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Re: [alsa-devel] [PATCH 4/5] ALSA: firewire-lib: add throttle for MIDI data rate
by Takashi Sakamoto 07 Oct '15

07 Oct '15

On Oct 7 2015 09:32, kbuild test robot wrote: > Hi Takashi, > > [auto build test ERROR on v4.3-rc4 -- if it's inappropriate base, please ignore] > > config: i386-randconfig-x009-201540 (attached as .config) > reproduce: > # save the attached .config to linux build tree > make ARCH=i386 > > Note: the linux-review/Takashi-Sakamoto/ALSA-add-helper-functions-for-MIDI-message-transmission-on-IEEE-1394-asynchronous-transaction HEAD f058371b5831baa9749177001cbbdcdf47753f3d builds fine. > It only hurts bisectibility. > > All errors (new ones prefixed by >>): > > sound/firewire/lib.c: In function 'snd_fw_async_midi_port_init': >>> sound/firewire/lib.c:181:6: error: 'struct snd_fw_async_midi_port' has no member named 'error' > port->error = false; > ^ > > vim +181 sound/firewire/lib.c > > 175 > 176 port->parent = fw_parent_device(unit); > 177 port->addr = addr; > 178 port->fill = fill; > 179 port->idling = true; > 180 port->next_ktime = ktime_set(0, 0); > > 181 port->error = false; > 182 > 183 INIT_WORK(&port->work, midi_port_work); > 184 Oops! I had wrong rebase operations to generate this patch. This line should be in the next patch, sorry... Regards Takashi Sakamoto

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[alsa-devel] [PATCH 0/5] ALSA: add helper functions for MIDI message transmission on IEEE 1394 asynchronous transaction
by Takashi Sakamoto 07 Oct '15

07 Oct '15

Hi, This patchset updates a part of my former post (patch 03-07): [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html This patchset adds some helper functions for MIDI transmission via IEEE 1394 asynchronous transaction. Changes: * Use workqueue instead of tasklet for better usage of system resources. * Use ktime_t instead of jiffies for better resolution of time counting. * Some minor fixes. Takashi Sakamoto (5): ALSA: firewire-lib: add helper functions for asynchronous transactions to transfer MIDI messages ALSA: firewire-lib: add a restriction for a transaction at once ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error sound/firewire/lib.c | 138 +++++++++++++++++++++++++++++++++++++++++++++++++++ sound/firewire/lib.h | 56 +++++++++++++++++++++ 2 files changed, 194 insertions(+) -- 2.1.4

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[alsa-devel] Supported (microphone!) external usb sound card?
by Frank Hartung 06 Oct '15

06 Oct '15

Hey guys, I bought the "ASUS Xonar U7" USB sound card about 2 years ago. When i switched from "Windows" to "Linux Mint Cinnamon", i noticed that ALSA wouldn't use the microphone in of my card. So I tried another external sound card this time, the "Creative Sound Blaster X-fi HD". Connected to my laptop (also computer) and having updated my Linux Mint Cinnamon to the version 17.1, ALSA wouldn't use the microphone in either! Audio out (headphone) is working fine for both. ------ Is there ANY (decent) external (usb) sound card supported by ALSA (headset AND microphone)? Please notice that I am really not a specialist in Linux or ALSA. Regards Frank

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[alsa-devel] [PATCH 0/5] Add ASoC support for AMD APUs [v4]
by Alex Deucher 06 Oct '15

06 Oct '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create mfd cells which we hang the i2s codec and DMA engine on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review V3 changes: - Update the designware driver to handle slave mode, amd specific features - Use the designware driver directly for i2s - Move the DMA handling from the GPU driver into the AMD ASoC driver - Change the license on the ASoC driver to GPL V4 changes: - get the master vs slave mode for the designware codec from registers - copy designware quirks from platform data in local struct Patch 4 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream4 Thanks, Alex Maruthi Bayyavarapu (1): drm/amd: add ACP driver support Maruthi Srinivas Bayyavarapu (4): ASoC : dwc : support dw i2s in slave mode ASoC : dwc : support dw i2s in AMD platform ASoC : AMD : add ACP 2.2 register headers ASoC: AMD: add AMD ASoC ACP-I2S driver drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 10 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 127 ++ drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 269 +++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 41 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + include/linux/mfd/amd_acp.h | 43 + include/sound/designware_i2s.h | 5 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 4 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 518 ++++++ sound/soc/amd/acp.c | 736 +++++++++ sound/soc/amd/acp.h | 147 ++ sound/soc/amd/include/acp_2_2_d.h | 609 +++++++ sound/soc/amd/include/acp_2_2_enum.h | 1068 ++++++++++++ sound/soc/amd/include/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++++++++ sound/soc/dwc/designware_i2s.c | 276 ++-- 24 files changed, 6146 insertions(+), 102 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 include/linux/mfd/amd_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp.c create mode 100644 sound/soc/amd/acp.h create mode 100644 sound/soc/amd/include/acp_2_2_d.h create mode 100644 sound/soc/amd/include/acp_2_2_enum.h create mode 100644 sound/soc/amd/include/acp_2_2_sh_mask.h -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: codecs: tas2552: fix dBscale-min declaration
by Andreas Dannenberg 06 Oct '15

06 Oct '15

The minimum volume level for the TAS2552 (control register value 0x00) is -7dB however the driver declares it as -0.07dB. Running amixer before the patch reports: dBscale-min=-0.07dB,step=1.00dB,mute=0 Running amixer with the patch applied reports: dBscale-min=-7.00dB,step=1.00dB,mute=0 Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> --- sound/soc/codecs/tas2552.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index e3a0bca..cc1d398 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = { /* * DAC digital volumes. From -7 to 24 dB in 1 dB steps */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0); static const char * const tas2552_din_source_select[] = { "Muted", -- 1.9.1

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[alsa-devel] Applied "MAINTAINERS: Remove wm97xx entry" to the asoc tree
by Mark Brown 06 Oct '15

06 Oct '15

The patch MAINTAINERS: Remove wm97xx entry 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 ff6afc8d0df0347901e862e622b1efaeb6f1fe5d Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Fri, 2 Oct 2015 18:07:09 +0100 Subject: [PATCH] MAINTAINERS: Remove wm97xx entry Neither myself or Liam is especially interested in this driver any more and the devices are already covered by the general ex-Wolfson entry so just remove this. Signed-off-by: Mark Brown <broonie(a)kernel.org> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- MAINTAINERS | 9 --------- 1 file changed, 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 0581f47..07790df 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -11363,15 +11363,6 @@ W: http://oops.ghostprotocols.net:81/blog S: Maintained F: drivers/net/wireless/wl3501* -WM97XX TOUCHSCREEN DRIVERS -M: Mark Brown <broonie(a)kernel.org> -M: Liam Girdwood <lrg(a)slimlogic.co.uk> -L: linux-input(a)vger.kernel.org -W: https://github.com/CirrusLogic/linux-drivers/wiki -S: Supported -F: drivers/input/touchscreen/*wm97* -F: include/linux/wm97xx.h - WOLFSON MICROELECTRONICS DRIVERS L: patches(a)opensource.wolfsonmicro.com T: git https://github.com/CirrusLogic/linux-drivers.git -- 2.5.0

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[alsa-devel] [PATCH RFC] ASoC: simple-card: Update clocks binding for simple-card DAI subnodes
by Jyri Sarha 06 Oct '15

06 Oct '15

The updated binding provides a way to set clock-ID and direction parameters for DAI drivers set_sysclk() call back. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- I proposed something similar about a year ago, but Mark rejected that at the time. This RFC is to start that discussion again. This time before I do any code changes. Best regards, Jyri Documentation/devicetree/bindings/sound/simple-card.txt | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index cf3979e..d10bf2d 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -76,6 +76,7 @@ Optional CPU/CODEC subnodes properties: - dai-tdm-slot-num : Please refer to tdm-slot.txt. - dai-tdm-slot-width : Please refer to tdm-slot.txt. - clocks / system-clock-frequency : specify subnode's clock if needed. + it can be specified via "clocks" if system has clock node (= common clock), or "system-clock-frequency" (if system doens't support common clock) @@ -83,7 +84,21 @@ Optional CPU/CODEC subnodes properties: enabled with clk_prepare_enable() in dai startup() and disabled with clk_disable_unprepare() in dai - shutdown(). + shutdown(). "system-clock-frequency" + can also be an array if more than one + clock is described. +- clock-ids : An array of clock ID integers, + preferrably defined in DT header. + Each entry corresponds to the same + index postion first in "clocks" and + after the end of clocks array to + "system-clock-frequency" array. +- clock-dirs : An array of integers describing + clock directions: CLK_IN (= 0) or + OUT (= 1). Entries in the array + refer to clocks in the same way as + in clock-ids property. + Example 1 - single DAI link: -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rockchip: namespace rockchip i2s module name
by Sjoerd Simons 06 Oct '15

06 Oct '15

Change the rockchip i2s object name (and thus module name) from the rather generic snd-soc-i2s to the more specific snd-soc-rockchip-i2s Signed-off-by: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> --- sound/soc/rockchip/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 1bc1dc3..e9ba558 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,7 +1,7 @@ # ROCKCHIP Platform Support -snd-soc-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-i2s-objs := rockchip_i2s.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o -- 2.6.0

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[alsa-devel] Applied "ASoC: tas2552: fix dBscale-min declaration" to the asoc tree
by Mark Brown 06 Oct '15

06 Oct '15

The patch ASoC: tas2552: fix dBscale-min declaration 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 e2600460bc3aa14ca1df86318a327cbbabedf9a8 Mon Sep 17 00:00:00 2001 From: Andreas Dannenberg <dannenberg(a)ti.com> Date: Mon, 5 Oct 2015 15:00:14 -0500 Subject: [PATCH] ASoC: tas2552: fix dBscale-min declaration The minimum volume level for the TAS2552 (control register value 0x00) is -7dB however the driver declares it as -0.07dB. Running amixer before the patch reports: dBscale-min=-0.07dB,step=1.00dB,mute=0 Running amixer with the patch applied reports: dBscale-min=-7.00dB,step=1.00dB,mute=0 Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/tas2552.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index e3a0bca..cc1d398 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = { /* * DAC digital volumes. From -7 to 24 dB in 1 dB steps */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0); static const char * const tas2552_din_source_select[] = { "Muted", -- 2.5.0

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[alsa-devel] [PATCH v2] ASoC: rt5645: Add the HW EQ for the customized speaker output of Google Celes
by Oder Chiou 06 Oct '15

06 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5645.h | 1 + sound/soc/codecs/rt5645.c | 70 ++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5645.h | 5 ++++ 3 files changed, 75 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf615..cf944e4 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,7 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Invert JD when jack insert */ bool jd_invert; + unsigned int eq_mode; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c4fe6b..def9caa 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -224,6 +224,36 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5647_eq_parameter { + unsigned int reg; + unsigned int val; +}; + +#define EQ_REG_NUM 56 +struct hweq_s { + struct rt5647_eq_parameter par[EQ_REG_NUM]; + unsigned int ctrl; +}; + +static const struct hweq_s hweq_param[] = { + { /* EQ_NONE */ + { + {0, 0}, + }, + 0x0000, + }, + { /* EQ_GOOGLE_CELES */ + { + {0x1c0, 0x04b8}, + {0x1c1, 0xfd02}, + {0x1c2, 0x04b8}, + {0x1c3, 0xfd02}, + {0, 0}, + }, + 0x0020, + }, +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -631,6 +661,27 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_update_eqmode(struct snd_soc_codec *codec, + unsigned int mode) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int i; + + for (i = 0; i < EQ_REG_NUM; i++) { + if (hweq_param[mode].par[i].reg) + regmap_write(rt5645->regmap, + hweq_param[mode].par[i].reg, + hweq_param[mode].par[i].val); + else + break; + } + + snd_soc_update_bits(codec, RT5645_EQ_CTRL2, 0x33fe, + hweq_param[mode].ctrl); + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1532,9 +1583,11 @@ static int rt5645_spk_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 rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: + rt5645_update_eqmode(codec, rt5645->pdata.eq_mode); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1596,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + rt5645_update_eqmode(codec, EQ_NONE); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3205,6 +3259,20 @@ static int strago_quirk_cb(const struct dmi_system_id *id) return 1; } +static struct rt5645_platform_data celes_platform_data = { + .dmic1_data_pin = RT5645_DMIC1_DISABLE, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, + .eq_mode = EQ_GOOGLE_CELES, +}; + +static int celes_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &celes_platform_data; + + return 1; +} + static const struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", @@ -3215,7 +3283,7 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { }, { .ident = "Google Celes", - .callback = strago_quirk_cb, + .callback = celes_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 13ac732..89f25b8 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2187,6 +2187,11 @@ enum { RT5645_AD_MONO_R_FILTER = (0x1 << 5), }; +enum { + EQ_NONE, + EQ_GOOGLE_CELES, +}; + int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec, unsigned int filter_mask, unsigned int clk_src); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: Document DAI signal polarity
by Anatol Pomozov 06 Oct '15

06 Oct '15

Currently there is no clear definition of what is FSYNC polarity. Different drivers use its own definition of what is "normal" and what is "inverted" fsync. This leads to compatibility problems between drivers. For example TegraX1 driver assumes that DSP-A format with frames starting at rising FSYNC edge has "negative" polarity, while RT5677 assumes it is "positive" polarity. Explicitly specify meaning of BCLK/FSYNC polarity to avoid future compatibility problems. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- include/sound/soc-dai.h | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..45198db 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,22 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * - * Specifies whether the DAI can also support inverted clocks for the specified - * format. + * BCLK: + * - "normal" polarity means signal is available at rising edge of BCLK + * - "inverted" polarity means signal is available at falling edge of BCLK + * + * FSYNC "normal" polarity depends on the frame format: + * - I2S: frame consists of left then right channel data. Left channel starts + * with falling FSYNC edge, right channel starts with rising FSYNC edge. + * - I2S right/left justified: frame consists of left then right channel data. + * Left channel starts with rising FSYNC edge, right channel starts with + * falling FSYNC edge. + * - DSP A/B: Frame starts with rising FSYNC edge. + * - AC97: Frame starts with rising FSYNC edge. + * + * "Negative" FSYNC polarity is the one opposite of "normal" polarity. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- 2.6.0.rc2.230.g3dd15c0

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[alsa-devel] [PATCH alsa-lib] hwdep: add FireWire digi00x/tascam driver support
by Takashi Sakamoto 06 Oct '15

06 Oct '15

Linux 4.4 newly support these drivers. Their hardware dependent interfaces give functionalities to get hardware information, to lock/unlock kernel streaming and to receive lock status event. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/hwdep.h | 4 +++- include/sound/asound.h | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/include/hwdep.h b/include/hwdep.h index 3d3c31b..2fe78cd 100644 --- a/include/hwdep.h +++ b/include/hwdep.h @@ -74,8 +74,10 @@ typedef enum _snd_hwdep_iface { SND_HWDEP_IFACE_FW_FIREWORKS, /**< Echo Audio Fireworks based device */ SND_HWDEP_IFACE_FW_BEBOB, /**< BridgeCo BeBoB based device */ SND_HWDEP_IFACE_FW_OXFW, /**< Oxford OXFW970/971 based device */ + SND_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ + SND_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */ - SND_HWDEP_IFACE_LAST = SND_HWDEP_IFACE_FW_OXFW /**< last known hwdep interface */ + SND_HWDEP_IFACE_LAST = SND_HWDEP_IFACE_FW_TASCAM /**< last known hwdep interface */ } snd_hwdep_iface_t; /** open for reading */ diff --git a/include/sound/asound.h b/include/sound/asound.h index a45be6b..a82108e 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -100,9 +100,11 @@ enum { SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */ SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */ + SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ + SNDRV_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_OXFW + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_TASCAM }; struct snd_hwdep_info { -- 2.1.4

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[alsa-devel] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered
by Rick Mann 05 Oct '15

05 Oct '15

UPDATE: I was able to read the McASP registers (turns out, its clocks are only enabled while speaker-test is running, so that's the only time the registers are mapped). AHCLKXCTL is 0x00008000, which corresponds to having it set as an output, not inverted, and clock divide-by-1. This seems wrong. I'm having trouble determining what the AUXCLK that feeds the McASP is, but it's not 12 MHz, and without division, I don't see how to get there. For reference, here are the actual values of the McASP transmit registers: GBLCTL 0x00000000 Lots of stuff held in reset (red flag?) XGBLCTL 0x00000000 Same XMASK 0x0000FFFF I think this masks out 16-bits? XFMT 0x00008074 MSB bit first, slot size 16-bit, rot right 16 bits AFSXCTL 0x00000100 2-slot TDM (I2S mode) ACLKXCTL 0x00000000 Sync xmit, ACLKX is from pin AHCLKXCTL 0x00008000 Internal high-freq clock from clock divider XTDM 0x00000003 2 TDM slots active XINTCTL 0x00000000 XSTAT 0x0000010C Xmit error occurred, cur TDM slot is even, xmit clock failure XSLOT 0x0000017F Current time slot (shouldn't this be 0 or 1?) XCLKCHK 0x00000000 XEVTCTL 0x00000000 Thanks, Rick ---- I've been at this for weeks and I'm still unable to get linux to work correctly with my TI codec. This is on a Beaglebone Black running a 4.1.4 kernel (later versions are worse). I've tried multiple versions of the kernel trying to narrow it down. I keep coming back to 4.1.4 as the most recent that at least generates some audio clocks. The AM335x mcasp0 should be speaking i2s with the tlv320aic3104. I have my own build from source of the BBB 4.1.4-ti-r9 release (mainline plus TI plus Beaglebone stuff). I've sprinkled some logging around the audio drivers, but so far I don't see much, and I really don't understand how all the pieces fit together. One thing I found with my logging is the mcasp MCLK (AHCLKXDIV) is never set. I have logging calls in __davinci_mcasp_set_clkdiv() (sound/soc/davinci/davinci-mcasp.c), which is the only code I could find that sets the McASP clock, and it's never called. I tried to use devmem2 to read the registers in the SoC to see if they were set correctly, but I just get a bus error (other reads do work). Errors in dmesg when I load the device tree overlay for my cape suggest I still don't have something configured correctly (see below). I'd sure appreciate any insight into these latest issues. Thanks! Symptoms: MCLK should be 12 MHz, but is 24.00 MHz. Buffer underflows when running speaker-test. Errors to console upon loading cape: [ 38.502299] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.517917] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.530425] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.536996] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.564279] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT dmesg upon loading cape overlay (https://github.com/JetForMe/podtique/blob/v1/bbb/cape/Podtique1/BB-ENABLE-P…): [ 568.323587] bone_capemgr bone_capemgr: part_number 'BB-ENABLE-PRU', version 'N/A' [ 568.323629] bone_capemgr bone_capemgr: slot #4: override [ 568.323646] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 568.323663] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-ENABLE-PRU' [ 568.323804] bone_capemgr:capemgr_load_slot:1148: bone_capemgr bone_capemgr: slot #4: Requesting part number/version based 'BB-ENABLE-PRU-00A0.dtbo [ 568.323825] bone_capemgr:capemgr_load_slot:1162: bone_capemgr bone_capemgr: slot #4: Requesting firmware 'BB-ENABLE-PRU-00A0.dtbo' for board-name 'Override Board Name', version '00A0' [ 568.330369] bone_capemgr:capemgr_load_slot:1171: bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; converting to live tree [ 568.341685] gpio-of-helper ocp:gpio_helper: ready [ 568.344695] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; overlay id #0 [ 568.387640] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.387695] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.387712] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.394284] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.410560] snd_soc_core:snd_soc_register_codec:3064: tlv320aic3x-codec 2-0018: codec register 2-0018 [ 568.410614] snd_soc_core:snd_soc_register_dais:2590: tlv320aic3x-codec 2-0018: ASoC: dai register 2-0018 #1 [ 568.410633] snd_soc_core:snd_soc_register_dais:2634: tlv320aic3x-codec 2-0018: ASoC: Registered DAI 'tlv320aic3x-hifi' [ 568.410653] snd_soc_core:snd_soc_register_codec:3138: tlv320aic3x-codec 2-0018: ASoC: Registered codec 'tlv320aic3x-codec.2-0018' [ 568.421529] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.421588] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.421605] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.431455] **** RLM davinci-mcasp.c:1734: davinci_mcasp_probe [ 568.431507] base:of_property_match_string:1398: comparing tx with tx [ 568.431544] base:of_property_match_string:1398: comparing rx with tx [ 568.431555] base:of_property_match_string:1398: comparing rx with rx [ 568.431608] base:of_property_match_string:1398: comparing common with tx [ 568.431621] base:of_property_match_string:1398: comparing common with rx [ 568.431633] base:of_property_match_string:1398: comparing rx with tx [ 568.431643] base:of_property_match_string:1398: comparing rx with rx [ 568.434643] base:of_property_match_string:1398: comparing tx with tx [ 568.437971] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.446651] snd_soc_core:snd_soc_register_dais:2590: davinci-mcasp 48038000.mcasp: ASoC: dai register 48038000.mcasp #1 [ 568.446682] snd_soc_core:snd_soc_register_dais:2634: davinci-mcasp 48038000.mcasp: ASoC: Registered DAI '48038000.mcasp' [ 568.446840] snd_soc_core:snd_soc_add_platform:2892: davinci-mcasp 48038000.mcasp: ASoC: Registered platform '48038000.mcasp' [ 568.450046] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.450100] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.450425] snd_soc_core:snd_soc_dapm_new_dai_widgets:3494: tlv320aic3x-codec 2-0018: ASoC: adding Playback widget [ 568.450448] snd_soc_core:snd_soc_dapm_new_dai_widgets:3513: tlv320aic3x-codec 2-0018: ASoC: adding Capture widget [ 568.450995] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -2 [ 568.451015] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -1 [ 568.451030] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 0 [ 568.451045] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 1 [ 568.451059] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 2 [ 568.451099] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT [ 568.465530] snd_soc_core:soc_new_pcm:2510: davinci_evm ocp:sound: ASoC: registered pcm #0 AIC3X tlv320aic3x-hifi-0 [ 568.466155] davinci_evm ocp:sound: tlv320aic3x-hifi <-> 48038000.mcasp mapping ok [ 568.466185] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Right ADC -> Capture [ 568.466206] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Left ADC -> Capture [ 568.466226] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Right DAC [ 568.466241] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Left DAC upon starting speaker-test: [ 572.786071] snd_soc_core:soc_pcm_open:535: ASoC: tlv320aic3x-hifi <-> 48038000.mcasp info: [ 572.786103] snd_soc_core:soc_pcm_open:536: ASoC: rate mask 0x7fe [ 572.786114] snd_soc_core:soc_pcm_open:538: ASoC: min ch 2 max ch 2 [ 572.786126] snd_soc_core:soc_pcm_open:540: ASoC: min rate 8000 max rate 96000 [ 572.841514] davinci-mcasp 48038000.mcasp: Transmit buffer underflow ...repeated [ 574.087162] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 579.135283] snd_soc_core:close_delayed_work:602: TLV320AIC3X: ASoC: pop wq checking: Playback status: inactive waiting: yes -- Rick Mann rmann(a)latencyzero.com

2 2

[alsa-devel] [asoc:topic/fsl-mega-fast 4/4] sound/soc/fsl/fsl_esai.c:903:5: warning: "CONFIG_PM_SLEEP" is not defined
by kbuild test robot 05 Oct '15

05 Oct '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/fsl-mega-fast head: c64c60763b4e3c72a3520c8d51be858cd67bacb5 commit: c64c60763b4e3c72a3520c8d51be858cd67bacb5 [4/4] ASoC: fsl_esai: Add driver suspend and resume to support MEGA Fast config: x86_64-randconfig-x015-201540 (attached as .config) reproduce: git checkout c64c60763b4e3c72a3520c8d51be858cd67bacb5 # save the attached .config to linux build tree make ARCH=x86_64 All warnings (new ones prefixed by >>): >> sound/soc/fsl/fsl_esai.c:903:5: warning: "CONFIG_PM_SLEEP" is not defined [-Wundef] #if CONFIG_PM_SLEEP ^ vim +/CONFIG_PM_SLEEP +903 sound/soc/fsl/fsl_esai.c 887 } 888 889 ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); 890 if (ret) 891 dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); 892 893 return ret; 894 } 895 896 static const struct of_device_id fsl_esai_dt_ids[] = { 897 { .compatible = "fsl,imx35-esai", }, 898 { .compatible = "fsl,vf610-esai", }, 899 {} 900 }; 901 MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); 902 > 903 #if CONFIG_PM_SLEEP 904 static int fsl_esai_suspend(struct device *dev) 905 { 906 struct fsl_esai *esai = dev_get_drvdata(dev); 907 908 regcache_cache_only(esai->regmap, true); 909 regcache_mark_dirty(esai->regmap); 910 911 return 0; --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 1/4] ASoC: imx-spdif: add snd_soc_pm_ops for spdif machine driver
by Zidan Wang 05 Oct '15

05 Oct '15

Add snd_soc_pm_ops in machine driver to make the trigger suspend/resume be called in suspend/resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/imx-spdif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 33da26a..a407e83 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -89,6 +89,7 @@ MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); static struct platform_driver imx_spdif_driver = { .driver = { .name = "imx-spdif", + .pm = &snd_soc_pm_ops, .of_match_table = imx_spdif_dt_ids, }, .probe = imx_spdif_audio_probe, -- 1.9.1

3 9

[alsa-devel] [PATCH v3 1/4] ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 05 Oct '15

05 Oct '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..5c737f1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -802,10 +805,40 @@ static const struct of_device_id fsl_sai_ids[] = { { /* sentinel */ } }; +#if CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; -- 1.9.1

4 15

[alsa-devel] [PATCH linux-next v3 1/1] ASoC: ad193x: add support to ad1934
by Cyrille Pitchen 05 Oct '15

05 Oct '15

The AD1934 codec has no DAC feature. Hence it register mapping is slightly different from the register mapping of other members of the AD193x family. Some ASoC controls and widgets are related to the DAC feature so are not relevant in the case of an AD1934 codec. Signed-off-by: Cyrille Pitchen <cyrille.pitchen(a)atmel.com> --- Hi all, This patch adds support to the AD1934 codec, which is DAC only. Best Regards, Cyrille ChangeLog v3: - rebase patch for linux-next: don't use the CODEC's dapm field, which no longer exists, but call snd_soc_codec_get_dapm() instead. v2: - replace AD193x by AD193X in ad193x-i2c.c - add ad193x_ prefix to adc_audio_paths --- sound/soc/codecs/ad193x-i2c.c | 8 +-- sound/soc/codecs/ad193x-spi.c | 16 +++++- sound/soc/codecs/ad193x.c | 122 ++++++++++++++++++++++++++++++++---------- sound/soc/codecs/ad193x.h | 9 +++- 4 files changed, 121 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c index df3a1a415825..171313664bc8 100644 --- a/sound/soc/codecs/ad193x-i2c.c +++ b/sound/soc/codecs/ad193x-i2c.c @@ -15,8 +15,8 @@ #include "ad193x.h" static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, { } }; MODULE_DEVICE_TABLE(i2c, ad193x_id); @@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client, config.val_bits = 8; config.reg_bits = 8; - return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); + return ad193x_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 390cef9b9dc2..364e82112e8c 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -16,6 +16,7 @@ static int ad193x_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct regmap_config config; config = ad193x_regmap_config; @@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi) config.read_flag_mask = 0x09; config.write_flag_mask = 0x08; - return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_spi_remove(struct spi_device *spi) @@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id ad193x_spi_id[] = { + { "ad193x", AD193X }, + { "ad1933", AD1933 }, + { "ad1934", AD1934 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, + { "ad1938", AD193X }, + { } +}; +MODULE_DEVICE_TABLE(spi, ad193x_spi_id); + static struct spi_driver ad193x_spi_driver = { .driver = { .name = "ad193x", @@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = { }, .probe = ad193x_spi_probe, .remove = ad193x_spi_remove, + .id_table = ad193x_spi_id, }; module_spi_driver(ad193x_spi_driver); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 17c953595660..76d7966bd72c 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,6 +23,7 @@ /* codec private data */ struct ad193x_priv { struct regmap *regmap; + enum ad193x_type type; int sysclk; }; @@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, - AD193X_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, - AD193X_ADCR2_MUTE, 1, 1), - /* DAC switch control */ SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, AD193X_DACR1_MUTE, 1, 1), @@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, AD193X_DACR4_MUTE, 1, 1), + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = { + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, AD193X_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_INPUT("ADC1IN"), SND_SOC_DAPM_INPUT("ADC2IN"), }; @@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, { "DAC Output", NULL, "DAC" }, { "DAC Output", NULL, "VMID" }, - { "ADC", NULL, "SYSCLK" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC Output" }, { "DAC2OUT", NULL, "DAC Output" }, { "DAC3OUT", NULL, "DAC Output" }, { "DAC4OUT", NULL, "DAC Output" }, + { "SYSCLK", NULL, "PLL_PWR" }, +}; + +static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = { + { "ADC", NULL, "SYSCLK" }, + { "ADC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, - { "SYSCLK", NULL, "PLL_PWR" }, }; +static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) +{ + switch (ad193x->type) { + case AD1933: + case AD1934: + return false; + default: + break; + } + + return true; +} + /* * DAI ops entries */ @@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: - return -EINVAL; + if (ad193x_has_adc(ad193x)) + return -EINVAL; + break; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_SERFMT_MASK, adc_serfmt); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_FMT_MASK, adc_fmt); + if (ad193x_has_adc(ad193x)) { + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + } regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, dac_fmt); @@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_WORD_LEN_MASK, word_len); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -326,6 +354,8 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int num, ret; /* default setting for ad193x */ @@ -335,14 +365,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* dac in tdm mode */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* high-pass filter enable */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + } + /* pll input: mclki/xi */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* add adc controls */ + num = ARRAY_SIZE(ad193x_adc_snd_controls); + ret = snd_soc_add_codec_controls(codec, + ad193x_adc_snd_controls, + num); + if (ret) + return ret; + + /* add adc widgets */ + num = ARRAY_SIZE(ad193x_adc_widgets); + ret = snd_soc_dapm_new_controls(dapm, + ad193x_adc_widgets, + num); + if (ret) + return ret; + + /* add adc routes */ + num = ARRAY_SIZE(ad193x_adc_audio_paths); + ret = snd_soc_dapm_add_routes(dapm, + ad193x_adc_audio_paths, + num); + if (ret) + return ret; + } + return 0; } @@ -367,7 +429,8 @@ const struct regmap_config ad193x_regmap_config = { }; EXPORT_SYMBOL_GPL(ad193x_regmap_config); -int ad193x_probe(struct device *dev, struct regmap *regmap) +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type) { struct ad193x_priv *ad193x; @@ -379,6 +442,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; ad193x->regmap = regmap; + ad193x->type = type; dev_set_drvdata(dev, ad193x); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index ab9a998f15be..8b1e65f928d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -13,8 +13,15 @@ struct device; +enum ad193x_type { + AD193X, + AD1933, + AD1934, +}; + extern const struct regmap_config ad193x_regmap_config; -int ad193x_probe(struct device *dev, struct regmap *regmap); +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type); #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -- 1.8.2.2

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[alsa-devel] [PATCH 0/4] ALSA: hdac: fixes and support for hda codecs
by Vinod Koul 05 Oct '15

05 Oct '15

We had two fixes found on testing. First one checks if stream is in use or not before releasing and second fixs wrong stread id map Then we add structure defination for hdac_ext_dma_params and lastly add helpers for read/write for codecs Jeeja KP (1): ALSA: hdac: Fix to check if stream not in use in release Subhransu S. Prusty (3): ALSA: hdac: Fix incorrect update of stream id mapping ALSA: hdac: structure definition for ext_dma_params ALSA: hdac: Add codec read/write and check power state for widgets include/sound/hdaudio.h | 6 ++++++ include/sound/hdaudio_ext.h | 6 ++++++ sound/hda/ext/hdac_ext_stream.c | 9 ++++---- sound/hda/hdac_device.c | 47 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 63 insertions(+), 5 deletions(-) -- 2.4.3

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: Prevent the weird sound of the headphone while rebooting
by Oder Chiou 05 Oct '15

05 Oct '15

The patch adds the codec reset setting in the shutdown function to prevent the weird sound of the headphone happened by rebooting. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index f436f50..4c4fe6b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3517,6 +3517,8 @@ static void rt5645_i2c_shutdown(struct i2c_client *i2c) RT5645_CBJ_MN_JD); regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, 0); + msleep(20); + regmap_write(rt5645->regmap, RT5645_RESET, 0); } static struct i2c_driver rt5645_i2c_driver = { -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] Applied "ASoC: da7219: Improve error handling for regulator supplies" to the asoc tree
by Mark Brown 05 Oct '15

05 Oct '15

The patch ASoC: da7219: Improve error handling for regulator supplies 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 4e929134eb8271abc9c52c371e056debfea6898b Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Mon, 5 Oct 2015 15:10:04 +0100 Subject: [PATCH] ASoC: da7219: Improve error handling for regulator supplies Currently if bulk_enable() of supplies fails, the code still goes on to try and put the device into active state, and set expected IO voltage of the device. This doesn't really make sense so code now returns on bulk_enable() failure, with an error message. Also, to help with debug, failure to get supplies also provides an error message. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7219.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 76f8fc2..c86a833 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1554,8 +1554,10 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES, da7219->supplies); - if (ret) + if (ret) { + dev_err(codec->dev, "Failed to get supplies"); return ret; + } /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; @@ -1567,6 +1569,10 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) /* Enable main supplies */ ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies"); + return ret; + } /* Ensure device in active mode */ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); @@ -1574,7 +1580,7 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) /* Update IO voltage level range */ snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl); - return ret; + return 0; } static void da7219_handle_pdata(struct snd_soc_codec *codec) -- 2.5.0

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[alsa-devel] sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0'
by Michael Shell 05 Oct '15

05 Oct '15

Greetings, I recently upgraded (an admittedly ancient K6-2 PCI/ISA machine) to kernel version 4.2.2 (compiled from source). It went surprising well, but I did run into a minor issue which did not occur under the older kernel 3.16.3. Maybe the info here will help someone else out in the future or even reveal a bug that might have greater implications that have not yet been seen. The ancient machine has one of those ISA SoundBlaster cards whose drivers are compiled as modules (to allow isapnp to operate) before module loading. Upon issuing a /sbin/modprobe snd-sbawe port=0x220 mpu_port=0x330 awe_port=0x620 irq=5 dma8=1 dma16=5 mic_agc=1 csp=0 dmesg (under 4.2.2) spits out: EMU8000 [0x620]: 4096 KiB on-board DRAM detected ------------[ cut here ]------------ WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80() sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0' Modules linked in: snd_emu8000_synth(+) snd_emux_synth snd_seq_virmidi snd_util_mem snd_opl3_synth snd_seq_midi_emul snd_sbawe snd_opl3_lib snd_sb16_dsp snd_sb16_csp snd_sb_common snd_mpu401_uart lp parport_pc parport CPU: 0 PID: 2791 Comm: modprobe Not tainted 4.2.2 #1 . . ---- (full output with trace at the end of this message) The /sys/devices/isa/sbawe.0 directory did not exist prior to the attempt to load snd-sbawe and there are no other sound drivers in the system other than those for the Soundblaster. I do still have OSS emulation selected in the kernel config - I was surpised that still is the recommended default. I probably could avoid this bug by deselecting that. I doubt I have any software that still requires OSS emulation, which is probably going to be removed from future kernels anyway (and Debian systems don't even carry it anymore). Nevertheless, a bug is a bug and this problem should not happen. I think snd_seq_oss and snd_emux_synth are both trying to register the same sysfs name or a single driver is trying to register the same thing twice. For the record, what is the correct procedure to find out exactly which drivers are in conflict in duplicate sysfs filename cases like this? Cheers and thanks in advance, Mike Shell ---- EMU8000 [0x620]: 4096 KiB on-board DRAM detected ------------[ cut here ]------------ WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80() sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0' Modules linked in: snd_emu8000_synth(+) snd_emux_synth snd_seq_virmidi snd_util_mem snd_opl3_synth snd_seq_midi_emul snd_sbawe snd_opl3_lib snd_sb16_dsp snd_sb16_csp snd_sb_common snd_mpu401_uart lp parport_pc parport CPU: 0 PID: 2791 Comm: modprobe Not tainted 4.2.2 #1 c1604e78 c1031c40 c175c544 dd6dfc48 00000ae7 c1758c38 0000001f c1103391 c1103391 df493000 dd6ef2c0 dd6ec6b0 dd6bfe9c c1031d13 00000009 dd6dfc30 c175c544 dd6dfc48 c1103391 c1758c38 0000001f c175c544 df493fdf dd6ef2c0 Call Trace: [<c1604e78>] ? dump_stack+0xa/0x13 [<c1031c40>] ? warn_slowpath_common+0x80/0xc0 [<c1103391>] ? sysfs_warn_dup+0x51/0x80 [<c1103391>] ? sysfs_warn_dup+0x51/0x80 [<c1031d13>] ? warn_slowpath_fmt+0x33/0x40 [<c1103391>] ? sysfs_warn_dup+0x51/0x80 [<c110346b>] ? sysfs_create_dir_ns+0x6b/0x80 [<c11c156c>] ? kobject_add_internal+0x6c/0x2e0 [<c11c86cb>] ? vsnprintf+0xcb/0x400 [<c11c1956>] ? kobject_add+0x36/0x80 [<c134de21>] ? device_add+0xc1/0x500 [<c1441669>] ? snd_device_new+0x29/0xa0 [<c1460a56>] ? snd_seq_device_new+0xb6/0x100 [<c146096c>] ? snd_seq_device_dev_register+0xc/0x40 [<c1441604>] ? snd_device_register+0x44/0x80 [<e0bd1fc4>] ? snd_emux_init_seq_oss+0x84/0xa0 [snd_emux_synth] [<e0bcd1e3>] ? snd_emux_register+0x103/0x140 [snd_emux_synth] [<e0bcd000>] ? 0xe0bcd000 [<e0bcd020>] ? sf_sample_new+0x20/0x20 [snd_emux_synth] [<e0bcd040>] ? sf_sample_free+0x20/0x20 [snd_emux_synth] [<e0bdf113>] ? snd_emu8000_probe+0xd3/0x180 [snd_emu8000_synth] [<c11036dd>] ? sysfs_create_link+0x1d/0x40 [<c13504fb>] ? driver_probe_device+0x15b/0x220 [<c1350631>] ? __driver_attach+0x71/0x80 [<c13505c0>] ? driver_probe_device+0x220/0x220 [<c134f000>] ? bus_for_each_dev+0x40/0x80 [<c134ff7a>] ? driver_attach+0x1a/0x20 [<c13505c0>] ? driver_probe_device+0x220/0x220 [<c134fcf4>] ? bus_add_driver+0xd4/0x200 [<c1350b91>] ? driver_register+0x51/0xe0 [<e0be3000>] ? 0xe0be3000 [<e0be3000>] ? 0xe0be3000 [<c10003f2>] ? do_one_initcall+0x72/0x1a0 [<e0be3000>] ? 0xe0be3000 [<c14ac646>] ? netlink_broadcast_filtered+0x166/0x3c0 [<c106c4a5>] ? do_init_module+0x45/0x200 [<c10a8425>] ? __vunmap+0x85/0xe0 [<c106dfaa>] ? load_module+0x186a/0x1ea0 [<c106e6aa>] ? SyS_init_module+0xca/0x100 [<c160a2dd>] ? syscall_call+0x7/0x7 ---[ end trace 826ff8353957cad3 ]--- ------------[ cut here ]------------ WARNING: CPU: 0 PID: 2791 at lib/kobject.c:240 kobject_add_internal+0x252/0x2e0() kobject_add_internal failed for seq-oss-0-0 with -EEXIST, don't try to register things with the same name in the same directory. Modules linked in: snd_emu8000_synth(+) snd_emux_synth snd_seq_virmidi snd_util_mem snd_opl3_synth snd_seq_midi_emul snd_sbawe snd_opl3_lib snd_sb16_dsp snd_sb16_csp snd_sb_common snd_mpu401_uart lp parport_pc parport CPU: 0 PID: 2791 Comm: modprobe Tainted: G W 4.2.2 #1 c1604e78 c1031c40 c1768d78 dd6dfc84 00000ae7 c1768e8e 000000f0 c11c1752 c11c1752 dd66c274 00000000 dd66c274 dd6bfe9c c1031d13 00000009 dd6dfc6c c1768d78 dd6dfc84 c11c1752 c1768e8e 000000f0 c1768d78 c162bbd4 dd6ef2c0 Call Trace: [<c1604e78>] ? dump_stack+0xa/0x13 [<c1031c40>] ? warn_slowpath_common+0x80/0xc0 [<c11c1752>] ? kobject_add_internal+0x252/0x2e0 [<c11c1752>] ? kobject_add_internal+0x252/0x2e0 [<c1031d13>] ? warn_slowpath_fmt+0x33/0x40 [<c11c1752>] ? kobject_add_internal+0x252/0x2e0 [<c11c1956>] ? kobject_add+0x36/0x80 [<c134de21>] ? device_add+0xc1/0x500 [<c1441669>] ? snd_device_new+0x29/0xa0 [<c1460a56>] ? snd_seq_device_new+0xb6/0x100 [<c146096c>] ? snd_seq_device_dev_register+0xc/0x40 [<c1441604>] ? snd_device_register+0x44/0x80 [<e0bd1fc4>] ? snd_emux_init_seq_oss+0x84/0xa0 [snd_emux_synth] [<e0bcd1e3>] ? snd_emux_register+0x103/0x140 [snd_emux_synth] [<e0bcd000>] ? 0xe0bcd000 [<e0bcd020>] ? sf_sample_new+0x20/0x20 [snd_emux_synth] [<e0bcd040>] ? sf_sample_free+0x20/0x20 [snd_emux_synth] [<e0bdf113>] ? snd_emu8000_probe+0xd3/0x180 [snd_emu8000_synth] [<c11036dd>] ? sysfs_create_link+0x1d/0x40 [<c13504fb>] ? driver_probe_device+0x15b/0x220 [<c1350631>] ? __driver_attach+0x71/0x80 [<c13505c0>] ? driver_probe_device+0x220/0x220 [<c134f000>] ? bus_for_each_dev+0x40/0x80 [<c134ff7a>] ? driver_attach+0x1a/0x20 [<c13505c0>] ? driver_probe_device+0x220/0x220 [<c134fcf4>] ? bus_add_driver+0xd4/0x200 [<c1350b91>] ? driver_register+0x51/0xe0 [<e0be3000>] ? 0xe0be3000 [<e0be3000>] ? 0xe0be3000 [<c10003f2>] ? do_one_initcall+0x72/0x1a0 [<e0be3000>] ? 0xe0be3000 [<c14ac646>] ? netlink_broadcast_filtered+0x166/0x3c0 [<c106c4a5>] ? do_init_module+0x45/0x200 [<c10a8425>] ? __vunmap+0x85/0xe0 [<c106dfaa>] ? load_module+0x186a/0x1ea0 [<c106e6aa>] ? SyS_init_module+0xca/0x100 [<c160a2dd>] ? syscall_call+0x7/0x7 ---[ end trace 826ff8353957cad4 ]---

