867_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max9867_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max9867_priv *max9867; + int ret = 0, reg; + + max9867 = devm_kzalloc(&i2c->dev, + sizeof(*max9867), GFP_KERNEL); + if (!max9867) + return -ENOMEM; + + i2c_set_clientdata(i2c, max9867); + max9867->regmap = devm_regmap_init_i2c(i2c, &max9867_regmap); + if (IS_ERR(max9867->regmap)) { + ret = PTR_ERR(max9867->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + ret = regmap_read(max9867->regmap, + MAX9867_REVISION, &reg); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read: %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device revision: %x\n", reg); + ret = snd_soc_register_codec(&i2c->dev, &max9867_codec, + max9867_dai, ARRAY_SIZE(max9867_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max9867_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max9867_i2c_id[] = { + { "max9867", 0 }, +}; + +static const struct of_device_id max9867_of_match[] = { + { .compatible = "maxim,max9867", }, + { } +}; + +MODULE_DEVICE_TABLE(i2c, max9867_i2c_id); + +static const struct dev_pm_ops max9867_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(max9867_suspend, max9867_resume) +}; + +static struct i2c_driver max9867_i2c_driver = { + .driver = { + .name = "max9867", + .of_match_table = of_match_ptr(max9867_of_match), + .pm = &max9867_pm_ops, + }, + .probe = max9867_i2c_probe, + .remove = max9867_i2c_remove, + .id_table = max9867_i2c_id, +}; + +module_i2c_driver(max9867_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX9867 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h new file mode 100755 index 0000000..65590b4 --- /dev/null +++ b/sound/soc/codecs/max9867.h @@ -0,0 +1,83 @@ +/* + * max9867.h -- MAX9867 ALSA SoC Audio driver + * + * Copyright 2013-2015 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX9867_H +#define _MAX9867_H + +/* MAX9867 register space */ + +#define MAX9867_STATUS 0x00 +#define MAX9867_JACKSTATUS 0x01 +#define MAX9867_AUXHIGH 0x02 +#define MAX9867_AUXLOW 0x03 +#define MAX9867_INTEN 0x04 +#define MAX9867_SYSCLK 0x05 +#define MAX9867_FREQ_MASK 0xF +#define MAX9867_PSCLK_SHIFT 0x4 +#define MAX9867_PSCLK_WIDTH 0x2 +#define MAX9867_PSCLK_MASK (0x03<<MAX9867_PSCLK_SHIFT) +#define MAX9867_PSCLK_10_20 0x1 +#define MAX9867_PSCLK_20_40 0x2 +#define MAX9867_PSCLK_40_60 0x3 +#define MAX9867_AUDIOCLKHIGH 0x06 +#define MAX9867_NI_HIGH_WIDTH 0x7 +#define MAX9867_NI_HIGH_MASK 0x7F +#define MAX9867_NI_LOW_MASK 0x7F +#define MAX9867_NI_LOW_SHIFT 0x1 +#define MAX9867_PLL (1<<7) +#define MAX9867_AUDIOCLKLOW 0x07 +#define MAX9867_RAPID_LOCK 0x01 +#define MAX9867_IFC1A 0x08 +#define MAX9867_MASTER (1<<7) +#define MAX9867_I2S_DLY (1<<4) +#define MAX9867_SDOUT_HIZ (1<<3) +#define MAX9867_TDM_MODE (1<<2) +#define MAX9867_WCI_MODE (1<<6) +#define MAX9867_BCI_MODE (1<<5) +#define MAX9867_IFC1B 0x09 +#define MAX9867_IFC1B_BCLK_MASK 7 +#define MAX9867_IFC1B_32BIT 0x01 +#define MAX9867_IFC1B_24BIT 0x02 +#define MAX9867_IFC1B_PCLK_2 4 +#define MAX9867_IFC1B_PCLK_4 5 +#define MAX9867_IFC1B_PCLK_8 6 +#define MAX9867_IFC1B_PCLK_16 7 +#define MAX9867_CODECFLTR 0x0a +#define MAX9867_DACGAIN 0x0b +#define MAX9867_DACLEVEL 0x0c +#define MAX9867_DAC_MUTE_SHIFT 0x6 +#define MAX9867_DAC_MUTE_WIDTH 0x1 +#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) +#define MAX9867_ADCLEVEL 0x0d +#define MAX9867_LEFTLINELVL 0x0e +#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_LEFTVOL 0x10 +#define MAX9867_RIGHTVOL 0x11 +#define MAX9867_LEFTMICGAIN 0x12 +#define MAX9867_RIGHTMICGAIN 0x13 +#define MAX9867_INPUTCONFIG 0x14 +#define MAX9867_INPUT_SHIFT 0x6 +#define MAX9867_MICCONFIG 0x15 +#define MAX9867_MODECONFIG 0x16 +#define MAX9867_PWRMAN 0x17 +#define MAX9867_SHTDOWN_MASK (1<<7) +#define MAX9867_REVISION 0xff + +#define MAX9867_CACHEREGNUM 10 + +/* codec private data */ +struct max9867_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + unsigned int sysclk; + unsigned int pclk; + unsigned int master; +}; +#endif -- 1.9.3

