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

[alsa-devel] [PATCH 00/19] Rework OMAP4+ HDMI audio support
by Jyri Sarha 23 May '14

23 May '14

Since RFC version of the patch set: - Split callbacks removal patch away from "Integrated ASoC DAI component driver implementation" patches for easier reading This set of patches fixes OMAP4+ HDMI audio. The structure of the implementatin looks a bit different than before. Instead of creating a driver specific API for a separate ASoC component driver to connect to, this implementation integrates an the component driver into the HDMI driver. The idea is to use an existing ASoC component driver API instead of creating a new custom API for each HDMI IP and to avoid splitting the driver to half for separate video and audio parts connected with the API. The new implementation also uses simple-audio-card for a machine driver instead of having its own HW specific machine driver. The patches are based on 3.15-rc2 merged with git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next and git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git work/dss-dt-omap5 Everything is pushed here here: git://git.ti.com/~jyrisarha/ti-linux-kernel/jyrisarhas-audio-video-linux-fe… omap-hdmi-audio Best regards, Jyri Jyri Sarha (19): ASoC: omap-pcm: Move omap-pcm under include/sound OMAPDSS: hdmi-common: Add hdmi_dss_audio_from_hw_params() OMAPDSS: hdmi4: Remove callbacks for an external ASoC DAI driver OMAPDSS: hdmi4: Integrated ASoC DAI component driver implementation OMAPDSS: Kconfig: Add depencies and help section to OMAP4_DSS_HDMI_AUDIO OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi5_core: Fix compilation with OMAP5_DSS_HDMI_AUDIO OMAPDSS: hdmi5: Remove callbacks for an external ASoC DAI driver OMAPDSS: hdmi5: Integrated ASoC DAI component driver implementation OMAPDSS: Kconfig: Add depencies and help section to OMAP5_DSS_HDMI_AUDIO ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove obsolete audio code ARM: omap4.dtsi: Add audio related parametes to hdmi node ARM: omap4-panda-common.dtsi: Add HDMI audio nodes ARM: omap5.dtsi: Add audio related parameters to hdmi node ARM: omap5-uevm.dts: Add hdmi audio related nodes ARM: omap2plus_defconfig: Build DSS HDMI support for OMAP4 in kernel ARM: omap2plus_defconfig: Enable OMAP5 HDMI support ARM: omap2plus_defconfig: Enable OMAP4+ HDMI audio support .../devicetree/bindings/video/ti,omap4-dss.txt | 4 + .../devicetree/bindings/video/ti,omap5-dss.txt | 4 + arch/arm/boot/dts/omap4-panda-common.dtsi | 21 +- arch/arm/boot/dts/omap4.dtsi | 2 + arch/arm/boot/dts/omap5-uevm.dts | 19 + arch/arm/boot/dts/omap5.dtsi | 2 + arch/arm/configs/omap2plus_defconfig | 20 +- .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ------ .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 19 +- drivers/video/fbdev/omap2/dss/hdmi.h | 8 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 216 ++++++++---- drivers/video/fbdev/omap2/dss/hdmi5.c | 216 ++++++++---- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 2 +- drivers/video/fbdev/omap2/dss/hdmi_common.c | 117 ++++++- include/sound/omap-pcm.h | 30 ++ include/video/omapdss.h | 31 -- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/omap/Kconfig | 13 - sound/soc/omap/Makefile | 4 - sound/soc/omap/omap-dmic.c | 2 +- sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 -------------------- sound/soc/omap/omap-hdmi.h | 38 -- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/omap/omap-pcm.h | 30 -- 27 files changed, 540 insertions(+), 870 deletions(-) create mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h delete mode 100644 sound/soc/omap/omap-pcm.h -- 1.7.9.5

5 34

[alsa-devel] [PATCH] hda-emu: Add new HDMI commands
by David Henningsson 23 May '14