2 3

[alsa-devel] [PATCH] ASoC: sun4i-codec: Rework and fix headphone routing
by Maxime Ripard 05 Oct '15

05 Oct '15

Most of the boards have their headphone jack directly connected to the matching pins of the SoCs. Since most of the time we will have the same routing path, it makes sense to put that in the driver, and only have a property describing whether that route is enabled or not. It also fixes the following warning messages that were seen so far: sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Left -> direct -> Headphone Jack sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Right -> direct -> Headphone Jack Reported-by: Priit Laes <plaes(a)plaes.org> Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- .../devicetree/bindings/sound/sun4i-codec.txt | 9 ++++---- sound/soc/sunxi/sun4i-codec.c | 25 +++++++++++++++++----- 2 files changed, 24 insertions(+), 10 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt index 680144b74ae9..470823845698 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-codec.txt @@ -13,10 +13,10 @@ Required properties: - clock-names: should contain followings: - "apb": the parent APB clock for this controller - "codec": the parent module clock -- 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. +Optional Properties: +- allwinner,hp-jack-attached: Indicates that the headphone jack on the + board is directly routed to the SoC pins Example: codec: codec@01c22c00 { @@ -28,6 +28,5 @@ codec: codec@01c22c00 { clock-names = "apb", "codec"; dmas = <&dma 0 19>, <&dma 0 19>; dma-names = "rx", "tx"; - routing = "Headphone Jack", "HP Right", - "Headphone Jack", "HP Left"; + allwinner,hp-jack-attached; }; diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 47780552dcd0..e86b8fcede5a 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -568,10 +568,18 @@ static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev, return link; }; +static const struct snd_soc_dapm_widget sun4i_codec_sink_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static const struct snd_soc_dapm_route sun4i_codec_sink_routes[] = { + { "Headphone Jack", NULL, "HP Right" }, + { "Headphone Jack", NULL, "HP Left" }, +}; + static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) { struct snd_soc_card *card; - int ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) @@ -584,10 +592,17 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) card->dev = dev; card->name = "sun4i-codec"; - ret = snd_soc_of_parse_audio_routing(card, "routing"); - if (ret) { - dev_err(dev, "Failed to create our audio routing\n"); - return NULL; + /* + * FIXME: This will probably require more logic once we will + * support other sinks (for capture for example), or + * encounter more complicated setups where the + * headphone jack is not directly connected to the SoC. + */ + if (of_property_read_bool(dev->of_node, "allwinner,hp-jack-attached")) { + card->dapm_widgets = sun4i_codec_sink_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_sink_widgets); + card->dapm_routes = sun4i_codec_sink_routes; + card->num_dapm_routes = ARRAY_SIZE(sun4i_codec_sink_routes); } return card; -- 2.5.3

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[alsa-devel] [PATCH 1/1] ASoC: da7219: Improve error handling for regulator supplies
by Adam Thomson 05 Oct '15

05 Oct '15

Currently if bulk_enable() of supplies fails, the code still goes on to try and put the device into active state, and set expected IO voltage of the device. This doesn't really make sense so code now returns on bulk_enable() failure, with an error message. Also, to help with debug, failure to get supplies also provides an error message. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- This patch is based on changes introduced under patch: 'ASoC: codecs: Add da7219 codec driver' (Commit 6d817c0e9fd7536be76690bfdee88e8a81c16f7d, kernel/git/broonie/sound.git) sound/soc/codecs/da7219.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 76f8fc2..c86a833 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1554,8 +1554,10 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES, da7219->supplies); - if (ret) + if (ret) { + dev_err(codec->dev, "Failed to get supplies"); return ret; + } /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; @@ -1567,6 +1569,10 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) /* Enable main supplies */ ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies"); + return ret; + } /* Ensure device in active mode */ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); @@ -1574,7 +1580,7 @@ static int da7219_handle_supplies(struct snd_soc_codec *codec) /* Update IO voltage level range */ snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl); - return ret; + return 0; } static void da7219_handle_pdata(struct snd_soc_codec *codec) -- 1.9.3

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[alsa-devel] [PATCH v2 0/2] ASoC: add driver for Atmel I2S controller
by Cyrille Pitchen 05 Oct '15

05 Oct '15

This series of patches adds support to the new Atmel I2S controller embedded on sama5d2 SoCs. ChangeLog v2: - initialize dev->dev before calling dev->caps->mck_init(). Cyrille Pitchen (2): ASoC: atmel-i2s: add DT bindings for I2S controller ASoC: atmel-i2s: add driver for the new Atmel I2S controller .../devicetree/bindings/sound/atmel-i2s.txt | 43 ++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-i2s.c | 760 +++++++++++++++++++++ 4 files changed, 815 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-i2s.txt create mode 100644 sound/soc/atmel/atmel-i2s.c -- 1.8.2.2

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[alsa-devel] Missing return check of of_property_read_*()
by Takashi Iwai 05 Oct '15

05 Oct '15

Hi, I hoped that I wouldn't need to write this kind of post, but since nothing happened so far, so here we go: The following compile warnings are present for quite some time. They are due to the lack of return check from of_property_read*(). 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] switch (val) { ^ sound/soc/codecs/cs35l32.c:272:15: note: ‘val’ was declared here unsigned int val; ^ sound/soc/sti/uniperif_player.c: In function ‘uni_player_init’: sound/soc/sti/uniperif_player.c:1006:6: warning: ‘mode’ may be used uninitialized in this function [-Wmaybe-uninitialized] if (strcasecmp(mode, "hdmi") == 0) ^ sound/soc/sti/uniperif_player.c:985:14: note: ‘mode’ was declared here const char *mode; ^ Both should be trivial to fix, but I won't bother to fix these things by myself, as it results always in a matter of taste how to write. So, please fix them appropriately. Although not shown as compile warning like the above, lots of codec drivers have no return value checks from of_property_read*(), too. A quick grep reveals: % git grep -B1 '^[[:space:]]*of_property_read_' sound/soc sound/soc/codecs/adau1701.c- sound/soc/codecs/adau1701.c: of_property_read_u32(dev->of_node, "adi,pll-clkdiv", -- sound/soc/codecs/adau1701.c- sound/soc/codecs/adau1701.c: of_property_read_u8_array(dev->of_node, "adi,pin-config", -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,boost-manager", &val); -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,sdout-datacfg", &val); -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,battery-threshold", &val); -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,battery-recovery", &val); -- sound/soc/codecs/sta32x.c- sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,output-conf", sound/soc/codecs/sta32x.c- &pdata->output_conf); sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,ch1-output-mapping", sound/soc/codecs/sta32x.c- &pdata->ch1_output_mapping); sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,ch2-output-mapping", sound/soc/codecs/sta32x.c- &pdata->ch2_output_mapping); sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,ch3-output-mapping", -- sound/soc/codecs/sta32x.c- tmp = 140; sound/soc/codecs/sta32x.c: of_property_read_u16(np, "st,drop-compensation-ns", &tmp); -- sound/soc/codecs/sta350.c- sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,output-conf", sound/soc/codecs/sta350.c- &pdata->output_conf); sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,ch1-output-mapping", sound/soc/codecs/sta350.c- &pdata->ch1_output_mapping); sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,ch2-output-mapping", sound/soc/codecs/sta350.c- &pdata->ch2_output_mapping); sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,ch3-output-mapping", -- sound/soc/codecs/sta350.c- tmp = 140; sound/soc/codecs/sta350.c: of_property_read_u16(np, "st,drop-compensation-ns", &tmp); -- sound/soc/codecs/tas5086.c- sound/soc/codecs/tas5086.c: of_property_read_u32(of_node, "ti,charge-period", -- sound/soc/codecs/tlv320aic31xx.c- sound/soc/codecs/tlv320aic31xx.c: of_property_read_u32(np, "ai31xx-micbias-vg", &value); -- sound/soc/codecs/twl4030.c- sound/soc/codecs/twl4030.c: of_property_read_u32(node, "ti,digimic_delay", sound/soc/codecs/twl4030.c- &pdata->digimic_delay); sound/soc/codecs/twl4030.c: of_property_read_u32(node, "ti,ramp_delay_value", sound/soc/codecs/twl4030.c- &pdata->ramp_delay_value); sound/soc/codecs/twl4030.c: of_property_read_u32(node, "ti,offset_cncl_path", -- sound/soc/fsl/fsl_esai.c- esai_priv->synchronous = sound/soc/fsl/fsl_esai.c: of_property_read_bool(np, "fsl,esai-synchronous"); -- sound/soc/omap/omap-abe-twl6040.c- priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); sound/soc/omap/omap-abe-twl6040.c: of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); -- sound/soc/samsung/i2s.c- } sound/soc/samsung/i2s.c: of_property_read_string_index(dev->of_node, -- sound/soc/sh/rcar/rsrc-card.c- /* sampling rate convert */ sound/soc/sh/rcar/rsrc-card.c: of_property_read_u32(node, "convert-rate", &priv->convert_rate); -- sound/soc/sti/uniperif_player.c- sound/soc/sti/uniperif_player.c: of_property_read_u32(pnode, "version", &player->ver); -- sound/soc/sti/uniperif_player.c- sound/soc/sti/uniperif_player.c: of_property_read_u32(pnode, "uniperiph-id", &info->id); -- sound/soc/sti/uniperif_player.c- /* Read the device mode property */ sound/soc/sti/uniperif_player.c: of_property_read_string(pnode, "mode", &mode); -- sound/soc/sti/uniperif_reader.c- sound/soc/sti/uniperif_reader.c: of_property_read_u32(node, "version", &reader->ver); Many of them look really buggy (while some have the default value to fall back). And many codes have even no value checks. These should be fixed as well. thanks, Takashi

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[alsa-devel] Fwd: Fwd: [PATCH 1/1] SPI : spi-pxa2xx : fix spi init of WM510205 codec via ACPI (resend)
by Christian Hartmann 05 Oct '15

05 Oct '15

Hi, this is a resend with all recipients in cc now ---------- Forwarded message ---------- From: Christian Hartmann <cornogle(a)googlemail.com> Date: 2015-08-28 9:41 GMT+02:00 Subject: Re: [alsa-devel] Fwd: [PATCH 1/1] SPI : spi-pxa2xx : fix spi init of WM510205 codec via ACPI (resend) To: Mark Brown <broonie(a)kernel.org> hi Mark et al., 2015-08-27 16:56 GMT+02:00 Mark Brown <broonie(a)kernel.org>: > I can't see how that would work, the interrupt output isn't a GPIO. yes I think also that this is the GPIO pin number. I am not familiar with the interrupt handling, have again to look (in other drivers or similar), to understand how things should work, never done that before. I have added three dev_info() into spi-pxa2xx pxa2xx_spi_probe() to found out more whats wrong with the slave/master: commit 4b6ff73610deef478e53b958a02617f024bbef22 Author: Christian Hartmann <cornogle(a)googlemail.com> Date: Thu Aug 27 14:51:48 2015 +0200 spi/spi-pxa2xx.c : added three new dev_infos, which prints irq, bus_num and num_chipselect Signed-off-by: Christian Hartmann <cornogle(a)googlemail.com> diff --git a/drivers/spi/spi-pxa2xx.c b/drivers/spi/spi-pxa2xx.c index 6d795a5..cf02094 100644 --- a/drivers/spi/spi-pxa2xx.c +++ b/drivers/spi/spi-pxa2xx.c @@ -1389,6 +1389,9 @@ static int pxa2xx_spi_probe(struct platform_device *pdev) master->unprepare_transfer_hardware = pxa2xx_spi_unprepare_transfer; master->auto_runtime_pm = true; + dev_info(&pdev->dev, "bus_num = %d\n",master->bus_num); + dev_info(&pdev->dev, "num_chipselect = %d\n",master->num_chipselect); + drv_data->ssp_type = ssp->type; drv_data->null_dma_buf = (u32 *)PTR_ALIGN(&drv_data[1], DMA_ALIGNMENT); @@ -1423,6 +1426,8 @@ static int pxa2xx_spi_probe(struct platform_device *pdev) } + dev_info(&pdev->dev, "IRQ %d\n", ssp->irq); + /* Setup DMA if requested */ drv_data->tx_channel = -1; drv_data->rx_channel = -1; and thats what I got: [ 6.182903] pxa2xx-spi 80860F0E:00: bus_num = -1 [ 6.182909] pxa2xx-spi 80860F0E:00: num_chipselect = 2 [ 6.182928] pxa2xx-spi 80860F0E:00: IRQ 194 [ 6.183049] pxa2xx-spi 80860F0E:00: no DMA channels available, using PIO [ 6.183111] pxa2xx-spi 80860F0E:00: registered master spi32766 (dynamic) [ 6.192761] spi spi-WM510205:00: 8333333 Hz actual, PIO .... the rest of the dmesg is currently the same.... Q: this thread is about the patch for the pxa2xx-spi where the num_chipselect was raised by one to so that the slave connects here at all. otherwise the cs1 >= max will be hit. Do you think the bus_number is ok ? And shouldn't it be set the cs1=0 and num_chipselect=1 ?? The DSDT has wrongly SPI=1 as chipselect, either the firmware (DSDT) must be updated or a correct patch enables the spi master/slave init here. I hope or expect that when the spi slave will be init correctly, that the device would also work. The ACPI stuff is already done here, and I am running out of time. In the last resort I have to give up or have to do that at night :) cheers chris

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[alsa-devel] Regarding ASoC: rt286: Fix the runtime error in the booting
by Axel Lin 05 Oct '15

05 Oct '15

Hi Order, I just notice your patch on the ML: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/098390.ht… I didn't subscribe the ML and I'm not in CC so I cannot directly reply to you. I think you should make a copy of memory for index_cache rather than directly use the rt286_index_def if the index_cache is necessary. diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1fbdb4f..210ab415 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -37,7 +37,7 @@ #define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { - const struct reg_default *index_cache; + struct reg_default *index_cache; int index_cache_size; struct regmap *regmap; struct snd_soc_codec *codec; @@ -1160,8 +1160,12 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return -ENODEV; } - rt286->index_cache = rt286_index_def; rt286->index_cache_size = INDEX_CACHE_SIZE; + rt286->index_cache = devm_kmalloc(&i2c->dev, INDEX_CACHE_SIZE, + GFP_KERNEL); + if (!rt286->index_cache) + return -ENOMEM; + rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286);

1 1

[alsa-devel] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered
by Rick Mann 05 Oct '15

05 Oct '15

I've been at this for weeks and I'm still unable to get linux to work correctly with my TI codec. This is on a Beaglebone Black running a 4.1.4 kernel (later versions are worse). I've tried multiple versions of the kernel trying to narrow it down. I keep coming back to 4.1.4 as the most recent that at least generates some audio clocks. The AM335x mcasp0 should be speaking i2s with the tlv320aic3104. I have my own build from source of the BBB 4.1.4-ti-r9 release (mainline plus TI plus Beaglebone stuff). I've sprinkled some logging around the audio drivers, but so far I don't see much, and I really don't understand how all the pieces fit together. One thing I found with my logging is the mcasp MCLK (AHCLKXDIV) is never set. I have logging calls in __davinci_mcasp_set_clkdiv() (sound/soc/davinci/davinci-mcasp.c), which is the only code I could find that sets the McASP clock, and it's never called. I tried to use devmem2 to read the registers in the SoC to see if they were set correctly, but I just get a bus error (other reads do work). Errors in dmesg when I load the device tree overlay for my cape suggest I still don't have something configured correctly (see below). I'd sure appreciate any insight into these latest issues. Thanks! Symptoms: MCLK should be 12 MHz, but is 24.00 MHz. Buffer underflows when running speaker-test. Errors to console upon loading cape: [ 38.502299] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.517917] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.530425] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 38.536996] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 38.564279] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT dmesg upon loading cape overlay (https://github.com/JetForMe/podtique/blob/v1/bbb/cape/Podtique1/BB-ENABLE-P…): [ 568.323587] bone_capemgr bone_capemgr: part_number 'BB-ENABLE-PRU', version 'N/A' [ 568.323629] bone_capemgr bone_capemgr: slot #4: override [ 568.323646] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 568.323663] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-ENABLE-PRU' [ 568.323804] bone_capemgr:capemgr_load_slot:1148: bone_capemgr bone_capemgr: slot #4: Requesting part number/version based 'BB-ENABLE-PRU-00A0.dtbo [ 568.323825] bone_capemgr:capemgr_load_slot:1162: bone_capemgr bone_capemgr: slot #4: Requesting firmware 'BB-ENABLE-PRU-00A0.dtbo' for board-name 'Override Board Name', version '00A0' [ 568.330369] bone_capemgr:capemgr_load_slot:1171: bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; converting to live tree [ 568.341685] gpio-of-helper ocp:gpio_helper: ready [ 568.344695] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; overlay id #0 [ 568.387640] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.387695] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.387712] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.394284] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.410560] snd_soc_core:snd_soc_register_codec:3064: tlv320aic3x-codec 2-0018: codec register 2-0018 [ 568.410614] snd_soc_core:snd_soc_register_dais:2590: tlv320aic3x-codec 2-0018: ASoC: dai register 2-0018 #1 [ 568.410633] snd_soc_core:snd_soc_register_dais:2634: tlv320aic3x-codec 2-0018: ASoC: Registered DAI 'tlv320aic3x-hifi' [ 568.410653] snd_soc_core:snd_soc_register_codec:3138: tlv320aic3x-codec 2-0018: ASoC: Registered codec 'tlv320aic3x-codec.2-0018' [ 568.421529] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.421588] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.421605] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 568.431455] **** RLM davinci-mcasp.c:1734: davinci_mcasp_probe [ 568.431507] base:of_property_match_string:1398: comparing tx with tx [ 568.431544] base:of_property_match_string:1398: comparing rx with tx [ 568.431555] base:of_property_match_string:1398: comparing rx with rx [ 568.431608] base:of_property_match_string:1398: comparing common with tx [ 568.431621] base:of_property_match_string:1398: comparing common with rx [ 568.431633] base:of_property_match_string:1398: comparing rx with tx [ 568.431643] base:of_property_match_string:1398: comparing rx with rx [ 568.434643] base:of_property_match_string:1398: comparing tx with tx [ 568.437971] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 568.446651] snd_soc_core:snd_soc_register_dais:2590: davinci-mcasp 48038000.mcasp: ASoC: dai register 48038000.mcasp #1 [ 568.446682] snd_soc_core:snd_soc_register_dais:2634: davinci-mcasp 48038000.mcasp: ASoC: Registered DAI '48038000.mcasp' [ 568.446840] snd_soc_core:snd_soc_add_platform:2892: davinci-mcasp 48038000.mcasp: ASoC: Registered platform '48038000.mcasp' [ 568.450046] **** RLM davinci-evm.c:480: RLM: clock rate: 12000000 [ 568.450100] snd_soc_core:soc_bind_dai_link:938: davinci_evm ocp:sound: ASoC: binding TLV320AIC3X at idx 0 [ 568.450425] snd_soc_core:snd_soc_dapm_new_dai_widgets:3494: tlv320aic3x-codec 2-0018: ASoC: adding Playback widget [ 568.450448] snd_soc_core:snd_soc_dapm_new_dai_widgets:3513: tlv320aic3x-codec 2-0018: ASoC: adding Capture widget [ 568.450995] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -2 [ 568.451015] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late -1 [ 568.451030] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 0 [ 568.451045] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 1 [ 568.451059] snd_soc_core:soc_probe_link_dais:1334: davinci_evm ocp:sound: ASoC: probe Podtique DaVinci EVM dai link 0 late 2 [ 568.451099] davinci_evm ocp:sound: ASoC: DAPM unknown pin MONO_LOUT [ 568.465530] snd_soc_core:soc_new_pcm:2510: davinci_evm ocp:sound: ASoC: registered pcm #0 AIC3X tlv320aic3x-hifi-0 [ 568.466155] davinci_evm ocp:sound: tlv320aic3x-hifi <-> 48038000.mcasp mapping ok [ 568.466185] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Right ADC -> Capture [ 568.466206] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Left ADC -> Capture [ 568.466226] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Right DAC [ 568.466241] snd_soc_core:snd_soc_dapm_link_dai_widgets:3570: tlv320aic3x-codec 2-0018: Playback -> Left DAC upon starting speaker-test: [ 572.786071] snd_soc_core:soc_pcm_open:535: ASoC: tlv320aic3x-hifi <-> 48038000.mcasp info: [ 572.786103] snd_soc_core:soc_pcm_open:536: ASoC: rate mask 0x7fe [ 572.786114] snd_soc_core:soc_pcm_open:538: ASoC: min ch 2 max ch 2 [ 572.786126] snd_soc_core:soc_pcm_open:540: ASoC: min rate 8000 max rate 96000 [ 572.841514] davinci-mcasp 48038000.mcasp: Transmit buffer underflow ...repeated [ 574.087162] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 579.135283] snd_soc_core:close_delayed_work:602: TLV320AIC3X: ASoC: pop wq checking: Playback status: inactive waiting: yes -- Rick Mann rmann(a)latencyzero.com

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[alsa-devel] [PATCH v2] ASoC: rt286: Fix the runtime error in the booting
by Oder Chiou 05 Oct '15

05 Oct '15

The struct rt286_index_def was used by the cache function, so it cannot be declared as const. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt286.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1fbdb4f..a7178ac 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -37,7 +37,7 @@ #define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { - const struct reg_default *index_cache; + struct reg_default *index_cache; int index_cache_size; struct regmap *regmap; struct snd_soc_codec *codec; @@ -49,7 +49,7 @@ struct rt286_priv { int clk_id; }; -static const struct reg_default rt286_index_def[] = { +static struct reg_default rt286_index_def[] = { { 0x01, 0xaaaa }, { 0x02, 0x8aaa }, { 0x03, 0x0002 }, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt286: Fix the runtime error in the booting
by Oder Chiou 05 Oct '15

05 Oct '15

The struct rt286_index_def was used by the cache function, so it cannot be declared as const. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1fbdb4f..79107dc 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -49,7 +49,7 @@ struct rt286_priv { int clk_id; }; -static const struct reg_default rt286_index_def[] = { +static struct reg_default rt286_index_def[] = { { 0x01, 0xaaaa }, { 0x02, 0x8aaa }, { 0x03, 0x0002 }, -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] baytrail crv2 audio mic no data
by Lemon 05 Oct '15

05 Oct '15

Hello everyone, I have a Z3735F board which use ssp0 connect to rt5640, I using linux mainline 4.2 and fw_sst_0f28.bin from https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/?h=byt the audio sound is ok, I tried many alsa config but the analog mic record data is all 0, kernel no error message. I don't know why mic record data is 0, does anyone knows ? regards LemonZou ======================================== arecord -f dat -t wav -d 5 foobar.wav hexdump foobar.wav 0000000 4952 4646 a624 000e 4157 4556 6d66 2074 0000010 0010 0000 0001 0002 bb80 0000 ee00 0002 0000020 0004 0010 6164 6174 a600 000e 0000 0000 0000030 0000 0000 0000 0000 0000 0000 0000 0000 * 00ea62c my board mic connect to alc5640 in2p, my main config include: "Stereo ADC MIXL ADC1" on "Stereo ADC1 Mux" "ADCL" "RECMIXL INL" on "RECMIXL BST2" on "pcm1_out mix 0 codec_in0" on "pcm1_out mix 0 codec_in1" on kernel message: [ 5329.874448] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp0-port [ 5329.979365] intel_sst_acpi 80860F28:00: Alloc for str 3 pipe 0xe [ 5335.250574] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp0-port [ 5335.253528] intel_sst_acpi 80860F28:00: Free for str 3 pipe 0xe ========================================

1 0

[alsa-devel] [REANNOUNCE] Audio Mini Summit 2015, 8th October, Dublin
by Mark Brown 05 Oct '15

05 Oct '15

Like previous years, we're going to hold a meeting to discuss lowlevel audio on Linux. This will be held the day after ELC Europe on 8th October at CCD (The Convention Centre) in Dublin. If you're interested please sign up in the attendee list: https://docs.google.com/spreadsheets/d/1bi-8GHsqlzt41FR20WiMuWILe_cZowFireV… and please follow up to this mail with any topics you'd like to see raised so we can start collecting them (probably another Google doc will be forthcoming for them). Previously suggested topics include: - Reducing kernel size for IoT applications - ACPI simple-card - Testing tools Thanks, Mark

9 25

[alsa-devel] [PATCH] ALSA: hda - Disable power_save_node for IDT 92HD73xx chips
by Takashi Iwai 04 Oct '15

04 Oct '15

The recent widget power saving introduced some unavoidable click noises on old IDT 92HD73xx chips while it still seems working on the compatible new chips. In the bugzilla, we tried lots of tests and workarounds, but they didn't help much. So, let's disable the feature for these specific chips as the least (but safest) fix. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=104981 Cc: <stable(a)vger.kernel.org> # v4.1+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9d947aef2c8b..def5cc8dff02 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4520,7 +4520,11 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_save_node = 1; + /* enable power_save_node only for new 92HD89xx chips, as it causes + * click noises on old 92HD73xx chips. + */ + if ((codec->core.vendor_id & 0xfffffff0) != 0x111d7670) + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; -- 2.5.3

1 0

[alsa-devel] [PATCH 0/7 v3] firewire-tascam: new driver for TASCAM FireWire series
by Takashi Sakamoto 04 Oct '15

04 Oct '15

Hi, This patchset updates a part of my former post: [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html This patchset adds support for a part of TASCAM FireWire series. This patchset supports their streaming and PCM functionalities, while their MIDI functionalities are not supported yet. In future work, the functionalities will be supported. Takashi Sakamoto (7): ALSA: firewire-tascam: add skeleton for TASCAM FireWire series ALSA: firewire-tascam: add a structure for model-dependent parameters. ALSA: firewire-tascam: add proc node to show firmware information ALSA: firewire-tascam: add data block processing layer ALSA: firewire-tascam: add streaming functionality ALSA: firewire-tascam: add PCM functionality ALSA: firewire-tascam: add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 12 + sound/firewire/Makefile | 1 + sound/firewire/tascam/Makefile | 3 + sound/firewire/tascam/amdtp-tascam.c | 243 +++++++++++++++++ sound/firewire/tascam/tascam-hwdep.c | 201 ++++++++++++++ sound/firewire/tascam/tascam-pcm.c | 312 +++++++++++++++++++++ sound/firewire/tascam/tascam-proc.c | 88 ++++++ sound/firewire/tascam/tascam-stream.c | 496 ++++++++++++++++++++++++++++++++++ sound/firewire/tascam/tascam.c | 213 +++++++++++++++ sound/firewire/tascam/tascam.h | 117 ++++++++ 12 files changed, 1689 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/tascam/Makefile create mode 100644 sound/firewire/tascam/amdtp-tascam.c create mode 100644 sound/firewire/tascam/tascam-hwdep.c create mode 100644 sound/firewire/tascam/tascam-pcm.c create mode 100644 sound/firewire/tascam/tascam-proc.c create mode 100644 sound/firewire/tascam/tascam-stream.c create mode 100644 sound/firewire/tascam/tascam.c create mode 100644 sound/firewire/tascam/tascam.h -- 2.1.4

5 19

[alsa-devel] [PATCH V5 1/2] ACPI / EC: Fix broken 64bit big-endian users of 'global_lock'
by Viresh Kumar 04 Oct '15

04 Oct '15

global_lock is defined as an unsigned long and accessing only its lower 32 bits from sysfs is incorrect, as we need to consider other 32 bits for big endian 64-bit systems. There are no such platforms yet, but the code needs to be robust for such a case. Fix that by changing type of 'global_lock' to u32. Signed-off-by: Viresh Kumar <viresh.kumar(a)linaro.org> --- BCC'd a lot of people (rather than cc'ing them) to make sure - the series reaches them - mailing lists do not block the patchset due to long cc list - and we don't spam the BCC'd people for every reply V4->V5: - Switch back to the original solution of making global_lock u32. --- drivers/acpi/ec_sys.c | 2 +- drivers/acpi/internal.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/acpi/ec_sys.c b/drivers/acpi/ec_sys.c index b4c216bab22b..bea8e425a8de 100644 --- a/drivers/acpi/ec_sys.c +++ b/drivers/acpi/ec_sys.c @@ -128,7 +128,7 @@ static int acpi_ec_add_debugfs(struct acpi_ec *ec, unsigned int ec_device_count) if (!debugfs_create_x32("gpe", 0444, dev_dir, (u32 *)&first_ec->gpe)) goto error; if (!debugfs_create_bool("use_global_lock", 0444, dev_dir, - (u32 *)&first_ec->global_lock)) + &first_ec->global_lock)) goto error; if (write_support) diff --git a/drivers/acpi/internal.h b/drivers/acpi/internal.h index 9e426210c2a8..9db196de003c 100644 --- a/drivers/acpi/internal.h +++ b/drivers/acpi/internal.h @@ -138,7 +138,7 @@ struct acpi_ec { unsigned long gpe; unsigned long command_addr; unsigned long data_addr; - unsigned long global_lock; + u32 global_lock; unsigned long flags; unsigned long reference_count; struct mutex mutex; -- 2.4.0

3 3

[alsa-devel] [RFC] MIDI over Bluetooth Low Energy
by Felipe Tonello 03 Oct '15

03 Oct '15

Hi guys, I am planning to start the support of MIDI BLE profile[1]. This profile is not officially supported yet, but it will most likely be very similar, so development efforts are still valid. I suggest two main goals: * To be transparent to applications, i.e., use rawmidi and sequencer ALSA interfaces to interact. * To support peripheral and central BLE roles. My question is: what is the best way possible of doing it? My initial though is to write a GATT BlueZ profile plugin that will load snd-virtmidi module with id and midi_devs parameters, then read and write seq events from/to it. I am not sure if this is really possible. Another way of implementing is as a rawmidi and a seq plugin using the BlueZ GATT D-Bus interface. IMO this is not ideal because it requires a lot more work (rawmidi and seq plugins, maybe even a library to avoid code duplication) and has an overhead of using dbus. It is also possible to write a kernel module to handle ALSA card/device setup and reads and writes from the bluez plugin (perhaps this simplifies things because it has less dependencies). They all have the problem of context switching between bluez plugin and alsa midi driver. I would prefer to use a shared ring buffer between ALSA e BlueZ. Any ideas and comments? [1] https://developer.apple.com/bluetooth/Apple-Bluetooth-Low-Energy-MIDI-Speci… PS: I am at #bluez and #alsa as ftonello. Felipe Tonello

5 15

[alsa-devel] [PATCH] ALSA: hda - Apply SPDIF pin ctl to MacBookPro 12, 1
by Takashi Iwai 03 Oct '15

03 Oct '15

From: John Flatness <john(a)zerocrates.org> The MacBookPro 12,1 has the same setup as the 11 for controlling the status of the optical audio light. Simply apply the existing workaround to the subsystem ID for the 12,1. [sorted the fixup entry by tiwai] Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=105401 Signed-off-by: John Flatness <john(a)zerocrates.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 584a0343ab0c..85813de26da8 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -633,6 +633,7 @@ static const struct snd_pci_quirk cs4208_mac_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x5e00, "MacBookPro 11,2", CS4208_MBP11), SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6), SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6), + SND_PCI_QUIRK(0x106b, 0x7b00, "MacBookPro 12,1", CS4208_MBP11), {} /* terminator */ }; -- 2.5.3

1 0

[alsa-devel] [PATCH] ALSA: hda: Add dock support for ThinkPad T550
by Laura Abbott 03 Oct '15

03 Oct '15

Much like all the other Lenovo laptops, add a quirk to make sound work with docking. Reported-and-tested-by: lacknerflo(a)gmail.com Signed-off-by: Laura Abbott <labbott(a)fedoraproject.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index afec6dc..16b8dcb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5306,6 +5306,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.4.3

2 1

[alsa-devel] [RESEND PATCH v2 1/1] ASoC: dwc: fix dma stop transferring issue
by yitian 03 Oct '15

03 Oct '15

Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music continues to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally causes music playing will be stopped once XRUN occurs for the first time. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- changes in V2: - add definition for i and irq --- sound/soc/dwc/designware_i2s.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..76b2e19 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -141,13 +141,22 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { - + u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 1.7.12.4

2 6

[alsa-devel] Kernel Device Tree entries for simple-audio-card
by Rick Mann 03 Oct '15