2 2

[alsa-devel] [RFC v4 0/8] ASoC: Add mediatek HDMI codec support
by Philipp Zabel 18 Feb '16

18 Feb '16

Hi, these patches add an interface for Jyri's generic hdmi-codec driver [1] to the mediatek hdmi bridge driver. The accesses to the hcp->eld field are now locked by a mutex, and I have added a patch to use the hdmi helper function [2] for N/CTS calculation. Changes since v3: - Add mutex to lock hcp->eld - Use helper function for N and CTS calculation The mediatek drm patches apply on top of https://patchwork.kernel.org/patch/8337071/ ("drm/mediatek: Add HDMI support"), and need this patch: [2] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") The asoc patches depend on these two patches: https://patchwork.kernel.org/patch/7215121/ ("ALSA: pcm: add IEC958 channel status helper for hw_params") [1] https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the machine driver Philipp Zabel (7): drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver ASoC: mediatek: address dai link array entries by enum video: rmk's HDMI notification prototype drm/mediatek: hdmi: issue notifications ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: hdmi-codec: Add ELD control drm/mediatek: hdmi: use helper function for N and CTS calculation .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- drivers/gpu/drm/mediatek/Kconfig | 1 + drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_drm_hdmi_drv.c | 148 +++++++++++++++++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 89 +++++----- drivers/gpu/drm/mediatek/mtk_hdmi.h | 10 +- drivers/gpu/drm/mediatek/mtk_hdmi_hw.c | 179 ++++----------------- drivers/gpu/drm/mediatek/mtk_hdmi_hw.h | 4 +- drivers/video/Makefile | 2 +- drivers/video/hdmi-not.c | 61 +++++++ include/linux/hdmi-not.h | 39 +++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/hdmi-codec.c | 123 +++++++++++++- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 66 +++++++- 14 files changed, 528 insertions(+), 216 deletions(-) create mode 100644 drivers/video/hdmi-not.c create mode 100644 include/linux/hdmi-not.h -- 2.7.0

2 11

[alsa-devel] [PATCH] ALSA: oxfw: use workqueue instead of tasklet for scs1x
by Takashi Sakamoto 18 Feb '16

18 Feb '16

This commit replaces tasklet with workqueue for scs1x functionality of ALSA oxfw driver. This driver transfers MIDI message specific for SCS.1m and SCS.1d. This task is currently done in software IRQ context of tasklet. In a view of system, this context is limited resources and some important drivers (at least, more important than ALSA oxfw driver) use the context as its bottom-harf. If the work to transfer MIDI messages is done within a time, it's better to use the other context for the work. Actually, with recent CPUs, the work will be scheduled within a time. This is a reason of this commit. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-scs1x.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c index bb53eb3..f7ac124 100644 --- a/sound/firewire/oxfw/oxfw-scs1x.c +++ b/sound/firewire/oxfw/oxfw-scs1x.c @@ -26,7 +26,7 @@ struct fw_scs1x { u8 output_bytes; bool output_escaped; bool output_escape_high_nibble; - struct tasklet_struct tasklet; + struct work_struct work; wait_queue_head_t idle_wait; u8 buffer[HSS1394_MAX_PACKET_SIZE]; bool transaction_running; @@ -129,7 +129,7 @@ static void scs_write_callback(struct fw_card *card, int rcode, ; /* TODO: retry this packet */ scs->transaction_running = false; - tasklet_schedule(&scs->tasklet); + schedule_work(&scs->work); } static bool is_valid_running_status(u8 status) @@ -165,9 +165,9 @@ static bool is_invalid_cmd(u8 status) status == 0xfd; } -static void scs_output_tasklet(unsigned long data) +static void scs_output_work(struct work_struct *work) { - struct fw_scs1x *scs = (struct fw_scs1x *)data; + struct fw_scs1x *scs = container_of(work, struct fw_scs1x, work); struct snd_rawmidi_substream *stream; unsigned int i; u8 byte; @@ -311,7 +311,7 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *stream, int up) scs->output_idle = false; ACCESS_ONCE(scs->output) = stream; - tasklet_schedule(&scs->tasklet); + schedule_work(&scs->work); } else { ACCESS_ONCE(scs->output) = NULL; } @@ -395,7 +395,7 @@ int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw) snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &midi_playback_ops); - tasklet_init(&scs->tasklet, scs_output_tasklet, (unsigned long)scs); + INIT_WORK(&scs->work, scs_output_work); init_waitqueue_head(&scs->idle_wait); scs->output_idle = true; -- 2.5.0