23 May '14

Some new HDMI commands were added recently to the HDA driver. This patch makes hda-emu support these new commands (or at least, not output errors). Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-int.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ include/hda-types.h | 2 ++ 2 files changed, 55 insertions(+) diff --git a/hda-int.c b/hda-int.c index ef90fe8..6dd862d 100644 --- a/hda-int.c +++ b/hda-int.c @@ -823,6 +823,52 @@ static int get_cvt_channel_count(struct xhda_codec *codec, return node->cvt_channel_count; } +static int get_asp_channel_slot(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + return (node->asp_channel_slot[cmd & 0xf] << 4) + (cmd & 0xf); +} + +static int set_asp_channel_slot(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + node->asp_channel_slot[cmd & 0xf] = (cmd & 0xf0) >> 4; + return 0; +} + +static int set_dip_index(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + return 0; /* FIXME */ +} + +static int set_dip_data(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + return 0; /* FIXME */ +} + +static int get_dip_size(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + if (cmd & 0x8) + return 128; /* ELD buffer size */ + return 32; /* DIP buffer size */ +} + +static int set_dip_xmitctrl(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + node->dip_xmitctrl = cmd & 0xc0; + return 0; +} + +static int get_dip_xmitctrl(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + return node->dip_xmitctrl; +} + /* * parameters @@ -1063,6 +1109,10 @@ static struct xhda_verb_table verb_tbl[] = { { 0x71e, set_config_def_2, "set_config_def_2" }, { 0x71f, set_config_def_3, "set_config_def_3" }, { 0x72d, set_cvt_channel_count, "set_cvt_channel_count" }, + { 0x730, set_dip_index, "set_dip_index" }, + { 0x731, set_dip_data, "set_dip_data" }, + { 0x732, set_dip_xmitctrl, "set_dip_xmitctrl" }, + { 0x734, set_asp_channel_slot, "set_asp_channel_slot" }, { 0x7ff, set_codec_reset, "set_codec_reset" }, { 0xf00, get_parameters, "get_parameters" }, { 0xf01, get_connect_sel, "get_connect_sel" }, @@ -1087,6 +1137,9 @@ static struct xhda_verb_table verb_tbl[] = { { 0xf1c, get_config_default, "get_config_default" }, { 0xf20, get_ssid, "get_ssid" }, { 0xf2d, get_cvt_channel_count, "get_cvt_channel_count" }, + { 0xf2e, get_dip_size, "get_dip_size" }, + { 0xf32, get_dip_xmitctrl, "get_dip_xmitctrl" }, + { 0xf34, get_asp_channel_slot, "get_asp_channel_slot" }, {} }; diff --git a/include/hda-types.h b/include/hda-types.h index fe6b31f..46bfc67 100644 --- a/include/hda-types.h +++ b/include/hda-types.h @@ -58,6 +58,8 @@ struct xhda_node { unsigned char gpio_wake, gpio_unsol, gpio_sticky; unsigned int coef_idx; unsigned char cvt_channel_count; + unsigned char asp_channel_slot[16]; + unsigned char dip_xmitctrl; struct xhda_coef_table *coef_tbl; struct xhda_node *next; }; -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix the cache handling
by Fabio Estevam 23 May '14