03 Oct '15

Hi. I've built a Beaglebone Black Cape that has, among other things, a TI Audio CODEC on it (TLV320AIC3104). This is the same CODEC as the BBB Audio Cape (http://elinux.org/CircuitCo:Audio_Cape_RevB), and it's connected in almost the exact same way. The last few days I've been figuring out the Device Tree overlay needed to get Linux (4.1.4) to recognize it. It's finding the device and instantiating the low-level driver (tlv320aic3x-codec), although I'm not sure at this point if it's actually talking to it (I have verified that low-level i2c commands do work). But when it comes to the sound card setup, I'm having trouble getting the Device Tree right. I'm hoping someone on this list can offer some insight. The DT is here: https://github.com/JetForMe/podtique/blob/master/bbb/cape/Podtique1/BB-ENAB… And this is dmesg when I try to load that (note the name of my overlay is BB-ENABLE-PRU, which is an artifact of the experimentation I've been doing; it'll change to something more appropriate): [ 65.859275] bone_capemgr bone_capemgr: part_number 'BB-ENABLE-PRU', version 'N/A' [ 65.859318] bone_capemgr bone_capemgr: slot #4: override [ 65.859336] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 65.859355] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-ENABLE-PRU' [ 65.859488] bone_capemgr: bone_capemgr bone_capemgr: slot #4: Requesting part number/version based 'BB-ENABLE-PRU-00A0.dtbo [ 65.859506] bone_capemgr: bone_capemgr bone_capemgr: slot #4: Requesting firmware 'BB-ENABLE-PRU-00A0.dtbo' for board-name 'Override Board Name', version '00A0' [ 65.861105] bone_capemgr: bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; converting to live tree [ 65.875618] gpio-of-helper ocp:gpio_helper: ready [ 65.876327] gpiolib_of: of_get_named_gpiod_flags: parsed 'gpio' property of node '//fixedregulator@1[0]' - status (0) [ 65.876619] core: lz-codec-reg: no parameters [ 65.876713] fixed: reg-fixed-voltage fixedregulator@1: lz-codec-reg supplying 0uV [ 65.883284] pruss_uio 4a300000.pruss: pins are not configured from the driver [ 65.900080] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-ENABLE-PRU-00A0.dtbo' loaded; overlay id #0 [ 65.946254] gpiolib_of: of_get_named_gpiod_flags: parsed 'gpio-reset' property of node '/ocp/i2c@4819c000/tlv320aic3104@0[0]' - status (0) [ 65.946317] core: tlv320aic3x-codec 2-0018: Looking up IOVDD-supply from device tree [ 65.946405] core: tlv320aic3x-codec 2-0018: Looking up DVDD-supply from device tree [ 65.946468] core: tlv320aic3x-codec 2-0018: Looking up AVDD-supply from device tree [ 65.946544] core: tlv320aic3x-codec 2-0018: Looking up DRVDD-supply from device tree [ 65.946606] snd_soc_core: tlv320aic3x-codec 2-0018: codec register 2-0018 [ 65.946640] snd_soc_core: tlv320aic3x-codec 2-0018: ASoC: dai register 2-0018 #1 [ 65.946655] snd_soc_core: tlv320aic3x-codec 2-0018: ASoC: Registered DAI 'tlv320aic3x-hifi' [ 65.946671] snd_soc_core: tlv320aic3x-codec 2-0018: ASoC: Registered codec 'tlv320aic3x-codec.2-0018' [ 65.956491] //sound/simple-audio-card,cpu: could not get #sound-dai-cells for /ocp/mcasp@48038000 [ 65.978759] snd_soc_core: davinci-mcasp 48038000.mcasp: ASoC: dai register 48038000.mcasp #1 [ 65.978795] snd_soc_core: davinci-mcasp 48038000.mcasp: ASoC: Registered DAI '48038000.mcasp' [ 65.978954] snd_soc_core: davinci-mcasp 48038000.mcasp: ASoC: Registered platform '48038000.mcasp' [ 66.012427] asoc-simple-card sound: parse error -22 [ 66.042609] asoc-simple-card: probe of sound failed with error -22 Note the bit about "#sound-dai-cells". I'm not sure what's wrong here, and it seems like it might be something in the distro-provided DTB, not my own. aplay lists nothing (I think in part because I have yet to put in a proper asound.conf file): # aplay -l aplay: device_list:268: no soundcards found... Can someone suggest some diagnostics I can do to see what the state of things is? I'm just not sure what to look for. Thanks! -- Rick Mann rmann(a)latencyzero.com

2 18

[alsa-devel] [PATCH MC Next Gen 00/20] Update ALSA, and au0828 drivers to use Managed Media Controller API
by Shuah Khan 03 Oct '15

03 Oct '15

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API to share tuner. Please note that Managed Media Controller API and DVB and V4L2 drivers updates to use Media Controller API have been added in a prior patch series. Media Controller API is enhanced with two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. Please find the graphs generated using mc_nextgen_test tool at various points during testing at: https://drive.google.com/folderview?id=0B0NIL0BQg-Alb3JFb2diMXRoQlU&usp=sha… This patch series is the port of the same work that was done in the following patch series to use Media Controller Next Gen API. Patch v3 - Media Controller API: https://www.mail-archive.com/linux-media@vger.kernel.org/msg92572.html References and History: Changes since v2: http://www.spinics.net/lists/linux-media/msg91926.html 1. An important change in this patch series is made to ALSA to check for resources and hold in snd_usb_hw_params(), and release from snd_usb_hw_free(). This change fixed the lockdep warnings seen when resources were held in TRIGGER_START and released from TRIGGER_STOP which could run in IRQ context. I acknowledge Clemens Ladisch for suggesting the correct places to hold/free resources to avoid IRQ path complications. 2. With the above change, the patch series is simpler without the need to change the graph_mutex into a spinlock. 3. I split the patches up differently for easy reviews - no code bloat from v2. 4. A second important change is now the Bridge driver (au0828) owns and drives the graph creation as well as enabling and disabling tuner. It also keeps state information to avoid graph walks in enable_source and disable_source handler. I acknowledge Hans Verkuil for his suggestions and ideas for this change. History: This patch series has been updated to address comments from 3 previous versions of this series. Links to v3 version for reference are: https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89491.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg89492.html https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89493.html Shuah Khan (20): media: Media Controller register/unregister entity_notify API media: Add ALSA Media Controller function entities media: Media Controller enable/disable source handler API media: Media Controller fix to not let stream_count go negative media: Media Controller export non locking __media_entity_setup_link() media: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add v4l_enable/disable_media_tuner() helper functions media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: au0828 Use au8522_media_pads enum for pad defines media: au0828 fix au0828_create_media_graph() entity checks media: Change v4l-core to check for tuner availability media: dvb-frontend invoke enable/disable_source handlers media: au0828 video remove au0828_enable_analog_tuner() media: au0828 video change to use v4l_enable_media_tuner() media: au0828 change to use Managed Media Controller API media: au0828 change to register/unregister entity_notify hook media: au0828 implement enable_source and disable_source handlers media: dvb-core create tuner to demod pad link in disabled state sound/usb: Update ALSA driver to use Managed Media Controller API drivers/media/dvb-core/dvb_frontend.c | 139 ++--------- drivers/media/dvb-core/dvb_frontend.h | 3 + drivers/media/dvb-core/dvbdev.c | 3 +- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 43 ++++ drivers/media/media-entity.c | 64 +++-- drivers/media/usb/au0828/au0828-core.c | 337 ++++++++++++++++++++------- drivers/media/usb/au0828/au0828-video.c | 75 +----- drivers/media/usb/au0828/au0828.h | 8 + drivers/media/v4l2-core/v4l2-dev.c | 27 +++ drivers/media/v4l2-core/v4l2-fh.c | 1 + drivers/media/v4l2-core/v4l2-ioctl.c | 29 +++ drivers/media/v4l2-core/videobuf2-core.c | 3 + include/media/media-device.h | 44 ++++ include/media/media-entity.h | 3 + include/media/v4l2-dev.h | 4 + include/uapi/linux/media.h | 9 +- sound/usb/Makefile | 15 +- sound/usb/card.c | 5 + sound/usb/card.h | 1 + sound/usb/media.c | 201 ++++++++++++++++ sound/usb/media.h | 53 +++++ sound/usb/mixer.c | 1 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 1 + 30 files changed, 823 insertions(+), 304 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

1 20

[alsa-devel] [PATCH 0/8] Add support for Wolfson WM8998 and WM1814 codecs
by Richard Fitzgerald 02 Oct '15

02 Oct '15

This set of patches adds support for the Wolfson Microelectronics WM8998 and WM1814 codecs. Richard Fitzgerald (8): mfd: arizona: Split INx_MODE into two fields ASoC: arizona: add defines for single-input gain control mfd: arizona: Add support for WM8998 and WM1814 gpio: arizona: add support for WM8998 and WM1814 regulator: arizona: add support for WM8998 and WM1814 extcon: arizona: Add support for WM8998 and WM1814 ASoC: wm8998: Initial WM8998 codec driver Documentation: Add WM8998/WM1814 device tree bindings Documentation/devicetree/bindings/mfd/arizona.txt | 3 + drivers/extcon/extcon-arizona.c | 33 +- drivers/gpio/gpio-arizona.c | 2 + drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 106 ++- drivers/mfd/arizona-i2c.c | 8 + drivers/mfd/arizona-irq.c | 9 + drivers/mfd/arizona.h | 5 + drivers/mfd/wm8998-tables.c | 1592 +++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 2 + include/linux/mfd/arizona/core.h | 3 + include/linux/mfd/arizona/pdata.h | 5 +- include/linux/mfd/arizona/registers.h | 247 +++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 7 + sound/soc/codecs/arizona.h | 7 +- sound/soc/codecs/wm8998.c | 1392 ++++++++++++++++++ sound/soc/codecs/wm8998.h | 23 + 20 files changed, 3432 insertions(+), 29 deletions(-) create mode 100644 drivers/mfd/wm8998-tables.c create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h -- 1.7.2.5

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[alsa-devel] [PATCH v2 0/3] Add support for Wolfson Micro WM8998 codec
by Richard Fitzgerald 02 Oct '15

02 Oct '15

This set of patches is based on broonie/regulator.git/for-next and broonie/sound.git/for-next branches. Richard Fitzgerald (3): regulator: arizona: add support for WM8998 and WM1814 ASoC: arizona: add defines for single-input gain control ASoC: wm8998: Initial WM8998 codec driver drivers/regulator/arizona-ldo1.c | 2 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 33 + sound/soc/codecs/arizona.h | 12 +- sound/soc/codecs/wm8998.c | 1430 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8998.h | 23 + 7 files changed, 1507 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h -- 1.9.1

3 6

[alsa-devel] [PATCH 0/8] ASoC: topology: ABI - Add support for PCM, BE & CC link
by mengdong.lin＠intel.com 02 Oct '15

02 Oct '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This series includes topology ABI update to support PCM (FE DAI/DAI links), BE and CC DAI links. Bump the ABI version to 4. Mengdong Lin (2): ASoC: topology: ABI - Change stream formats to a bitwise flag ASoC: topology: ABI - Remove tdm_slot & dai_fmt from snd_soc_tplg_stream Vedang Patel (6): ASoC: topology: ABI - Add PCM Support and bump ABI version to 4 ASoC: topology: ABI - Remove unused struct snd_soc_tplg_stream_config ASoC: topology: ABI - Use __le32 instead of __u32 in snd_soc_tplg_dapm_widget ASoC: topology: ABI - Add name element to snd_soc_tplg_stream ASoC: topology: ABI - Add the type for BE DAI link ASoC: topology: ABI - Add configuration for BE & Codec-Codec DAI Links include/uapi/sound/asoc.h | 76 +++++++++++++++++++++++++---------------------- sound/soc/soc-topology.c | 4 +-- 2 files changed, 43 insertions(+), 37 deletions(-) -- 1.9.1

3 17

[alsa-devel] Applied "ASoC: dwc: fix dma stop transferring issue" to the asoc tree
by Mark Brown 02 Oct '15

02 Oct '15

The patch ASoC: dwc: fix dma stop transferring issue 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 924eb475126fd6bc23c475ac742a69fc466c0b75 Mon Sep 17 00:00:00 2001 From: yitian <yitian.bu(a)tangramtek.com> Date: Tue, 29 Sep 2015 22:43:17 +0800 Subject: [PATCH] ASoC: dwc: fix dma stop transferring issue Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music continues to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally causes music playing will be stopped once XRUN occurs for the first time. [On list discussion suggests this may be partly a race condition on slow systems -- broonie] Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/dwc/designware_i2s.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 0d28e3b..ba34252 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -141,13 +141,22 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { - + u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 2.5.0

1 0

[alsa-devel] Applied "ASoC: codecs: Add da7219 codec driver" to the asoc tree
by Mark Brown 02 Oct '15