2 1

[alsa-devel] [PATCH v5] portman2x4 - use new parport device model
by Sudip Mukherjee 18 Feb '16

18 Feb '16

Modify portman driver to use the new parallel port device model. The advantage of using the device model is that the device gets binded to the hardware, we get the feature of hotplug, we can bind/unbind the driver at runtime. The changes are in the way the driver gets registered with the parallel port subsystem and the temporary device to probe portman card is removed and portman_probe() is used in the probe callback. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v5: claimed flag and one call to parport_claim() removed. v4: temporary device to probe is removed. v3: changed commit message v2: 1. pardev_cb is initialized while declaring, thus removing the use of memset. 2. used pdev->id. 3. v1 did not have the parport probe callback, but we will need the probe callback for binding as the name of the driver and the name of the device is different. 4. in v1 I missed modifying snd_portman_probe_port(). sound/drivers/portman2x4.c | 93 +++++++++++++++++++++------------------------- 1 file changed, 42 insertions(+), 51 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 464385a..362533f 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -83,8 +83,6 @@ struct portman { struct snd_card *card; struct snd_rawmidi *rmidi; struct pardevice *pardev; - int pardev_claimed; - int open_count; int mode[PORTMAN_NUM_INPUT_PORTS]; struct snd_rawmidi_substream *midi_input[PORTMAN_NUM_INPUT_PORTS]; @@ -648,30 +646,6 @@ static void snd_portman_interrupt(void *userdata) spin_unlock(&pm->reg_lock); } -static int snd_portman_probe_port(struct parport *p) -{ - struct pardevice *pardev; - int res; - - pardev = parport_register_device(p, DRIVER_NAME, - NULL, NULL, NULL, - 0, NULL); - if (!pardev) - return -EIO; - - if (parport_claim(pardev)) { - parport_unregister_device(pardev); - return -EIO; - } - - res = portman_probe(p); - - parport_release(pardev); - parport_unregister_device(pardev); - - return res ? -EIO : 0; -} - static void snd_portman_attach(struct parport *p) { struct platform_device *device; @@ -705,10 +679,20 @@ static void snd_portman_detach(struct parport *p) /* nothing to do here */ } +static int snd_portman_dev_probe(struct pardevice *pardev) +{ + if (strcmp(pardev->name, DRIVER_NAME)) + return -ENODEV; + + return 0; +} + static struct parport_driver portman_parport_driver = { - .name = "portman2x4", - .attach = snd_portman_attach, - .detach = snd_portman_detach + .name = "portman2x4", + .probe = snd_portman_dev_probe, + .match_port = snd_portman_attach, + .detach = snd_portman_detach, + .devmodel = true, }; /********************************************************************* @@ -720,8 +704,7 @@ static void snd_portman_card_private_free(struct snd_card *card) struct pardevice *pardev = pm->pardev; if (pardev) { - if (pm->pardev_claimed) - parport_release(pardev); + parport_release(pardev); parport_unregister_device(pardev); } @@ -736,6 +719,12 @@ static int snd_portman_probe(struct platform_device *pdev) struct snd_card *card = NULL; struct portman *pm = NULL; int err; + struct pardev_cb portman_cb = { + .preempt = NULL, + .wakeup = NULL, + .irq_func = snd_portman_interrupt, /* ISR */ + .flags = PARPORT_DEV_EXCL, /* flags */ + }; p = platform_get_drvdata(pdev); platform_set_drvdata(pdev, NULL); @@ -745,9 +734,6 @@ static int snd_portman_probe(struct platform_device *pdev) if (!enable[dev]) return -ENOENT; - if ((err = snd_portman_probe_port(p)) < 0) - return err; - err = snd_card_new(&pdev->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -759,23 +745,32 @@ static int snd_portman_probe(struct platform_device *pdev) sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); - pardev = parport_register_device(p, /* port */ - DRIVER_NAME, /* name */ - NULL, /* preempt */ - NULL, /* wakeup */ - snd_portman_interrupt, /* ISR */ - PARPORT_DEV_EXCL, /* flags */ - (void *)card); /* private */ + portman_cb.private = card; /* private */ + pardev = parport_register_dev_model(p, /* port */ + DRIVER_NAME, /* name */ + &portman_cb, /* callbacks */ + pdev->id); /* device number */ if (pardev == NULL) { snd_printd("Cannot register pardevice\n"); err = -EIO; goto __err; } + /* claim parport */ + if (parport_claim(pardev)) { + snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); + err = -EIO; + goto free_pardev; + } + err = portman_probe(p); + if (err) { + err = -EIO; + goto release_pardev; + } + if ((err = portman_create(card, pardev, &pm)) < 0) { snd_printd("Cannot create main component\n"); - parport_unregister_device(pardev); - goto __err; + goto release_pardev; } card->private_data = pm; card->private_free = snd_portman_card_private_free; @@ -785,14 +780,6 @@ static int snd_portman_probe(struct platform_device *pdev) goto __err; } - /* claim parport */ - if (parport_claim(pardev)) { - snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); - err = -EIO; - goto __err; - } - pm->pardev_claimed = 1; - /* init device */ if ((err = portman_device_init(pm)) < 0) goto __err; @@ -808,6 +795,10 @@ static int snd_portman_probe(struct platform_device *pdev) snd_printk(KERN_INFO "Portman 2x4 on 0x%lx\n", p->base); return 0; +release_pardev: + parport_release(pardev); +free_pardev: + parport_unregister_device(pardev); __err: snd_card_free(card); return err; -- 1.9.1