23 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. According to Mark Brown: "Yes, reg_cache isn't there if we're not using ASoC level caching. The fix should just be to replace the direct cache references with snd_soc_read()s which will end up in a cache lookup if the register is cached." Do as suggested and also complete the cache array with the missing registers so that sgtl5000_restore_regs() can properly cache the all the registers it needs. Reported-by: Shawn Guo <shawn.guo(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Also fix sgtl5000_set_bias_level so that suspend/resume/play sequence does not fail after several interactions sound/soc/codecs/sgtl5000.c | 50 +++++++++++++++++++++++++-------------------- 1 file changed, 28 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9626ee0..53511c6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -36,18 +36,32 @@ /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_DIG_POWER, 0x0000 }, { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_ADCDAC_CTRL, 0x020c }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_ADC_CTRL, 0x0000 }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINREG_CTRL, 0x0000 }, + { SGTL5000_CHIP_REF_CTRL, 0x0000 }, + { SGTL5000_CHIP_MIC_CTRL, 0x0000 }, + { SGTL5000_CHIP_LINE_OUT_CTRL, 0x0000 }, { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, { SGTL5000_CHIP_ANA_POWER, 0x7060 }, { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_CHIP_CLK_TOP_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_STATUS, 0x0000 }, + { SGTL5000_CHIP_SHORT_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_TEST2, 0x0000 }, + { SGTL5000_DAP_CTRL, 0x0000 }, + { SGTL5000_DAP_PEQ, 0x0000 }, { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_AUDIO_EQ, 0x0000 }, { SGTL5000_DAP_SURROUND, 0x0040 }, { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, @@ -55,6 +69,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_MIX_CHAN, 0x0000 }, { SGTL5000_DAP_AVC_CTRL, 0x0510 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, @@ -924,20 +939,6 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; udelay(10); - - regcache_cache_only(sgtl5000->regmap, false); - - ret = regcache_sync(sgtl5000->regmap); - if (ret != 0) { - dev_err(codec->dev, - "Failed to restore cache: %d\n", ret); - - regcache_cache_only(sgtl5000->regmap, true); - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - return ret; - } } break; @@ -1063,7 +1064,10 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) #ifdef CONFIG_SUSPEND static int sgtl5000_suspend(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(sgtl5000->regmap, true); return 0; } @@ -1075,7 +1079,6 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec) */ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; u16 reg; /* restore regular registers */ @@ -1089,12 +1092,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) reg == SGTL5000_CHIP_REF_CTRL) continue; - snd_soc_write(codec, reg, cache[reg]); + snd_soc_write(codec, reg, snd_soc_read(codec, reg)); } /* restore dap registers */ for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) - snd_soc_write(codec, reg, cache[reg]); + snd_soc_write(codec, reg, snd_soc_read(codec, reg)); /* * restore these regs according to the power setting sequence in @@ -1110,29 +1113,32 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER]); + snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER)); snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_CLK_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_REF_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_LINE_OUT_CTRL)); return 0; } static int sgtl5000_resume(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* Bring the codec back up to standby to enable regulators */ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Restore registers by cached in memory */ sgtl5000_restore_regs(codec); + + regcache_cache_only(sgtl5000->regmap, false); return 0; } #else -- 1.8.3.2

2 2

[alsa-devel] [PATCH 0/2] ASoC: max98090/max98095: Add master clock handing
by Tushar Behera 23 May '14

23 May '14

These CODECs are used on Snow/Peach-pit boards and the master clock needs to be set at 24MHz. Adding support in respective codec drivers so that we can update the master clock frequency from sound card driver through set_sysclk. These patches are prepared as per review comments on following patch. [PATCH] ASoC: samsung: Add 'mclk' handling for Snow sound-card driver https://lkml.org/lkml/2014/5/20/4 Tushar Behera (2): ASoC: max98090: Add master clock handling ASoC: max98095: Add master clock handling .../devicetree/bindings/sound/max98090.txt | 6 ++++++ .../devicetree/bindings/sound/max98095.txt | 6 ++++++ sound/soc/codecs/max98090.c | 13 +++++++++++++ sound/soc/codecs/max98090.h | 1 + sound/soc/codecs/max98095.c | 14 ++++++++++++++ 5 files changed, 40 insertions(+) -- 1.7.9.5

3 15

Re: [alsa-devel] [PATCH 2/2] ASoC: samsung: Add machine driver for odroidx2
by Mark Brown 23 May '14

23 May '14

On Fri, May 23, 2014 at 02:18:39AM +0000, Zhen Chen wrote: > <HTML xmlns:o = "urn:schemas-microsoft-com:office:office"><HEAD><TITLE>Samsung Enterprise Portal mySingle</TITLE> > <META content=IE=5 http-equiv=X-UA-Compatible> > <META content="text/html; charset=windows-1252" http-equiv=Content-Type> Please fix your mail configuration to send in plain text, I can't read this - it's not even got a multipart/alternative plain text bit.

1 0

[alsa-devel] [PATCH] ALSA: Remove deprecated snd_card_create()
by Takashi Iwai 23 May '14

23 May '14

Now all calls have been fixed, let's get rid of the old definition. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 7 ------- 1 file changed, 7 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index d3f5f818e0b9..eedda2cdfe57 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -282,13 +282,6 @@ int snd_card_new(struct device *parent, int idx, const char *xid, struct module *module, int extra_size, struct snd_card **card_ret); -static inline int __deprecated -snd_card_create(int idx, const char *id, struct module *module, int extra_size, - struct snd_card **ret) -{ - return snd_card_new(NULL, idx, id, module, extra_size, ret); -} - int snd_card_disconnect(struct snd_card *card); int snd_card_free(struct snd_card *card); int snd_card_free_when_closed(struct snd_card *card); -- 1.9.3