02 Oct '15

The patch ASoC: codecs: Add da7219 codec driver 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 6d817c0e9fd7536be76690bfdee88e8a81c16f7d Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Tue, 29 Sep 2015 16:44:01 +0100 Subject: [PATCH] ASoC: codecs: Add da7219 codec driver This adds support for the DA7219 audio codec with built-in advanced accessory detect features. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/da7219-aad.h | 99 +++ include/sound/da7219.h | 55 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 823 +++++++++++++++++ sound/soc/codecs/da7219-aad.h | 212 +++++ sound/soc/codecs/da7219.c | 1940 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 +++++++++++++++++ 8 files changed, 3955 insertions(+) create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h diff --git a/include/sound/da7219-aad.h b/include/sound/da7219-aad.h new file mode 100644 index 0000000..17802fb --- /dev/null +++ b/include/sound/da7219-aad.h @@ -0,0 +1,99 @@ +/* + * da7219-aad.h - DA7322 ASoC Codec AAD Driver Platform Data + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_AAD_PDATA_H +#define __DA7219_AAD_PDATA_H + +enum da7219_aad_micbias_pulse_lvl { + DA7219_AAD_MICBIAS_PULSE_LVL_OFF = 0, + DA7219_AAD_MICBIAS_PULSE_LVL_2_8V = 6, + DA7219_AAD_MICBIAS_PULSE_LVL_2_9V, +}; + +enum da7219_aad_btn_cfg { + DA7219_AAD_BTN_CFG_2MS = 1, + DA7219_AAD_BTN_CFG_5MS, + DA7219_AAD_BTN_CFG_10MS, + DA7219_AAD_BTN_CFG_50MS, + DA7219_AAD_BTN_CFG_100MS, + DA7219_AAD_BTN_CFG_200MS, + DA7219_AAD_BTN_CFG_500MS, +}; + +enum da7219_aad_mic_det_thr { + DA7219_AAD_MIC_DET_THR_200_OHMS = 0, + DA7219_AAD_MIC_DET_THR_500_OHMS, + DA7219_AAD_MIC_DET_THR_750_OHMS, + DA7219_AAD_MIC_DET_THR_1000_OHMS, +}; + +enum da7219_aad_jack_ins_deb { + DA7219_AAD_JACK_INS_DEB_5MS = 0, + DA7219_AAD_JACK_INS_DEB_10MS, + DA7219_AAD_JACK_INS_DEB_20MS, + DA7219_AAD_JACK_INS_DEB_50MS, + DA7219_AAD_JACK_INS_DEB_100MS, + DA7219_AAD_JACK_INS_DEB_200MS, + DA7219_AAD_JACK_INS_DEB_500MS, + DA7219_AAD_JACK_INS_DEB_1S, +}; + +enum da7219_aad_jack_det_rate { + DA7219_AAD_JACK_DET_RATE_32_64MS = 0, + DA7219_AAD_JACK_DET_RATE_64_128MS, + DA7219_AAD_JACK_DET_RATE_128_256MS, + DA7219_AAD_JACK_DET_RATE_256_512MS, +}; + +enum da7219_aad_jack_rem_deb { + DA7219_AAD_JACK_REM_DEB_1MS = 0, + DA7219_AAD_JACK_REM_DEB_5MS, + DA7219_AAD_JACK_REM_DEB_10MS, + DA7219_AAD_JACK_REM_DEB_20MS, +}; + +enum da7219_aad_btn_avg { + DA7219_AAD_BTN_AVG_1 = 0, + DA7219_AAD_BTN_AVG_2, + DA7219_AAD_BTN_AVG_4, + DA7219_AAD_BTN_AVG_8, +}; + +enum da7219_aad_adc_1bit_rpt { + DA7219_AAD_ADC_1BIT_RPT_1 = 0, + DA7219_AAD_ADC_1BIT_RPT_2, + DA7219_AAD_ADC_1BIT_RPT_4, + DA7219_AAD_ADC_1BIT_RPT_8, +}; + +struct da7219_aad_pdata { + int irq; + + enum da7219_aad_micbias_pulse_lvl micbias_pulse_lvl; + u32 micbias_pulse_time; + enum da7219_aad_btn_cfg btn_cfg; + enum da7219_aad_mic_det_thr mic_det_thr; + enum da7219_aad_jack_ins_deb jack_ins_deb; + enum da7219_aad_jack_det_rate jack_det_rate; + enum da7219_aad_jack_rem_deb jack_rem_deb; + + u8 a_d_btn_thr; + u8 d_b_btn_thr; + u8 b_c_btn_thr; + u8 c_mic_btn_thr; + + enum da7219_aad_btn_avg btn_avg; + enum da7219_aad_adc_1bit_rpt adc_1bit_rpt; +}; + +#endif /* __DA7219_AAD_PDATA_H */ diff --git a/include/sound/da7219.h b/include/sound/da7219.h new file mode 100644 index 0000000..3f39e13 --- /dev/null +++ b/include/sound/da7219.h @@ -0,0 +1,55 @@ +/* + * da7219.h - DA7219 ASoC Codec Driver Platform Data + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_PDATA_H +#define __DA7219_PDATA_H + +/* LDO */ +enum da7219_ldo_lvl_sel { + DA7219_LDO_LVL_SEL_1_05V = 0, + DA7219_LDO_LVL_SEL_1_10V, + DA7219_LDO_LVL_SEL_1_20V, + DA7219_LDO_LVL_SEL_1_40V, +}; + +/* Mic Bias */ +enum da7219_micbias_voltage { + DA7219_MICBIAS_1_8V = 1, + DA7219_MICBIAS_2_0V, + DA7219_MICBIAS_2_2V, + DA7219_MICBIAS_2_4V, + DA7219_MICBIAS_2_6V, +}; + +/* Mic input type */ +enum da7219_mic_amp_in_sel { + DA7219_MIC_AMP_IN_SEL_DIFF = 0, + DA7219_MIC_AMP_IN_SEL_SE_P, + DA7219_MIC_AMP_IN_SEL_SE_N, +}; + +struct da7219_aad_pdata; + +struct da7219_pdata { + /* Internal LDO */ + enum da7219_ldo_lvl_sel ldo_lvl_sel; + + /* Mic */ + enum da7219_micbias_voltage micbias_lvl; + enum da7219_mic_amp_in_sel mic_amp_in_sel; + + /* AAD */ + struct da7219_aad_pdata *aad_pdata; +}; + +#endif /* __DA7219_PDATA_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..c9a895c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C + select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC @@ -430,6 +431,9 @@ config SND_SOC_DA7210 config SND_SOC_DA7213 tristate +config SND_SOC_DA7219 + tristate + config SND_SOC_DA732X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077..eb10a34 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -49,6 +49,7 @@ snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o +snd-soc-da7219-objs := da7219.o da7219-aad.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-bt-sco-objs := bt-sco.o @@ -241,6 +242,7 @@ obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o +obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c new file mode 100644 index 0000000..9459593 --- /dev/null +++ b/sound/soc/codecs/da7219-aad.c @@ -0,0 +1,823 @@ +/* + * da7219-aad.c - Dialog DA7219 ALSA SoC AAD Driver + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> +#include <linux/pm_wakeirq.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/workqueue.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/da7219.h> + +#include "da7219.h" +#include "da7219-aad.h" + + +/* + * Detection control + */ + +void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + da7219->aad->jack = jack; + da7219->aad->jack_inserted = false; + + /* Send an initial empty report */ + snd_soc_jack_report(jack, 0, DA7219_AAD_REPORT_ALL_MASK); + + /* Enable/Disable jack detection */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_ACCDET_EN_MASK, + (jack ? DA7219_ACCDET_EN_MASK : 0)); +} +EXPORT_SYMBOL_GPL(da7219_aad_jack_det); + +/* + * Button/HPTest work + */ + +static void da7219_aad_btn_det_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, btn_det_work); + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + u8 statusa, micbias_ctrl; + bool micbias_up = false; + int retries = 0; + + /* Drive headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK, + DA7219_HP_L_AMP_OE_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK, + DA7219_HP_R_AMP_OE_MASK); + + /* Make sure mic bias is up */ + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + + do { + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); + if (statusa & DA7219_MICBIAS_UP_STS_MASK) + micbias_up = true; + else if (retries++ < DA7219_AAD_MICBIAS_CHK_RETRIES) + msleep(DA7219_AAD_MICBIAS_CHK_DELAY); + } while ((!micbias_up) && (retries < DA7219_AAD_MICBIAS_CHK_RETRIES)); + + if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES) + dev_warn(codec->dev, "Mic bias status check timed out"); + + /* + * Mic bias pulse required to enable mic, must be done before enabling + * button detection to prevent erroneous button readings. + */ + if (da7219_aad->micbias_pulse_lvl && da7219_aad->micbias_pulse_time) { + /* Pulse higher level voltage */ + micbias_ctrl = snd_soc_read(codec, DA7219_MICBIAS_CTRL); + snd_soc_update_bits(codec, DA7219_MICBIAS_CTRL, + DA7219_MICBIAS1_LEVEL_MASK, + da7219_aad->micbias_pulse_lvl); + msleep(da7219_aad->micbias_pulse_time); + snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_ctrl); + + } + + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, + da7219_aad->btn_cfg); +} + +static void da7219_aad_hptest_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, hptest_work); + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + u16 tonegen_freq_hptest; + u8 accdet_cfg8; + int report = 0; + + /* Lock DAPM and any Kcontrols that are affected by this test */ + snd_soc_dapm_mutex_lock(dapm); + mutex_lock(&da7219->lock); + + /* Bypass cache so it saves current settings */ + regcache_cache_bypass(da7219->regmap, true); + + /* Make sure Tone Generator is disabled */ + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0); + + /* Enable HPTest block, 1KOhms check */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8, + DA7219_HPTEST_EN_MASK | DA7219_HPTEST_RES_SEL_MASK, + DA7219_HPTEST_EN_MASK | + DA7219_HPTEST_RES_SEL_1KOHMS); + + /* Set gains to 0db */ + snd_soc_write(codec, DA7219_DAC_L_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB); + snd_soc_write(codec, DA7219_DAC_R_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB); + snd_soc_write(codec, DA7219_HP_L_GAIN, DA7219_HP_AMP_GAIN_0DB); + snd_soc_write(codec, DA7219_HP_R_GAIN, DA7219_HP_AMP_GAIN_0DB); + + /* Disable DAC filters, EQs and soft mute */ + snd_soc_update_bits(codec, DA7219_DAC_FILTERS1, DA7219_HPF_MODE_MASK, + 0); + snd_soc_update_bits(codec, DA7219_DAC_FILTERS4, DA7219_DAC_EQ_EN_MASK, + 0); + snd_soc_update_bits(codec, DA7219_DAC_FILTERS5, + DA7219_DAC_SOFTMUTE_EN_MASK, 0); + + /* Enable HP left & right paths */ + snd_soc_update_bits(codec, DA7219_CP_CTRL, DA7219_CP_EN_MASK, + DA7219_CP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DIG_ROUTING_DAC, + DA7219_DAC_L_SRC_MASK | DA7219_DAC_R_SRC_MASK, + DA7219_DAC_L_SRC_TONEGEN | + DA7219_DAC_R_SRC_TONEGEN); + snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, + DA7219_DAC_L_EN_MASK | DA7219_DAC_L_MUTE_EN_MASK, + DA7219_DAC_L_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, + DA7219_DAC_R_EN_MASK | DA7219_DAC_R_MUTE_EN_MASK, + DA7219_DAC_R_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_L_MIX_SELECT_MASK, + DA7219_MIXOUT_L_MIX_SELECT_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_R_SELECT, + DA7219_MIXOUT_R_MIX_SELECT_MASK, + DA7219_MIXOUT_R_MIX_SELECT_MASK); + snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1L, + DA7219_OUTFILT_ST_1L_SRC_MASK, + DA7219_DMIX_ST_SRC_OUTFILT1L); + snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1R, + DA7219_OUTFILT_ST_1R_SRC_MASK, + DA7219_DMIX_ST_SRC_OUTFILT1R); + snd_soc_update_bits(codec, DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_MASK, + DA7219_MIXOUT_L_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_MASK, + DA7219_MIXOUT_R_AMP_EN_MASK); + snd_soc_write(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK); + snd_soc_write(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK); + + /* Configure & start Tone Generator */ + snd_soc_write(codec, DA7219_TONE_GEN_ON_PER, DA7219_BEEP_ON_PER_MASK); + tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ); + regmap_raw_write(da7219->regmap, DA7219_TONE_GEN_FREQ1_L, + &tonegen_freq_hptest, sizeof(tonegen_freq_hptest)); + snd_soc_update_bits(codec, DA7219_TONE_GEN_CFG2, + DA7219_SWG_SEL_MASK | DA7219_TONE_GEN_GAIN_MASK, + DA7219_SWG_SEL_SRAMP | + DA7219_TONE_GEN_GAIN_MINUS_15DB); + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, DA7219_START_STOPN_MASK); + + msleep(DA7219_AAD_HPTEST_PERIOD); + + /* Grab comparator reading */ + accdet_cfg8 = snd_soc_read(codec, DA7219_ACCDET_CONFIG_8); + if (accdet_cfg8 & DA7219_HPTEST_COMP_MASK) + report |= SND_JACK_HEADPHONE; + else + report |= SND_JACK_LINEOUT; + + /* Stop tone generator */ + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0); + + msleep(DA7219_AAD_HPTEST_PERIOD); + + /* Restore original settings from cache */ + regcache_mark_dirty(da7219->regmap); + regcache_sync_region(da7219->regmap, DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DROUTING_ST_OUTFILT_1L, + DA7219_DROUTING_ST_OUTFILT_1R); + regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_R_SELECT); + regcache_sync_region(da7219->regmap, DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DIG_ROUTING_DAC, + DA7219_DIG_ROUTING_DAC); + regcache_sync_region(da7219->regmap, DA7219_CP_CTRL, DA7219_CP_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS5, + DA7219_DAC_FILTERS5); + regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS4, + DA7219_DAC_FILTERS1); + regcache_sync_region(da7219->regmap, DA7219_HP_L_GAIN, + DA7219_HP_R_GAIN); + regcache_sync_region(da7219->regmap, DA7219_DAC_L_GAIN, + DA7219_DAC_R_GAIN); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_ON_PER, + DA7219_TONE_GEN_ON_PER); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_FREQ1_L, + DA7219_TONE_GEN_FREQ1_U); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_CFG1, + DA7219_TONE_GEN_CFG2); + + regcache_cache_bypass(da7219->regmap, false); + + /* Disable HPTest block */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8, + DA7219_HPTEST_EN_MASK, 0); + + /* Drive Headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK, + DA7219_HP_L_AMP_OE_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK, + DA7219_HP_R_AMP_OE_MASK); + + mutex_unlock(&da7219->lock); + snd_soc_dapm_mutex_unlock(dapm); + + /* + * Only send report if jack hasn't been removed during process, + * otherwise it's invalid and we drop it. + */ + if (da7219_aad->jack_inserted) + snd_soc_jack_report(da7219_aad->jack, report, + SND_JACK_HEADSET | SND_JACK_LINEOUT); +} + + +/* + * IRQ + */ + +static irqreturn_t da7219_aad_irq_thread(int irq, void *data) +{ + struct da7219_aad_priv *da7219_aad = data; + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 events[DA7219_AAD_IRQ_REG_MAX]; + u8 statusa; + int i, report = 0, mask = 0; + + /* Read current IRQ events */ + regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + + if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) + return IRQ_NONE; + + /* Read status register for jack insertion & type status */ + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); + + /* Clear events */ + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + + dev_dbg(codec->dev, "IRQ events = 0x%x|0x%x, status = 0x%x\n", + events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B], + statusa); + + if (statusa & DA7219_JACK_INSERTION_STS_MASK) { + /* Jack Insertion */ + if (events[DA7219_AAD_IRQ_REG_A] & + DA7219_E_JACK_INSERTED_MASK) { + report |= SND_JACK_MECHANICAL; + mask |= SND_JACK_MECHANICAL; + da7219_aad->jack_inserted = true; + } + + /* Jack type detection */ + if (events[DA7219_AAD_IRQ_REG_A] & + DA7219_E_JACK_DETECT_COMPLETE_MASK) { + /* + * If 4-pole, then enable button detection, else perform + * HP impedance test to determine output type to report. + * + * We schedule work here as the tasks themselves can + * take time to complete, and in particular for hptest + * we want to be able to check if the jack was removed + * during the procedure as this will invalidate the + * result. By doing this as work, the IRQ thread can + * handle a removal, and we can check at the end of + * hptest if we have a valid result or not. + */ + if (statusa & DA7219_JACK_TYPE_STS_MASK) { + report |= SND_JACK_HEADSET; + mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT; + schedule_work(&da7219_aad->btn_det_work); + } else { + schedule_work(&da7219_aad->hptest_work); + } + } + + /* Button support for 4-pole jack */ + if (statusa & DA7219_JACK_TYPE_STS_MASK) { + for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) { + /* Button Press */ + if (events[DA7219_AAD_IRQ_REG_B] & + (DA7219_E_BUTTON_A_PRESSED_MASK << i)) { + report |= SND_JACK_BTN_0 >> i; + mask |= SND_JACK_BTN_0 >> i; + } + } + snd_soc_jack_report(da7219_aad->jack, report, mask); + + for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) { + /* Button Release */ + if (events[DA7219_AAD_IRQ_REG_B] & + (DA7219_E_BUTTON_A_RELEASED_MASK >> i)) { + report &= ~(SND_JACK_BTN_0 >> i); + mask |= SND_JACK_BTN_0 >> i; + } + } + } + } else { + /* Jack removal */ + if (events[DA7219_AAD_IRQ_REG_A] & DA7219_E_JACK_REMOVED_MASK) { + report = 0; + mask |= DA7219_AAD_REPORT_ALL_MASK; + da7219_aad->jack_inserted = false; + + /* Un-drive headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK, 0); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK, 0); + + /* Ensure button detection disabled */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, 0); + + /* Disable mic bias */ + snd_soc_dapm_disable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + + /* Cancel any pending work */ + cancel_work_sync(&da7219_aad->btn_det_work); + cancel_work_sync(&da7219_aad->hptest_work); + } + } + + snd_soc_jack_report(da7219_aad->jack, report, mask); + + return IRQ_HANDLED; +} + +/* + * DT to pdata conversion + */ + +static enum da7219_aad_micbias_pulse_lvl + da7219_aad_of_micbias_pulse_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 2800: + return DA7219_AAD_MICBIAS_PULSE_LVL_2_8V; + case 2900: + return DA7219_AAD_MICBIAS_PULSE_LVL_2_9V; + default: + dev_warn(codec->dev, "Invalid micbias pulse level"); + return DA7219_AAD_MICBIAS_PULSE_LVL_OFF; + } +} + +static enum da7219_aad_btn_cfg + da7219_aad_of_btn_cfg(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 2: + return DA7219_AAD_BTN_CFG_2MS; + case 5: + return DA7219_AAD_BTN_CFG_5MS; + case 10: + return DA7219_AAD_BTN_CFG_10MS; + case 50: + return DA7219_AAD_BTN_CFG_50MS; + case 100: + return DA7219_AAD_BTN_CFG_100MS; + case 200: + return DA7219_AAD_BTN_CFG_200MS; + case 500: + return DA7219_AAD_BTN_CFG_500MS; + default: + dev_warn(codec->dev, "Invalid button config"); + return DA7219_AAD_BTN_CFG_10MS; + } +} + +static enum da7219_aad_mic_det_thr + da7219_aad_of_mic_det_thr(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 200: + return DA7219_AAD_MIC_DET_THR_200_OHMS; + case 500: + return DA7219_AAD_MIC_DET_THR_500_OHMS; + case 750: + return DA7219_AAD_MIC_DET_THR_750_OHMS; + case 1000: + return DA7219_AAD_MIC_DET_THR_1000_OHMS; + default: + dev_warn(codec->dev, "Invalid mic detect threshold"); + return DA7219_AAD_MIC_DET_THR_500_OHMS; + } +} + +static enum da7219_aad_jack_ins_deb + da7219_aad_of_jack_ins_deb(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 5: + return DA7219_AAD_JACK_INS_DEB_5MS; + case 10: + return DA7219_AAD_JACK_INS_DEB_10MS; + case 20: + return DA7219_AAD_JACK_INS_DEB_20MS; + case 50: + return DA7219_AAD_JACK_INS_DEB_50MS; + case 100: + return DA7219_AAD_JACK_INS_DEB_100MS; + case 200: + return DA7219_AAD_JACK_INS_DEB_200MS; + case 500: + return DA7219_AAD_JACK_INS_DEB_500MS; + case 1000: + return DA7219_AAD_JACK_INS_DEB_1S; + default: + dev_warn(codec->dev, "Invalid jack insert debounce"); + return DA7219_AAD_JACK_INS_DEB_20MS; + } +} + +static enum da7219_aad_jack_det_rate + da7219_aad_of_jack_det_rate(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "32ms_64ms")) { + return DA7219_AAD_JACK_DET_RATE_32_64MS; + } else if (!strcmp(str, "64ms_128ms")) { + return DA7219_AAD_JACK_DET_RATE_64_128MS; + } else if (!strcmp(str, "128ms_256ms")) { + return DA7219_AAD_JACK_DET_RATE_128_256MS; + } else if (!strcmp(str, "256ms_512ms")) { + return DA7219_AAD_JACK_DET_RATE_256_512MS; + } else { + dev_warn(codec->dev, "Invalid jack detect rate"); + return DA7219_AAD_JACK_DET_RATE_256_512MS; + } +} + +static enum da7219_aad_jack_rem_deb + da7219_aad_of_jack_rem_deb(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_JACK_REM_DEB_1MS; + case 5: + return DA7219_AAD_JACK_REM_DEB_5MS; + case 10: + return DA7219_AAD_JACK_REM_DEB_10MS; + case 20: + return DA7219_AAD_JACK_REM_DEB_20MS; + default: + dev_warn(codec->dev, "Invalid jack removal debounce"); + return DA7219_AAD_JACK_REM_DEB_1MS; + } +} + +static enum da7219_aad_btn_avg + da7219_aad_of_btn_avg(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_BTN_AVG_1; + case 2: + return DA7219_AAD_BTN_AVG_2; + case 4: + return DA7219_AAD_BTN_AVG_4; + case 8: + return DA7219_AAD_BTN_AVG_8; + default: + dev_warn(codec->dev, "Invalid button average value"); + return DA7219_AAD_BTN_AVG_2; + } +} + +static enum da7219_aad_adc_1bit_rpt + da7219_aad_of_adc_1bit_rpt(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_ADC_1BIT_RPT_1; + case 2: + return DA7219_AAD_ADC_1BIT_RPT_2; + case 4: + return DA7219_AAD_ADC_1BIT_RPT_4; + case 8: + return DA7219_AAD_ADC_1BIT_RPT_8; + default: + dev_warn(codec->dev, "Invalid ADC 1-bit repeat value"); + return DA7219_AAD_ADC_1BIT_RPT_1; + } +} + +static struct da7219_aad_pdata *da7219_aad_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct device_node *aad_np = of_find_node_by_name(np, "da7219_aad"); + struct da7219_aad_pdata *aad_pdata; + const char *of_str; + u32 of_val32; + + if (!aad_np) + return NULL; + + aad_pdata = devm_kzalloc(codec->dev, sizeof(*aad_pdata), GFP_KERNEL); + if (!aad_pdata) + goto out; + + aad_pdata->irq = irq_of_parse_and_map(np, 0); + + if (of_property_read_u32(aad_np, "dlg,micbias-pulse-lvl", + &of_val32) >= 0) + aad_pdata->micbias_pulse_lvl = + da7219_aad_of_micbias_pulse_lvl(codec, of_val32); + else + aad_pdata->micbias_pulse_lvl = DA7219_AAD_MICBIAS_PULSE_LVL_OFF; + + if (of_property_read_u32(aad_np, "dlg,micbias-pulse-time", + &of_val32) >= 0) + aad_pdata->micbias_pulse_time = of_val32; + + if (of_property_read_u32(aad_np, "dlg,btn-cfg", &of_val32) >= 0) + aad_pdata->btn_cfg = da7219_aad_of_btn_cfg(codec, of_val32); + else + aad_pdata->btn_cfg = DA7219_AAD_BTN_CFG_10MS; + + if (of_property_read_u32(aad_np, "dlg,mic-det-thr", &of_val32) >= 0) + aad_pdata->mic_det_thr = + da7219_aad_of_mic_det_thr(codec, of_val32); + else + aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS; + + if (of_property_read_u32(aad_np, "dlg,jack-ins-deb", &of_val32) >= 0) + aad_pdata->jack_ins_deb = + da7219_aad_of_jack_ins_deb(codec, of_val32); + else + aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS; + + if (!of_property_read_string(aad_np, "dlg,jack-det-rate", &of_str)) + aad_pdata->jack_det_rate = + da7219_aad_of_jack_det_rate(codec, of_str); + else + aad_pdata->jack_det_rate = DA7219_AAD_JACK_DET_RATE_256_512MS; + + if (of_property_read_u32(aad_np, "dlg,jack-rem-deb", &of_val32) >= 0) + aad_pdata->jack_rem_deb = + da7219_aad_of_jack_rem_deb(codec, of_val32); + else + aad_pdata->jack_rem_deb = DA7219_AAD_JACK_REM_DEB_1MS; + + if (of_property_read_u32(aad_np, "dlg,a-d-btn-thr", &of_val32) >= 0) + aad_pdata->a_d_btn_thr = (u8) of_val32; + else + aad_pdata->a_d_btn_thr = 0xA; + + if (of_property_read_u32(aad_np, "dlg,d-b-btn-thr", &of_val32) >= 0) + aad_pdata->d_b_btn_thr = (u8) of_val32; + else + aad_pdata->d_b_btn_thr = 0x16; + + if (of_property_read_u32(aad_np, "dlg,b-c-btn-thr", &of_val32) >= 0) + aad_pdata->b_c_btn_thr = (u8) of_val32; + else + aad_pdata->b_c_btn_thr = 0x21; + + if (of_property_read_u32(aad_np, "dlg,c-mic-btn-thr", &of_val32) >= 0) + aad_pdata->c_mic_btn_thr = (u8) of_val32; + else + aad_pdata->c_mic_btn_thr = 0x3E; + + if (of_property_read_u32(aad_np, "dlg,btn-avg", &of_val32) >= 0) + aad_pdata->btn_avg = da7219_aad_of_btn_avg(codec, of_val32); + else + aad_pdata->btn_avg = DA7219_AAD_BTN_AVG_2; + + if (of_property_read_u32(aad_np, "dlg,adc-1bit-rpt", &of_val32) >= 0) + aad_pdata->adc_1bit_rpt = + da7219_aad_of_adc_1bit_rpt(codec, of_val32); + else + aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1; + +out: + of_node_put(aad_np); + + return aad_pdata; +} + +static void da7219_aad_handle_pdata(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + struct da7219_pdata *pdata = da7219->pdata; + + if ((pdata) && (pdata->aad_pdata)) { + struct da7219_aad_pdata *aad_pdata = pdata->aad_pdata; + u8 cfg, mask; + + da7219_aad->irq = aad_pdata->irq; + + switch (aad_pdata->micbias_pulse_lvl) { + case DA7219_AAD_MICBIAS_PULSE_LVL_2_8V: + case DA7219_AAD_MICBIAS_PULSE_LVL_2_9V: + da7219_aad->micbias_pulse_lvl = + (aad_pdata->micbias_pulse_lvl << + DA7219_MICBIAS1_LEVEL_SHIFT); + break; + default: + break; + } + + da7219_aad->micbias_pulse_time = aad_pdata->micbias_pulse_time; + + switch (aad_pdata->btn_cfg) { + case DA7219_AAD_BTN_CFG_2MS: + case DA7219_AAD_BTN_CFG_5MS: + case DA7219_AAD_BTN_CFG_10MS: + case DA7219_AAD_BTN_CFG_50MS: + case DA7219_AAD_BTN_CFG_100MS: + case DA7219_AAD_BTN_CFG_200MS: + case DA7219_AAD_BTN_CFG_500MS: + da7219_aad->btn_cfg = (aad_pdata->btn_cfg << + DA7219_BUTTON_CONFIG_SHIFT); + } + + cfg = 0; + mask = 0; + switch (aad_pdata->mic_det_thr) { + case DA7219_AAD_MIC_DET_THR_200_OHMS: + case DA7219_AAD_MIC_DET_THR_500_OHMS: + case DA7219_AAD_MIC_DET_THR_750_OHMS: + case DA7219_AAD_MIC_DET_THR_1000_OHMS: + cfg |= (aad_pdata->mic_det_thr << + DA7219_MIC_DET_THRESH_SHIFT); + mask |= DA7219_MIC_DET_THRESH_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, mask, cfg); + + cfg = 0; + mask = 0; + switch (aad_pdata->jack_ins_deb) { + case DA7219_AAD_JACK_INS_DEB_5MS: + case DA7219_AAD_JACK_INS_DEB_10MS: + case DA7219_AAD_JACK_INS_DEB_20MS: + case DA7219_AAD_JACK_INS_DEB_50MS: + case DA7219_AAD_JACK_INS_DEB_100MS: + case DA7219_AAD_JACK_INS_DEB_200MS: + case DA7219_AAD_JACK_INS_DEB_500MS: + case DA7219_AAD_JACK_INS_DEB_1S: + cfg |= (aad_pdata->jack_ins_deb << + DA7219_JACKDET_DEBOUNCE_SHIFT); + mask |= DA7219_JACKDET_DEBOUNCE_MASK; + } + switch (aad_pdata->jack_det_rate) { + case DA7219_AAD_JACK_DET_RATE_32_64MS: + case DA7219_AAD_JACK_DET_RATE_64_128MS: + case DA7219_AAD_JACK_DET_RATE_128_256MS: + case DA7219_AAD_JACK_DET_RATE_256_512MS: + cfg |= (aad_pdata->jack_det_rate << + DA7219_JACK_DETECT_RATE_SHIFT); + mask |= DA7219_JACK_DETECT_RATE_MASK; + } + switch (aad_pdata->jack_rem_deb) { + case DA7219_AAD_JACK_REM_DEB_1MS: + case DA7219_AAD_JACK_REM_DEB_5MS: + case DA7219_AAD_JACK_REM_DEB_10MS: + case DA7219_AAD_JACK_REM_DEB_20MS: + cfg |= (aad_pdata->jack_rem_deb << + DA7219_JACKDET_REM_DEB_SHIFT); + mask |= DA7219_JACKDET_REM_DEB_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_2, mask, cfg); + + snd_soc_write(codec, DA7219_ACCDET_CONFIG_3, + aad_pdata->a_d_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_4, + aad_pdata->d_b_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_5, + aad_pdata->b_c_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_6, + aad_pdata->c_mic_btn_thr); + + cfg = 0; + mask = 0; + switch (aad_pdata->btn_avg) { + case DA7219_AAD_BTN_AVG_1: + case DA7219_AAD_BTN_AVG_2: + case DA7219_AAD_BTN_AVG_4: + case DA7219_AAD_BTN_AVG_8: + cfg |= (aad_pdata->btn_avg << + DA7219_BUTTON_AVERAGE_SHIFT); + mask |= DA7219_BUTTON_AVERAGE_MASK; + } + switch (aad_pdata->adc_1bit_rpt) { + case DA7219_AAD_ADC_1BIT_RPT_1: + case DA7219_AAD_ADC_1BIT_RPT_2: + case DA7219_AAD_ADC_1BIT_RPT_4: + case DA7219_AAD_ADC_1BIT_RPT_8: + cfg |= (aad_pdata->adc_1bit_rpt << + DA7219_ADC_1_BIT_REPEAT_SHIFT); + mask |= DA7219_ADC_1_BIT_REPEAT_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_7, mask, cfg); + } +} + + +/* + * Init/Exit + */ + +int da7219_aad_init(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad; + u8 mask[DA7219_AAD_IRQ_REG_MAX]; + int ret; + + da7219_aad = devm_kzalloc(codec->dev, sizeof(*da7219_aad), GFP_KERNEL); + if (!da7219_aad) + return -ENOMEM; + + da7219->aad = da7219_aad; + da7219_aad->codec = codec; + + /* Handle any DT/platform data */ + if ((codec->dev->of_node) && (da7219->pdata)) + da7219->pdata->aad_pdata = da7219_aad_of_to_pdata(codec); + + da7219_aad_handle_pdata(codec); + + /* Disable button detection */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, 0); + + INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work); + INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); + + ret = request_threaded_irq(da7219_aad->irq, NULL, + da7219_aad_irq_thread, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "da7219-aad", da7219_aad); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + return ret; + } + + /* Unmask AAD IRQs */ + memset(mask, 0, DA7219_AAD_IRQ_REG_MAX); + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, + &mask, DA7219_AAD_IRQ_REG_MAX); + + return 0; +} +EXPORT_SYMBOL_GPL(da7219_aad_init); + +void da7219_aad_exit(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + u8 mask[DA7219_AAD_IRQ_REG_MAX]; + + /* Mask off AAD IRQs */ + memset(mask, DA7219_BYTE_MASK, DA7219_AAD_IRQ_REG_MAX); + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, + mask, DA7219_AAD_IRQ_REG_MAX); + + free_irq(da7219_aad->irq, da7219_aad); + + cancel_work_sync(&da7219_aad->btn_det_work); + cancel_work_sync(&da7219_aad->hptest_work); +} +EXPORT_SYMBOL_GPL(da7219_aad_exit); + +MODULE_DESCRIPTION("ASoC DA7219 AAD Driver"); +MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h new file mode 100644 index 0000000..4fccf67 --- /dev/null +++ b/sound/soc/codecs/da7219-aad.h @@ -0,0 +1,212 @@ +/* + * da7219-aad.h - DA7322 ASoC AAD Driver + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_AAD_H +#define __DA7219_AAD_H + +#include <linux/timer.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/da7219-aad.h> + +/* + * Registers + */ + +#define DA7219_ACCDET_STATUS_A 0xC0 +#define DA7219_ACCDET_STATUS_B 0xC1 +#define DA7219_ACCDET_IRQ_EVENT_A 0xC2 +#define DA7219_ACCDET_IRQ_EVENT_B 0xC3 +#define DA7219_ACCDET_IRQ_MASK_A 0xC4 +#define DA7219_ACCDET_IRQ_MASK_B 0xC5 +#define DA7219_ACCDET_CONFIG_1 0xC6 +#define DA7219_ACCDET_CONFIG_2 0xC7 +#define DA7219_ACCDET_CONFIG_3 0xC8 +#define DA7219_ACCDET_CONFIG_4 0xC9 +#define DA7219_ACCDET_CONFIG_5 0xCA +#define DA7219_ACCDET_CONFIG_6 0xCB +#define DA7219_ACCDET_CONFIG_7 0xCC +#define DA7219_ACCDET_CONFIG_8 0xCD + + +/* + * Bit Fields + */ + +/* DA7219_ACCDET_STATUS_A = 0xC0 */ +#define DA7219_JACK_INSERTION_STS_SHIFT 0 +#define DA7219_JACK_INSERTION_STS_MASK (0x1 << 0) +#define DA7219_JACK_TYPE_STS_SHIFT 1 +#define DA7219_JACK_TYPE_STS_MASK (0x1 << 1) +#define DA7219_JACK_PIN_ORDER_STS_SHIFT 2 +#define DA7219_JACK_PIN_ORDER_STS_MASK (0x1 << 2) +#define DA7219_MICBIAS_UP_STS_SHIFT 3 +#define DA7219_MICBIAS_UP_STS_MASK (0x1 << 3) + +/* DA7219_ACCDET_STATUS_B = 0xC1 */ +#define DA7219_BUTTON_TYPE_STS_SHIFT 0 +#define DA7219_BUTTON_TYPE_STS_MASK (0xFF << 0) + +/* DA7219_ACCDET_IRQ_EVENT_A = 0xC2 */ +#define DA7219_E_JACK_INSERTED_SHIFT 0 +#define DA7219_E_JACK_INSERTED_MASK (0x1 << 0) +#define DA7219_E_JACK_REMOVED_SHIFT 1 +#define DA7219_E_JACK_REMOVED_MASK (0x1 << 1) +#define DA7219_E_JACK_DETECT_COMPLETE_SHIFT 2 +#define DA7219_E_JACK_DETECT_COMPLETE_MASK (0x1 << 2) + +/* DA7219_ACCDET_IRQ_EVENT_B = 0xC3 */ +#define DA7219_E_BUTTON_A_PRESSED_SHIFT 0 +#define DA7219_E_BUTTON_A_PRESSED_MASK (0x1 << 0) +#define DA7219_E_BUTTON_B_PRESSED_SHIFT 1 +#define DA7219_E_BUTTON_B_PRESSED_MASK (0x1 << 1) +#define DA7219_E_BUTTON_C_PRESSED_SHIFT 2 +#define DA7219_E_BUTTON_C_PRESSED_MASK (0x1 << 2) +#define DA7219_E_BUTTON_D_PRESSED_SHIFT 3 +#define DA7219_E_BUTTON_D_PRESSED_MASK (0x1 << 3) +#define DA7219_E_BUTTON_D_RELEASED_SHIFT 4 +#define DA7219_E_BUTTON_D_RELEASED_MASK (0x1 << 4) +#define DA7219_E_BUTTON_C_RELEASED_SHIFT 5 +#define DA7219_E_BUTTON_C_RELEASED_MASK (0x1 << 5) +#define DA7219_E_BUTTON_B_RELEASED_SHIFT 6 +#define DA7219_E_BUTTON_B_RELEASED_MASK (0x1 << 6) +#define DA7219_E_BUTTON_A_RELEASED_SHIFT 7 +#define DA7219_E_BUTTON_A_RELEASED_MASK (0x1 << 7) + +/* DA7219_ACCDET_IRQ_MASK_A = 0xC4 */ +#define DA7219_M_JACK_INSERTED_SHIFT 0 +#define DA7219_M_JACK_INSERTED_MASK (0x1 << 0) +#define DA7219_M_JACK_REMOVED_SHIFT 1 +#define DA7219_M_JACK_REMOVED_MASK (0x1 << 1) +#define DA7219_M_JACK_DETECT_COMPLETE_SHIFT 2 +#define DA7219_M_JACK_DETECT_COMPLETE_MASK (0x1 << 2) + +/* DA7219_ACCDET_IRQ_MASK_B = 0xC5 */ +#define DA7219_M_BUTTON_A_PRESSED_SHIFT 0 +#define DA7219_M_BUTTON_A_PRESSED_MASK (0x1 << 0) +#define DA7219_M_BUTTON_B_PRESSED_SHIFT 1 +#define DA7219_M_BUTTON_B_PRESSED_MASK (0x1 << 1) +#define DA7219_M_BUTTON_C_PRESSED_SHIFT 2 +#define DA7219_M_BUTTON_C_PRESSED_MASK (0x1 << 2) +#define DA7219_M_BUTTON_D_PRESSED_SHIFT 3 +#define DA7219_M_BUTTON_D_PRESSED_MASK (0x1 << 3) +#define DA7219_M_BUTTON_D_RELEASED_SHIFT 4 +#define DA7219_M_BUTTON_D_RELEASED_MASK (0x1 << 4) +#define DA7219_M_BUTTON_C_RELEASED_SHIFT 5 +#define DA7219_M_BUTTON_C_RELEASED_MASK (0x1 << 5) +#define DA7219_M_BUTTON_B_RELEASED_SHIFT 6 +#define DA7219_M_BUTTON_B_RELEASED_MASK (0x1 << 6) +#define DA7219_M_BUTTON_A_RELEASED_SHIFT 7 +#define DA7219_M_BUTTON_A_RELEASED_MASK (0x1 << 7) + +/* DA7219_ACCDET_CONFIG_1 = 0xC6 */ +#define DA7219_ACCDET_EN_SHIFT 0 +#define DA7219_ACCDET_EN_MASK (0x1 << 0) +#define DA7219_BUTTON_CONFIG_SHIFT 1 +#define DA7219_BUTTON_CONFIG_MASK (0x7 << 1) +#define DA7219_MIC_DET_THRESH_SHIFT 4 +#define DA7219_MIC_DET_THRESH_MASK (0x3 << 4) +#define DA7219_JACK_TYPE_DET_EN_SHIFT 6 +#define DA7219_JACK_TYPE_DET_EN_MASK (0x1 << 6) +#define DA7219_PIN_ORDER_DET_EN_SHIFT 7 +#define DA7219_PIN_ORDER_DET_EN_MASK (0x1 << 7) + +/* DA7219_ACCDET_CONFIG_2 = 0xC7 */ +#define DA7219_ACCDET_PAUSE_SHIFT 0 +#define DA7219_ACCDET_PAUSE_MASK (0x1 << 0) +#define DA7219_JACKDET_DEBOUNCE_SHIFT 1 +#define DA7219_JACKDET_DEBOUNCE_MASK (0x7 << 1) +#define DA7219_JACK_DETECT_RATE_SHIFT 4 +#define DA7219_JACK_DETECT_RATE_MASK (0x3 << 4) +#define DA7219_JACKDET_REM_DEB_SHIFT 6 +#define DA7219_JACKDET_REM_DEB_MASK (0x3 << 6) + +/* DA7219_ACCDET_CONFIG_3 = 0xC8 */ +#define DA7219_A_D_BUTTON_THRESH_SHIFT 0 +#define DA7219_A_D_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_4 = 0xC9 */ +#define DA7219_D_B_BUTTON_THRESH_SHIFT 0 +#define DA7219_D_B_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_5 = 0xCA */ +#define DA7219_B_C_BUTTON_THRESH_SHIFT 0 +#define DA7219_B_C_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_6 = 0xCB */ +#define DA7219_C_MIC_BUTTON_THRESH_SHIFT 0 +#define DA7219_C_MIC_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_7 = 0xCC */ +#define DA7219_BUTTON_AVERAGE_SHIFT 0 +#define DA7219_BUTTON_AVERAGE_MASK (0x3 << 0) +#define DA7219_ADC_1_BIT_REPEAT_SHIFT 2 +#define DA7219_ADC_1_BIT_REPEAT_MASK (0x3 << 2) +#define DA7219_PIN_ORDER_FORCE_SHIFT 4 +#define DA7219_PIN_ORDER_FORCE_MASK (0x1 << 4) +#define DA7219_JACK_TYPE_FORCE_SHIFT 5 +#define DA7219_JACK_TYPE_FORCE_MASK (0x1 << 5) + +/* DA7219_ACCDET_CONFIG_8 = 0xCD */ +#define DA7219_HPTEST_EN_SHIFT 0 +#define DA7219_HPTEST_EN_MASK (0x1 << 0) +#define DA7219_HPTEST_RES_SEL_SHIFT 1 +#define DA7219_HPTEST_RES_SEL_MASK (0x3 << 1) +#define DA7219_HPTEST_RES_SEL_1KOHMS (0x0 << 1) +#define DA7219_HPTEST_COMP_SHIFT 4 +#define DA7219_HPTEST_COMP_MASK (0x1 << 4) + + +#define DA7219_AAD_MAX_BUTTONS 4 +#define DA7219_AAD_REPORT_ALL_MASK (SND_JACK_MECHANICAL | \ + SND_JACK_HEADSET | SND_JACK_LINEOUT | \ + SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +#define DA7219_AAD_MICBIAS_CHK_DELAY 10 +#define DA7219_AAD_MICBIAS_CHK_RETRIES 5 + +#define DA7219_AAD_HPTEST_RAMP_FREQ 0x28 +#define DA7219_AAD_HPTEST_PERIOD 65 + +enum da7219_aad_event_regs { + DA7219_AAD_IRQ_REG_A = 0, + DA7219_AAD_IRQ_REG_B, + DA7219_AAD_IRQ_REG_MAX, +}; + +/* Private data */ +struct da7219_aad_priv { + struct snd_soc_codec *codec; + int irq; + + u8 micbias_pulse_lvl; + u32 micbias_pulse_time; + + u8 btn_cfg; + + struct work_struct btn_det_work; + struct work_struct hptest_work; + + struct snd_soc_jack *jack; + bool jack_inserted; +}; + +/* AAD control */ +void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +/* Init/Exit */ +int da7219_aad_init(struct snd_soc_codec *codec); +void da7219_aad_exit(struct snd_soc_codec *codec); + +#endif /* __DA7219_AAD_H */ diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c new file mode 100644 index 0000000..76f8fc2 --- /dev/null +++ b/sound/soc/codecs/da7219.c @@ -0,0 +1,1940 @@ +/* + * da7219.c - DA7219 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/pm.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> + +#include <sound/da7219.h> +#include "da7219.h" +#include "da7219-aad.h" + + +/* + * TLVs and Enums + */ + +/* Input TLVs */ +static const DECLARE_TLV_DB_SCALE(da7219_mic_gain_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7219_adc_dig_gain_tlv, -8325, 75, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_ana_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_sidetone_gain_tlv, -4200, 300, 0); +static const DECLARE_TLV_DB_SCALE(da7219_tonegen_gain_tlv, -4500, 300, 0); + +/* Output TLVs */ +static const DECLARE_TLV_DB_SCALE(da7219_dac_eq_band_tlv, -1050, 150, 0); + +static const DECLARE_TLV_DB_RANGE(da7219_dac_dig_gain_tlv, + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -77.25dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7725, 75, 0) +); + +static const DECLARE_TLV_DB_SCALE(da7219_dac_ng_threshold_tlv, -10200, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_hp_gain_tlv, -5700, 100, 0); + +/* Input Enums */ +static const char * const da7219_alc_attack_rate_txt[] = { + "7.33/fs", "14.66/fs", "29.32/fs", "58.64/fs", "117.3/fs", "234.6/fs", + "469.1/fs", "938.2/fs", "1876/fs", "3753/fs", "7506/fs", "15012/fs", + "30024/fs" +}; + +static const struct soc_enum da7219_alc_attack_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_ATTACK_SHIFT, + DA7219_ALC_ATTACK_MAX, da7219_alc_attack_rate_txt); + +static const char * const da7219_alc_release_rate_txt[] = { + "28.66/fs", "57.33/fs", "114.6/fs", "229.3/fs", "458.6/fs", "917.1/fs", + "1834/fs", "3668/fs", "7337/fs", "14674/fs", "29348/fs" +}; + +static const struct soc_enum da7219_alc_release_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_RELEASE_SHIFT, + DA7219_ALC_RELEASE_MAX, da7219_alc_release_rate_txt); + +static const char * const da7219_alc_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da7219_alc_hold_time = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_HOLD_SHIFT, + DA7219_ALC_HOLD_MAX, da7219_alc_hold_time_txt); + +static const char * const da7219_alc_env_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da7219_alc_env_attack_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_ATTACK_SHIFT, + DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt); + +static const struct soc_enum da7219_alc_env_release_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_RELEASE_SHIFT, + DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt); + +static const char * const da7219_alc_anticlip_step_txt[] = { + "0.034dB/fs", "0.068dB/fs", "0.136dB/fs", "0.272dB/fs" +}; + +static const struct soc_enum da7219_alc_anticlip_step = + SOC_ENUM_SINGLE(DA7219_ALC_ANTICLIP_CTRL, + DA7219_ALC_ANTICLIP_STEP_SHIFT, + DA7219_ALC_ANTICLIP_STEP_MAX, + da7219_alc_anticlip_step_txt); + +/* Input/Output Enums */ +static const char * const da7219_gain_ramp_rate_txt[] = { + "Nominal Rate * 8", "Nominal Rate", "Nominal Rate / 8", + "Nominal Rate / 16" +}; + +static const struct soc_enum da7219_gain_ramp_rate = + SOC_ENUM_SINGLE(DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_SHIFT, + DA7219_GAIN_RAMP_RATE_MAX, da7219_gain_ramp_rate_txt); + +static const char * const da7219_hpf_mode_txt[] = { + "Disabled", "Audio", "Voice" +}; + +static const unsigned int da7219_hpf_mode_val[] = { + DA7219_HPF_DISABLED, DA7219_HPF_AUDIO_EN, DA7219_HPF_VOICE_EN, +}; + +static const struct soc_enum da7219_adc_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_ADC_FILTERS1, DA7219_HPF_MODE_SHIFT, + DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX, + da7219_hpf_mode_txt, da7219_hpf_mode_val); + +static const struct soc_enum da7219_dac_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_DAC_FILTERS1, DA7219_HPF_MODE_SHIFT, + DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX, + da7219_hpf_mode_txt, da7219_hpf_mode_val); + +static const char * const da7219_audio_hpf_corner_txt[] = { + "2Hz", "4Hz", "8Hz", "16Hz" +}; + +static const struct soc_enum da7219_adc_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1, + DA7219_ADC_AUDIO_HPF_CORNER_SHIFT, + DA7219_AUDIO_HPF_CORNER_MAX, + da7219_audio_hpf_corner_txt); + +static const struct soc_enum da7219_dac_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1, + DA7219_DAC_AUDIO_HPF_CORNER_SHIFT, + DA7219_AUDIO_HPF_CORNER_MAX, + da7219_audio_hpf_corner_txt); + +static const char * const da7219_voice_hpf_corner_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7219_adc_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1, + DA7219_ADC_VOICE_HPF_CORNER_SHIFT, + DA7219_VOICE_HPF_CORNER_MAX, + da7219_voice_hpf_corner_txt); + +static const struct soc_enum da7219_dac_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1, + DA7219_DAC_VOICE_HPF_CORNER_SHIFT, + DA7219_VOICE_HPF_CORNER_MAX, + da7219_voice_hpf_corner_txt); + +static const char * const da7219_tonegen_dtmf_key_txt[] = { + "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D", + "*", "#" +}; + +static const struct soc_enum da7219_tonegen_dtmf_key = + SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG1, DA7219_DTMF_REG_SHIFT, + DA7219_DTMF_REG_MAX, da7219_tonegen_dtmf_key_txt); + +static const char * const da7219_tonegen_swg_sel_txt[] = { + "Sum", "SWG1", "SWG2", "SWG1_1-Cos" +}; + +static const struct soc_enum da7219_tonegen_swg_sel = + SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG2, DA7219_SWG_SEL_SHIFT, + DA7219_SWG_SEL_MAX, da7219_tonegen_swg_sel_txt); + +/* Output Enums */ +static const char * const da7219_dac_softmute_rate_txt[] = { + "1 Sample", "2 Samples", "4 Samples", "8 Samples", "16 Samples", + "32 Samples", "64 Samples" +}; + +static const struct soc_enum da7219_dac_softmute_rate = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS5, DA7219_DAC_SOFTMUTE_RATE_SHIFT, + DA7219_DAC_SOFTMUTE_RATE_MAX, + da7219_dac_softmute_rate_txt); + +static const char * const da7219_dac_ng_setup_time_txt[] = { + "256 Samples", "512 Samples", "1024 Samples", "2048 Samples" +}; + +static const struct soc_enum da7219_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_SETUP_TIME_SHIFT, + DA7219_DAC_NG_SETUP_TIME_MAX, + da7219_dac_ng_setup_time_txt); + +static const char * const da7219_dac_ng_rampup_txt[] = { + "0.22ms/dB", "0.0138ms/dB" +}; + +static const struct soc_enum da7219_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_RAMPUP_RATE_SHIFT, + DA7219_DAC_NG_RAMP_RATE_MAX, + da7219_dac_ng_rampup_txt); + +static const char * const da7219_dac_ng_rampdown_txt[] = { + "0.88ms/dB", "14.08ms/dB" +}; + +static const struct soc_enum da7219_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_RAMPDN_RATE_SHIFT, + DA7219_DAC_NG_RAMP_RATE_MAX, + da7219_dac_ng_rampdown_txt); + + +static const char * const da7219_cp_track_mode_txt[] = { + "Largest Volume", "DAC Volume", "Signal Magnitude" +}; + +static const unsigned int da7219_cp_track_mode_val[] = { + DA7219_CP_MCHANGE_LARGEST_VOL, DA7219_CP_MCHANGE_DAC_VOL, + DA7219_CP_MCHANGE_SIG_MAG +}; + +static const struct soc_enum da7219_cp_track_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_CP_CTRL, DA7219_CP_MCHANGE_SHIFT, + DA7219_CP_MCHANGE_REL_MASK, DA7219_CP_MCHANGE_MAX, + da7219_cp_track_mode_txt, + da7219_cp_track_mode_val); + + +/* + * Control Functions + */ + +/* Locked Kcontrol calls */ +static int da7219_volsw_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_get_volsw(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_volsw_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_put_volsw(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_enum_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_get_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_put_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +/* ALC */ +static void da7219_alc_calib(struct snd_soc_codec *codec) +{ + u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl; + + /* Save current state of mic control register */ + mic_ctrl = snd_soc_read(codec, DA7219_MIC_1_CTRL); + + /* Save current state of input mixer control register */ + mixin_ctrl = snd_soc_read(codec, DA7219_MIXIN_L_CTRL); + + /* Save current state of input ADC control register */ + adc_ctrl = snd_soc_read(codec, DA7219_ADC_L_CTRL); + + /* Enable then Mute MIC PGAs */ + snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK, + DA7219_MIC_1_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_MUTE_EN_MASK, + DA7219_MIC_1_AMP_MUTE_EN_MASK); + + /* Enable input mixers unmuted */ + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_MASK | + DA7219_MIXIN_L_AMP_MUTE_EN_MASK, + DA7219_MIXIN_L_AMP_EN_MASK); + + /* Enable input filters unmuted */ + snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, + DA7219_ADC_L_MUTE_EN_MASK | DA7219_ADC_L_EN_MASK, + DA7219_ADC_L_EN_MASK); + + /* Perform auto calibration */ + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_AUTO_CALIB_EN_MASK, + DA7219_ALC_AUTO_CALIB_EN_MASK); + do { + calib_ctrl = snd_soc_read(codec, DA7219_ALC_CTRL1); + } while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK); + + /* If auto calibration fails, disable DC offset, hybrid ALC */ + if (calib_ctrl & DA7219_ALC_CALIB_OVERFLOW_MASK) { + dev_warn(codec->dev, + "ALC auto calibration failed with overflow\n"); + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK, 0); + } else { + /* Enable DC offset cancellation, hybrid mode */ + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK); + } + + /* Restore input filter control register to original state */ + snd_soc_write(codec, DA7219_ADC_L_CTRL, adc_ctrl); + + /* Restore input mixer control registers to original state */ + snd_soc_write(codec, DA7219_MIXIN_L_CTRL, mixin_ctrl); + + /* Restore MIC control registers to original states */ + snd_soc_write(codec, DA7219_MIC_1_CTRL, mic_ctrl); +} + +static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + + /* + * If ALC in operation and value of control has been updated, + * make sure calibrated offsets are updated. + */ + if ((ret == 1) && (da7219->alc_en)) + da7219_alc_calib(codec); + + return ret; +} + +static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + + /* Force ALC offset calibration if enabling ALC */ + if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) { + da7219_alc_calib(codec); + da7219->alc_en = false; + } else { + da7219->alc_en = false; + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* ToneGen */ +static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + int ret; + + mutex_lock(&da7219->lock); + ret = regmap_raw_read(da7219->regmap, reg, &val, sizeof(val)); + mutex_unlock(&da7219->lock); + + if (ret) + return ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to host endianness here. + */ + ucontrol->value.integer.value[0] = le16_to_cpu(val); + + return 0; +} + +static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + int ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to little endian here to align with + * HW registers. + */ + val = cpu_to_le16(ucontrol->value.integer.value[0]); + + mutex_lock(&da7219->lock); + ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val)); + mutex_unlock(&da7219->lock); + + return ret; +} + + +/* + * KControls + */ + +static const struct snd_kcontrol_new da7219_snd_controls[] = { + /* Mics */ + SOC_SINGLE_TLV("Mic Volume", DA7219_MIC_1_GAIN, + DA7219_MIC_1_AMP_GAIN_SHIFT, DA7219_MIC_1_AMP_GAIN_MAX, + DA7219_NO_INVERT, da7219_mic_gain_tlv), + SOC_SINGLE("Mic Switch", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + + /* Mixer Input */ + SOC_SINGLE_EXT_TLV("Mixin Volume", DA7219_MIXIN_L_GAIN, + DA7219_MIXIN_L_AMP_GAIN_SHIFT, + DA7219_MIXIN_L_AMP_GAIN_MAX, DA7219_NO_INVERT, + snd_soc_get_volsw, da7219_mixin_gain_put, + da7219_mixin_gain_tlv), + SOC_SINGLE("Mixin Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + SOC_SINGLE("Mixin Gain Ramp Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + SOC_SINGLE("Mixin ZC Gain Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_ZC_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + + /* ADC */ + SOC_SINGLE_TLV("Capture Digital Volume", DA7219_ADC_L_GAIN, + DA7219_ADC_L_DIGITAL_GAIN_SHIFT, + DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, + da7219_adc_dig_gain_tlv), + SOC_SINGLE("Capture Digital Switch", DA7219_ADC_L_CTRL, + DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + SOC_SINGLE("Capture Digital Gain Ramp Switch", DA7219_ADC_L_CTRL, + DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + + /* ALC */ + SOC_ENUM("ALC Attack Rate", da7219_alc_attack_rate), + SOC_ENUM("ALC Release Rate", da7219_alc_release_rate), + SOC_ENUM("ALC Hold Time", da7219_alc_hold_time), + SOC_ENUM("ALC Envelope Attack Rate", da7219_alc_env_attack_rate), + SOC_ENUM("ALC Envelope Release Rate", da7219_alc_env_release_rate), + SOC_SINGLE_TLV("ALC Noise Threshold", DA7219_ALC_NOISE, + DA7219_ALC_NOISE_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Min Threshold", DA7219_ALC_TARGET_MIN, + DA7219_ALC_THRESHOLD_MIN_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold", DA7219_ALC_TARGET_MAX, + DA7219_ALC_THRESHOLD_MAX_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS, + DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Volume", DA7219_ALC_GAIN_LIMITS, + DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Min Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS, + DA7219_ALC_ANA_GAIN_MIN_SHIFT, + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Max Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS, + DA7219_ALC_ANA_GAIN_MAX_SHIFT, + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), + SOC_ENUM("ALC Anticlip Step", da7219_alc_anticlip_step), + SOC_SINGLE("ALC Anticlip Switch", DA7219_ALC_ANTICLIP_CTRL, + DA7219_ALC_ANTIPCLIP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + SOC_SINGLE_EXT("ALC Switch", DA7219_ALC_CTRL1, DA7219_ALC_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT, + snd_soc_get_volsw, da7219_alc_sw_put), + + /* Input High-Pass Filters */ + SOC_ENUM("ADC HPF Mode", da7219_adc_hpf_mode), + SOC_ENUM("ADC HPF Corner Audio", da7219_adc_audio_hpf_corner), + SOC_ENUM("ADC HPF Corner Voice", da7219_adc_voice_hpf_corner), + + /* Sidetone Filter */ + SOC_SINGLE_TLV("Sidetone Volume", DA7219_SIDETONE_GAIN, + DA7219_SIDETONE_GAIN_SHIFT, DA7219_SIDETONE_GAIN_MAX, + DA7219_NO_INVERT, da7219_sidetone_gain_tlv), + SOC_SINGLE("Sidetone Switch", DA7219_SIDETONE_CTRL, + DA7219_SIDETONE_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + + /* Tone Generator */ + SOC_SINGLE_EXT_TLV("ToneGen Volume", DA7219_TONE_GEN_CFG2, + DA7219_TONE_GEN_GAIN_SHIFT, DA7219_TONE_GEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put, da7219_tonegen_gain_tlv), + SOC_ENUM_EXT("ToneGen DTMF Key", da7219_tonegen_dtmf_key, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_SINGLE_EXT("ToneGen DTMF Switch", DA7219_TONE_GEN_CFG1, + DA7219_DTMF_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_ENUM_EXT("ToneGen Sinewave Gen Type", da7219_tonegen_swg_sel, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_SINGLE_EXT("ToneGen Sinewave1 Freq", DA7219_TONE_GEN_FREQ1_L, + DA7219_FREQ1_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT, + da7219_tonegen_freq_get, da7219_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen Sinewave2 Freq", DA7219_TONE_GEN_FREQ2_L, + DA7219_FREQ2_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT, + da7219_tonegen_freq_get, da7219_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen On Time", DA7219_TONE_GEN_ON_PER, + DA7219_BEEP_ON_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_SINGLE("ToneGen Off Time", DA7219_TONE_GEN_OFF_PER, + DA7219_BEEP_OFF_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX, + DA7219_NO_INVERT), + + /* Gain ramping */ + SOC_ENUM("Gain Ramp Rate", da7219_gain_ramp_rate), + + /* DAC High-Pass Filter */ + SOC_ENUM_EXT("DAC HPF Mode", da7219_dac_hpf_mode, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_ENUM("DAC HPF Corner Audio", da7219_dac_audio_hpf_corner), + SOC_ENUM("DAC HPF Corner Voice", da7219_dac_voice_hpf_corner), + + /* DAC 5-Band Equaliser */ + SOC_SINGLE_TLV("DAC EQ Band1 Volume", DA7219_DAC_FILTERS2, + DA7219_DAC_EQ_BAND1_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band2 Volume", DA7219_DAC_FILTERS2, + DA7219_DAC_EQ_BAND2_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band3 Volume", DA7219_DAC_FILTERS3, + DA7219_DAC_EQ_BAND3_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band4 Volume", DA7219_DAC_FILTERS3, + DA7219_DAC_EQ_BAND4_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band5 Volume", DA7219_DAC_FILTERS4, + DA7219_DAC_EQ_BAND5_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_EXT("DAC EQ Switch", DA7219_DAC_FILTERS4, + DA7219_DAC_EQ_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + + /* DAC Softmute */ + SOC_ENUM("DAC Soft Mute Rate", da7219_dac_softmute_rate), + SOC_SINGLE_EXT("DAC Soft Mute Switch", DA7219_DAC_FILTERS5, + DA7219_DAC_SOFTMUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + + /* DAC Noise Gate */ + SOC_ENUM("DAC NG Setup Time", da7219_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da7219_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da7219_dac_ng_rampdown_rate), + SOC_SINGLE_TLV("DAC NG Off Threshold", DA7219_DAC_NG_OFF_THRESH, + DA7219_DAC_NG_OFF_THRESHOLD_SHIFT, + DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT, + da7219_dac_ng_threshold_tlv), + SOC_SINGLE_TLV("DAC NG On Threshold", DA7219_DAC_NG_ON_THRESH, + DA7219_DAC_NG_ON_THRESHOLD_SHIFT, + DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT, + da7219_dac_ng_threshold_tlv), + SOC_SINGLE("DAC NG Switch", DA7219_DAC_NG_CTRL, DA7219_DAC_NG_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + + /* DACs */ + SOC_DOUBLE_R_EXT_TLV("Playback Digital Volume", DA7219_DAC_L_GAIN, + DA7219_DAC_R_GAIN, DA7219_DAC_L_DIGITAL_GAIN_SHIFT, + DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put, + da7219_dac_dig_gain_tlv), + SOC_DOUBLE_R_EXT("Playback Digital Switch", DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL, DA7219_DAC_L_MUTE_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put), + SOC_DOUBLE_R("Playback Digital Gain Ramp Switch", DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + + /* CP */ + SOC_ENUM("Charge Pump Track Mode", da7219_cp_track_mode), + SOC_SINGLE("Charge Pump Threshold", DA7219_CP_VOL_THRESHOLD1, + DA7219_CP_THRESH_VDD2_SHIFT, DA7219_CP_THRESH_VDD2_MAX, + DA7219_NO_INVERT), + + /* Headphones */ + SOC_DOUBLE_R_EXT_TLV("Headphone Volume", DA7219_HP_L_GAIN, + DA7219_HP_R_GAIN, DA7219_HP_L_AMP_GAIN_SHIFT, + DA7219_HP_AMP_GAIN_MAX, DA7219_NO_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put, + da7219_hp_gain_tlv), + SOC_DOUBLE_R_EXT("Headphone Switch", DA7219_HP_L_CTRL, DA7219_HP_R_CTRL, + DA7219_HP_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_DOUBLE_R("Headphone Gain Ramp Switch", DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL, DA7219_HP_L_AMP_RAMP_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + SOC_DOUBLE_R("Headphone ZC Gain Switch", DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL, DA7219_HP_L_AMP_ZC_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + + +/* + * DAPM Mux Controls + */ + +static const char * const da7219_out_sel_txt[] = { + "ADC", "Tone Generator", "DAIL", "DAIR" +}; + +static const struct soc_enum da7219_out_dail_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI, + DA7219_DAI_L_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dail_sel_mux = + SOC_DAPM_ENUM("Out DAIL Mux", da7219_out_dail_sel); + +static const struct soc_enum da7219_out_dair_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI, + DA7219_DAI_R_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dair_sel_mux = + SOC_DAPM_ENUM("Out DAIR Mux", da7219_out_dair_sel); + +static const struct soc_enum da7219_out_dacl_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC, + DA7219_DAC_L_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dacl_sel_mux = + SOC_DAPM_ENUM("Out DACL Mux", da7219_out_dacl_sel); + +static const struct soc_enum da7219_out_dacr_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC, + DA7219_DAC_R_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dacr_sel_mux = + SOC_DAPM_ENUM("Out DACR Mux", da7219_out_dacr_sel); + + +/* + * DAPM Mixer Controls + */ + +static const struct snd_kcontrol_new da7219_mixin_controls[] = { + SOC_DAPM_SINGLE("Mic Switch", DA7219_MIXIN_L_SELECT, + DA7219_MIXIN_L_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +static const struct snd_kcontrol_new da7219_mixout_l_controls[] = { + SOC_DAPM_SINGLE("DACL Switch", DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_L_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +static const struct snd_kcontrol_new da7219_mixout_r_controls[] = { + SOC_DAPM_SINGLE("DACR Switch", DA7219_MIXOUT_R_SELECT, + DA7219_MIXOUT_R_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +#define DA7219_DMIX_ST_CTRLS(reg) \ + SOC_DAPM_SINGLE("Out FilterL Switch", reg, \ + DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \ + SOC_DAPM_SINGLE("Out FilterR Switch", reg, \ + DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \ + SOC_DAPM_SINGLE("Sidetone Switch", reg, \ + DA7219_DMIX_ST_SRC_SIDETONE_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT) \ + +static const struct snd_kcontrol_new da7219_st_out_filtl_mix_controls[] = { + DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1L), +}; + +static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = { + DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1R), +}; + + +/* + * DAPM Events + */ + +static int da7219_dai_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 da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 pll_ctrl, pll_status; + int i = 0; + bool srm_lock = false; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (da7219->master) + /* Enable DAI clks for master mode */ + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, + DA7219_DAI_CLK_EN_MASK); + + /* PC synchronised to DAI */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, 0); + + /* Slave mode, if SRM not enabled no need for status checks */ + pll_ctrl = snd_soc_read(codec, DA7219_PLL_CTRL); + if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) + return 0; + + /* Check SRM has locked */ + do { + pll_status = snd_soc_read(codec, DA7219_PLL_SRM_STS); + if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { + srm_lock = true; + } else { + ++i; + msleep(50); + } + } while ((i < DA7219_SRM_CHECK_RETRIES) & (!srm_lock)); + + if (!srm_lock) + dev_warn(codec->dev, "SRM failed to lock\n"); + + return 0; + case SND_SOC_DAPM_POST_PMD: + /* PC free-running */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, + DA7219_PC_FREERUN_MASK); + + /* Disable DAI clks if in master mode */ + if (da7219->master) + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, 0); + return 0; + default: + return -EINVAL; + } +} + + +/* + * DAPM Widgets + */ + +static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = { + /* Input Supplies */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7219_MICBIAS_CTRL, + DA7219_MICBIAS1_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Input Filters */ + SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT, + DA7219_NO_INVERT), + + /* Tone Generator */ + SND_SOC_DAPM_SIGGEN("TONE"), + SND_SOC_DAPM_PGA("Tone Generator", DA7219_TONE_GEN_CFG1, + DA7219_START_STOPN_SHIFT, DA7219_NO_INVERT, NULL, 0), + + /* Sidetone Input */ + SND_SOC_DAPM_ADC("Sidetone Filter", NULL, DA7219_SIDETONE_CTRL, + DA7219_SIDETONE_EN_SHIFT, DA7219_NO_INVERT), + + /* Input Mixer Supply */ + SND_SOC_DAPM_SUPPLY("Mixer In Supply", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_MIX_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Input Mixer */ + SND_SOC_DAPM_MIXER("Mixer In", SND_SOC_NOPM, 0, 0, + da7219_mixin_controls, + ARRAY_SIZE(da7219_mixin_controls)), + + /* Input Muxes */ + SND_SOC_DAPM_MUX("Out DAIL Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dail_sel_mux), + SND_SOC_DAPM_MUX("Out DAIR Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dair_sel_mux), + + /* DAI Supply */ + SND_SOC_DAPM_SUPPLY("DAI", DA7219_DAI_CTRL, DA7219_DAI_EN_SHIFT, + DA7219_NO_INVERT, da7219_dai_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* DAI */ + SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DAIIN", "Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Muxes */ + SND_SOC_DAPM_MUX("Out DACL Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dacl_sel_mux), + SND_SOC_DAPM_MUX("Out DACR Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dacr_sel_mux), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7219_mixout_l_controls, + ARRAY_SIZE(da7219_mixout_l_controls)), + SND_SOC_DAPM_MIXER("Mixer Out FilterR", SND_SOC_NOPM, 0, 0, + da7219_mixout_r_controls, + ARRAY_SIZE(da7219_mixout_r_controls)), + + /* Sidetone Mixers */ + SND_SOC_DAPM_MIXER("ST Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7219_st_out_filtl_mix_controls, + ARRAY_SIZE(da7219_st_out_filtl_mix_controls)), + SND_SOC_DAPM_MIXER("ST Mixer Out FilterR", SND_SOC_NOPM, 0, + 0, da7219_st_out_filtr_mix_controls, + ARRAY_SIZE(da7219_st_out_filtr_mix_controls)), + + /* DACs */ + SND_SOC_DAPM_DAC("DACL", NULL, DA7219_DAC_L_CTRL, DA7219_DAC_L_EN_SHIFT, + DA7219_NO_INVERT), + SND_SOC_DAPM_DAC("DACR", NULL, DA7219_DAC_R_CTRL, DA7219_DAC_R_EN_SHIFT, + DA7219_NO_INVERT), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("Mixout Left PGA", DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixout Right PGA", DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left PGA", DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right PGA", DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + + /* Output Supplies */ + SND_SOC_DAPM_SUPPLY("Charge Pump", DA7219_CP_CTRL, DA7219_CP_EN_SHIFT, + DA7219_NO_INVERT, NULL, 0), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + + +/* + * DAPM Mux Routes + */ + +#define DA7219_OUT_DAI_MUX_ROUTES(name) \ + {name, "ADC", "Mixer In"}, \ + {name, "Tone Generator", "Tone Generator"}, \ + {name, "DAIL", "DAIOUT"}, \ + {name, "DAIR", "DAIOUT"} + +#define DA7219_OUT_DAC_MUX_ROUTES(name) \ + {name, "ADC", "Mixer In"}, \ + {name, "Tone Generator", "Tone Generator"}, \ + {name, "DAIL", "DAIIN"}, \ + {name, "DAIR", "DAIIN"} + +/* + * DAPM Mixer Routes + */ + +#define DA7219_DMIX_ST_ROUTES(name) \ + {name, "Out FilterL Switch", "Mixer Out FilterL"}, \ + {name, "Out FilterR Switch", "Mixer Out FilterR"}, \ + {name, "Sidetone Switch", "Sidetone Filter"} + + +/* + * DAPM audio route definition + */ + +static const struct snd_soc_dapm_route da7219_audio_map[] = { + /* Input paths */ + {"MIC", NULL, "Mic Bias"}, + {"Mic PGA", NULL, "MIC"}, + {"Mixin PGA", NULL, "Mic PGA"}, + {"ADC", NULL, "Mixin PGA"}, + + {"Sidetone Filter", NULL, "ADC"}, + {"Mixer In", NULL, "Mixer In Supply"}, + {"Mixer In", "Mic Switch", "ADC"}, + + {"Tone Generator", NULL, "TONE"}, + + DA7219_OUT_DAI_MUX_ROUTES("Out DAIL Mux"), + DA7219_OUT_DAI_MUX_ROUTES("Out DAIR Mux"), + + {"DAIOUT", NULL, "Out DAIL Mux"}, + {"DAIOUT", NULL, "Out DAIR Mux"}, + {"DAIOUT", NULL, "DAI"}, + + /* Output paths */ + {"DAIIN", NULL, "DAI"}, + + DA7219_OUT_DAC_MUX_ROUTES("Out DACL Mux"), + DA7219_OUT_DAC_MUX_ROUTES("Out DACR Mux"), + + {"Mixer Out FilterL", "DACL Switch", "Out DACL Mux"}, + {"Mixer Out FilterR", "DACR Switch", "Out DACR Mux"}, + + DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterL"), + DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterR"), + + {"DACL", NULL, "ST Mixer Out FilterL"}, + {"DACR", NULL, "ST Mixer Out FilterR"}, + + {"Mixout Left PGA", NULL, "DACL"}, + {"Mixout Right PGA", NULL, "DACR"}, + + {"Headphone Left PGA", NULL, "Mixout Left PGA"}, + {"Headphone Right PGA", NULL, "Mixout Right PGA"}, + + {"HPL", NULL, "Headphone Left PGA"}, + {"HPR", NULL, "Headphone Right PGA"}, + + {"HPL", NULL, "Charge Pump"}, + {"HPR", NULL, "Charge Pump"}, +}; + + +/* + * DAI operations + */ + +static int da7219_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 da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) + return 0; + + if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + + switch (clk_id) { + case DA7219_CLKSRC_MCLK_SQR: + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_MCLK_SQR_EN_MASK, + DA7219_PLL_MCLK_SQR_EN_MASK); + break; + case DA7219_CLKSRC_MCLK: + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_MCLK_SQR_EN_MASK, 0); + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } + + da7219->clk_src = clk_id; + + if (da7219->mclk) { + freq = clk_round_rate(da7219->mclk, freq); + ret = clk_set_rate(da7219->mclk, freq); + if (ret) { + dev_err(codec_dai->dev, "Failed to set clock rate %d\n", + freq); + return ret; + } + } + + da7219->mclk_rate = freq; + + return 0; +} + +static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + u8 pll_ctrl, indiv_bits, indiv; + u8 pll_frac_top, pll_frac_bot, pll_integer; + u32 freq_ref; + u64 frac_div; + + /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ + if (da7219->mclk_rate == 32768) { + indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; + indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; + } else if (da7219->mclk_rate < 2000000) { + dev_err(codec->dev, "PLL input clock %d below valid range\n", + da7219->mclk_rate); + return -EINVAL; + } else if (da7219->mclk_rate <= 5000000) { + indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; + indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; + } else if (da7219->mclk_rate <= 10000000) { + indiv_bits = DA7219_PLL_INDIV_5_10_MHZ; + indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL; + } else if (da7219->mclk_rate <= 20000000) { + indiv_bits = DA7219_PLL_INDIV_10_20_MHZ; + indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL; + } else if (da7219->mclk_rate <= 40000000) { + indiv_bits = DA7219_PLL_INDIV_20_40_MHZ; + indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7219->mclk_rate <= 54000000) { + indiv_bits = DA7219_PLL_INDIV_40_54_MHZ; + indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL; + } else { + dev_err(codec->dev, "PLL input clock %d above valid range\n", + da7219->mclk_rate); + return -EINVAL; + } + freq_ref = (da7219->mclk_rate / indiv); + pll_ctrl = indiv_bits; + + /* Configure PLL */ + switch (source) { + case DA7219_SYSCLK_MCLK: + pll_ctrl |= DA7219_PLL_MODE_BYPASS; + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_INDIV_MASK | + DA7219_PLL_MODE_MASK, pll_ctrl); + return 0; + case DA7219_SYSCLK_PLL: + pll_ctrl |= DA7219_PLL_MODE_NORMAL; + break; + case DA7219_SYSCLK_PLL_SRM: + pll_ctrl |= DA7219_PLL_MODE_SRM; + break; + case DA7219_SYSCLK_PLL_32KHZ: + pll_ctrl |= DA7219_PLL_MODE_32KHZ; + break; + default: + dev_err(codec->dev, "Invalid PLL config\n"); + return -EINVAL; + } + + /* Calculate dividers for PLL */ + pll_integer = fout / freq_ref; + frac_div = (u64)(fout % freq_ref) * 8192ULL; + do_div(frac_div, freq_ref); + pll_frac_top = (frac_div >> DA7219_BYTE_SHIFT) & DA7219_BYTE_MASK; + pll_frac_bot = (frac_div) & DA7219_BYTE_MASK; + + /* Write PLL config & dividers */ + snd_soc_write(codec, DA7219_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA7219_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA7219_PLL_INTEGER, pll_integer); + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK, + pll_ctrl); + + return 0; +} + +static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 dai_clk_mode = 0, dai_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + da7219->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + da7219->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7219_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV | + DA7219_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_ctrl |= DA7219_DAI_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + dai_ctrl |= DA7219_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_ctrl |= DA7219_DAI_FORMAT_RIGHT_J; + break; + case SND_SOC_DAIFMT_DSP_B: + dai_ctrl |= DA7219_DAI_FORMAT_DSP; + break; + default: + return -EINVAL; + } + + /* By default 64 BCLKs per WCLK is supported */ + dai_clk_mode |= DA7219_DAI_BCLKS_PER_WCLK_64; + + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK | + DA7219_DAI_CLK_POL_MASK | DA7219_DAI_WCLK_POL_MASK, + dai_clk_mode); + snd_soc_update_bits(codec, DA7219_DAI_CTRL, DA7219_DAI_FORMAT_MASK, + dai_ctrl); + + return 0; +} + +static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 dai_bclks_per_wclk; + u16 offset; + u32 frame_size; + + /* No channels enabled so disable TDM, revert to 64-bit frames */ + if (!tx_mask) { + snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL, + DA7219_DAI_TDM_CH_EN_MASK | + DA7219_DAI_TDM_MODE_EN_MASK, 0); + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + DA7219_DAI_BCLKS_PER_WCLK_64); + return 0; + } + + /* Check we have valid slots */ + if (fls(tx_mask) > DA7219_DAI_TDM_MAX_SLOTS) { + dev_err(codec->dev, "Invalid number of slots, max = %d\n", + DA7219_DAI_TDM_MAX_SLOTS); + return -EINVAL; + } + + /* Check we have a valid offset given */ + if (rx_mask > DA7219_DAI_OFFSET_MAX) { + dev_err(codec->dev, "Invalid slot offset, max = %d\n", + DA7219_DAI_OFFSET_MAX); + return -EINVAL; + } + + /* Calculate & validate frame size based on slot info provided. */ + frame_size = slots * slot_width; + switch (frame_size) { + case 32: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; + break; + case 64: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; + break; + case 128: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; + break; + case 256: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; + break; + default: + dev_err(codec->dev, "Invalid frame size %d\n", frame_size); + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + dai_bclks_per_wclk); + + offset = cpu_to_le16(rx_mask); + regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, + &offset, sizeof(offset)); + + snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL, + DA7219_DAI_TDM_CH_EN_MASK | + DA7219_DAI_TDM_MODE_EN_MASK, + (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | + DA7219_DAI_TDM_MODE_EN_MASK); + + return 0; +} + +static int da7219_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; + u8 dai_ctrl = 0, fs; + unsigned int channels; + + switch (params_width(params)) { + case 16: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S16_LE; + break; + case 20: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S20_LE; + break; + case 24: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S24_LE; + break; + case 32: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S32_LE; + break; + default: + return -EINVAL; + } + + channels = params_channels(params); + if ((channels < 1) | (channels > DA7219_DAI_CH_NUM_MAX)) { + dev_err(codec->dev, + "Invalid number of channels, only 1 to %d supported\n", + DA7219_DAI_CH_NUM_MAX); + return -EINVAL; + } + dai_ctrl |= channels << DA7219_DAI_CH_NUM_SHIFT; + + switch (params_rate(params)) { + case 8000: + fs = DA7219_SR_8000; + break; + case 11025: + fs = DA7219_SR_11025; + break; + case 12000: + fs = DA7219_SR_12000; + break; + case 16000: + fs = DA7219_SR_16000; + break; + case 22050: + fs = DA7219_SR_22050; + break; + case 24000: + fs = DA7219_SR_24000; + break; + case 32000: + fs = DA7219_SR_32000; + break; + case 44100: + fs = DA7219_SR_44100; + break; + case 48000: + fs = DA7219_SR_48000; + break; + case 88200: + fs = DA7219_SR_88200; + break; + case 96000: + fs = DA7219_SR_96000; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7219_DAI_CTRL, + DA7219_DAI_WORD_LENGTH_MASK | + DA7219_DAI_CH_NUM_MASK, + dai_ctrl); + snd_soc_write(codec, DA7219_SR, fs); + + return 0; +} + +static const struct snd_soc_dai_ops da7219_dai_ops = { + .hw_params = da7219_hw_params, + .set_sysclk = da7219_set_dai_sysclk, + .set_pll = da7219_set_dai_pll, + .set_fmt = da7219_set_dai_fmt, + .set_tdm_slot = da7219_set_dai_tdm_slot, +}; + +#define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver da7219_dai = { + .name = "da7219-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = DA7219_DAI_CH_NUM_MAX, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7219_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = DA7219_DAI_CH_NUM_MAX, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7219_FORMATS, + }, + .ops = &da7219_dai_ops, + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, +}; + + +/* + * DT + */ + +static const struct of_device_id da7219_of_match[] = { + { .compatible = "dlg,da7219", }, + { } +}; +MODULE_DEVICE_TABLE(of, da7219_of_match); + +static enum da7219_ldo_lvl_sel da7219_of_ldo_lvl(struct snd_soc_codec *codec, + u32 val) +{ + switch (val) { + case 1050: + return DA7219_LDO_LVL_SEL_1_05V; + case 1100: + return DA7219_LDO_LVL_SEL_1_10V; + case 1200: + return DA7219_LDO_LVL_SEL_1_20V; + case 1400: + return DA7219_LDO_LVL_SEL_1_40V; + default: + dev_warn(codec->dev, "Invalid LDO level"); + return DA7219_LDO_LVL_SEL_1_05V; + } +} + +static enum da7219_micbias_voltage + da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1800: + return DA7219_MICBIAS_1_8V; + case 2000: + return DA7219_MICBIAS_2_0V; + case 2200: + return DA7219_MICBIAS_2_2V; + case 2400: + return DA7219_MICBIAS_2_4V; + case 2600: + return DA7219_MICBIAS_2_6V; + default: + dev_warn(codec->dev, "Invalid micbias level"); + return DA7219_MICBIAS_2_2V; + } +} + +static enum da7219_mic_amp_in_sel + da7219_of_mic_amp_in_sel(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "diff")) { + return DA7219_MIC_AMP_IN_SEL_DIFF; + } else if (!strcmp(str, "se_p")) { + return DA7219_MIC_AMP_IN_SEL_SE_P; + } else if (!strcmp(str, "se_n")) { + return DA7219_MIC_AMP_IN_SEL_SE_N; + } else { + dev_warn(codec->dev, "Invalid mic input type selection"); + return DA7219_MIC_AMP_IN_SEL_DIFF; + } +} + +static struct da7219_pdata *da7219_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct da7219_pdata *pdata; + const char *of_str; + u32 of_val32; + + pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return NULL; + + if (of_property_read_u32(np, "dlg,ldo-lvl", &of_val32) >= 0) + pdata->ldo_lvl_sel = da7219_of_ldo_lvl(codec, of_val32); + + if (of_property_read_u32(np, "dlg,micbias-lvl", &of_val32) >= 0) + pdata->micbias_lvl = da7219_of_micbias_lvl(codec, of_val32); + else + pdata->micbias_lvl = DA7219_MICBIAS_2_2V; + + if (!of_property_read_string(np, "dlg,mic-amp-in-sel", &of_str)) + pdata->mic_amp_in_sel = da7219_of_mic_amp_in_sel(codec, of_str); + else + pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + + return pdata; +} + + +/* + * Codec driver functions + */ + +static int da7219_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* MCLK */ + clk_prepare_enable(da7219->mclk); + + /* Master bias */ + snd_soc_update_bits(codec, DA7219_REFERENCES, + DA7219_BIAS_EN_MASK, + DA7219_BIAS_EN_MASK); + + /* Enable Internal Digital LDO */ + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_EN_MASK, + DA7219_LDO_EN_MASK); + } + break; + case SND_SOC_BIAS_OFF: + /* Only disable if jack detection not active */ + if (!da7219->aad->jack) { + /* Bypass Internal Digital LDO */ + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_EN_MASK, 0); + + /* Master bias */ + snd_soc_update_bits(codec, DA7219_REFERENCES, + DA7219_BIAS_EN_MASK, 0); + } + + /* MCLK */ + clk_disable_unprepare(da7219->mclk); + break; + } + + return 0; +} + +static const char *da7219_supply_names[DA7219_NUM_SUPPLIES] = { + [DA7219_SUPPLY_VDD] = "VDD", + [DA7219_SUPPLY_VDDMIC] = "VDDMIC", + [DA7219_SUPPLY_VDDIO] = "VDDIO", +}; + +static int da7219_handle_supplies(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct regulator *vddio; + u8 io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V; + int i, ret; + + /* Get required supplies */ + for (i = 0; i < DA7219_NUM_SUPPLIES; ++i) + da7219->supplies[i].supply = da7219_supply_names[i]; + + ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES, + da7219->supplies); + if (ret) + return ret; + + /* Determine VDDIO voltage provided */ + vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; + ret = regulator_get_voltage(vddio); + if (ret < 1200000) + dev_warn(codec->dev, "Invalid VDDIO voltage\n"); + else if (ret < 2800000) + io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; + + /* Enable main supplies */ + ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); + + /* Ensure device in active mode */ + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); + + /* Update IO voltage level range */ + snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl); + + return ret; +} + +static void da7219_handle_pdata(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_pdata *pdata = da7219->pdata; + + if (pdata) { + u8 micbias_lvl = 0; + + /* Internal LDO */ + switch (pdata->ldo_lvl_sel) { + case DA7219_LDO_LVL_SEL_1_05V: + case DA7219_LDO_LVL_SEL_1_10V: + case DA7219_LDO_LVL_SEL_1_20V: + case DA7219_LDO_LVL_SEL_1_40V: + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_LEVEL_SELECT_MASK, + (pdata->ldo_lvl_sel << + DA7219_LDO_LEVEL_SELECT_SHIFT)); + break; + } + + /* Mic Bias voltages */ + switch (pdata->micbias_lvl) { + case DA7219_MICBIAS_1_8V: + case DA7219_MICBIAS_2_0V: + case DA7219_MICBIAS_2_2V: + case DA7219_MICBIAS_2_4V: + case DA7219_MICBIAS_2_6V: + micbias_lvl |= (pdata->micbias_lvl << + DA7219_MICBIAS1_LEVEL_SHIFT); + break; + } + + snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_lvl); + + /* Mic */ + switch (pdata->mic_amp_in_sel) { + case DA7219_MIC_AMP_IN_SEL_DIFF: + case DA7219_MIC_AMP_IN_SEL_SE_P: + case DA7219_MIC_AMP_IN_SEL_SE_N: + snd_soc_write(codec, DA7219_MIC_1_SELECT, + pdata->mic_amp_in_sel); + break; + } + } +} + +static int da7219_probe(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_init(&da7219->lock); + + /* Regulator configuration */ + ret = da7219_handle_supplies(codec); + if (ret) + return ret; + + /* Handle DT/Platform data */ + if (codec->dev->of_node) + da7219->pdata = da7219_of_to_pdata(codec); + else + da7219->pdata = dev_get_platdata(codec->dev); + + da7219_handle_pdata(codec); + + /* Check if MCLK provided */ + da7219->mclk = devm_clk_get(codec->dev, "mclk"); + if (IS_ERR(da7219->mclk)) { + if (PTR_ERR(da7219->mclk) != -ENOENT) + return PTR_ERR(da7219->mclk); + else + da7219->mclk = NULL; + } + + /* Default PC counter to free-running */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, + DA7219_PC_FREERUN_MASK); + + /* Default gain ramping */ + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_RAMP_EN_MASK, + DA7219_MIXIN_L_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, DA7219_ADC_L_RAMP_EN_MASK, + DA7219_ADC_L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, DA7219_DAC_L_RAMP_EN_MASK, + DA7219_DAC_L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, DA7219_DAC_R_RAMP_EN_MASK, + DA7219_DAC_R_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_RAMP_EN_MASK, + DA7219_HP_L_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_RAMP_EN_MASK, + DA7219_HP_R_AMP_RAMP_EN_MASK); + + /* Default infinite tone gen, start/stop by Kcontrol */ + snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK); + + /* Initialise AAD block */ + return da7219_aad_init(codec); +} + +static int da7219_remove(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + da7219_aad_exit(codec); + + /* Supplies */ + return regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies); +} + +#ifdef CONFIG_PM +static int da7219_suspend(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + + /* Put device into standby mode if jack detection disabled */ + if (!da7219->aad->jack) + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, 0); + + return 0; +} + +static int da7219_resume(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + /* Put device into active mode if previously pushed to standby */ + if (!da7219->aad->jack) + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, + DA7219_SYSTEM_ACTIVE_MASK); + + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define da7219_suspend NULL +#define da7219_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_da7219 = { + .probe = da7219_probe, + .remove = da7219_remove, + .suspend = da7219_suspend, + .resume = da7219_resume, + .set_bias_level = da7219_set_bias_level, + + .controls = da7219_snd_controls, + .num_controls = ARRAY_SIZE(da7219_snd_controls), + + .dapm_widgets = da7219_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7219_dapm_widgets), + .dapm_routes = da7219_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7219_audio_map), +}; + + +/* + * Regmap configs + */ + +static struct reg_default da7219_reg_defaults[] = { + { DA7219_MIC_1_SELECT, 0x00 }, + { DA7219_CIF_TIMEOUT_CTRL, 0x01 }, + { DA7219_SR_24_48, 0x00 }, + { DA7219_SR, 0x0A }, + { DA7219_CIF_I2C_ADDR_CFG, 0x02 }, + { DA7219_PLL_CTRL, 0x10 }, + { DA7219_PLL_FRAC_TOP, 0x00 }, + { DA7219_PLL_FRAC_BOT, 0x00 }, + { DA7219_PLL_INTEGER, 0x20 }, + { DA7219_DIG_ROUTING_DAI, 0x10 }, + { DA7219_DAI_CLK_MODE, 0x01 }, + { DA7219_DAI_CTRL, 0x28 }, + { DA7219_DAI_TDM_CTRL, 0x40 }, + { DA7219_DIG_ROUTING_DAC, 0x32 }, + { DA7219_DAI_OFFSET_LOWER, 0x00 }, + { DA7219_DAI_OFFSET_UPPER, 0x00 }, + { DA7219_REFERENCES, 0x00 }, + { DA7219_MIXIN_L_SELECT, 0x00 }, + { DA7219_MIXIN_L_GAIN, 0x03 }, + { DA7219_ADC_L_GAIN, 0x6F }, + { DA7219_ADC_FILTERS1, 0x80 }, + { DA7219_MIC_1_GAIN, 0x01 }, + { DA7219_SIDETONE_CTRL, 0x40 }, + { DA7219_SIDETONE_GAIN, 0x0E }, + { DA7219_DROUTING_ST_OUTFILT_1L, 0x01 }, + { DA7219_DROUTING_ST_OUTFILT_1R, 0x02 }, + { DA7219_DAC_FILTERS5, 0x00 }, + { DA7219_DAC_FILTERS2, 0x88 }, + { DA7219_DAC_FILTERS3, 0x88 }, + { DA7219_DAC_FILTERS4, 0x08 }, + { DA7219_DAC_FILTERS1, 0x80 }, + { DA7219_DAC_L_GAIN, 0x6F }, + { DA7219_DAC_R_GAIN, 0x6F }, + { DA7219_CP_CTRL, 0x20 }, + { DA7219_HP_L_GAIN, 0x39 }, + { DA7219_HP_R_GAIN, 0x39 }, + { DA7219_MIXOUT_L_SELECT, 0x00 }, + { DA7219_MIXOUT_R_SELECT, 0x00 }, + { DA7219_MICBIAS_CTRL, 0x03 }, + { DA7219_MIC_1_CTRL, 0x40 }, + { DA7219_MIXIN_L_CTRL, 0x40 }, + { DA7219_ADC_L_CTRL, 0x40 }, + { DA7219_DAC_L_CTRL, 0x40 }, + { DA7219_DAC_R_CTRL, 0x40 }, + { DA7219_HP_L_CTRL, 0x40 }, + { DA7219_HP_R_CTRL, 0x40 }, + { DA7219_MIXOUT_L_CTRL, 0x10 }, + { DA7219_MIXOUT_R_CTRL, 0x10 }, + { DA7219_CHIP_ID1, 0x23 }, + { DA7219_CHIP_ID2, 0x93 }, + { DA7219_CHIP_REVISION, 0x00 }, + { DA7219_LDO_CTRL, 0x00 }, + { DA7219_IO_CTRL, 0x00 }, + { DA7219_GAIN_RAMP_CTRL, 0x00 }, + { DA7219_PC_COUNT, 0x02 }, + { DA7219_CP_VOL_THRESHOLD1, 0x0E }, + { DA7219_DIG_CTRL, 0x00 }, + { DA7219_ALC_CTRL2, 0x00 }, + { DA7219_ALC_CTRL3, 0x00 }, + { DA7219_ALC_NOISE, 0x3F }, + { DA7219_ALC_TARGET_MIN, 0x3F }, + { DA7219_ALC_TARGET_MAX, 0x00 }, + { DA7219_ALC_GAIN_LIMITS, 0xFF }, + { DA7219_ALC_ANA_GAIN_LIMITS, 0x71 }, + { DA7219_ALC_ANTICLIP_CTRL, 0x00 }, + { DA7219_ALC_ANTICLIP_LEVEL, 0x00 }, + { DA7219_DAC_NG_SETUP_TIME, 0x00 }, + { DA7219_DAC_NG_OFF_THRESH, 0x00 }, + { DA7219_DAC_NG_ON_THRESH, 0x00 }, + { DA7219_DAC_NG_CTRL, 0x00 }, + { DA7219_TONE_GEN_CFG1, 0x00 }, + { DA7219_TONE_GEN_CFG2, 0x00 }, + { DA7219_TONE_GEN_CYCLES, 0x00 }, + { DA7219_TONE_GEN_FREQ1_L, 0x55 }, + { DA7219_TONE_GEN_FREQ1_U, 0x15 }, + { DA7219_TONE_GEN_FREQ2_L, 0x00 }, + { DA7219_TONE_GEN_FREQ2_U, 0x40 }, + { DA7219_TONE_GEN_ON_PER, 0x02 }, + { DA7219_TONE_GEN_OFF_PER, 0x01 }, + { DA7219_ACCDET_IRQ_MASK_A, 0x00 }, + { DA7219_ACCDET_IRQ_MASK_B, 0x00 }, + { DA7219_ACCDET_CONFIG_1, 0xD6 }, + { DA7219_ACCDET_CONFIG_2, 0x34 }, + { DA7219_ACCDET_CONFIG_3, 0x0A }, + { DA7219_ACCDET_CONFIG_4, 0x16 }, + { DA7219_ACCDET_CONFIG_5, 0x21 }, + { DA7219_ACCDET_CONFIG_6, 0x3E }, + { DA7219_ACCDET_CONFIG_7, 0x01 }, + { DA7219_SYSTEM_ACTIVE, 0x00 }, +}; + +static bool da7219_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA7219_MIC_1_GAIN_STATUS: + case DA7219_MIXIN_L_GAIN_STATUS: + case DA7219_ADC_L_GAIN_STATUS: + case DA7219_DAC_L_GAIN_STATUS: + case DA7219_DAC_R_GAIN_STATUS: + case DA7219_HP_L_GAIN_STATUS: + case DA7219_HP_R_GAIN_STATUS: + case DA7219_CIF_CTRL: + case DA7219_PLL_SRM_STS: + case DA7219_ALC_CTRL1: + case DA7219_SYSTEM_MODES_INPUT: + case DA7219_SYSTEM_MODES_OUTPUT: + case DA7219_ALC_OFFSET_AUTO_M_L: + case DA7219_ALC_OFFSET_AUTO_U_L: + case DA7219_TONE_GEN_CFG1: + case DA7219_ACCDET_STATUS_A: + case DA7219_ACCDET_STATUS_B: + case DA7219_ACCDET_IRQ_EVENT_A: + case DA7219_ACCDET_IRQ_EVENT_B: + case DA7219_ACCDET_CONFIG_8: + case DA7219_SYSTEM_STATUS: + return 1; + default: + return 0; + } +} + +static const struct regmap_config da7219_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DA7219_SYSTEM_ACTIVE, + .reg_defaults = da7219_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7219_reg_defaults), + .volatile_reg = da7219_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + + +/* + * I2C layer + */ + +static int da7219_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7219_priv *da7219; + int ret; + + da7219 = devm_kzalloc(&i2c->dev, sizeof(struct da7219_priv), + GFP_KERNEL); + if (!da7219) + return -ENOMEM; + + i2c_set_clientdata(i2c, da7219); + + da7219->regmap = devm_regmap_init_i2c(i2c, &da7219_regmap_config); + if (IS_ERR(da7219->regmap)) { + ret = PTR_ERR(da7219->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7219, + &da7219_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da7219 codec: %d\n", + ret); + } + return ret; +} + +static int da7219_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da7219_i2c_id[] = { + { "da7219", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7219_i2c_id); + +static struct i2c_driver da7219_i2c_driver = { + .driver = { + .name = "da7219", + .of_match_table = da7219_of_match, + }, + .probe = da7219_i2c_probe, + .remove = da7219_i2c_remove, + .id_table = da7219_i2c_id, +}; + +module_i2c_driver(da7219_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA7219 Codec Driver"); +MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h new file mode 100644 index 0000000..b514268 --- /dev/null +++ b/sound/soc/codecs/da7219.h @@ -0,0 +1,820 @@ +/* + * da7219.h - DA7219 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __DA7219_H +#define __DA7219_H + +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/da7219.h> + +/* + * Registers + */ + +#define DA7219_MIC_1_GAIN_STATUS 0x6 +#define DA7219_MIXIN_L_GAIN_STATUS 0x8 +#define DA7219_ADC_L_GAIN_STATUS 0xA +#define DA7219_DAC_L_GAIN_STATUS 0xC +#define DA7219_DAC_R_GAIN_STATUS 0xD +#define DA7219_HP_L_GAIN_STATUS 0xE +#define DA7219_HP_R_GAIN_STATUS 0xF +#define DA7219_MIC_1_SELECT 0x10 +#define DA7219_CIF_TIMEOUT_CTRL 0x12 +#define DA7219_CIF_CTRL 0x13 +#define DA7219_SR_24_48 0x16 +#define DA7219_SR 0x17 +#define DA7219_CIF_I2C_ADDR_CFG 0x1B +#define DA7219_PLL_CTRL 0x20 +#define DA7219_PLL_FRAC_TOP 0x22 +#define DA7219_PLL_FRAC_BOT 0x23 +#define DA7219_PLL_INTEGER 0x24 +#define DA7219_PLL_SRM_STS 0x25 +#define DA7219_DIG_ROUTING_DAI 0x2A +#define DA7219_DAI_CLK_MODE 0x2B +#define DA7219_DAI_CTRL 0x2C +#define DA7219_DAI_TDM_CTRL 0x2D +#define DA7219_DIG_ROUTING_DAC 0x2E +#define DA7219_ALC_CTRL1 0x2F +#define DA7219_DAI_OFFSET_LOWER 0x30 +#define DA7219_DAI_OFFSET_UPPER 0x31 +#define DA7219_REFERENCES 0x32 +#define DA7219_MIXIN_L_SELECT 0x33 +#define DA7219_MIXIN_L_GAIN 0x34 +#define DA7219_ADC_L_GAIN 0x36 +#define DA7219_ADC_FILTERS1 0x38 +#define DA7219_MIC_1_GAIN 0x39 +#define DA7219_SIDETONE_CTRL 0x3A +#define DA7219_SIDETONE_GAIN 0x3B +#define DA7219_DROUTING_ST_OUTFILT_1L 0x3C +#define DA7219_DROUTING_ST_OUTFILT_1R 0x3D +#define DA7219_DAC_FILTERS5 0x40 +#define DA7219_DAC_FILTERS2 0x41 +#define DA7219_DAC_FILTERS3 0x42 +#define DA7219_DAC_FILTERS4 0x43 +#define DA7219_DAC_FILTERS1 0x44 +#define DA7219_DAC_L_GAIN 0x45 +#define DA7219_DAC_R_GAIN 0x46 +#define DA7219_CP_CTRL 0x47 +#define DA7219_HP_L_GAIN 0x48 +#define DA7219_HP_R_GAIN 0x49 +#define DA7219_MIXOUT_L_SELECT 0x4B +#define DA7219_MIXOUT_R_SELECT 0x4C +#define DA7219_SYSTEM_MODES_INPUT 0x50 +#define DA7219_SYSTEM_MODES_OUTPUT 0x51 +#define DA7219_MICBIAS_CTRL 0x62 +#define DA7219_MIC_1_CTRL 0x63 +#define DA7219_MIXIN_L_CTRL 0x65 +#define DA7219_ADC_L_CTRL 0x67 +#define DA7219_DAC_L_CTRL 0x69 +#define DA7219_DAC_R_CTRL 0x6A +#define DA7219_HP_L_CTRL 0x6B +#define DA7219_HP_R_CTRL 0x6C +#define DA7219_MIXOUT_L_CTRL 0x6E +#define DA7219_MIXOUT_R_CTRL 0x6F +#define DA7219_CHIP_ID1 0x81 +#define DA7219_CHIP_ID2 0x82 +#define DA7219_CHIP_REVISION 0x83 +#define DA7219_LDO_CTRL 0x90 +#define DA7219_IO_CTRL 0x91 +#define DA7219_GAIN_RAMP_CTRL 0x92 +#define DA7219_PC_COUNT 0x94 +#define DA7219_CP_VOL_THRESHOLD1 0x95 +#define DA7219_CP_DELAY 0x96 +#define DA7219_DIG_CTRL 0x99 +#define DA7219_ALC_CTRL2 0x9A +#define DA7219_ALC_CTRL3 0x9B +#define DA7219_ALC_NOISE 0x9C +#define DA7219_ALC_TARGET_MIN 0x9D +#define DA7219_ALC_TARGET_MAX 0x9E +#define DA7219_ALC_GAIN_LIMITS 0x9F +#define DA7219_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA7219_ALC_ANTICLIP_CTRL 0xA1 +#define DA7219_ALC_ANTICLIP_LEVEL 0xA2 +#define DA7219_ALC_OFFSET_AUTO_M_L 0xA3 +#define DA7219_ALC_OFFSET_AUTO_U_L 0xA4 +#define DA7219_DAC_NG_SETUP_TIME 0xAF +#define DA7219_DAC_NG_OFF_THRESH 0xB0 +#define DA7219_DAC_NG_ON_THRESH 0xB1 +#define DA7219_DAC_NG_CTRL 0xB2 +#define DA7219_TONE_GEN_CFG1 0xB4 +#define DA7219_TONE_GEN_CFG2 0xB5 +#define DA7219_TONE_GEN_CYCLES 0xB6 +#define DA7219_TONE_GEN_FREQ1_L 0xB7 +#define DA7219_TONE_GEN_FREQ1_U 0xB8 +#define DA7219_TONE_GEN_FREQ2_L 0xB9 +#define DA7219_TONE_GEN_FREQ2_U 0xBA +#define DA7219_TONE_GEN_ON_PER 0xBB +#define DA7219_TONE_GEN_OFF_PER 0xBC +#define DA7219_SYSTEM_STATUS 0xE0 +#define DA7219_SYSTEM_ACTIVE 0xFD + + +/* + * Bit Fields + */ + +#define DA7219_SWITCH_EN_MAX 0x1 + +/* DA7219_MIC_1_GAIN_STATUS = 0x6 */ +#define DA7219_MIC_1_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_MIC_1_AMP_GAIN_STATUS_MASK (0x7 << 0) +#define DA7219_MIC_1_AMP_GAIN_MAX 0x7 + +/* DA7219_MIXIN_L_GAIN_STATUS = 0x8 */ +#define DA7219_MIXIN_L_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_MIXIN_L_AMP_GAIN_STATUS_MASK (0xF << 0) + +/* DA7219_ADC_L_GAIN_STATUS = 0xA */ +#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_DAC_L_GAIN_STATUS = 0xC */ +#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_DAC_R_GAIN_STATUS = 0xD */ +#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_HP_L_GAIN_STATUS = 0xE */ +#define DA7219_HP_L_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_HP_L_AMP_GAIN_STATUS_MASK (0x3F << 0) + +/* DA7219_HP_R_GAIN_STATUS = 0xF */ +#define DA7219_HP_R_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_HP_R_AMP_GAIN_STATUS_MASK (0x3F << 0) + +/* DA7219_MIC_1_SELECT = 0x10 */ +#define DA7219_MIC_1_AMP_IN_SEL_SHIFT 0 +#define DA7219_MIC_1_AMP_IN_SEL_MASK (0x3 << 0) + +/* DA7219_CIF_TIMEOUT_CTRL = 0x12 */ +#define DA7219_I2C_TIMEOUT_EN_SHIFT 0 +#define DA7219_I2C_TIMEOUT_EN_MASK (0x1 << 0) + +/* DA7219_CIF_CTRL = 0x13 */ +#define DA7219_CIF_I2C_WRITE_MODE_SHIFT 0 +#define DA7219_CIF_I2C_WRITE_MODE_MASK (0x1 << 0) +#define DA7219_CIF_REG_SOFT_RESET_SHIFT 7 +#define DA7219_CIF_REG_SOFT_RESET_MASK (0x1 << 7) + +/* DA7219_SR_24_48 = 0x16 */ +#define DA7219_SR_24_48_SHIFT 0 +#define DA7219_SR_24_48_MASK (0x1 << 0) + +/* DA7219_SR = 0x17 */ +#define DA7219_SR_SHIFT 0 +#define DA7219_SR_MASK (0xF << 0) +#define DA7219_SR_8000 (0x01 << 0) +#define DA7219_SR_11025 (0x02 << 0) +#define DA7219_SR_12000 (0x03 << 0) +#define DA7219_SR_16000 (0x05 << 0) +#define DA7219_SR_22050 (0x06 << 0) +#define DA7219_SR_24000 (0x07 << 0) +#define DA7219_SR_32000 (0x09 << 0) +#define DA7219_SR_44100 (0x0A << 0) +#define DA7219_SR_48000 (0x0B << 0) +#define DA7219_SR_88200 (0x0E << 0) +#define DA7219_SR_96000 (0x0F << 0) + +/* DA7219_CIF_I2C_ADDR_CFG = 0x1B */ +#define DA7219_CIF_I2C_ADDR_CFG_SHIFT 0 +#define DA7219_CIF_I2C_ADDR_CFG_MASK (0x3 << 0) + +/* DA7219_PLL_CTRL = 0x20 */ +#define DA7219_PLL_INDIV_SHIFT 2 +#define DA7219_PLL_INDIV_MASK (0x7 << 2) +#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2) +#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2) +#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2) +#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2) +#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2) +#define DA7219_PLL_MCLK_SQR_EN_SHIFT 5 +#define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5) +#define DA7219_PLL_MODE_SHIFT 6 +#define DA7219_PLL_MODE_MASK (0x3 << 6) +#define DA7219_PLL_MODE_BYPASS (0x0 << 6) +#define DA7219_PLL_MODE_NORMAL (0x1 << 6) +#define DA7219_PLL_MODE_SRM (0x2 << 6) +#define DA7219_PLL_MODE_32KHZ (0x3 << 6) + +/* DA7219_PLL_FRAC_TOP = 0x22 */ +#define DA7219_PLL_FBDIV_FRAC_TOP_SHIFT 0 +#define DA7219_PLL_FBDIV_FRAC_TOP_MASK (0x1F << 0) + +/* DA7219_PLL_FRAC_BOT = 0x23 */ +#define DA7219_PLL_FBDIV_FRAC_BOT_SHIFT 0 +#define DA7219_PLL_FBDIV_FRAC_BOT_MASK (0xFF << 0) + +/* DA7219_PLL_INTEGER = 0x24 */ +#define DA7219_PLL_FBDIV_INTEGER_SHIFT 0 +#define DA7219_PLL_FBDIV_INTEGER_MASK (0x7F << 0) + +/* DA7219_PLL_SRM_STS = 0x25 */ +#define DA7219_PLL_SRM_STATE_SHIFT 0 +#define DA7219_PLL_SRM_STATE_MASK (0xF << 0) +#define DA7219_PLL_SRM_STATUS_SHIFT 4 +#define DA7219_PLL_SRM_STATUS_MASK (0xF << 4) +#define DA7219_PLL_SRM_STS_SRM_LOCK (0x1 << 7) + +/* DA7219_DIG_ROUTING_DAI = 0x2A */ +#define DA7219_DAI_L_SRC_SHIFT 0 +#define DA7219_DAI_L_SRC_MASK (0x3 << 0) +#define DA7219_DAI_R_SRC_SHIFT 4 +#define DA7219_DAI_R_SRC_MASK (0x3 << 4) +#define DA7219_OUT_SRC_MAX 4 + +/* DA7219_DAI_CLK_MODE = 0x2B */ +#define DA7219_DAI_BCLKS_PER_WCLK_SHIFT 0 +#define DA7219_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_32 (0x0 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_64 (0x1 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_128 (0x2 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) +#define DA7219_DAI_CLK_POL_SHIFT 2 +#define DA7219_DAI_CLK_POL_MASK (0x1 << 2) +#define DA7219_DAI_CLK_POL_INV (0x1 << 2) +#define DA7219_DAI_WCLK_POL_SHIFT 3 +#define DA7219_DAI_WCLK_POL_MASK (0x1 << 3) +#define DA7219_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7219_DAI_WCLK_TRI_STATE_SHIFT 4 +#define DA7219_DAI_WCLK_TRI_STATE_MASK (0x1 << 4) +#define DA7219_DAI_CLK_EN_SHIFT 7 +#define DA7219_DAI_CLK_EN_MASK (0x1 << 7) + +/* DA7219_DAI_CTRL = 0x2C */ +#define DA7219_DAI_FORMAT_SHIFT 0 +#define DA7219_DAI_FORMAT_MASK (0x3 << 0) +#define DA7219_DAI_FORMAT_I2S (0x0 << 0) +#define DA7219_DAI_FORMAT_LEFT_J (0x1 << 0) +#define DA7219_DAI_FORMAT_RIGHT_J (0x2 << 0) +#define DA7219_DAI_FORMAT_DSP (0x3 << 0) +#define DA7219_DAI_WORD_LENGTH_SHIFT 2 +#define DA7219_DAI_WORD_LENGTH_MASK (0x3 << 2) +#define DA7219_DAI_WORD_LENGTH_S16_LE (0x0 << 2) +#define DA7219_DAI_WORD_LENGTH_S20_LE (0x1 << 2) +#define DA7219_DAI_WORD_LENGTH_S24_LE (0x2 << 2) +#define DA7219_DAI_WORD_LENGTH_S32_LE (0x3 << 2) +#define DA7219_DAI_CH_NUM_SHIFT 4 +#define DA7219_DAI_CH_NUM_MASK (0x3 << 4) +#define DA7219_DAI_CH_NUM_MAX 2 +#define DA7219_DAI_EN_SHIFT 7 +#define DA7219_DAI_EN_MASK (0x1 << 7) + +/* DA7219_DAI_TDM_CTRL = 0x2D */ +#define DA7219_DAI_TDM_CH_EN_SHIFT 0 +#define DA7219_DAI_TDM_CH_EN_MASK (0x3 << 0) +#define DA7219_DAI_OE_SHIFT 6 +#define DA7219_DAI_OE_MASK (0x1 << 6) +#define DA7219_DAI_TDM_MODE_EN_SHIFT 7 +#define DA7219_DAI_TDM_MODE_EN_MASK (0x1 << 7) +#define DA7219_DAI_TDM_MAX_SLOTS 2 + +/* DA7219_DIG_ROUTING_DAC = 0x2E */ +#define DA7219_DAC_L_SRC_SHIFT 0 +#define DA7219_DAC_L_SRC_MASK (0x3 << 0) +#define DA7219_DAC_L_SRC_TONEGEN (0x1 << 0) +#define DA7219_DAC_L_MONO_SHIFT 3 +#define DA7219_DAC_L_MONO_MASK (0x1 << 3) +#define DA7219_DAC_R_SRC_SHIFT 4 +#define DA7219_DAC_R_SRC_MASK (0x3 << 4) +#define DA7219_DAC_R_SRC_TONEGEN (0x1 << 4) +#define DA7219_DAC_R_MONO_SHIFT 7 +#define DA7219_DAC_R_MONO_MASK (0x1 << 7) + +/* DA7219_ALC_CTRL1 = 0x2F */ +#define DA7219_ALC_OFFSET_EN_SHIFT 0 +#define DA7219_ALC_OFFSET_EN_MASK (0x1 << 0) +#define DA7219_ALC_SYNC_MODE_SHIFT 1 +#define DA7219_ALC_SYNC_MODE_MASK (0x1 << 1) +#define DA7219_ALC_EN_SHIFT 3 +#define DA7219_ALC_EN_MASK (0x1 << 3) +#define DA7219_ALC_AUTO_CALIB_EN_SHIFT 4 +#define DA7219_ALC_AUTO_CALIB_EN_MASK (0x1 << 4) +#define DA7219_ALC_CALIB_OVERFLOW_SHIFT 5 +#define DA7219_ALC_CALIB_OVERFLOW_MASK (0x1 << 5) + +/* DA7219_DAI_OFFSET_LOWER = 0x30 */ +#define DA7219_DAI_OFFSET_LOWER_SHIFT 0 +#define DA7219_DAI_OFFSET_LOWER_MASK (0xFF << 0) + +/* DA7219_DAI_OFFSET_UPPER = 0x31 */ +#define DA7219_DAI_OFFSET_UPPER_SHIFT 0 +#define DA7219_DAI_OFFSET_UPPER_MASK (0x7 << 0) +#define DA7219_DAI_OFFSET_MAX 0x2FF + +/* DA7219_REFERENCES = 0x32 */ +#define DA7219_BIAS_EN_SHIFT 3 +#define DA7219_BIAS_EN_MASK (0x1 << 3) +#define DA7219_VMID_FAST_CHARGE_SHIFT 4 +#define DA7219_VMID_FAST_CHARGE_MASK (0x1 << 4) + +/* DA7219_MIXIN_L_SELECT = 0x33 */ +#define DA7219_MIXIN_L_MIX_SELECT_SHIFT 0 +#define DA7219_MIXIN_L_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_MIXIN_L_GAIN = 0x34 */ +#define DA7219_MIXIN_L_AMP_GAIN_SHIFT 0 +#define DA7219_MIXIN_L_AMP_GAIN_MASK (0xF << 0) +#define DA7219_MIXIN_L_AMP_GAIN_MAX 0xF + +/* DA7219_ADC_L_GAIN = 0x36 */ +#define DA7219_ADC_L_DIGITAL_GAIN_SHIFT 0 +#define DA7219_ADC_L_DIGITAL_GAIN_MASK (0x7F << 0) +#define DA7219_ADC_L_DIGITAL_GAIN_MAX 0x7F + +/* DA7219_ADC_FILTERS1 = 0x38 */ +#define DA7219_ADC_VOICE_HPF_CORNER_SHIFT 0 +#define DA7219_ADC_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7219_VOICE_HPF_CORNER_MAX 8 +#define DA7219_ADC_VOICE_EN_SHIFT 3 +#define DA7219_ADC_VOICE_EN_MASK (0x1 << 3) +#define DA7219_ADC_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7219_ADC_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7219_AUDIO_HPF_CORNER_MAX 4 +#define DA7219_ADC_HPF_EN_SHIFT 7 +#define DA7219_ADC_HPF_EN_MASK (0x1 << 7) +#define DA7219_HPF_MODE_SHIFT 0 +#define DA7219_HPF_DISABLED ((0x0 << 3) | (0x0 << 7)) +#define DA7219_HPF_AUDIO_EN ((0x0 << 3) | (0x1 << 7)) +#define DA7219_HPF_VOICE_EN ((0x1 << 3) | (0x1 << 7)) +#define DA7219_HPF_MODE_MASK ((0x1 << 3) | (0x1 << 7)) +#define DA7219_HPF_MODE_MAX 3 + +/* DA7219_MIC_1_GAIN = 0x39 */ +#define DA7219_MIC_1_AMP_GAIN_SHIFT 0 +#define DA7219_MIC_1_AMP_GAIN_MASK (0x7 << 0) + +/* DA7219_SIDETONE_CTRL = 0x3A */ +#define DA7219_SIDETONE_MUTE_EN_SHIFT 6 +#define DA7219_SIDETONE_MUTE_EN_MASK (0x1 << 6) +#define DA7219_SIDETONE_EN_SHIFT 7 +#define DA7219_SIDETONE_EN_MASK (0x1 << 7) + +/* DA7219_SIDETONE_GAIN = 0x3B */ +#define DA7219_SIDETONE_GAIN_SHIFT 0 +#define DA7219_SIDETONE_GAIN_MASK (0xF << 0) +#define DA7219_SIDETONE_GAIN_MAX 0xE + +/* DA7219_DROUTING_ST_OUTFILT_1L = 0x3C */ +#define DA7219_OUTFILT_ST_1L_SRC_SHIFT 0 +#define DA7219_OUTFILT_ST_1L_SRC_MASK (0x7 << 0) +#define DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT 0 +#define DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT 1 +#define DA7219_DMIX_ST_SRC_SIDETONE_SHIFT 2 +#define DA7219_DMIX_ST_SRC_OUTFILT1L (0x1 << 0) +#define DA7219_DMIX_ST_SRC_OUTFILT1R (0x1 << 1) + +/* DA7219_DROUTING_ST_OUTFILT_1R = 0x3D */ +#define DA7219_OUTFILT_ST_1R_SRC_SHIFT 0 +#define DA7219_OUTFILT_ST_1R_SRC_MASK (0x7 << 0) + +/* DA7219_DAC_FILTERS5 = 0x40 */ +#define DA7219_DAC_SOFTMUTE_RATE_SHIFT 4 +#define DA7219_DAC_SOFTMUTE_RATE_MASK (0x7 << 4) +#define DA7219_DAC_SOFTMUTE_RATE_MAX 7 +#define DA7219_DAC_SOFTMUTE_EN_SHIFT 7 +#define DA7219_DAC_SOFTMUTE_EN_MASK (0x1 << 7) + +/* DA7219_DAC_FILTERS2 = 0x41 */ +#define DA7219_DAC_EQ_BAND1_SHIFT 0 +#define DA7219_DAC_EQ_BAND1_MASK (0xF << 0) +#define DA7219_DAC_EQ_BAND2_SHIFT 4 +#define DA7219_DAC_EQ_BAND2_MASK (0xF << 4) +#define DA7219_DAC_EQ_BAND_MAX 0xF + +/* DA7219_DAC_FILTERS3 = 0x42 */ +#define DA7219_DAC_EQ_BAND3_SHIFT 0 +#define DA7219_DAC_EQ_BAND3_MASK (0xF << 0) +#define DA7219_DAC_EQ_BAND4_SHIFT 4 +#define DA7219_DAC_EQ_BAND4_MASK (0xF << 4) + +/* DA7219_DAC_FILTERS4 = 0x43 */ +#define DA7219_DAC_EQ_BAND5_SHIFT 0 +#define DA7219_DAC_EQ_BAND5_MASK (0xF << 0) +#define DA7219_DAC_EQ_EN_SHIFT 7 +#define DA7219_DAC_EQ_EN_MASK (0x1 << 7) + +/* DA7219_DAC_FILTERS1 = 0x44 */ +#define DA7219_DAC_VOICE_HPF_CORNER_SHIFT 0 +#define DA7219_DAC_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7219_DAC_VOICE_EN_SHIFT 3 +#define DA7219_DAC_VOICE_EN_MASK (0x1 << 3) +#define DA7219_DAC_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7219_DAC_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7219_DAC_HPF_EN_SHIFT 7 +#define DA7219_DAC_HPF_EN_MASK (0x1 << 7) + +/* DA7219_DAC_L_GAIN = 0x45 */ +#define DA7219_DAC_L_DIGITAL_GAIN_SHIFT 0 +#define DA7219_DAC_L_DIGITAL_GAIN_MASK (0x7F << 0) +#define DA7219_DAC_DIGITAL_GAIN_MAX 0x7F +#define DA7219_DAC_DIGITAL_GAIN_0DB (0x6F << 0) + +/* DA7219_DAC_R_GAIN = 0x46 */ +#define DA7219_DAC_R_DIGITAL_GAIN_SHIFT 0 +#define DA7219_DAC_R_DIGITAL_GAIN_MASK (0x7F << 0) + +/* DA7219_CP_CTRL = 0x47 */ +#define DA7219_CP_MCHANGE_SHIFT 4 +#define DA7219_CP_MCHANGE_MASK (0x3 << 4) +#define DA7219_CP_MCHANGE_REL_MASK 0x3 +#define DA7219_CP_MCHANGE_MAX 3 +#define DA7219_CP_MCHANGE_LARGEST_VOL 0x1 +#define DA7219_CP_MCHANGE_DAC_VOL 0x2 +#define DA7219_CP_MCHANGE_SIG_MAG 0x3 +#define DA7219_CP_EN_SHIFT 7 +#define DA7219_CP_EN_MASK (0x1 << 7) + +/* DA7219_HP_L_GAIN = 0x48 */ +#define DA7219_HP_L_AMP_GAIN_SHIFT 0 +#define DA7219_HP_L_AMP_GAIN_MASK (0x3F << 0) +#define DA7219_HP_AMP_GAIN_MAX 0x3F +#define DA7219_HP_AMP_GAIN_0DB (0x39 << 0) + +/* DA7219_HP_R_GAIN = 0x49 */ +#define DA7219_HP_R_AMP_GAIN_SHIFT 0 +#define DA7219_HP_R_AMP_GAIN_MASK (0x3F << 0) + +/* DA7219_MIXOUT_L_SELECT = 0x4B */ +#define DA7219_MIXOUT_L_MIX_SELECT_SHIFT 0 +#define DA7219_MIXOUT_L_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_MIXOUT_R_SELECT = 0x4C */ +#define DA7219_MIXOUT_R_MIX_SELECT_SHIFT 0 +#define DA7219_MIXOUT_R_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_SYSTEM_MODES_INPUT = 0x50 */ +#define DA7219_MODE_SUBMIT_SHIFT 0 +#define DA7219_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7219_ADC_MODE_SHIFT 1 +#define DA7219_ADC_MODE_MASK (0x7F << 1) + +/* DA7219_SYSTEM_MODES_OUTPUT = 0x51 */ +#define DA7219_MODE_SUBMIT_SHIFT 0 +#define DA7219_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7219_DAC_MODE_SHIFT 1 +#define DA7219_DAC_MODE_MASK (0x7F << 1) + +/* DA7219_MICBIAS_CTRL = 0x62 */ +#define DA7219_MICBIAS1_LEVEL_SHIFT 0 +#define DA7219_MICBIAS1_LEVEL_MASK (0x7 << 0) +#define DA7219_MICBIAS1_EN_SHIFT 3 +#define DA7219_MICBIAS1_EN_MASK (0x1 << 3) + +/* DA7219_MIC_1_CTRL = 0x63 */ +#define DA7219_MIC_1_AMP_RAMP_EN_SHIFT 5 +#define DA7219_MIC_1_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_MIC_1_AMP_MUTE_EN_SHIFT 6 +#define DA7219_MIC_1_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_MIC_1_AMP_EN_SHIFT 7 +#define DA7219_MIC_1_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXIN_L_CTRL = 0x65 */ +#define DA7219_MIXIN_L_MIX_EN_SHIFT 3 +#define DA7219_MIXIN_L_MIX_EN_MASK (0x1 << 3) +#define DA7219_MIXIN_L_AMP_ZC_EN_SHIFT 4 +#define DA7219_MIXIN_L_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT 5 +#define DA7219_MIXIN_L_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT 6 +#define DA7219_MIXIN_L_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_MIXIN_L_AMP_EN_SHIFT 7 +#define DA7219_MIXIN_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_ADC_L_CTRL = 0x67 */ +#define DA7219_ADC_L_BIAS_SHIFT 0 +#define DA7219_ADC_L_BIAS_MASK (0x3 << 0) +#define DA7219_ADC_L_RAMP_EN_SHIFT 5 +#define DA7219_ADC_L_RAMP_EN_MASK (0x1 << 5) +#define DA7219_ADC_L_MUTE_EN_SHIFT 6 +#define DA7219_ADC_L_MUTE_EN_MASK (0x1 << 6) +#define DA7219_ADC_L_EN_SHIFT 7 +#define DA7219_ADC_L_EN_MASK (0x1 << 7) + +/* DA7219_DAC_L_CTRL = 0x69 */ +#define DA7219_DAC_L_RAMP_EN_SHIFT 5 +#define DA7219_DAC_L_RAMP_EN_MASK (0x1 << 5) +#define DA7219_DAC_L_MUTE_EN_SHIFT 6 +#define DA7219_DAC_L_MUTE_EN_MASK (0x1 << 6) +#define DA7219_DAC_L_EN_SHIFT 7 +#define DA7219_DAC_L_EN_MASK (0x1 << 7) + +/* DA7219_DAC_R_CTRL = 0x6A */ +#define DA7219_DAC_R_RAMP_EN_SHIFT 5 +#define DA7219_DAC_R_RAMP_EN_MASK (0x1 << 5) +#define DA7219_DAC_R_MUTE_EN_SHIFT 6 +#define DA7219_DAC_R_MUTE_EN_MASK (0x1 << 6) +#define DA7219_DAC_R_EN_SHIFT 7 +#define DA7219_DAC_R_EN_MASK (0x1 << 7) + +/* DA7219_HP_L_CTRL = 0x6B */ +#define DA7219_HP_L_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7219_HP_L_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7219_HP_L_AMP_OE_SHIFT 3 +#define DA7219_HP_L_AMP_OE_MASK (0x1 << 3) +#define DA7219_HP_L_AMP_ZC_EN_SHIFT 4 +#define DA7219_HP_L_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_HP_L_AMP_RAMP_EN_SHIFT 5 +#define DA7219_HP_L_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_HP_L_AMP_MUTE_EN_SHIFT 6 +#define DA7219_HP_L_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_HP_L_AMP_EN_SHIFT 7 +#define DA7219_HP_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_HP_R_CTRL = 0x6C */ +#define DA7219_HP_R_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7219_HP_R_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7219_HP_R_AMP_OE_SHIFT 3 +#define DA7219_HP_R_AMP_OE_MASK (0x1 << 3) +#define DA7219_HP_R_AMP_ZC_EN_SHIFT 4 +#define DA7219_HP_R_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_HP_R_AMP_RAMP_EN_SHIFT 5 +#define DA7219_HP_R_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_HP_R_AMP_MUTE_EN_SHIFT 6 +#define DA7219_HP_R_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_HP_R_AMP_EN_SHIFT 7 +#define DA7219_HP_R_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXOUT_L_CTRL = 0x6E */ +#define DA7219_MIXOUT_L_AMP_EN_SHIFT 7 +#define DA7219_MIXOUT_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXOUT_R_CTRL = 0x6F */ +#define DA7219_MIXOUT_R_AMP_EN_SHIFT 7 +#define DA7219_MIXOUT_R_AMP_EN_MASK (0x1 << 7) + +/* DA7219_CHIP_ID1 = 0x81 */ +#define DA7219_CHIP_ID1_SHIFT 0 +#define DA7219_CHIP_ID1_MASK (0xFF << 0) + +/* DA7219_CHIP_ID2 = 0x82 */ +#define DA7219_CHIP_ID2_SHIFT 0 +#define DA7219_CHIP_ID2_MASK (0xFF << 0) + +/* DA7219_CHIP_REVISION = 0x83 */ +#define DA7219_CHIP_MINOR_SHIFT 0 +#define DA7219_CHIP_MINOR_MASK (0xF << 0) +#define DA7219_CHIP_MAJOR_SHIFT 4 +#define DA7219_CHIP_MAJOR_MASK (0xF << 4) + +/* DA7219_LDO_CTRL = 0x90 */ +#define DA7219_LDO_LEVEL_SELECT_SHIFT 4 +#define DA7219_LDO_LEVEL_SELECT_MASK (0x3 << 4) +#define DA7219_LDO_EN_SHIFT 7 +#define DA7219_LDO_EN_MASK (0x1 << 7) + +/* DA7219_IO_CTRL = 0x91 */ +#define DA7219_IO_VOLTAGE_LEVEL_SHIFT 0 +#define DA7219_IO_VOLTAGE_LEVEL_MASK (0x1 << 0) +#define DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V 0 +#define DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V 1 + +/* DA7219_GAIN_RAMP_CTRL = 0x92 */ +#define DA7219_GAIN_RAMP_RATE_SHIFT 0 +#define DA7219_GAIN_RAMP_RATE_MASK (0x3 << 0) +#define DA7219_GAIN_RAMP_RATE_MAX 4 + +/* DA7219_PC_COUNT = 0x94 */ +#define DA7219_PC_FREERUN_SHIFT 0 +#define DA7219_PC_FREERUN_MASK (0x1 << 0) +#define DA7219_PC_RESYNC_AUTO_SHIFT 1 +#define DA7219_PC_RESYNC_AUTO_MASK (0x1 << 1) + +/* DA7219_CP_VOL_THRESHOLD1 = 0x95 */ +#define DA7219_CP_THRESH_VDD2_SHIFT 0 +#define DA7219_CP_THRESH_VDD2_MASK (0x3F << 0) +#define DA7219_CP_THRESH_VDD2_MAX 0x3F + +/* DA7219_DIG_CTRL = 0x99 */ +#define DA7219_DAC_L_INV_SHIFT 3 +#define DA7219_DAC_L_INV_MASK (0x1 << 3) +#define DA7219_DAC_R_INV_SHIFT 7 +#define DA7219_DAC_R_INV_MASK (0x1 << 7) + +/* DA7219_ALC_CTRL2 = 0x9A */ +#define DA7219_ALC_ATTACK_SHIFT 0 +#define DA7219_ALC_ATTACK_MASK (0xF << 0) +#define DA7219_ALC_ATTACK_MAX 13 +#define DA7219_ALC_RELEASE_SHIFT 4 +#define DA7219_ALC_RELEASE_MASK (0xF << 4) +#define DA7219_ALC_RELEASE_MAX 11 + +/* DA7219_ALC_CTRL3 = 0x9B */ +#define DA7219_ALC_HOLD_SHIFT 0 +#define DA7219_ALC_HOLD_MASK (0xF << 0) +#define DA7219_ALC_HOLD_MAX 16 +#define DA7219_ALC_INTEG_ATTACK_SHIFT 4 +#define DA7219_ALC_INTEG_ATTACK_MASK (0x3 << 4) +#define DA7219_ALC_INTEG_RELEASE_SHIFT 6 +#define DA7219_ALC_INTEG_RELEASE_MASK (0x3 << 6) +#define DA7219_ALC_INTEG_MAX 4 + +/* DA7219_ALC_NOISE = 0x9C */ +#define DA7219_ALC_NOISE_SHIFT 0 +#define DA7219_ALC_NOISE_MASK (0x3F << 0) +#define DA7219_ALC_THRESHOLD_MAX 0x3F + +/* DA7219_ALC_TARGET_MIN = 0x9D */ +#define DA7219_ALC_THRESHOLD_MIN_SHIFT 0 +#define DA7219_ALC_THRESHOLD_MIN_MASK (0x3F << 0) + +/* DA7219_ALC_TARGET_MAX = 0x9E */ +#define DA7219_ALC_THRESHOLD_MAX_SHIFT 0 +#define DA7219_ALC_THRESHOLD_MAX_MASK (0x3F << 0) + +/* DA7219_ALC_GAIN_LIMITS = 0x9F */ +#define DA7219_ALC_ATTEN_MAX_SHIFT 0 +#define DA7219_ALC_ATTEN_MAX_MASK (0xF << 0) +#define DA7219_ALC_GAIN_MAX_SHIFT 4 +#define DA7219_ALC_GAIN_MAX_MASK (0xF << 4) +#define DA7219_ALC_ATTEN_GAIN_MAX 0xF + +/* DA7219_ALC_ANA_GAIN_LIMITS = 0xA0 */ +#define DA7219_ALC_ANA_GAIN_MIN_SHIFT 0 +#define DA7219_ALC_ANA_GAIN_MIN_MASK (0x7 << 0) +#define DA7219_ALC_ANA_GAIN_MIN 0x1 +#define DA7219_ALC_ANA_GAIN_MAX_SHIFT 4 +#define DA7219_ALC_ANA_GAIN_MAX_MASK (0x7 << 4) +#define DA7219_ALC_ANA_GAIN_MAX 0x7 + +/* DA7219_ALC_ANTICLIP_CTRL = 0xA1 */ +#define DA7219_ALC_ANTICLIP_STEP_SHIFT 0 +#define DA7219_ALC_ANTICLIP_STEP_MASK (0x3 << 0) +#define DA7219_ALC_ANTICLIP_STEP_MAX 4 +#define DA7219_ALC_ANTIPCLIP_EN_SHIFT 7 +#define DA7219_ALC_ANTIPCLIP_EN_MASK (0x1 << 7) + +/* DA7219_ALC_ANTICLIP_LEVEL = 0xA2 */ +#define DA7219_ALC_ANTICLIP_LEVEL_SHIFT 0 +#define DA7219_ALC_ANTICLIP_LEVEL_MASK (0x7F << 0) + +/* DA7219_ALC_OFFSET_AUTO_M_L = 0xA3 */ +#define DA7219_ALC_OFFSET_AUTO_M_L_SHIFT 0 +#define DA7219_ALC_OFFSET_AUTO_M_L_MASK (0xFF << 0) + +/* DA7219_ALC_OFFSET_AUTO_U_L = 0xA4 */ +#define DA7219_ALC_OFFSET_AUTO_U_L_SHIFT 0 +#define DA7219_ALC_OFFSET_AUTO_U_L_MASK (0xF << 0) + +/* DA7219_DAC_NG_SETUP_TIME = 0xAF */ +#define DA7219_DAC_NG_SETUP_TIME_SHIFT 0 +#define DA7219_DAC_NG_SETUP_TIME_MASK (0x3 << 0) +#define DA7219_DAC_NG_SETUP_TIME_MAX 4 +#define DA7219_DAC_NG_RAMPUP_RATE_SHIFT 2 +#define DA7219_DAC_NG_RAMPUP_RATE_MASK (0x1 << 2) +#define DA7219_DAC_NG_RAMPDN_RATE_SHIFT 3 +#define DA7219_DAC_NG_RAMPDN_RATE_MASK (0x1 << 3) +#define DA7219_DAC_NG_RAMP_RATE_MAX 2 + +/* DA7219_DAC_NG_OFF_THRESH = 0xB0 */ +#define DA7219_DAC_NG_OFF_THRESHOLD_SHIFT 0 +#define DA7219_DAC_NG_OFF_THRESHOLD_MASK (0x7 << 0) +#define DA7219_DAC_NG_THRESHOLD_MAX 0x7 + +/* DA7219_DAC_NG_ON_THRESH = 0xB1 */ +#define DA7219_DAC_NG_ON_THRESHOLD_SHIFT 0 +#define DA7219_DAC_NG_ON_THRESHOLD_MASK (0x7 << 0) + +/* DA7219_DAC_NG_CTRL = 0xB2 */ +#define DA7219_DAC_NG_EN_SHIFT 7 +#define DA7219_DAC_NG_EN_MASK (0x1 << 7) + +/* DA7219_TONE_GEN_CFG1 = 0xB4 */ +#define DA7219_DTMF_REG_SHIFT 0 +#define DA7219_DTMF_REG_MASK (0xF << 0) +#define DA7219_DTMF_REG_MAX 16 +#define DA7219_DTMF_EN_SHIFT 4 +#define DA7219_DTMF_EN_MASK (0x1 << 4) +#define DA7219_START_STOPN_SHIFT 7 +#define DA7219_START_STOPN_MASK (0x1 << 7) + +/* DA7219_TONE_GEN_CFG2 = 0xB5 */ +#define DA7219_SWG_SEL_SHIFT 0 +#define DA7219_SWG_SEL_MASK (0x3 << 0) +#define DA7219_SWG_SEL_MAX 4 +#define DA7219_SWG_SEL_SRAMP (0x3 << 0) +#define DA7219_TONE_GEN_GAIN_SHIFT 4 +#define DA7219_TONE_GEN_GAIN_MASK (0xF << 4) +#define DA7219_TONE_GEN_GAIN_MAX 0xF +#define DA7219_TONE_GEN_GAIN_MINUS_9DB (0x3 << 4) +#define DA7219_TONE_GEN_GAIN_MINUS_15DB (0x5 << 4) + +/* DA7219_TONE_GEN_CYCLES = 0xB6 */ +#define DA7219_BEEP_CYCLES_SHIFT 0 +#define DA7219_BEEP_CYCLES_MASK (0x7 << 0) + +/* DA7219_TONE_GEN_FREQ1_L = 0xB7 */ +#define DA7219_FREQ1_L_SHIFT 0 +#define DA7219_FREQ1_L_MASK (0xFF << 0) +#define DA7219_FREQ_MAX 0xFFFF + +/* DA7219_TONE_GEN_FREQ1_U = 0xB8 */ +#define DA7219_FREQ1_U_SHIFT 0 +#define DA7219_FREQ1_U_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_FREQ2_L = 0xB9 */ +#define DA7219_FREQ2_L_SHIFT 0 +#define DA7219_FREQ2_L_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_FREQ2_U = 0xBA */ +#define DA7219_FREQ2_U_SHIFT 0 +#define DA7219_FREQ2_U_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_ON_PER = 0xBB */ +#define DA7219_BEEP_ON_PER_SHIFT 0 +#define DA7219_BEEP_ON_PER_MASK (0x3F << 0) +#define DA7219_BEEP_ON_OFF_MAX 0x3F + +/* DA7219_TONE_GEN_OFF_PER = 0xBC */ +#define DA7219_BEEP_OFF_PER_SHIFT 0 +#define DA7219_BEEP_OFF_PER_MASK (0x3F << 0) + +/* DA7219_SYSTEM_STATUS = 0xE0 */ +#define DA7219_SC1_BUSY_SHIFT 0 +#define DA7219_SC1_BUSY_MASK (0x1 << 0) +#define DA7219_SC2_BUSY_SHIFT 1 +#define DA7219_SC2_BUSY_MASK (0x1 << 1) + +/* DA7219_SYSTEM_ACTIVE = 0xFD */ +#define DA7219_SYSTEM_ACTIVE_SHIFT 0 +#define DA7219_SYSTEM_ACTIVE_MASK (0x1 << 0) + + +/* + * General defines & data + */ + +/* Register inversion */ +#define DA7219_NO_INVERT 0 +#define DA7219_INVERT 1 + +/* Byte related defines */ +#define DA7219_BYTE_SHIFT 8 +#define DA7219_BYTE_MASK 0xFF + +/* PLL Output Frequencies */ +#define DA7219_PLL_FREQ_OUT_90316 90316800 +#define DA7219_PLL_FREQ_OUT_98304 98304000 + +/* PLL Frequency Dividers */ +#define DA7219_PLL_INDIV_2_5_MHZ_VAL 1 +#define DA7219_PLL_INDIV_5_10_MHZ_VAL 2 +#define DA7219_PLL_INDIV_10_20_MHZ_VAL 4 +#define DA7219_PLL_INDIV_20_40_MHZ_VAL 8 +#define DA7219_PLL_INDIV_40_54_MHZ_VAL 16 + +/* SRM */ +#define DA7219_SRM_CHECK_RETRIES 8 + +enum da7219_clk_src { + DA7219_CLKSRC_MCLK = 0, + DA7219_CLKSRC_MCLK_SQR, +}; + +enum da7219_sys_clk { + DA7219_SYSCLK_MCLK = 0, + DA7219_SYSCLK_PLL, + DA7219_SYSCLK_PLL_SRM, + DA7219_SYSCLK_PLL_32KHZ +}; + +/* Regulators */ +enum da7219_supplies { + DA7219_SUPPLY_VDD = 0, + DA7219_SUPPLY_VDDMIC, + DA7219_SUPPLY_VDDIO, + DA7219_NUM_SUPPLIES, +}; + +struct da7219_aad_priv; + +/* Private data */ +struct da7219_priv { + struct da7219_aad_priv *aad; + struct da7219_pdata *pdata; + + struct regulator_bulk_data supplies[DA7219_NUM_SUPPLIES]; + struct regmap *regmap; + struct mutex lock; + + struct clk *mclk; + unsigned int mclk_rate; + int clk_src; + + bool master; + bool alc_en; +}; + +#endif /* __DA7219_H */ -- 2.5.0