2 2

[alsa-devel] another locking issue with nonatomic ops?
by Vinod Koul 18 Feb '16

18 Feb '16

Hi Takashi, We found another issue with non atomic ops. This occurs when we have snd_pcm_do_prepare being invoked which acquires the lock snd_pcm_link_rwsem twice, first in snd_pcm_common_ioctl1, and then again in dpcm_dapm_stream_event. Normally this is fine, but if we have another stream get closed in between two read accesses we get stuck. First thread: [ 419.657259] AudioOut_2 D ffff8800704a3a60 0 3691 1 0x20020004 [ 419.665946] ffff8800704a3a60 00000000704a3a60 ffff88006d1fddd0 ffffffff8252ffd8 [ 419.674678] ffff8800704a4000 ffffffff8252ffc0 ffffffffffffffff ffffffff8252ffd8 [ 419.683513] 0000000000000000 ffff8800704a3a80 ffffffff81ca2987 ffffffffffffffff [ 419.683574] Call Trace: [ 419.692290] [<ffffffff81ca2987>] schedule+0x37/0x90 [ 419.692306] [<ffffffff81ca6f6d>] rwsem_down_read_failed+0xdd/0x130 [ 419.701050] [<ffffffff814df454>] call_rwsem_down_read_failed+0x14/0x30 [ 419.709758] [<ffffffff810e793b>] ? down_read_nested+0x5b/0x70 [ 419.709805] [<ffffffff81adfda9>] ? snd_pcm_stream_lock+0xa9/0x120 [ 419.723012] [<ffffffff81adfda9>] snd_pcm_stream_lock+0xa9/0x120 < ============ 2nd Read lock is acquired here, without releasing previous one. [ 419.723026] [<ffffffff81adfe4c>] snd_pcm_stream_lock_irq+0x2c/0x30 [ 419.731801] [<ffffffff81b01f5e>] ? dpcm_dapm_stream_event+0xbe/0xd0 [ 419.740565] [<ffffffff81b0079c>] dpcm_set_fe_update_state+0x3c/0xb0 [ 419.749252] [<ffffffff81b04514>] dpcm_fe_dai_prepare+0xc4/0x150 [ 419.749301] [<ffffffff81adf7eb>] snd_pcm_do_prepare+0x1b/0x30 [ 419.758083] [<ffffffff81adf3bf>] snd_pcm_action_single+0x2f/0x70 [ 419.766897] [<ffffffff81ca6827>] ? down_read+0x47/0x60 [ 419.766962] [<ffffffff81adf577>] ? snd_pcm_action_nonatomic+0x27/0x80 [ 419.775565] [<ffffffff81adf5c6>] snd_pcm_action_nonatomic+0x76/0x80 < ============ 1st Read lock is acquired here [ 419.784419] [<ffffffff81ae2c62>] snd_pcm_common_ioctl1+0x802/0xd30 [ 419.784495] [<ffffffff81ae332b>] snd_pcm_playback_ioctl1+0x19b/0x280 [ 419.793106] [<ffffffff81230915>] ? __fget+0x5/0x210 [ 419.801943] [<ffffffff81ae3dca>] snd_pcm_ioctl_compat+0x41a/0x770 [ 419.801959] [<ffffffff81230915>] ? __fget+0x5/0x210 [ 419.810793] [<ffffffff8126d670>] compat_SyS_ioctl+0xd0/0x13b0 [ 419.819491] [<ffffffff81ca7bc4>] ? do_nanosleep+0x94/0x190 [ 419.819535] [<ffffffff810eb696>] ? trace_hardirqs_on_caller+0x16/0x210 [ 419.828306] [<ffffffff814df4f8>] ? trace_hardirqs_on_thunk+0x17/0x19 [ 419.837104] [<ffffffff81cabf0e>] sysenter_dispatch+0x7/0x1f Second thread: [ 419.543062] mediaserver D ffff880170957cd8 0 3690 1 0x20020004 [ 419.543191] ffff880170957cd8 0000000070957cd8 ffff88006de49ad0 ffffffff8252ffd8 [ 419.551920] ffff880170958000 ffff88006de49ad0 ffffffff8252ffc0 ffffffff00000001 [ 419.560555] ffffffff8252ffd8 