1 0

Re: [alsa-devel] [ALC1150] combined with Intel H97/Z97
by Takashi Iwai 23 May '14

23 May '14

At Thu, 22 May 2014 21:34:27 +0200, Arthur Borsboom wrote: > > Hi Takashi, > > I have tested the patch and it fixes the issue. > Tested on kernel 3.15.0-rc6 > > Do you think you can get this patch in 3.15? > Or will it be 3.16? I'll queue the patch below for 3.15. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix onboard audio on Intel H97/Z97 chipsets The recent Intel H97/Z97 chipsets need the similar setups like other Intel chipsets for snooping, etc. Especially without snooping, the audio playback stutters or gets corrupted. This fix patch just adds the corresponding PCI ID entry with the proper flags. Reported-and-tested-by: Arthur Borsboom <arthurborsboom(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2c54629d62d1..6cc3cf285558 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1743,6 +1743,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + /* 9 Series */ + { PCI_DEVICE(0x8086, 0x8ca0), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Wellsburg */ { PCI_DEVICE(0x8086, 0x8d20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, -- 1.9.3

1 0

[alsa-devel] [RFC PATCH] ALSA: hda - Add a new quirk match based on default pin configuration
by David Henningsson 23 May '14

23 May '14

Normally, we match on pci ssid only. This works but needs new code for every machine. To catch more machines in the same quirk, let's add a new type of quirk, where we match on 1) PCI Subvendor ID (i e, not device, just vendor) 2) Codec ID 3) Pin configuration default If all these three match, we could be reasonably certain that the quirk should apply to the machine even though it might not be the exact same device. In case a quirk matches both the pci ssid quirk and this type of quirk, then the pci ssid quirk is chosen. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Hi Takashi, Do you think this could be working? If so I'll ask Hui to move some machines over and come up with a patch for that. Two questions: 1) I'm unsure if the "name" entry makes sense here - I mean, since the same quirk hopefully matches many machines, we can't use the machine name here. Should I just skip it? 2) Should I make a macro (like SND_PCI_QUIRK) to make the actual pinquirk tables look more compact? sound/pci/hda/hda_auto_parser.c | 39 ++++++++++++++++++++++++++++++++++----- sound/pci/hda/hda_local.h | 27 +++++++++++++++++++++++---- 2 files changed, 57 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 90d2fda..6904779 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -839,10 +839,22 @@ void snd_hda_apply_fixup(struct hda_codec *codec, int action) } EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); -void snd_hda_pick_fixup(struct hda_codec *codec, - const struct hda_model_fixup *models, - const struct snd_pci_quirk *quirk, - const struct hda_fixup *fixlist) +static bool pin_config_match(struct hda_codec *codec, + const struct hda_pintbl *pins) +{ + for (; pins->nid; pins++) { + u32 def_conf = snd_hda_codec_get_pincfg(codec, pins->nid); + if (pins->val != def_conf) + return false; + } + return true; +} + +void snd_hda_pick_pin_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct snd_hda_pin_quirk *pin_quirk, + const struct hda_fixup *fixlist) { const struct snd_pci_quirk *q; int id = -1; @@ -889,10 +901,27 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } + if (id < 0 && pin_quirk) { + const struct snd_hda_pin_quirk *pq; + for (pq = pin_quirk; pq->subvendor; pq++) { + if (codec->bus->pci->subsystem_vendor != pq->subvendor) + continue; + if (codec->vendor_id != pq->codec) + continue; + if (pin_config_match(codec, pq->pins)) { + id = pq->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + name = pq->name; +#endif + break; + } + } + } + codec->fixup_id = id; if (id >= 0) { codec->fixup_list = fixlist; codec->fixup_name = name; } } -EXPORT_SYMBOL_GPL(snd_hda_pick_fixup); +EXPORT_SYMBOL_GPL(snd_hda_pick_pin_fixup); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e51d155..41a6817 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -425,15 +425,34 @@ enum { HDA_FIXUP_ACT_FREE, }; +struct snd_hda_pin_quirk { + unsigned int codec; /* Codec vendor/device ID */ + unsigned short subvendor; /* PCI subvendor ID */ + const struct hda_pintbl *pins; /* list of matching pins */ +#ifdef CONFIG_SND_DEBUG_VERBOSE + const char *name; +#endif + int value; /* quirk value */ +}; + int snd_hda_add_verbs(struct hda_codec *codec, const struct hda_verb *list); void snd_hda_apply_verbs(struct hda_codec *codec); void snd_hda_apply_pincfgs(struct hda_codec *codec, const struct hda_pintbl *cfg); void snd_hda_apply_fixup(struct hda_codec *codec, int action); -void snd_hda_pick_fixup(struct hda_codec *codec, - const struct hda_model_fixup *models, - const struct snd_pci_quirk *quirk, - const struct hda_fixup *fixlist); +void snd_hda_pick_pin_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct snd_hda_pin_quirk *pin_quirk, + const struct hda_fixup *fixlist); + +static inline void snd_hda_pick_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct hda_fixup *fixlist) +{ + snd_hda_pick_pin_fixup(codec, models, quirk, NULL, fixlist); +} /* * unsolicited event handler -- 1.9.1