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[alsa-devel] [PATCH v2 0/4] ASoC: Add support for DA7219 audio codec
by Adam Thomson 02 Oct '15

02 Oct '15

This patch set adds support for the DA7219 audio codec with built-in advanced accessory detection functionality. Patch set includes codec driver, associated DT bindings documentation and MAINTAINERS file updates to cover new bindings. This patch set is baselined against the v4.3-rc2 kernel version. Changes in v2: - Add SOC_DOUBLE_R_EXT support to ASoC core. - Use of upper-case naming for controls, where missing. - Use of Playback/Capture Digital naming for certain ADC/DAC controls. - CP mchange and vol_thresh DT bindings removed, now Kcontrols - Kcontrols and DAPM now locked as part of hptest work, and specific controls affected have had locking introduced. - Timeout for btn_det work when checking micbias status. - Added comments to describe reasoning for work use with hptest, btn_det. - Added comments to tonegen_freq_put/get to clarify endian conversion, and code now uses regmap_raw_read/write calls instead of bulk. - DT parsing code now prints dev_warn() for invalid settings. - SOC_VALUE_ENUM_SINGLE now used for hpf_mode handling, instead of customer get/put handlers. - DTMF key control now a SOC_ENUM to correctly describe supported keys, including non-numeric. - MIXIN_L_MIX_EN bit set as part of DAPM now (SUPPLY), rather than in probe. - Fix dai_sysclk() function to correctly check for no change, using clk_id as well as freq. - Error checking added for clk_set_rate() call. - Improved error checking for devm_clk_get() call. - Removed explicit use of device_may_wakeup() and enable/disable_irq_wake(). Now taken care of by I2C core so DT bindings updated accordingly to document this. - hptest work now uses cache to keep state, instead of reading registers individually. Cache is bypassed, then when test executed and complete, regcache_sync_region() used to restore settings to pre-test values. - Regulator consumer framework usage added, in particular to determine VDDIO level in use so device can be configured accordingly. DT documents updated to reflect this. Adam Thomson (4): ASoC: Add SOC_DOUBLE_R_EXT ASoC: codecs: Add da7219 codec driver ASoC: da7219: Add bindings documentation for DA7219 audio codec MAINTAINERS: da7219: Add entry to cover DA7219 bindings document Documentation/devicetree/bindings/sound/da7219.txt | 106 ++ MAINTAINERS | 1 + include/sound/da7219-aad.h | 99 + include/sound/da7219.h | 55 + include/sound/soc.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 823 +++++++++ sound/soc/codecs/da7219-aad.h | 212 +++ sound/soc/codecs/da7219.c | 1938 ++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 +++++++++ 11 files changed, 4067 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h -- 1.9.3

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[alsa-devel] [PATCH 0/3] ASoC: Add support for DA7219 audio codec
by Adam Thomson 02 Oct '15

02 Oct '15

This patch set adds support for the DA7219 audio codec with built-in advanced accessory detection functionality. Patch set includes codec driver, associated DT bindings documentation and MAINTAINERS file updates to cover new bindings. Adam Thomson (3): ASoC: codecs: Add da7219 codec driver ASoC: da7219: Add bindings documentation for DA7219 audio codec MAINTAINERS: da7219: Add entry to cover DA7219 bindings document Documentation/devicetree/bindings/sound/da7219.txt | 106 ++ MAINTAINERS | 1 + include/sound/da7219-aad.h | 99 ++ include/sound/da7219.h | 76 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 800 +++++++++ sound/soc/codecs/da7219-aad.h | 209 +++ sound/soc/codecs/da7219.c | 1877 ++++++++++++++++++++ sound/soc/codecs/da7219.h | 807 +++++++++ 10 files changed, 3981 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h -- 1.9.3

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[alsa-devel] [PATCH] ASoC: rt5645: Allow 4 channel recording on AIF1
by Ben Zhang 02 Oct '15

02 Oct '15

The codec supports 4 channel recording with TDM on AIF1. This patch modifies the DAI capability to allow it. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index b0d96b6..0d092bf 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3101,7 +3101,7 @@ static struct snd_soc_dai_driver rt5645_dai[] = { .capture = { .stream_name = "AIF1 Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 4, .rates = RT5645_STEREO_RATES, .formats = RT5645_FORMATS, }, -- 2.6.0.rc2.230.g3dd15c0

2 3

[alsa-devel] ATMEL sam9g25 SSC in I2S slave mode on linux 4.1.1 play only right channel
by Paolo Tagliavini 02 Oct '15

02 Oct '15

I'm working on my based sam9g25 design. To have SPDIF output I used cs4265 codec instead of wm8731 as sem9g25 reference design. Now I'm working on porting of cs4265 codec driver. It work for play only right channel, left channel is mute and not supplied by SSC. SSC is configured as I2S slave : SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM Do you know this behaviour ? There is a patch for fix it ? Thank you in advance.