ffff880170957cf8 ffffffff81ca2987 ffffffff8252ffc0 [ 419.569368] Call Trace: [ 419.569392] [<ffffffff81ca2987>] schedule+0x37/0x90 [ 419.578026] [<ffffffff81ca71f5>] rwsem_down_write_failed+0x235/0x450 [ 419.586884] [<ffffffff81ca7005>] ? rwsem_down_write_failed+0x45/0x450 [ 419.586941] [<ffffffff814df483>] call_rwsem_down_write_failed+0x13/0x20 [ 419.595620] [<ffffffff81ca689f>] ? down_write+0x5f/0x80 < ============ Write lock acquired here. If this occurs between two read lock's then it's hangs [ 419.604428] [<ffffffff81aded58>] ? snd_pcm_unlink+0x28/0x120 [ 419.604465] [<ffffffff81aded58>] snd_pcm_unlink+0x28/0x120 [ 419.613215] [<ffffffff81adee5e>] pcm_release_private+0xe/0x10 [ 419.622058] [<ffffffff81ae0d57>] snd_pcm_release_substream.part.34+0x67/0x90 [ 419.622143] [<ffffffff81ae0e68>] snd_pcm_release+0xb8/0xc0 [ 419.630889] [<ffffffff8120f1aa>] __fput+0xea/0x200 [ 419.639680] [<ffffffff8120f30e>] ____fput+0xe/0x10 [ 419.639744] [<ffffffff810ba43c>] task_work_run+0xcc/0x100 [ 419.648435] [<ffffffff81002e41>] do_notify_resume+0x51/0x70 [ 419.657205] [<ffffffff81ca986c>] int_signal+0x12/0x17 Thanks -- ~Vinod

4 13

[alsa-devel] [PATCH] ALSA: hda - hdmi get jack from hda_jack_tbl when not dyn_pcm_assign
by libin.yang＠linux.intel.com 18 Feb '16

18 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> On Intel platform, if !dyn_pcm_assign, spec->pcm_rec[].jack is not NULL even after snd_hda_jack_tbl_clear() is called to free snd_jack. This may cause access invalid memory when calling snd_jack_report. Please see more detail from: https://bugs.freedesktop.org/show_bug.cgi?id=94079 Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f4443b5..3b47101 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1962,6 +1962,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; + struct hda_jack_tbl *jack_tbl; struct snd_jack *jack = NULL; int size; @@ -1989,10 +1990,21 @@ static void sync_eld_via_acomp(struct hda_codec *codec, /* pcm_idx >=0 before update_eld() means it is in monitor * disconnected event. Jack must be fetched before update_eld() */ - if (per_pin->pcm_idx >= 0) + /* if !dyn_pcm_assign, get jack from hda_jack_tbl + * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not + * NULL even after snd_hda_jack_tbl_clear() is called to + * free snd_jack. This may cause access invalid memory + * when calling snd_jack_report + */ + if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) jack = spec->pcm_rec[per_pin->pcm_idx].jack; + else { + jack_tbl = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + if (jack_tbl) + jack = jack_tbl->jack; + } update_eld(codec, per_pin, eld); - if (jack == NULL && per_pin->pcm_idx >= 0) + if (jack == NULL && per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) jack = spec->pcm_rec[per_pin->pcm_idx].jack; if (jack == NULL) goto unlock; -- 1.9.1