2 8

[alsa-devel] [PATCH RFT] ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls
by Axel Lin 23 May '14

23 May '14

The new value argument needs proper shift to match the mask bit fields. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Brian, This patch replaces [PATCH RFT] ASoC: cs42l56: Fix update mute register bits in cs42l56_digital_mute. I just found more places of snd_soc_update_bits calls needs fix. Regards, Axel sound/soc/codecs/cs42l56.c | 71 ++++++++++++++++++---------------------------- sound/soc/codecs/cs42l56.h | 14 ++++----- 2 files changed, 35 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 5bb134b..c0be526 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -763,14 +763,14 @@ static int cs42l56_set_sysclk(struct snd_soc_dai *codec_dai, case CS42L56_MCLK_11P2896MHZ: case CS42L56_MCLK_12MHZ: case CS42L56_MCLK_12P288MHZ: - cs42l56->mclk_div2 = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; cs42l56->mclk_prediv = 0; break; case CS42L56_MCLK_22P5792MHZ: case CS42L56_MCLK_24MHZ: case CS42L56_MCLK_24P576MHZ: - cs42l56->mclk_div2 = 1; - cs42l56->mclk_prediv = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; + cs42l56->mclk_prediv = CS42L56_MCLK_PREDIV; break; default: return -EINVAL; @@ -839,62 +839,47 @@ static int cs42l56_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + unsigned int mask; if (mute) { /* Hit the DSP Mixer first */ - snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, - CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_MUTE); + mask = CS42L56_ADCAMIX_MUTE_MASK | CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | CS42L56_MSTA_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, mask, mask); + /* Mute ADC's */ - snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_MUTE); + mask = CS42L56_ADCA_MUTE_MASK | CS42L56_ADCB_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, mask, mask); + /* HP And LO */ snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); } else { - snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, - CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_UNMUTE); - snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_UNMUTE); + mask = CS42L56_ADCAMIX_MUTE_MASK | CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | CS42L56_MSTA_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, mask , 0); + + mask = CS42L56_ADCA_MUTE_MASK | CS42L56_ADCB_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, mask, 0); + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, 0); } return 0; } diff --git a/sound/soc/codecs/cs42l56.h b/sound/soc/codecs/cs42l56.h index ad2b50a..d2f68c6 100644 --- a/sound/soc/codecs/cs42l56.h +++ b/sound/soc/codecs/cs42l56.h @@ -80,19 +80,21 @@ #define CS42L56_PDN_HPB_MASK 0xc0 /* serial port and clk masks */ -#define CS42L56_MASTER_MODE 1 -#define CS42L56_SLAVE_MODE 0 +#define CS42L56_MASTER_MODE (1 << 6) +#define CS42L56_SLAVE_MODE (0 << 6) #define CS42L56_MS_MODE_MASK 0x40 -#define CS42L56_SCLK_INV 1 +#define CS42L56_SCLK_INV (1 << 5) #define CS42L56_SCLK_INV_MASK 0x20 #define CS42L56_SCLK_MCLK_MASK 0x18 +#define CS42L56_MCLK_PREDIV (1 << 6) #define CS42L56_MCLK_PREDIV_MASK 0x04 +#define CS42L56_MCLK_DIV2 (1 << 1) #define CS42L56_MCLK_DIV2_MASK 0x02 #define CS42L56_MCLK_DIS_MASK 0x01 #define CS42L56_CLK_AUTO_MASK 0x20 #define CS42L56_CLK_RATIO_MASK 0x1f -#define CS42L56_DIG_FMT_I2S 0 -#define CS42L56_DIG_FMT_LEFT_J 1 +#define CS42L56_DIG_FMT_I2S (0 << 3) +#define CS42L56_DIG_FMT_LEFT_J (1 << 3) #define CS42L56_DIG_FMT_MASK 0x08 /* Class H and misc ctl masks */ @@ -116,8 +118,6 @@ #define CS42L56_DEEMPH_MASK 0x40 #define CS42L56_PLYBCK_GANG_MASK 0x10 #define CS42L56_PCM_INV_MASK 0x0c -#define CS42L56_MUTE 1 -#define CS42L56_UNMUTE 0 #define CS42L56_ADCAMIX_MUTE_MASK 0x40 #define CS42L56_ADCBMIX_MUTE_MASK 0x80 #define CS42L56_PCMAMIX_MUTE_MASK 0x10 -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: sgtl5000: Fix the cache handling
by Fabio Estevam 23 May '14