1 0

[alsa-devel] [PATCH 1/1] ASoc: dwc: correct irq clear method
by yitian 02 Oct '15

02 Oct '15

from Designware I2S datasheet, irq is cleared by reading from TOR/ROR registers, rather than by writing into them. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..0d28e3b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -131,10 +131,10 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, TOR(i), 0); + i2s_read_reg(dev->i2s_base, TOR(i)); } else { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, ROR(i), 0); + i2s_read_reg(dev->i2s_base, ROR(i)); } } -- 1.7.12.4

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Re: [alsa-devel] Bug: LADSPA implementation - no up/downmix handling
by Takashi Iwai 02 Oct '15

02 Oct '15

On Fri, 25 Sep 2015 20:51:38 +0200, Andreas Hartmann wrote: > > Am Donnerstag, 24. September 2015, 15:07:14 schrieben Sie: > > On Thu, 24 Sep 2015 12:44:53 +0200, > > > > Andreas Hartmann wrote: > > > Am Mittwoch, 23. September 2015, 16:26:58 schrieb Takashi Iwai: > > > > On Sat, 12 Sep 2015 22:10:40 +0200, > > > > > > > > Andreas Hartmann wrote: > > > > > Hi, > > > > > > > > > > Using the surround_encoder of the swh-plugins I noticed, that the alsa > > > > > implementation only assumes an equal number of in- and output channels > > > > > for > > > > > ladspa plugins. This occurs from Line 752 onwards in pcm_ladspa.c and > > > > > leads to a Segfault shortly after. > > > > > There needs to be a check and correct handling for up/downmix > > > > > scenarios. > > > > > > > > Could you give the back trace? Or at best submit a fix patch? > > > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > _______________________________________________ > > > > Alsa-devel mailing list > > > > Alsa-devel(a)alsa-project.org > > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > > Sure here you go > > > > > > #0 0x00007ffff78623ef in snd_pcm_ladspa_allocate_memory (pcm=0x626fa0, > > > pcm=0x626fa0, pcm=0x626fa0, ladspa=0x621ad0) at pcm_ladspa.c:753 > > > #1 snd_pcm_ladspa_init (pcm=0x626fa0) at pcm_ladspa.c:834 > > > #2 0x00007ffff7842946 in snd_pcm_plugin_prepare (pcm=0x626fa0) at > > > pcm_plugin.c:171 > > > #3 0x00007ffff784290f in snd_pcm_plugin_prepare (pcm=0x62c760) at > > > pcm_plugin.c:162 > > > #4 0x000000000040256a in ?? () > > > #5 0x00007ffff7222ec5 in __libc_start_main (main=0x401d80, argc=4, > > > argv=0x7fffffffde28, init=<optimized out>, fini=<optimized out>, > > > rtld_fini=<optimized out>, > > > > > > stack_end=0x7fffffffde18) at libc-start.c:287 > > > > > > #6 0x0000000000402fdd in ?? () > > > > It looks rather like a simply typo. Does the patch below work? > > > > > > Takashi > > > > diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c > > index 631ee0f35b73..6552b4379ec0 100644 > > --- a/src/pcm/pcm_ladspa.c > > +++ b/src/pcm/pcm_ladspa.c > > @@ -749,7 +749,7 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t > > *pcm, snd_pcm_ladspa_t *lads return -ENOMEM; > > } > > for (idx = 0; idx < instance->input.channels.size; idx++) { > > - chn = instance->output.channels.array[idx]; > > + chn = instance->input.channels.array[idx]; > > if (pchannels[chn] == NULL && chn < ichannels) { > > instance->input.data[idx] = NULL; > > continue; > > It seems to work Sorry, I forgot to apply this to upstream. Now pushed out. thanks, Takashi

1 0

[alsa-devel] [RESEND PATCH 1/1] ASoC: dwc: correct irq clear method
by yitian 02 Oct '15

02 Oct '15

from Designware I2S datasheet, irq is cleared by reading from TOR/ROR registers, rather than by writing into them. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..0d28e3b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -131,10 +131,10 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, TOR(i), 0); + i2s_read_reg(dev->i2s_base, TOR(i)); } else { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, ROR(i), 0); + i2s_read_reg(dev->i2s_base, ROR(i)); } } -- 1.7.12.4

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[alsa-devel] [PATCH 1/1] ASoC: dwc: correct irq clear method
by Yitian Bu 02 Oct '15

02 Oct '15

from Designware I2S datasheet, tx/rx XRUN irq is cleared by reading register TOR/ROR, rather than by writing into them. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..0d28e3b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -131,10 +131,10 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, TOR(i), 0); + i2s_read_reg(dev->i2s_base, TOR(i)); } else { for (i = 0; i < 4; i++) - i2s_write_reg(dev->i2s_base, ROR(i), 0); + i2s_read_reg(dev->i2s_base, ROR(i)); } } -- 1.7.12.4

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[alsa-devel] [PATCH 0/2] CONFIG_OF vs DMA_OF
by Kuninori Morimoto 02 Oct '15

02 Oct '15

Hi Vinod, Mark Current OF base DMAEngine API will be compiled if .config has CONFIG_DMA_OF. but of-dma.h based on CONFIG_OF instead of CONFIG_DMA_OF. These patches tidyup this issue. Please check 1) patch carefully, since DMAEngine API has many dependency to other drivers. > Mark I think 2) patch is based on 1). Please check it Kuninori Morimoto (2): 1) dmaengine: OF DMAEngine API based on CONFIG_DMA_OF instead of CONFIG_OF 2) ASoC: rsnd: SND_SOC_RCAR doesn't depend on DMA_OF include/linux/of_dma.h | 2 +- sound/soc/sh/Kconfig | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] Additional bugs in tlv320aic3x.c driver (register defaults)
by Rick Mann 01 Oct '15

01 Oct '15

Benoît, I've discovered some additional minor bugs in the tlv320aic3104 support in the tlv320aic3x driver. In this instance, the reg_default values aren't quite right. I've written a corrected list, but the set specified includes the reserved registers. I don't know if devm_regmap_init_i2c() attempts to write the default values, or uses them as the initial values in a cache (and attempts to avoid writes unless specific registers change). In my debugging, it does not appear to write the entire register map at any point to the IC, so I think it's the latter. I also don't know how the regmap is intended to be used. It is passed to devm_regmap_init_i2c() as .reg_defaults, implying the array provides starting values, but other parts of the code refer to the array as if the values are changed in that array as they're read and written. Documentation for regmap suggests regmaps can be "sparse," which I presume would be done by omitting the reserved registers from the defaults array. In any case, the fix is not trivial, and I don't want to go down that road not knowing if it's the right approach. Does this make sense? For reference, the registers with different default values (than the ones specified in that file) are 32, 33, 51, 58, 65, 72, 93. Thanks, -- Rick Mann rmann(a)latencyzero.com

2 1

[alsa-devel] Enabling kernel debugging in sound before devices are loaded
by Rick Mann 01 Oct '15

01 Oct '15

I'm trying to enable kernel debug logging like this: # echo "file sound* +pflm" > /sys/kernel/debug/dynamic_debug/control But it has no effect until after I load the Device Tree Overlay that creates my codec and sound card nodes. Unfortunately at this point, it's too late for the debugging to do any good. This is with kernel 4.1.9 on a Beaglebone Black Debian Jessie distro. I don't really know how the kernel dyndebug system works; I am able to enable other debugging locations, but none in sound/* Appreciate any guidance. Thank you. -- Rick Mann rmann(a)latencyzero.com

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Correct the naming and setting of ADC Boost Volume Control
by Oder Chiou 01 Oct '15

01 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 +++--- sound/soc/codecs/rt5645.h | 16 +++++++++------- 2 files changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3eb184c..080cc1c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -519,11 +519,11 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { RT5645_L_VOL_SFT + 1, RT5645_R_VOL_SFT + 1, 63, 0, adc_vol_tlv), /* ADC Boost Volume Control */ - SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1, + SOC_DOUBLE_TLV("ADC Boost Capture Volume", RT5645_ADC_BST_VOL1, RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1, - RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0, + SOC_DOUBLE_TLV("Mono ADC Boost Capture Volume", RT5645_ADC_BST_VOL2, + RT5645_MONO_ADC_L_BST_SFT, RT5645_MONO_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), /* I2S2 function select */ diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index f45861c..61bc8ab 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -39,8 +39,8 @@ #define RT5645_STO1_ADC_DIG_VOL 0x1c #define RT5645_MONO_ADC_DIG_VOL 0x1d #define RT5645_ADC_BST_VOL1 0x1e -/* Mixer - D-D */ #define RT5645_ADC_BST_VOL2 0x20 +/* Mixer - D-D */ #define RT5645_STO1_ADC_MIXER 0x27 #define RT5645_MONO_ADC_MIXER 0x28 #define RT5645_AD_DA_MIXER 0x29 @@ -315,12 +315,14 @@ #define RT5645_STO1_ADC_R_BST_SFT 12 #define RT5645_STO1_ADC_COMP_MASK (0x3 << 10) #define RT5645_STO1_ADC_COMP_SFT 10 -#define RT5645_STO2_ADC_L_BST_MASK (0x3 << 8) -#define RT5645_STO2_ADC_L_BST_SFT 8 -#define RT5645_STO2_ADC_R_BST_MASK (0x3 << 6) -#define RT5645_STO2_ADC_R_BST_SFT 6 -#define RT5645_STO2_ADC_COMP_MASK (0x3 << 4) -#define RT5645_STO2_ADC_COMP_SFT 4 + +/* ADC Boost Volume Control (0x20) */ +#define RT5645_MONO_ADC_L_BST_MASK (0x3 << 14) +#define RT5645_MONO_ADC_L_BST_SFT 14 +#define RT5645_MONO_ADC_R_BST_MASK (0x3 << 12) +#define RT5645_MONO_ADC_R_BST_SFT 12 +#define RT5645_MONO_ADC_COMP_MASK (0x3 << 10) +#define RT5645_MONO_ADC_COMP_SFT 10 /* Stereo2 ADC Mixer Control (0x26) */ #define RT5645_STO2_ADC_SRC_MASK (0x1 << 15) -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] Prevent writing reserved registers on tlv320aic3104 CODECs
by Rick Mann 01 Oct '15

01 Oct '15

The current code writes a set of registers that are reserved on the tlc320aic3104. The change skips those registers for that IC. Signed-off-by: Rick Mann <rmann(a)latencyzero.com> --- sound/soc/codecs/tlv320aic3x.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f1c9fffcd3a6..8d893e94e78d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1531,14 +1531,17 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); - /* Line2 Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + /* On tlv320aic3104, these registers are reserved and must not be written */ + if (aic3x->model != AIC3X_MODEL_3104) { + /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + /* Line2 Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + } switch (aic3x->model) { case AIC3X_MODEL_3X: -- 1.9.1

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[alsa-devel] Driver for audio power amplifier ICs?
by Rick Mann 01 Oct '15

01 Oct '15

I've been working on this Beaglebone Black custom audio cape. It has a tlv320aic3104 codec (that I'm still trying to get working properly), and connected to its line outs is a tpa3140d2 analog 2x10W power amplifier. This chip has very little in the way of CPU interface, but it does have /shutdown and /fault lines. I'm wondering how ALSA becomes aware of that chip, in order to enable it when it enables certain routes in the tlv320aic3104, and to inform application software if the /fault line asserts. I don't mind writing a driver, if necessary, I just don't know where such a thing would fit. I also wonder what the device tree entries for the amp might look like. Thanks! -- Rick Mann rmann(a)latencyzero.com

2 3

[alsa-devel] aic31xx_set_bias_level not called
by Matthias Weißer 01 Oct '15

01 Oct '15

Hi I currently try to port an existing alsa driver to a new hardware. The only difference between these to board is the used codec. Where the original board (iMX6SX SDB from Freescale) uses a wm8962 our custom board uses a tlv320aic3110. So I took the file sound\soc\fsl\imx-wm8962.c and changed it according to the new codec. From what I understand the main things where just renaming. I then setup my device tree according to the new environment and the kernel seemed to setup stuff correctly. But when I try to play some sound using speaker-test I see some errors and don't hear anything. Tracking stuff down using dev_dbg outputs I see that aic31xx_set_bias_level is never called with a value > 1 which causes that the clocks of the codec are never enabled. Maybe someone can give me some hints on what can be wrong. Here are some things which might be useful: ~ # dmesg | grep -i -A 2 ALSA [ 5.058905] ALSA device list: [ 5.061885] #0: tlv320aic31xx-audio ~ # dmesg | grep tlv [ 1.598495] tlv320aic31xx-codec 2-0018: ## aic31xx_i2c_probe: tlv320aic3110 codec_type = 1 [ 1.613239] tlv320aic31xx-codec 2-0018: ## aic31xx_codec_probe [ 1.621895] tlv320aic31xx-codec 2-0018: ## aic31xx_set_dai_fmt: fmt = 0x1001 [ 1.630390] imx-tlv320aic31xx sound: tlv320aic31xx-hifi <-> 202c000.ssi mapping ok [ 1.638145] tlv320aic31xx-codec 2-0018: ## aic31xx_set_dai_sysclk: clk_id = 0, freq = 24000000, dir = 0 [ 1.663566] tlv320aic31xx-codec 2-0018: ## aic31xx_set_bias_level: 0 -> 1 [ 5.061885] #0: tlv320aic31xx-audio [ 36.860404] tlv320aic31xx-codec 2-0018: ## aic31xx_hw_params: width 16 rate 48000 [ 36.886777] tlv320aic31xx-codec 2-0018: pll 8.1920/2 dosr 128 n 8 m 2 aosr 128 n 8 m 2 bclk_n 16 # aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=tlv320aic31xxau tlv320aic31xx-audio, Default Audio Device sysdefault:CARD=tlv320aic31xxau tlv320aic31xx-audio, Default Audio Device # speaker-test -Ddefault speaker-test 1.0.29 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 32768 Period size range from 64 to 16384 Using max buffer size 32768 Periods = 4 was set period_size = 8192 was set buffer_size = 32768 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error # uname -a Linux imx6sx-g343-devel 4.1.4-dirty #49 Thu Oct 1 10:39:44 CEST 2015 armv7l GNU/Linux /* * Copyright 2013 Freescale Semiconductor, Inc. * * Based on imx-wm8962.c * Copyright 2012 Freescale Semiconductor, Inc. * Copyright 2012 Linaro Ltd. * * The code contained herein is licensed under the GNU General Public * License. You may obtain a copy of the GNU General Public License * Version 2 or later at the following locations: * * http://www.opensource.org/licenses/gpl-license.html * http://www.gnu.org/copyleft/gpl.html */ #define DEBUG #include <linux/module.h> #include <linux/of_platform.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/clk.h> #include <sound/soc.h> #include <sound/pcm_params.h> #include <sound/soc-dapm.h> #include <linux/pinctrl/consumer.h> #include "../codecs/tlv320aic31xx.h" #include "imx-audmux.h" #define DAI_NAME_SIZE 32 struct imx_tlv320aic31xx_data { struct snd_soc_dai_link dai; struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; struct clk *codec_clk; unsigned int clk_frequency; }; struct imx_priv { struct platform_device *pdev; }; static struct imx_priv card_priv; static const struct snd_soc_dapm_widget imx_tlv320aic31xx_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("AMIC", NULL), SND_SOC_DAPM_MIC("DMIC", NULL), }; static int sample_rate = 44100; static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { sample_rate = params_rate(params); sample_format = params_format(params); return 0; } static struct snd_soc_ops imx_hifi_ops = { .hw_params = imx_hifi_hw_params, }; static int imx_tlv320aic31xx_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; struct imx_tlv320aic31xx_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret = 0; ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->clk_frequency, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(dev, "failed to set sysclk in %s\n", __func__); return ret; } static int imx_tlv320aic31xx_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct imx_priv *priv = &card_priv; struct i2c_client *codec_dev; struct imx_tlv320aic31xx_data *data; int int_port, ext_port; int ret; priv->pdev = pdev; ret = of_property_read_u32(np, "mux-int-port", &int_port); if (ret) { dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); return ret; } ret = of_property_read_u32(np, "mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); return ret; } /* * The port numbering in the hardware manual starts at 1, while * the audmux API expects it starts at 0. */ int_port--; ext_port--; ret = imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | IMX_AUDMUX_V2_PTCR_TFSDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR, IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); if (ret) { dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { dev_err(&pdev->dev, "audmux external port setup failed\n"); return ret; } ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); if (!ssi_np || !codec_np) { dev_err(&pdev->dev, "phandle missing or invalid\n"); ret = -EINVAL; goto fail; } ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); ret = -EINVAL; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev || !codec_dev->dev.driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); ret = -EINVAL; goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { ret = -ENOMEM; goto fail; } data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); goto fail; } data->clk_frequency = clk_get_rate(data->codec_clk); ret = clk_prepare_enable(data->codec_clk); if (ret) { dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); goto fail; } data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; data->dai.codec_dai_name = "tlv320aic31xx-hifi"; data->dai.codec_of_node = codec_np; data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) goto clk_fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) goto clk_fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; data->card.dapm_widgets = imx_tlv320aic31xx_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_tlv320aic31xx_dapm_widgets); data->card.late_probe = imx_tlv320aic31xx_late_probe; 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 clk_fail; } of_node_put(ssi_np); of_node_put(codec_np); return 0; clk_fail: clk_disable_unprepare(data->codec_clk); fail: of_node_put(ssi_np); of_node_put(codec_np); return ret; } static int imx_tlv320aic31xx_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct imx_tlv320aic31xx_data *data = snd_soc_card_get_drvdata(card); if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); return 0; } static const struct of_device_id imx_tlv320aic31xx_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, imx_tlv320aic31xx_dt_ids); static struct platform_driver imx_tlv320aic31xx_driver = { .driver = { .name = "imx-tlv320aic31xx", .pm = &snd_soc_pm_ops, .of_match_table = imx_tlv320aic31xx_dt_ids, }, .probe = imx_tlv320aic31xx_probe, .remove = imx_tlv320aic31xx_remove, }; module_platform_driver(imx_tlv320aic31xx_driver); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("Freescale i.MX tlv320aic31xx ASoC machine driver"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:imx-tlv320aic31xx");

1 0

[alsa-devel] [PATCH] ASoC: Document DAI signal polarity
by Anatol Pomozov 01 Oct '15

01 Oct '15

Per discussion at [1] currently there is no clear definition of what is FSYNC polarity. Different drivers use its own definition of what is "normal" and what is "inverted" fsync in different modes. This leads to compatibility problems between drivers. Explicitly specify meaning of BCLK/FSYNC polarity. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/097121.html Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- include/sound/soc-dai.h | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..c8170c4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,15 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * - * Specifies whether the DAI can also support inverted clocks for the specified - * format. + * For BCLK: + * - "normal" polarity means signal sensing happens at rising edge of BCLK + * - "inverted" polarity means signal sensing happens at falling edge of BCLK + * + * For FSYNC: + * - "normal" polarity means frame starts at rising edge of FSYNC + * - "inverted" polarity means frame starts at falling edge of FSYNC */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- 2.5.0.457.gab17608

6 10

[alsa-devel] [PATCH] ASoC: pxa: pxa-pcm-lib: switch over to snd-soc-dmaengine-pcm
by Robert Jarzmik 01 Oct '15

01 Oct '15

From: Daniel Mack <zonque(a)gmail.com> This patch removes the old PXA DMA API usage and switches over to generic functions provided by snd-soc-dmaengine-pcm. More cleanups may be done on top of this, and some function stubs can now be removed completetly. However, the intention here was to keep the transition as small as possible. This was tested on the mioa701 pxa27x board. Signed-off-by: Daniel Mack <zonque(a)gmail.com> [trivial change from mmp-dma to pxa-dma] Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- include/sound/pxa2xx-lib.h | 1 - sound/arm/pxa2xx-ac97.c | 13 ++- sound/arm/pxa2xx-pcm-lib.c | 201 ++++++++------------------------------------ sound/arm/pxa2xx-pcm.c | 12 +-- sound/arm/pxa2xx-pcm.h | 2 - sound/soc/pxa/pxa2xx-ac97.c | 49 +++++++---- sound/soc/pxa/pxa2xx-i2s.c | 3 + sound/soc/pxa/pxa2xx-pcm.c | 32 ------- 8 files changed, 85 insertions(+), 228 deletions(-) diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 56e818e4a1cb..6ef629bde164 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -12,7 +12,6 @@ extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd); extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream); extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); -extern void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id); extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream); extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 38590b322c54..fbd5dad0c484 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,7 +44,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_reset, }; -static unsigned long pxa2xx_ac97_pcm_out_req = 12; +static struct pxad_param pxa2xx_ac97_pcm_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 12, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -51,7 +56,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .filter_data = &pxa2xx_ac97_pcm_out_req, }; -static unsigned long pxa2xx_ac97_pcm_in_req = 11; +static struct pxad_param pxa2xx_ac97_pcm_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 11, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 01f8fdc42b1b..e9b98af6b52c 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/dma-mapping.h> #include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <sound/core.h> #include <sound/pcm.h> @@ -15,8 +16,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include <mach/dma.h> - #include "pxa2xx-pcm.h" static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { @@ -31,7 +30,7 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .period_bytes_min = 32, .period_bytes_max = 8192 - 32, .periods_min = 1, - .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc), + .periods_max = 256, .buffer_bytes_max = 128 * 1024, .fifo_size = 32, }; @@ -39,65 +38,29 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd = runtime->private_data; - size_t totsize = params_buffer_bytes(params); - size_t period = params_period_bytes(params); - pxa_dma_desc *dma_desc; - dma_addr_t dma_buff_phys, next_desc_phys; - u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_dmaengine_dai_dma_data *dma_params; + struct dma_slave_config config; + int ret; - /* temporary transition hack */ - switch (rtd->params->addr_width) { - case DMA_SLAVE_BUSWIDTH_1_BYTE: - dcmd |= DCMD_WIDTH1; - break; - case DMA_SLAVE_BUSWIDTH_2_BYTES: - dcmd |= DCMD_WIDTH2; - break; - case DMA_SLAVE_BUSWIDTH_4_BYTES: - dcmd |= DCMD_WIDTH4; - break; - default: - /* can't happen */ - break; - } + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; - switch (rtd->params->maxburst) { - case 8: - dcmd |= DCMD_BURST8; - break; - case 16: - dcmd |= DCMD_BURST16; - break; - case 32: - dcmd |= DCMD_BURST32; - break; - } + ret = snd_hwparams_to_dma_slave_config(substream, params, &config); + if (ret) + return ret; - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = totsize; + snd_dmaengine_pcm_set_config_from_dai_data(substream, + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream), + &config); - dma_desc = rtd->dma_desc_array; - next_desc_phys = rtd->dma_desc_array_phys; - dma_buff_phys = runtime->dma_addr; - do { - next_desc_phys += sizeof(pxa_dma_desc); - dma_desc->ddadr = next_desc_phys; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dma_desc->dsadr = dma_buff_phys; - dma_desc->dtadr = rtd->params->addr; - } else { - dma_desc->dsadr = rtd->params->addr; - dma_desc->dtadr = dma_buff_phys; - } - if (period > totsize) - period = totsize; - dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN; - dma_desc++; - dma_buff_phys += period; - } while (totsize -= period); - dma_desc[-1].ddadr = rtd->dma_desc_array_phys; + ret = dmaengine_slave_config(chan, &config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; } @@ -105,13 +68,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_params); int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - - if (rtd && rtd->params && rtd->params->filter_data) { - unsigned long req = *(unsigned long *) rtd->params->filter_data; - DRCMR(req) = 0; - } - snd_pcm_set_runtime_buffer(substream, NULL); return 0; } @@ -119,100 +75,36 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_free); int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; - DCSR(prtd->dma_ch) = DCSR_RUN; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - DCSR(prtd->dma_ch) &= ~DCSR_RUN; - break; - - case SNDRV_PCM_TRIGGER_RESUME: - DCSR(prtd->dma_ch) |= DCSR_RUN; - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; - DCSR(prtd->dma_ch) |= DCSR_RUN; - break; - - default: - ret = -EINVAL; - } - - return ret; + return snd_dmaengine_pcm_trigger(substream, cmd); } EXPORT_SYMBOL(pxa2xx_pcm_trigger); snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *prtd = runtime->private_data; - - dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch); - snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr); - - if (x == runtime->buffer_size) - x = 0; - return x; + return snd_dmaengine_pcm_pointer(substream); } EXPORT_SYMBOL(pxa2xx_pcm_pointer); int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - unsigned long req; - - if (!prtd || !prtd->params) - return 0; - - if (prtd->dma_ch == -1) - return -EINVAL; - - DCSR(prtd->dma_ch) &= ~DCSR_RUN; - DCSR(prtd->dma_ch) = 0; - DCMD(prtd->dma_ch) = 0; - req = *(unsigned long *) prtd->params->filter_data; - DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD; - return 0; } EXPORT_SYMBOL(__pxa2xx_pcm_prepare); -void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) -{ - struct snd_pcm_substream *substream = dev_id; - int dcsr; - - dcsr = DCSR(dma_ch); - DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN; - - if (dcsr & DCSR_ENDINTR) { - snd_pcm_period_elapsed(substream); - } else { - printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n", - dma_ch, dcsr); - snd_pcm_stop_xrun(substream); - } -} -EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); - int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd; + struct snd_dmaengine_dai_dma_data *dma_params; int ret; runtime->hw = pxa2xx_pcm_hardware; + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; + /* * For mysterious reasons (and despite what the manual says) * playback samples are lost if the DMA count is not a multiple @@ -221,48 +113,27 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (ret) - goto out; + return ret; ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); if (ret) - goto out; + return ret; ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) - goto out; - - ret = -ENOMEM; - rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); - if (!rtd) - goto out; - rtd->dma_desc_array = - dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE, - &rtd->dma_desc_array_phys, GFP_KERNEL); - if (!rtd->dma_desc_array) - goto err1; + return ret; - rtd->dma_ch = -1; - runtime->private_data = rtd; - return 0; - - err1: - kfree(rtd); - out: - return ret; + return snd_dmaengine_pcm_open_request_chan(substream, + pxad_filter_fn, + dma_params->filter_data); } EXPORT_SYMBOL(__pxa2xx_pcm_open); int __pxa2xx_pcm_close(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd = runtime->private_data; - - dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, - rtd->dma_desc_array, rtd->dma_desc_array_phys); - kfree(rtd); - return 0; + return snd_dmaengine_pcm_close_release_chan(substream); } EXPORT_SYMBOL(__pxa2xx_pcm_close); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 83be8e3f095e..83fcfac97739 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -46,17 +46,13 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? client->playback_params : client->capture_params; - ret = pxa_request_dma("dma", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - goto err2; - rtd->dma_ch = ret; ret = client->startup(substream); if (!ret) - goto out; + goto err2; + + return 0; - pxa_free_dma(rtd->dma_ch); err2: __pxa2xx_pcm_close(substream); out: @@ -66,9 +62,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) { struct pxa2xx_pcm_client *client = substream->private_data; - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - pxa_free_dma(rtd->dma_ch); client->shutdown(substream); return __pxa2xx_pcm_close(substream); diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h index 00330985beec..8fa2b7c9e6b8 100644 --- a/sound/arm/pxa2xx-pcm.h +++ b/sound/arm/pxa2xx-pcm.h @@ -13,8 +13,6 @@ struct pxa2xx_runtime_data { int dma_ch; struct snd_dmaengine_dai_dma_data *params; - struct pxa_dma_desc *dma_desc_array; - dma_addr_t dma_desc_array_phys; }; struct pxa2xx_pcm_client { diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9e4b04e0fbd1..f3de615aacd7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <sound/core.h> #include <sound/ac97_codec.h> @@ -49,7 +50,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11; +static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 11, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -57,7 +62,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, }; -static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12; +static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 12, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -65,7 +74,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10; +static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 10, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -73,7 +85,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9; +static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 9, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -81,7 +96,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8; +static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 8, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -89,9 +107,8 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; -static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -105,9 +122,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_aux_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -121,9 +137,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -139,15 +154,15 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_48000) static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = { - .hw_params = pxa2xx_ac97_hw_params, + .startup = pxa2xx_ac97_hifi_startup, }; static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = { - .hw_params = pxa2xx_ac97_hw_aux_params, + .startup = pxa2xx_ac97_aux_startup, }; static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { - .hw_params = pxa2xx_ac97_hw_mic_params, + .startup = pxa2xx_ac97_mic_startup, }; /* diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6b4e40036910..0389cf7b4b1e 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -319,6 +319,9 @@ static int pxa2xx_i2s_probe(struct snd_soc_dai *dai) /* Along with FIFO servicing */ SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + snd_soc_dai_init_dma_data(dai, &pxa2xx_i2s_pcm_stereo_out, + &pxa2xx_i2s_pcm_stereo_in); + return 0; } diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 831ee37d2e3e..76ba0e55870d 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -15,8 +15,6 @@ #include <linux/dmaengine.h> #include <linux/of.h> -#include <mach/dma.h> - #include <sound/core.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> @@ -27,11 +25,8 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_dmaengine_dai_dma_data *dma; - int ret; dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); @@ -40,40 +35,13 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, if (!dma) return 0; - /* this may get called several times by oss emulation - * with different params */ - if (prtd->params == NULL) { - prtd->params = dma; - ret = pxa_request_dma("name", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - return ret; - prtd->dma_ch = ret; - } else if (prtd->params != dma) { - pxa_free_dma(prtd->dma_ch); - prtd->params = dma; - ret = pxa_request_dma("name", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - return ret; - prtd->dma_ch = ret; - } - return __pxa2xx_pcm_hw_params(substream, params); } static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - __pxa2xx_pcm_hw_free(substream); - if (prtd->dma_ch >= 0) { - pxa_free_dma(prtd->dma_ch); - prtd->dma_ch = -1; - prtd->params = NULL; - } - return 0; } -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: imx-ssi: Fix DAI hardware signal inversions
by Benoît Thébaudeau 30 Sep '15

30 Sep '15

SND_SOC_DAIFMT_{IB|NB}_{IF|NF} are defined as inverting or not BCLK or FRM relatively to what is standard for the specified DAI hardware audio format. Consequently, the absolute polarities of these signals cannot be derived only from these settings as this driver did. The format has to be taken into account too. This fixes inverted left/right channels in I²S mode. Signed-off-by: Benoît Thébaudeau <benoit(a)wsystem.com> --- sound/soc/fsl/imx-ssi.c | 19 +++++++++---------- 1 file changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 48b2d24..b95132e 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -95,7 +95,8 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: /* data on rising edge of bclk, frame low 1clk before data */ - strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSI | + SSI_STCR_TEFS; scr |= SSI_SCR_NET; if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) { scr &= ~SSI_I2S_MODE_MASK; @@ -104,33 +105,31 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) break; case SND_SOC_DAIFMT_LEFT_J: /* data on rising edge of bclk, frame high with data */ - strcr |= SSI_STCR_TXBIT0; + strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP; break; case SND_SOC_DAIFMT_DSP_B: /* data on rising edge of bclk, frame high with data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0; + strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL; break; case SND_SOC_DAIFMT_DSP_A: /* data on rising edge of bclk, frame high 1clk before data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; + strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL | + SSI_STCR_TEFS; break; } /* DAI clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_IB_IF: - strcr |= SSI_STCR_TFSI; - strcr &= ~SSI_STCR_TSCKP; + strcr ^= SSI_STCR_TSCKP | SSI_STCR_TFSI; break; case SND_SOC_DAIFMT_IB_NF: - strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + strcr ^= SSI_STCR_TSCKP; break; case SND_SOC_DAIFMT_NB_IF: - strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + strcr ^= SSI_STCR_TFSI; break; case SND_SOC_DAIFMT_NB_NF: - strcr &= ~SSI_STCR_TFSI; - strcr |= SSI_STCR_TSCKP; break; } -- 2.1.4

5 6

[alsa-devel] [PATCH] ASoC: Codec: wm8962: balance pm_runtime_enable
by Jiada Wang 30 Sep '15

30 Sep '15

pm_runtime_enable is called in probe to enable runtime PM for wm8962 codec, but pm_runtime_disable isn't called in remove callback, nor is called in error path if probe fails after runtime PM is enabled, this causes unbalanced pm_runtime_enable. This patch Adds pm_runtime_disable in remove callback and error path, to balance pm_runtime_enable. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/codecs/wm8962.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 293e47a..2fbc6ef 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3760,7 +3760,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8962, &wm8962_dai, 1); if (ret < 0) - goto err_enable; + goto err_pm_runtime; regcache_cache_only(wm8962->regmap, true); @@ -3769,6 +3769,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, return 0; +err_pm_runtime: + pm_runtime_disable(&i2c->dev); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); err: @@ -3778,6 +3780,7 @@ err: static int wm8962_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); + pm_runtime_disable(&client->dev); return 0; } -- 2.4.5

3 3

[alsa-devel] [PATCH 1/2] ASoC: sun4i-codec: pass through clk_set_rate error
by Maxime Ripard 30 Sep '15

30 Sep '15

The current code, disregarding the clk_set_rate error code, was always returning -EINVAL. Fix that and return the code in order to have more clue about what's going on. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/soc/sunxi/sun4i-codec.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 6e83e62ef039..28834129bea3 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -283,7 +283,7 @@ static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); unsigned long clk_freq; - int hwrate; + int ret, hwrate; u32 val; if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) @@ -293,8 +293,9 @@ static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, if (!clk_freq) return -EINVAL; - if (clk_set_rate(scodec->clk_module, clk_freq)) - return -EINVAL; + ret = clk_set_rate(scodec->clk_module, clk_freq); + if (ret) + return ret; hwrate = sun4i_codec_get_hw_rate(params); if (hwrate < 0) -- 2.5.3

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[alsa-devel] SoC/Codecs/SGTL5000 - Non working MIC_BIAS Voltage settings
by gianluca 30 Sep '15

30 Sep '15

Hello, in attachment there is a small patch to ensure the correct register value for MIC BIAS Voltage. Best regards -- Eurek s.r.l. | Electronic Engineering | http://www.eurek.it via Celletta 8/B, 40026 Imola, Italy | Phone: +39-(0)542-609120 p.iva 00690621206 - c.f. 04020030377 | Fax: +39-(0)542-609212

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[alsa-devel] Found bugs in tlv320aic3x.c driver, how do I report them?
by Rick Mann 30 Sep '15

30 Sep '15

Hi. I came across some bugs in sound/soc/codecs/tlv320aic3x.c, wherein it writes non-zero values to reserved registers on the tlv320aic3104 (I found it in 4.1.4, but it's still in 4.3-rc3). I have an untested proposed patch. But I don't know anything about how to report it or offer it. What should I do? Thanks. For reference: $ git diff -- sound/soc/codecs/tlv320aic3x.c diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 51c4713..79c0ca0 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1509,14 +1509,24 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); - /* Line2 Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + if (aic3x->model == AIC3X_MODEL_3104) { + /* On tlv320aic3014, these registers are reserved and must be written 0 */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, 0); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, 0); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, 0); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, 0); + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, 0); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, 0); + } else { + /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + /* Line2 Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + } switch (aic3x->model) { case AIC3X_MODEL_3X: -- Rick Mann rmann(a)latencyzero.com

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[alsa-devel] [PATCH 0/2] ASoC: Add support for the Allwinner A10 codec
by Maxime Ripard 30 Sep '15

30 Sep '15

Hi everyone, This patch set adds the support for what Allwinner calls the codec on their SoCs. This codec is actually a combination of a codec and DAI, tied together in a single memory-mapped IP. It is completely standalone, and outputs directly the analog signal. While it supports both playback and capture, the capture is not implemented in this patch, and will be posted eventually as a separate one. This set, in order to be functional, has a dependency on the audio clocks patch set posted separately. However, it doesn't needs this to compile properly, so I guess it can be merged without really caring for the merging status of the clock patches. Let me know what you think, Maxime Emilio López (1): ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs Maxime Ripard (1): Documentation: bindings: Add the Allwinner A10 codec bindings .../devicetree/bindings/sound/sun4i-codec.txt | 33 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-codec.c | 720 +++++++++++++++++++++ 6 files changed, 767 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt create mode 100644 sound/soc/sunxi/Kconfig create mode 100644 sound/soc/sunxi/Makefile create mode 100644 sound/soc/sunxi/sun4i-codec.c -- 2.5.1

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[alsa-devel] [PATCH 0/7 v3] firewire-digi00x: new driver for Digidesign 002/003 family
by Takashi Sakamoto 30 Sep '15