3 10

Re: [alsa-devel] Auto mute mode
by Takashi Iwai 18 Feb '16

18 Feb '16

On Wed, 17 Feb 2016 09:31:55 +0100, Kailang wrote: > > Hi Takashi, > > I add the ALC225 for Dell headset mode supported. > But it has the issue for speaker don't mute when headphone or headset was pluged. Is alc_update_headset_mode() called when you plug/unplug the headphone or the headset? If yes, which value is set to new_headset_mode? > The issue is occur to Mic in pluged. > When Mic in is pluged, auto mute mode will change to disable. > But when plug headphone or headset into jack, auto mute mode still keep on disable. > So, speaker will have output at the same time. The same question applies. This is a Dell machine with ALC255_FIXUP_DELL1_MIC_NO_PRESENCE fix up. The mic pin has no jack detection but the detection is triggered only by the headset (HP) pin. All these imply likely that the detection of the headset mode doesn't work properly. I haven't touched with these Dell quirks, but Canonical guys did it. They should have a better clue. Now Cc'ed. Takashi

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[alsa-devel] [PATCH] Provide sequencer sound card number / PID via alsa-lib
by Martin Koegler 17 Feb '16

17 Feb '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This changes adds support for exposing the card number for kernel clients as well as the PID for user client. It supports kernels with and without the required support. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- include/seq.h | 1 + include/sound/asequencer.h | 5 +++-- src/seq/seq.c | 11 +++++++++++ src/seq/seq_hw.c | 8 +++++++- 4 files changed, 22 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index 9576822..77996e5 100644 --- a/include/seq.h +++ b/include/seq.h @@ -143,6 +143,7 @@ snd_seq_client_type_t snd_seq_client_info_get_type(const snd_seq_client_info_t * const char *snd_seq_client_info_get_name(snd_seq_client_info_t *info); int snd_seq_client_info_get_broadcast_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_owner(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..7ebf7fd 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -24,7 +24,7 @@ /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +356,8 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int owner; /* RO: card number[kernel] / PID[user] */ + char reserved[64 - sizeof(int)]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..2505293 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,6 +1522,17 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** + * \brief Get the sound card number (kernel) or owning PID + * \param info client_info container + * \return card number/PID/-1 if value is not available. + */ +int snd_seq_client_info_get_owner(const snd_seq_client_info_t *info) +{ + assert(info); + return info->owner; +} + +/** * \brief (DEPRECATED) Get the event filter bitmap of a client_info container * \param info client_info container * \return NULL if no event filter, or pointer to event filter bitmap diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..a4c083a 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int micro_version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,8 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) + info->owner = -1; return 0; } @@ -368,6 +371,8 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) + info->owner = -1; return 0; } @@ -480,6 +485,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->micro_version = SNDRV_PROTOCOL_MICRO(ver); if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { -- 2.6.2

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: Dereference 'edev' after checking for NULL
by Fabio Estevam 17 Feb '16

17 Feb '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 4a3478debf36 ("ASoC: hdac_hdmi: Add jack reporting") leads to the following Smatch complaint: sound/soc/codecs/hdac_hdmi.c:971 hdac_hdmi_present_sense() warn: variable dereferenced before check 'edev' (see line 967) ,so deference 'edev' after checking for NULL. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/hdac_hdmi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 36f1200..920af2f 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -964,13 +964,15 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) { struct hdac_ext_device *edev = pin->edev; - struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_priv *hdmi; struct hdac_hdmi_pcm *pcm; int val; if (!edev) return; + hdmi = edev->private_data; + pin->repoll_count = repoll; pm_runtime_get_sync(&edev->hdac.dev); -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: hdac_hdmi: Remove 'edev' NULL check
by Fabio Estevam 17 Feb '16

17 Feb '16

Commit 4a3478debf36 ("ASoC: hdac_hdmi: Add jack reporting") leads to the following Smatch complaint: sound/soc/codecs/hdac_hdmi.c:971 hdac_hdmi_present_sense() warn: variable dereferenced before check 'edev' (see line 967) As explained by Subhransu S. Prusty: "This NULL check is not required, can be removed. edev object is expected to be present when the present_sense is called. This is called from places where this edev object is already available." So remove the evdev NULL check. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Remove NULL check (Subhransu) sound/soc/codecs/hdac_hdmi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 36f1200..2884dbf 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -968,9 +968,6 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) struct hdac_hdmi_pcm *pcm; int val; - if (!edev) - return; - pin->repoll_count = repoll; pm_runtime_get_sync(&edev->hdac.dev); -- 1.9.1