23 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. According to Mark Brown: "Yes, reg_cache isn't there if we're not using ASoC level caching. The fix should just be to replace the direct cache references with snd_soc_read()s which will end up in a cache lookup if the register is cached." Do as suggested and also complete the cache array with the missing registers so that sgtl5000_restore_regs() can properly cache the all the registers it needs. Reported-by: Shawn Guo <shawn.guo(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 39 ++++++++++++++++++++++++++++----------- 1 file changed, 28 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9626ee0..aac4fbc 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -36,18 +36,32 @@ /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_DIG_POWER, 0x0000 }, { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_ADCDAC_CTRL, 0x020c }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_ADC_CTRL, 0x0000 }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINREG_CTRL, 0x0000 }, + { SGTL5000_CHIP_REF_CTRL, 0x0000 }, + { SGTL5000_CHIP_MIC_CTRL, 0x0000 }, + { SGTL5000_CHIP_LINE_OUT_CTRL, 0x0000 }, { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, { SGTL5000_CHIP_ANA_POWER, 0x7060 }, { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_CHIP_CLK_TOP_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_STATUS, 0x0000 }, + { SGTL5000_CHIP_SHORT_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_TEST2, 0x0000 }, + { SGTL5000_DAP_CTRL, 0x0000 }, + { SGTL5000_DAP_PEQ, 0x0000 }, { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_AUDIO_EQ, 0x0000 }, { SGTL5000_DAP_SURROUND, 0x0040 }, { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, @@ -55,6 +69,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_MIX_CHAN, 0x0000 }, { SGTL5000_DAP_AVC_CTRL, 0x0510 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, @@ -925,14 +940,11 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, return ret; udelay(10); - regcache_cache_only(sgtl5000->regmap, false); - ret = regcache_sync(sgtl5000->regmap); if (ret != 0) { dev_err(codec->dev, "Failed to restore cache: %d\n", ret); - regcache_cache_only(sgtl5000->regmap, true); regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1063,7 +1075,10 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) #ifdef CONFIG_SUSPEND static int sgtl5000_suspend(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(sgtl5000->regmap, true); return 0; } @@ -1075,7 +1090,6 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec) */ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; u16 reg; /* restore regular registers */ @@ -1089,12 +1103,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) reg == SGTL5000_CHIP_REF_CTRL) continue; - snd_soc_write(codec, reg, cache[reg]); + snd_soc_write(codec, reg, snd_soc_read(codec, reg)); } /* restore dap registers */ for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) - snd_soc_write(codec, reg, cache[reg]); + snd_soc_write(codec, reg, snd_soc_read(codec, reg)); /* * restore these regs according to the power setting sequence in @@ -1110,29 +1124,32 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER]); + snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER)); snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_CLK_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_REF_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_LINE_OUT_CTRL)); return 0; } static int sgtl5000_resume(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* Bring the codec back up to standby to enable regulators */ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Restore registers by cached in memory */ sgtl5000_restore_regs(codec); + + regcache_cache_only(sgtl5000->regmap, false); return 0; } #else -- 1.8.3.2

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