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[alsa-devel] [PATCH RFT] ASoC: max98926: Fix blr_clk_ratio calculation
by Axel Lin 17 Feb '16

17 Feb '16

Current code calculates blr_clk_ratio before setting max98926->ch_size, fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Anish, I don't have this h/w, please review and test this patch. thanks, Axel sound/soc/codecs/max98926.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 66884ebd..5245e10 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -383,8 +383,7 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream, int rate = params_rate(params), i; struct snd_soc_codec *codec = dai->codec; struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); - /* BCLK/LRCLK ratio calculation */ - int blr_clk_ratio = params_channels(params) * max98926->ch_size; + int blr_clk_ratio; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -414,6 +413,9 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = params_channels(params) * max98926->ch_size; + switch (blr_clk_ratio) { case 32: regmap_update_bits(max98926->regmap, -- 2.1.4

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[alsa-devel] [RFC v1 0/9] ASoC: Add support to Qualcomm msm8x16-wcd codec
by Srinivas Kandagatla 17 Feb '16

17 Feb '16

This patchset aims at adding msm8x16-wcd codec support. msm8x16-wcd codec is found in Qualcomm msm8916 and apq8016 processors. This codec IP is split in to two parts(Digital & Analog), Analog part is integrated in to PMIC PM8916 and the digital part is integrated into Application processor. Register access to the analog part is done via SPMI interface to PMIC, and registers on the Application processor are memory mapped. Data transfer between Analog and Digital Die is done via a internal bus called PDM. I did split the driver into sensible patches to easy the review process. Most of this driver is ported from downstream Andriod v3.10 kernel. Jack support and Headset button detection code is not added in to this series. This codec support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary Mic. - 2 Digital Microphones. - 2 Mic Bias Circuits. - Earpiece - Headset - Loud Speaker. - Jack Detect. - Headset Button detection circuit. Tested this code on v4.5-rc4 with lpass driver capture support patches for Headset (Mic and speakers), Secondary Mic and Loudspeaker on DB410C. DMIC should work too. Thanks, srini Srinivas Kandagatla (9): ASoC: msm8x176: Add Device Tree bindings. ASoC: msm8x16: add driver structure ASoC: msm8x16: add codec registers definitions ASoC: msm8x16: add ranges for default, readonly ASoC: msm8x16: add codec intialization setup ASoC: msm8x16: add codec dais ASoC: msm8x16: Add sound mixer controls. ASoC: msm8x16: add dapm widgets ASoC: msm8x16: add dapm routes .../devicetree/bindings/sound/qcom,msm8x16-wcd.txt | 91 + include/dt-bindings/sound/msm8x16-wcd.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 +- sound/soc/codecs/msm8x16-wcd-registers.h | 518 +++++ sound/soc/codecs/msm8x16-wcd.c | 2132 ++++++++++++++++++++ sound/soc/codecs/msm8x16-wcd.h | 764 +++++++ 7 files changed, 3518 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8x16-wcd.txt create mode 100644 include/dt-bindings/sound/msm8x16-wcd.h create mode 100644 sound/soc/codecs/msm8x16-wcd-registers.h create mode 100644 sound/soc/codecs/msm8x16-wcd.c create mode 100644 sound/soc/codecs/msm8x16-wcd.h -- 1.9.1

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[alsa-devel] [PATCH v4] portman2x4 - use new parport device model
by Sudip Mukherjee 17 Feb '16

17 Feb '16

Modify portman driver to use the new parallel port device model. The advantage of using the device model is that the device gets binded to the hardware, we get the feature of hotplug, we can bind/unbind the driver at runtime. The changes are in the way the driver gets registered with the parallel port subsystem and the temporary device to probe portman card is removed and portman_probe() is used in the probe callback. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v4: temporary device to probe is removed. v3: changed commit message v2: 1. pardev_cb is initialized while declaring, thus removing the use of memset. 2. used pdev->id. 3. v1 did not have the parport probe callback, but we will need the probe callback for binding as the name of the driver and the name of the device is different. 4. in v1 I missed modifying snd_portman_probe_port(). sound/drivers/portman2x4.c | 79 +++++++++++++++++++++++----------------------- 1 file changed, 40 insertions(+), 39 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 464385a..0741a3b 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -648,30 +648,6 @@ static void snd_portman_interrupt(void *userdata) spin_unlock(&pm->reg_lock); } -static int snd_portman_probe_port(struct parport *p) -{ - struct pardevice *pardev; - int res; - - pardev = parport_register_device(p, DRIVER_NAME, - NULL, NULL, NULL, - 0, NULL); - if (!pardev) - return -EIO; - - if (parport_claim(pardev)) { - parport_unregister_device(pardev); - return -EIO; - } - - res = portman_probe(p); - - parport_release(pardev); - parport_unregister_device(pardev); - - return res ? -EIO : 0; -} - static void snd_portman_attach(struct parport *p) { struct platform_device *device; @@ -705,10 +681,20 @@ static void snd_portman_detach(struct parport *p) /* nothing to do here */ } +static int snd_portman_dev_probe(struct pardevice *pardev) +{ + if (strcmp(pardev->name, DRIVER_NAME)) + return -ENODEV; + + return 0; +} + static struct parport_driver portman_parport_driver = { - .name = "portman2x4", - .attach = snd_portman_attach, - .detach = snd_portman_detach + .name = "portman2x4", + .probe = snd_portman_dev_probe, + .match_port = snd_portman_attach, + .detach = snd_portman_detach, + .devmodel = true, }; /********************************************************************* @@ -736,6 +722,12 @@ static int snd_portman_probe(struct platform_device *pdev) struct snd_card *card = NULL; struct portman *pm = NULL; int err; + struct pardev_cb portman_cb = { + .preempt = NULL, + .wakeup = NULL, + .irq_func = snd_portman_interrupt, /* ISR */ + .flags = PARPORT_DEV_EXCL, /* flags */ + }; p = platform_get_drvdata(pdev); platform_set_drvdata(pdev, NULL); @@ -745,9 +737,6 @@ static int snd_portman_probe(struct platform_device *pdev) if (!enable[dev]) return -ENOENT; - if ((err = snd_portman_probe_port(p)) < 0) - return err; - err = snd_card_new(&pdev->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -759,23 +748,33 @@ static int snd_portman_probe(struct platform_device *pdev) sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); - pardev = parport_register_device(p, /* port */ - DRIVER_NAME, /* name */ - NULL, /* preempt */ - NULL, /* wakeup */ - snd_portman_interrupt, /* ISR */ - PARPORT_DEV_EXCL, /* flags */ - (void *)card); /* private */ + portman_cb.private = card; /* private */ + pardev = parport_register_dev_model(p, /* port */ + DRIVER_NAME, /* name */ + &portman_cb, /* callbacks */ + pdev->id); /* device number */ if (pardev == NULL) { snd_printd("Cannot register pardevice\n"); err = -EIO; goto __err; } + if (parport_claim(pardev)) { + err = -EIO; + goto free_pardev; + } + + err = portman_probe(p); + parport_release(pardev); + + if (err) { + err = -EIO; + goto free_pardev; + } + if ((err = portman_create(card, pardev, &pm)) < 0) { snd_printd("Cannot create main component\n"); - parport_unregister_device(pardev); - goto __err; + goto free_pardev; } card->private_data = pm; card->private_free = snd_portman_card_private_free; @@ -808,6 +807,8 @@ static int snd_portman_probe(struct platform_device *pdev) snd_printk(KERN_INFO "Portman 2x4 on 0x%lx\n", p->base); return 0; +free_pardev: + parport_unregister_device(pardev); __err: snd_card_free(card); return err; -- 1.9.1

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Re: [alsa-devel] ASoC: hdac_hdmi: Add jack reporting
by Dan Carpenter 17 Feb '16

17 Feb '16

Hello Jeeja KP, This is a semi-automatic email about new static checker warnings. The patch 4a3478debf36: "ASoC: hdac_hdmi: Add jack reporting" from Feb 12, 2016, leads to the following Smatch complaint: sound/soc/codecs/hdac_hdmi.c:971 hdac_hdmi_present_sense() warn: variable dereferenced before check 'edev' (see line 967) sound/soc/codecs/hdac_hdmi.c 966 struct hdac_ext_device *edev = pin->edev; 967 struct hdac_hdmi_priv *hdmi = edev->private_data; ^^^^^^^^^^^^^^^^^^ Patch introduces a new dereference. 968 struct hdac_hdmi_pcm *pcm; 969 int val; 970 971 if (!edev) ^^^^^ Existing code assumed that edev could be NULL. 972 return; 973 regards, dan carpenter

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