30 Sep '15

This patchset updates a part of my previous post: [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html This patchset adds support for Digidesign Digi 002/003 family without MIDI functionality. Although most part of AMDTP stream functionality is clear, a mechanism of clock synchronization with internal clock source is not clear (see the last patch). Takashi Sakamoto (7): ALSA: firewire-digi00x: add skeleton for Digi 002/003 family ALSA: firewire-digi00x: add data block processing layer ALSA: firewire-digi00x: add stream functionality ALSA: firewire-digi00x: add proc node to show clock status ALSA: firewire-digi00x: add PCM functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 8 + sound/firewire/Kconfig | 15 + sound/firewire/Makefile | 1 + sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 330 +++++++++++++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 373 ++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 +++++++ sound/firewire/digi00x/digi00x-stream.c | 420 +++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 81 ++++++ sound/firewire/digi00x/digi00x.c | 169 +++++++++++ sound/firewire/digi00x/digi00x.h | 146 ++++++++++ 13 files changed, 1848 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/amdtp-dot.c create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x-transaction.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h -- 2.1.4

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[alsa-devel] [PATCH] drm/i915: Add missing const to audio_rate_need_prog()
by Takashi Iwai 30 Sep '15

30 Sep '15

The lack of const leads to a compile warning after merging i915 upstream tree: drivers/gpu/drm/i915/intel_audio.c:147:13: note: expected 'struct drm_display_mode *' but argument is of type 'const struct drm_display_mode *' Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/drm/i915/intel_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index 30f6859dcb36..dffd0b4c5f17 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -144,7 +144,7 @@ static uint32_t audio_config_setup_n_reg(int n, uint32_t val) /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, - struct drm_display_mode *mode) + const struct drm_display_mode *mode) { if (((mode->clock == TMDS_297M) || (mode->clock == TMDS_296M)) && -- 2.5.3

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[alsa-devel] [PATCH RFC 0/7] ASoC: tlv320aic26: Add device tree support and bug fixes
by Cormier, Jonathan 30 Sep '15

30 Sep '15

Hi, This is my first patch so I'm including the RFC to help identify if I've missed something. The following patches were created to get the tlv320aic26 working on our AM335x SoM. This patch series relies on commit 93d0ad8f374c "ASoC: tlv320aic26: Convert to params_width()" for proper operation. Thanks -Jonathan Cormier, Jonathan (7): ASoC: tlv320aic26: Add device tree binding ASoC: tlv320aic26: Fix module autoload ASoC: tlv320aic26: Make SND_SOC_TLV320AIC26 user-visible ASoC: tlv320aic26: Fix regmap by setting reg_defaults and reg_stride ASoC: tlv320aic26: hw_params was unintentionally clearing AIC26 master mode ASoC: tlv320aic26: Add support for DSP_B data format ASoC: tlv320aic26: Change Capture Mute to Capture Switch to match alsa documentation. .../devicetree/bindings/sound/tlv320aic26.txt | 65 ++++++++++++++++++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/tlv320aic26.c | 60 ++++++++++++++++++-- 3 files changed, 120 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic26.txt -- 1.9.1

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[alsa-devel] speaker-test error using sgtl5000 and imx27
by Alexandre Schnegg 30 Sep '15

30 Sep '15

Hello, I would like to use the Freescale SGTl5000 audio codec with the Freescale imx27 processor. Unfortunately, I get this error when I try to run < speaker-test >: # speaker-test speaker-test 1.0.28 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 32768 Period size range from 64 to 16384 Using max buffer size 32768 Periods = 4 was set period_size = 8192 was set buffer_size = 32768 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error The codec is connected to SSI2 pins and I would like to use the SSI0 internal port in asynchronous mode (4 wires). The command interface is I2C1. The codec seems to work because it generates both bit and frame clocks (the codec is the master) but no data are coming from the processor. I'm using kernel 4.1.7 but I have got the same error with versions 3.19.8, 4.0.5 and 4.0.9. I have also tried the < imx-sgtl5000 > driver and I get the same error. I can use < alsamixer > with no issue. Here are my device tree bindings: sound { compatible = "simple-audio-card"; simple-audio-card,name = "sgtl5000-audio"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&dailink0_master>; simple-audio-card,frame-master = <&dailink0_master>; simple-audio-card,widgets = "Microphone", "Mic Jack", "Headphone", "Headphone Jack"; simple-audio-card,routing = "MIC_IN", "Mic Jack", "Mic Jack", "Mic Bias", "Headphone Jack", "HP_OUT"; simple-audio-card,cpu { sound-dai = <&ssi1 0>; }; dailink0_master: simple-audio-card,codec { sound-dai = <&sgtl5000>; clocks = <&mclk>; }; }; mclk: asvmpc { #clock-cells = <0>; compatible = "fixed-clock"; clock-frequency = <12288000>; }; &ssi1 { #sound-dai-cells = <1>; status = "okay"; fsl,audmux-port = <MX27_AUDMUX_PPCR2_SSI_PINS_2>; fsl,port-config = < (IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)) >; }; &audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_ssi2>; status = "okay"; fsl,audmux-port = <MX27_AUDMUX_HPCR1_SSI0>; fsl,port-config = < (IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR2_SSI_PINS_2) | IMX_AUDMUX_V1_PCR_TFSDIR | IMX_AUDMUX_V1_PCR_TCLKDIR | IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR2_SSI_PINS_2)) >; }; &i2c1 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c1>; status = "okay"; sgtl5000: sgtl5000@0a { #sound-dai-cells = <0>; compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&mclk>; micbias-resistor-k-ohms = <2>; micbias-voltage-m-volts = <2250>; VDDA-supply = <&reg_2p8v>; VDDIO-supply = <&reg_2p8v>; }; }; pinctrl_ssi2: ssi2grp { fsl,pins = < MX27_PAD_SSI2_FS__SSI2_FS 0x0 MX27_PAD_SSI2_RXDAT__SSI2_RXDAT 0x0 MX27_PAD_SSI2_TXDAT__SSI2_TXDAT 0x0 MX27_PAD_SSI2_CLK__SSI2_CLK 0x0 >; }; And the modules I have loaded before running < speaker test > : # modprobe snd-soc-sgtl5000 [ 23.274351] sgtl5000 0-000a: sgtl5000 revision 0x11 # modprobe snd-soc-simple-card # modprobe snd-soc-fsl-ssi [ 79.752288] sgtl5000 0-000a: Using internal LDO instead of VDDD # [ 79.853711] asoc-simple-card sound: sgtl5000 <-> 10010000.ssi mapping ok # modprobe snd-soc-imx-audmux Does anybody know what is the mistake? Thank you for your help. Best regards. Alexandre Schnegg

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[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: fix wrong register MIC_BIAS_VOLTAGE setup on probe
by Gianluca Renzi 29 Sep '15

29 Sep '15

Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Gianluca Renzi <gianlucarenzi(a)eurekelettronica.it> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- a/sound/soc/codecs/sgtl5000.c 2015-09-25 15:24:10.597051191 +0200 +++ b/sound/soc/codecs/sgtl5000.c 2015-09-25 21:26:07.102020194 +0200 @@ -1376,8 +1376,8 @@ static int sgtl5000_probe(struct snd_soc sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_MASK, - sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT); + SGTL5000_BIAS_VOLT_MASK, + sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT); /* * disable DAP * TODO:

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[alsa-devel] [PATCH alsa-utils] alsactl: Manage both save and restore in a single unit
by Takashi Iwai 29 Sep '15

29 Sep '15

With RemainAfterExit=true, we can manage both save and restore of the card state in a single unit file. This will fix also the case where systemd reloads the service; with two individual units, it will restore the previous state before saving, and may lead to inconsistent state suddenly. Also fix alsa-state.service as well to make both start and stop working in a simpler way. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=929619 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/Makefile.am | 14 ++------------ alsactl/alsa-restore.service.in | 7 +++---- alsactl/alsa-state.service.in | 3 --- alsactl/alsa-store.service.in | 16 ---------------- 4 files changed, 5 insertions(+), 35 deletions(-) delete mode 100644 alsactl/alsa-store.service.in diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index 47f06e906ced..cac8094687de 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -25,20 +25,15 @@ if HAVE_SYSTEMD systemdsystemunit_DATA = \ alsa-state.service \ - alsa-restore.service \ - alsa-store.service + alsa-restore.service install-data-hook: $(MKDIR_P) -m 0755 \ - $(DESTDIR)$(systemdsystemunitdir)/basic.target.wants \ - $(DESTDIR)$(systemdsystemunitdir)/shutdown.target.wants + $(DESTDIR)$(systemdsystemunitdir)/basic.target.wants ( cd $(DESTDIR)$(systemdsystemunitdir)/basic.target.wants && \ rm -f alsa-state.service alsa-restore.service && \ $(LN_S) ../alsa-state.service alsa-state.service && \ $(LN_S) ../alsa-restore.service alsa-restore.service) - ( cd $(DESTDIR)$(systemdsystemunitdir)/shutdown.target.wants && \ - rm -f alsa-store.service && \ - $(LN_S) ../alsa-store.service alsa-store.service ) endif @@ -54,22 +49,17 @@ alsa-state.service: alsa-state.service.in alsa-restore.service: alsa-restore.service.in $(edit) -alsa-store.service: alsa-store.service.in - $(edit) - 90-alsa-restore.rules: 90-alsa-restore.rules.in $(edit) EXTRA_DIST += \ alsa-state.service.in \ alsa-restore.service.in \ - alsa-store.service.in \ 90-alsa-restore.rules.in CLEANFILES = \ alsa-state.service \ alsa-restore.service \ - alsa-store.service \ 90-alsa-restore.rules %.7: %.xml diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index d1a74d637457..2884098c26f0 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -4,15 +4,14 @@ # [Unit] -Description=Restore Sound Card State +Description=Save/Restore Sound Card State ConditionPathExists=!@daemonswitch@ ConditionPathExistsGlob=/dev/snd/control* -DefaultDependencies=no After=alsa-state.service -Before=shutdown.target -Conflicts=shutdown.target [Service] Type=oneshot +RemainAfterExit=true ExecStart=-@sbindir@/alsactl restore +ExecStop=-@sbindir@/alsactl store StandardOutput=syslog diff --git a/alsactl/alsa-state.service.in b/alsactl/alsa-state.service.in index 56c12c104565..f1321d63a0ca 100644 --- a/alsactl/alsa-state.service.in +++ b/alsactl/alsa-state.service.in @@ -6,10 +6,7 @@ [Unit] Description=Manage Sound Card State (restore and store) ConditionPathExists=@daemonswitch@ -DefaultDependencies=no After=sysinit.target -Before=shutdown.target -Conflicts=shutdown.target [Service] Type=simple diff --git a/alsactl/alsa-store.service.in b/alsactl/alsa-store.service.in deleted file mode 100644 index c89cfff04f35..000000000000 --- a/alsactl/alsa-store.service.in +++ /dev/null @@ -1,16 +0,0 @@ -# -# Note that two different ALSA card state management schemes exist and they -# can be switched using a file exist check - /etc/alsa/state-daemon.conf . -# - -[Unit] -Description=Store Sound Card State -ConditionPathExists=!@daemonswitch@ -ConditionPathExistsGlob=/dev/snd/control* -DefaultDependencies=no -Before=shutdown.target - -[Service] -Type=oneshot -ExecStart=@sbindir@/alsactl store -StandardOutput=syslog -- 2.5.3

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[alsa-devel] [PATCH v3 0/4] ASoC: Add support for DA7219 audio codec
by Adam Thomson 29 Sep '15

29 Sep '15

This patch set adds support for the DA7219 audio codec with built-in advanced accessory detection functionality. Patch set includes codec driver, associated DT bindings documentation and MAINTAINERS file updates to cover new bindings. This patch set is baselined against the v4.3-rc3 kernel version. Changes in v3: - Actually set clk_src in dai_sysclk() function of codec, so future calls can correctly identify if work is to be done or not. Changes in v2: - Add SOC_DOUBLE_R_EXT support to ASoC core. - Use of upper-case naming for controls, where missing. - Use of Playback/Capture Digital naming for certain ADC/DAC controls. - CP mchange and vol_thresh DT bindings removed, now Kcontrols - Kcontrols and DAPM now locked as part of hptest work, and specific controls affected have had locking introduced. - Timeout for btn_det work when checking micbias status. - Added comments to describe reasoning for work use with hptest, btn_det. - Added comments to tonegen_freq_put/get to clarify endian conversion, and code now uses regmap_raw_read/write calls instead of bulk. - DT parsing code now prints dev_warn() for invalid settings. - SOC_VALUE_ENUM_SINGLE now used for hpf_mode handling, instead of customer get/put handlers. - DTMF key control now a SOC_ENUM to correctly describe supported keys, including non-numeric. - MIXIN_L_MIX_EN bit set as part of DAPM now (SUPPLY), rather than in probe. - Fix dai_sysclk() function to correctly check for no change, using clk_id as well as freq. - Error checking added for clk_set_rate() call. - Improved error checking for devm_clk_get() call. - Removed explicit use of device_may_wakeup() and enable/disable_irq_wake(). Now taken care of by I2C core so DT bindings updated accordingly to document this. - hptest work now uses cache to keep state, instead of reading registers individually. Cache is bypassed, then when test executed and complete, regcache_sync_region() used to restore settings to pre-test values. - Regulator consumer framework usage added, in particular to determine VDDIO level in use so device can be configured accordingly. DT documents updated to reflect this. Adam Thomson (4): ASoC: Add SOC_DOUBLE_R_EXT ASoC: codecs: Add da7219 codec driver ASoC: da7219: Add bindings documentation for DA7219 audio codec MAINTAINERS: da7219: Add entry to cover DA7219 bindings document Documentation/devicetree/bindings/sound/da7219.txt | 106 ++ MAINTAINERS | 1 + include/sound/da7219-aad.h | 99 + include/sound/da7219.h | 55 + include/sound/soc.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 823 +++++++++ sound/soc/codecs/da7219-aad.h | 212 +++ sound/soc/codecs/da7219.c | 1940 ++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 +++++++++ 11 files changed, 4069 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h -- 1.9.3

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[alsa-devel] [linux-sunxi][PATCH 3/3] ASOC: sunxi: Add support for the spdif block
by codekipper＠gmail.com 29 Sep '15

29 Sep '15

From: Marcus Cooper <codekipper(a)gmail.com> The sun4i, sun6i and sun7i SoC families have an SPDIF block which is capable of playback and capture. This patch enables the playback of this block for the sun4i and sun7i families. Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> --- sound/soc/sunxi/Kconfig | 10 + sound/soc/sunxi/Makefile | 4 + sound/soc/sunxi/sunxi-machine-spdif.c | 110 +++++ sound/soc/sunxi/sunxi-spdif.c | 801 ++++++++++++++++++++++++++++++++++ 4 files changed, 925 insertions(+) create mode 100644 sound/soc/sunxi/sunxi-machine-spdif.c create mode 100644 sound/soc/sunxi/sunxi-spdif.c diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 84c72ec..053db02 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -8,4 +8,14 @@ config SND_SUN4I_CODEC Select Y or M to add support for the Codec embedded in the Allwinner A10 and affiliated SoCs. +config SND_SOC_SUNXI_DAI_SPDIF + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + +config SND_SOC_SUNXI_MACHINE_SPDIF + tristate "APB on-chip sun4i/sun5i/sun7i SPDIF" + select SND_SOC_SUNXI_DAI_SPDIF + help + Say Y if you want to add support for SoC S/PDIF audio as simple audio card. endmenu diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index ea8a08c..7849a75 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -1,2 +1,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o +snd-soc-sunxi-dai-spdif-objs := sunxi-spdif.o +obj-$(CONFIG_SND_SOC_SUNXI_DAI_SPDIF) += snd-soc-sunxi-dai-spdif.o +snd-soc-sunxi-machine-spdif-objs := sunxi-machine-spdif.o +obj-$(CONFIG_SND_SOC_SUNXI_MACHINE_SPDIF) += snd-soc-sunxi-machine-spdif.o diff --git a/sound/soc/sunxi/sunxi-machine-spdif.c b/sound/soc/sunxi/sunxi-machine-spdif.c new file mode 100644 index 0000000..f8f6bd8 --- /dev/null +++ b/sound/soc/sunxi/sunxi-machine-spdif.c @@ -0,0 +1,110 @@ +/* + * Copyright (C) 2015 Andrea Venturi <be17068(a)iperbole.bo.it> + * From code by (C) 2013 Freescale Semiconductor, Inc. + * (sound/soc/fsl/imx-spdif.c) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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/of_platform.h> +#include <sound/soc.h> + +struct sunxi_machine_spdif_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; +}; + +static int sunxi_machine_spdif_audio_probe(struct platform_device *pdev) +{ + struct device_node *spdif_np, *np = pdev->dev.of_node; + struct sunxi_machine_spdif_data *data; + int ret = 0; + + dev_dbg(&pdev->dev, "%s: Looking for spdif-controller\n", __func__); + spdif_np = of_parse_phandle(np, "spdif-controller", 0); + if (!spdif_np) { + dev_err(&pdev->dev, "failed to find spdif-controller\n"); + ret = -EINVAL; + goto end; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto end; + } + + data->dai.name = "S/PDIF PCM"; + data->dai.stream_name = "S/PDIF PCM"; + data->dai.codec_dai_name = "snd-soc-dummy-dai"; + data->dai.codec_name = "snd-soc-dummy"; + data->dai.cpu_of_node = spdif_np; + data->dai.platform_of_node = spdif_np; + data->dai.playback_only = true; + data->dai.capture_only = true; + + if (of_property_read_bool(np, "spdif-out")) + data->dai.capture_only = false; + + if (of_property_read_bool(np, "spdif-in")) + data->dai.playback_only = false; + + if (data->dai.playback_only && data->dai.capture_only) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto end; + } + + data->card.dev = &pdev->dev; + data->card.dai_link = &data->dai; + data->card.num_links = 1; + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto end; + + 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 end; + } + + platform_set_drvdata(pdev, data); + +end: + of_node_put(spdif_np); + + return ret; +} + +#ifdef CONFIG_OF +static const struct of_device_id sunxi_machine_spdif_dt_ids[] = { + { .compatible = "allwinner,sunxi-audio-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sunxi_machine_spdif_dt_ids); +#endif + +static struct platform_driver sunxi_machine_spdif_driver = { + .driver = { + .name = "sunxi-machine-spdif", + .of_match_table = sunxi_machine_spdif_dt_ids, + }, + .probe = sunxi_machine_spdif_audio_probe, +}; + +module_platform_driver(sunxi_machine_spdif_driver); + +MODULE_AUTHOR("Marcus Cooper <codekipper(a)gmail.com>"); +MODULE_AUTHOR("Andrea Venturi, <be17068(a)iperbole.bo.it>"); +MODULE_DESCRIPTION("Allwinner sunxi S/PDIF machine driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/sunxi/sunxi-spdif.c b/sound/soc/sunxi/sunxi-spdif.c new file mode 100644 index 0000000..cae9261 --- /dev/null +++ b/sound/soc/sunxi/sunxi-spdif.c @@ -0,0 +1,801 @@ +/* + * ALSA SoC SPDIF Audio Layer + * + * Copyright 2015 Andrea Venturi <be17068(a)iperbole.bo.it> + * Copyright 2015 Marcus Cooper <codekipper(a)gmail.com> + * + * Based on the Allwinner SDK driver, released under the GPL. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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. + */ + +/* + * this is SPDIF sunxi simple audio card DAI driver that uses the codec + * "dummy driver" as per sound/soc/fsl/imx-spdif.c + */ +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/regmap.h> +#include <linux/of_address.h> +#include <linux/of_device.h> +#include <linux/ioport.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUNXI_SPDIF_CTL (0x00) + #define SUNXI_SPDIF_CTL_MCLKDIV(v) ((v) << 4) /* v even */ + #define SUNXI_SPDIF_CTL_MCLKOUTEN BIT(2) + #define SUNXI_SPDIF_CTL_GEN BIT(1) + #define SUNXI_SPDIF_CTL_RESET BIT(0) + +#define SUNXI_SPDIF_TXCFG (0x04) + #define SUNXI_SPDIF_TXCFG_SINGLEMOD BIT(31) + #define SUNXI_SPDIF_TXCFG_ASS BIT(17) + #define SUNXI_SPDIF_TXCFG_NONAUDIO BIT(16) + #define SUNXI_SPDIF_TXCFG_TXRATIO(v) ((v) << 4) + #define SUNXI_SPDIF_TXCFG_TXRATIO_MASK GENMASK(8, 4) + #define SUNXI_SPDIF_TXCFG_FMTRVD GENMASK(3, 2) + #define SUNXI_SPDIF_TXCFG_FMT16BIT (0 << 2) + #define SUNXI_SPDIF_TXCFG_FMT20BIT (1 << 2) + #define SUNXI_SPDIF_TXCFG_FMT24BIT (2 << 2) + #define SUNXI_SPDIF_TXCFG_CHSTMODE BIT(1) + #define SUNXI_SPDIF_TXCFG_TXEN BIT(0) + +#define SUNXI_SPDIF_RXCFG (0x08) + #define SUNXI_SPDIF_RXCFG_LOCKFLAG BIT(4) + #define SUNXI_SPDIF_RXCFG_CHSTSRC BIT(3) + #define SUNXI_SPDIF_RXCFG_CHSTCP BIT(1) + #define SUNXI_SPDIF_RXCFG_RXEN BIT(0) + +#define SUNXI_SPDIF_TXFIFO (0x0C) + +#define SUNXI_SPDIF_RXFIFO (0x10) + +#define SUNXI_SPDIF_FCTL (0x14) + #define SUNXI_SPDIF_FCTL_FIFOSRC BIT(31) + #define SUNXI_SPDIF_FCTL_FTX BIT(17) + #define SUNXI_SPDIF_FCTL_FRX BIT(16) + #define SUNXI_SPDIF_FCTL_TXTL(v) ((v) << 8) + #define SUNXI_SPDIF_FCTL_TXTL_MASK GENMASK(12, 8) + #define SUNXI_SPDIF_FCTL_RXTL(v) ((v) << 3) + #define SUNXI_SPDIF_FCTL_RXTL_MASK GENMASK(7, 3) + #define SUNXI_SPDIF_FCTL_TXIM BIT(2) + #define SUNXI_SPDIF_FCTL_RXOM(v) ((v) << 0) + #define SUNXI_SPDIF_FCTL_RXOM_MASK GENMASK(1, 0) + +#define SUNXI_SPDIF_FSTA (0x18) + #define SUNXI_SPDIF_FSTA_TXE BIT(14) + #define SUNXI_SPDIF_FSTA_TXECNTSHT (8) + #define SUNXI_SPDIF_FSTA_RXA BIT(6) + #define SUNXI_SPDIF_FSTA_RXACNTSHT (0) + +#define SUNXI_SPDIF_INT (0x1C) + #define SUNXI_SPDIF_INT_RXLOCKEN BIT(18) + #define SUNXI_SPDIF_INT_RXUNLOCKEN BIT(17) + #define SUNXI_SPDIF_INT_RXPARERREN BIT(16) + #define SUNXI_SPDIF_INT_TXDRQEN BIT(7) + #define SUNXI_SPDIF_INT_TXUIEN BIT(6) + #define SUNXI_SPDIF_INT_TXOIEN BIT(5) + #define SUNXI_SPDIF_INT_TXEIEN BIT(4) + #define SUNXI_SPDIF_INT_RXDRQEN BIT(2) + #define SUNXI_SPDIF_INT_RXOIEN BIT(1) + #define SUNXI_SPDIF_INT_RXAIEN BIT(0) + +#define SUNXI_SPDIF_ISTA (0x20) + #define SUNXI_SPDIF_ISTA_RXLOCKSTA BIT(18) + #define SUNXI_SPDIF_ISTA_RXUNLOCKSTA BIT(17) + #define SUNXI_SPDIF_ISTA_RXPARERRSTA BIT(16) + #define SUNXI_SPDIF_ISTA_TXUSTA BIT(6) + #define SUNXI_SPDIF_ISTA_TXOSTA BIT(5) + #define SUNXI_SPDIF_ISTA_TXESTA BIT(4) + #define SUNXI_SPDIF_ISTA_RXOSTA BIT(1) + #define SUNXI_SPDIF_ISTA_RXASTA BIT(0) + +#define SUNXI_SPDIF_TXCNT (0x24) + +#define SUNXI_SPDIF_RXCNT (0x28) + +#define SUNXI_SPDIF_TXCHSTA0 (0x2C) + #define SUNXI_SPDIF_TXCHSTA0_CLK(v) ((v) << 28) + #define SUNXI_SPDIF_TXCHSTA0_SAMFREQ(v) ((v) << 24) + #define SUNXI_SPDIF_TXCHSTA0_SAMFREQ_MASK GENMASK(27, 24) + #define SUNXI_SPDIF_TXCHSTA0_CHNUM(v) ((v) << 20) + #define SUNXI_SPDIF_TXCHSTA0_CHNUM_MASK GENMASK(23, 20) + #define SUNXI_SPDIF_TXCHSTA0_SRCNUM(v) ((v) << 16) + #define SUNXI_SPDIF_TXCHSTA0_CATACOD(v) ((v) << 8) + #define SUNXI_SPDIF_TXCHSTA0_MODE(v) ((v) << 6) + #define SUNXI_SPDIF_TXCHSTA0_EMPHASIS(v) ((v) << 3) + #define SUNXI_SPDIF_TXCHSTA0_CP BIT(2) + #define SUNXI_SPDIF_TXCHSTA0_AUDIO BIT(1) + #define SUNXI_SPDIF_TXCHSTA0_PRO BIT(0) + +#define SUNXI_SPDIF_TXCHSTA1 (0x30) + #define SUNXI_SPDIF_TXCHSTA1_CGMSA(v) ((v) << 8) + #define SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ(v) ((v) << 4) + #define SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ_MASK GENMASK(7, 4) + #define SUNXI_SPDIF_TXCHSTA1_SAMWORDLEN(v) ((v) << 1) + #define SUNXI_SPDIF_TXCHSTA1_MAXWORDLEN BIT(0) + +#define SUNXI_SPDIF_RXCHSTA0 (0x34) + #define SUNXI_SPDIF_RXCHSTA0_CLK(v) ((v) << 28) + #define SUNXI_SPDIF_RXCHSTA0_SAMFREQ(v) ((v) << 24) + #define SUNXI_SPDIF_RXCHSTA0_CHNUM(v) ((v) << 20) + #define SUNXI_SPDIF_RXCHSTA0_SRCNUM(v) ((v) << 16) + #define SUNXI_SPDIF_RXCHSTA0_CATACOD(v) ((v) << 8) + #define SUNXI_SPDIF_RXCHSTA0_MODE(v) ((v) << 6) + #define SUNXI_SPDIF_RXCHSTA0_EMPHASIS(v) ((v) << 3) + #define SUNXI_SPDIF_RXCHSTA0_CP BIT(2) + #define SUNXI_SPDIF_RXCHSTA0_AUDIO BIT(1) + #define SUNXI_SPDIF_RXCHSTA0_PRO BIT(0) + +#define SUNXI_SPDIF_RXCHSTA1 (0x38) + #define SUNXI_SPDIF_RXCHSTA1_CGMSA(v) ((v) << 8) + #define SUNXI_SPDIF_RXCHSTA1_ORISAMFREQ(v) ((v) << 4) + #define SUNXI_SPDIF_RXCHSTA1_SAMWORDLEN(v) ((v) << 1) + #define SUNXI_SPDIF_RXCHSTA1_MAXWORDLEN BIT(0) + +/* Defines for Sampling Frequency */ +enum sunxi_spdif_sampling_freq { + SUNXI_SPDIF_SAMFREQ_44_1KHZ = 0x0, + SUNXI_SPDIF_SAMFREQ_NOT_INDICATED = 0x1, + SUNXI_SPDIF_SAMFREQ_48KHZ = 0x2, + SUNXI_SPDIF_SAMFREQ_32KHZ = 0x3, + SUNXI_SPDIF_SAMFREQ_22_05KHZ = 0x4, + SUNXI_SPDIF_SAMFREQ_24KHZ = 0x6, + SUNXI_SPDIF_SAMFREQ_88_2KHZ = 0x8, + SUNXI_SPDIF_SAMFREQ_768KHZ = 0x9, + SUNXI_SPDIF_SAMFREQ_96KHZ = 0xa, + SUNXI_SPDIF_SAMFREQ_176_4KHZ = 0xc, + SUNXI_SPDIF_SAMFREQ_192KHZ = 0xe, +}; + +/* + * Original sampling frequency can be represented by inverting the value of the + * sampling frequency. + */ +#define ORIGINAL(v) ((~v) & 0xf) + +struct sunxi_spdif_dev { + struct platform_device *pdev; + struct clk *clk; + struct clk *apb_clk; + struct clk *audio_clk; + struct snd_soc_dai_driver cpu_dai_drv; + bool running; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct snd_dmaengine_dai_dma_data dma_params_rx; + bool playback_supported; + bool capture_supported; +}; + +static void sunxi_spdif_configure(struct sunxi_spdif_dev *host) +{ + u32 reg_val; + + /* soft reset SPDIF */ + regmap_write(host->regmap, SUNXI_SPDIF_CTL, SUNXI_SPDIF_CTL_RESET); + + /* MCLK OUTPUT enable */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_CTL, + SUNXI_SPDIF_CTL_MCLKOUTEN, SUNXI_SPDIF_CTL_MCLKOUTEN); + + /* flush TX FIFO */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_FCTL, SUNXI_SPDIF_FCTL_FTX, + SUNXI_SPDIF_FCTL_FTX); + + /* clear interrupt status */ + regmap_read(host->regmap, SUNXI_SPDIF_ISTA, &reg_val); + regmap_write(host->regmap, SUNXI_SPDIF_ISTA, reg_val); + + /* clear TX counter */ + regmap_write(host->regmap, SUNXI_SPDIF_TXCNT, 0); + +} + +void sunxi_snd_txctrl(struct snd_pcm_substream *substream, + struct sunxi_spdif_dev *host, int on) +{ + u32 tmp; + + regmap_read(host->regmap, SUNXI_SPDIF_TXCNT, &tmp); + if (substream->runtime->channels == 1) + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCFG, + SUNXI_SPDIF_TXCFG_SINGLEMOD, + SUNXI_SPDIF_TXCFG_SINGLEMOD); + + if (on) { + /* SPDIF TX ENABLE */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCFG, + SUNXI_SPDIF_TXCFG_TXEN, + SUNXI_SPDIF_TXCFG_TXEN); + + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_INT, + SUNXI_SPDIF_INT_TXDRQEN, + SUNXI_SPDIF_INT_TXDRQEN); + + /* Global enable */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_CTL, + SUNXI_SPDIF_CTL_GEN, + SUNXI_SPDIF_CTL_GEN); + } else { + /* SPDIF TX DISABLE */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCFG, + SUNXI_SPDIF_TXCFG_TXEN, 0); + + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_INT, + SUNXI_SPDIF_INT_TXDRQEN, 0); + + /* Global disable */ + regmap_update_bits(host->regmap, SUNXI_SPDIF_CTL, + SUNXI_SPDIF_CTL_GEN, 0); + } +} + +static int sunxi_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai); + int ret; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + /* Reset module and interrupts only for first initialization */ + if (!cpu_dai->active) { + ret = clk_prepare_enable(host->clk); + if (ret) + return ret; + } + + host->running = true; + sunxi_spdif_configure(host); + + return 0; +} + +static void sunxi_spdif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return; + + clk_disable_unprepare(host->clk); + host->running = false; +} + +static int sunxi_spdif_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + u32 reg_val; + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + reg_val = 0; + reg_val &= ~SUNXI_SPDIF_TXCFG_SINGLEMOD; + reg_val |= SUNXI_SPDIF_TXCFG_ASS; + reg_val |= fmt; /* set non audio and bit depth */ + reg_val |= SUNXI_SPDIF_TXCFG_CHSTMODE; + regmap_write(host->regmap, SUNXI_SPDIF_TXCFG, reg_val); + reg_val = 0; + reg_val &= ~SUNXI_SPDIF_FCTL_FIFOSRC; + reg_val |= SUNXI_SPDIF_FCTL_TXTL_MASK; + reg_val |= SUNXI_SPDIF_FCTL_RXTL_MASK; + reg_val |= SUNXI_SPDIF_FCTL_TXIM; + reg_val |= SUNXI_SPDIF_FCTL_RXOM_MASK; + regmap_write(host->regmap, SUNXI_SPDIF_FCTL, reg_val); + reg_val = 0; + reg_val |= (SUNXI_SPDIF_TXCHSTA0_CHNUM(2)); + regmap_write(host->regmap, SUNXI_SPDIF_TXCHSTA0, reg_val); + + reg_val = 0; + reg_val |= (SUNXI_SPDIF_TXCHSTA1_SAMWORDLEN(1)); + if ((fmt & SUNXI_SPDIF_TXCFG_FMT24BIT) == SUNXI_SPDIF_TXCFG_FMT24BIT) + reg_val |= SUNXI_SPDIF_TXCHSTA1_MAXWORDLEN; + regmap_write(host->regmap, SUNXI_SPDIF_TXCHSTA1, reg_val); + return 0; +} + +static int sunxi_spdif_set_sysclk(struct snd_soc_dai *cpu_dai, unsigned int freq) +{ + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct platform_device *pdev = host->pdev; + + if (clk_set_rate(host->audio_clk, freq)) + dev_err(&pdev->dev, + "Setting pll2 clock rate for %d Hz failed!\n", freq); + if (clk_set_rate(host->clk, freq)) + dev_err(&pdev->dev, + "Setting SPDIF clock rate for %d Hz failed!\n", freq); + return 0; +} + +static int sunxi_spdif_set_clkdiv(struct snd_soc_dai *cpu_dai, + unsigned int rate, int div) +{ + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + int sample_freq, original_sample_freq; + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCHSTA0, + SUNXI_SPDIF_TXCHSTA0_SAMFREQ_MASK, 0); + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCHSTA1, + SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ_MASK, 0); + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCFG, + SUNXI_SPDIF_TXCFG_TXRATIO_MASK, + SUNXI_SPDIF_TXCFG_TXRATIO(div - 1)); + + if (rate == 24576000) { + switch (div) { + /* 24KHZ */ + case 8: + sample_freq = SUNXI_SPDIF_SAMFREQ_24KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_24KHZ); + break; + + /* 32KHZ */ + case 6: + sample_freq = SUNXI_SPDIF_SAMFREQ_32KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_32KHZ); + break; + + /* 48KHZ */ + case 4: + sample_freq = SUNXI_SPDIF_SAMFREQ_48KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_48KHZ); + break; + + /* 96KHZ */ + case 2: + sample_freq = SUNXI_SPDIF_SAMFREQ_96KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_96KHZ); + break; + + /* 192KHZ */ + case 1: + sample_freq = SUNXI_SPDIF_SAMFREQ_192KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_192KHZ); + break; + + default: + sample_freq = SUNXI_SPDIF_SAMFREQ_NOT_INDICATED; + original_sample_freq = 0; + break; + } + } else { + /* 22.5792MHz */ + switch (div) { + /* 22.05KHZ */ + case 8: + sample_freq = SUNXI_SPDIF_SAMFREQ_22_05KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_22_05KHZ); + break; + + /* 44.1KHZ */ + case 4: + sample_freq = SUNXI_SPDIF_SAMFREQ_44_1KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_44_1KHZ); + break; + + /* 88.2KHZ */ + case 2: + sample_freq = SUNXI_SPDIF_SAMFREQ_88_2KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_88_2KHZ); + break; + + /* 176.4KHZ */ + case 1: + sample_freq = SUNXI_SPDIF_SAMFREQ_176_4KHZ; + original_sample_freq + = ORIGINAL(SUNXI_SPDIF_SAMFREQ_176_4KHZ); + break; + + default: + sample_freq = SUNXI_SPDIF_SAMFREQ_NOT_INDICATED; + original_sample_freq = 0; + break; + } + } + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCHSTA0, + SUNXI_SPDIF_TXCHSTA0_SAMFREQ_MASK, + SUNXI_SPDIF_TXCHSTA0_SAMFREQ(sample_freq)); + + regmap_update_bits(host->regmap, SUNXI_SPDIF_TXCHSTA1, + SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ_MASK, + SUNXI_SPDIF_TXCHSTA1_ORISAMFREQ(original_sample_freq)); + + return 0; +} + +static int sunxi_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int ret = 0; + int fmt; + unsigned long rate = params_rate(params); + u32 mclk_div = 0; + unsigned int mclk = 0; + + /* Add the PCM and raw data select interface */ + switch (params_channels(params)) { + case 1: /* PCM mode */ + case 2: + fmt = 0; + break; + case 4: /* raw data mode */ + fmt = SUNXI_SPDIF_TXCFG_NONAUDIO; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + fmt |= SUNXI_SPDIF_TXCFG_FMT16BIT; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + fmt |= SUNXI_SPDIF_TXCFG_FMT20BIT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + fmt |= SUNXI_SPDIF_TXCFG_FMT24BIT; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + switch (rate) { + case 22050: + case 44100: + case 88200: + case 176400: + mclk = 22579200; + break; + case 24000: + case 32000: + case 48000: + case 96000: + case 192000: + mclk = 24576000; + break; + default: + return -EINVAL; + } + + ret = sunxi_spdif_set_sysclk(cpu_dai, mclk); + if (ret < 0) + return ret; + + switch (rate) { + case 22050: + case 24000: + mclk_div = 8; + break; + case 32000: + mclk_div = 6; + break; + case 44100: + case 48000: + mclk_div = 4; + break; + case 88200: + case 96000: + mclk_div = 2; + break; + case 176400: + case 192000: + mclk_div = 1; + break; + default: + return -EINVAL; + } + + ret = sunxi_spdif_set_clkdiv(cpu_dai, mclk, mclk_div); + if (ret < 0) + return ret; + + return 0; +} + +static int sunxi_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sunxi_snd_txctrl(substream, host, 1); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sunxi_snd_txctrl(substream, host, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sunxi_spdif_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sunxi_spdif_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &host->dma_params_tx, + &host->dma_params_rx); + return 0; +} + +static const struct snd_soc_dai_ops sunxi_spdif_dai_ops = { + .startup = sunxi_spdif_startup, + .shutdown = sunxi_spdif_shutdown, + .trigger = sunxi_spdif_trigger, + .hw_params = sunxi_spdif_hw_params, + .set_fmt = sunxi_spdif_set_fmt, +}; + +static const struct regmap_config sunxi_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUNXI_SPDIF_RXCHSTA1, +}; + +#define SUNXI_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) +#define SUNXI_FORMATS (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ + SNDRV_PCM_FORMAT_S24_LE) + +static struct snd_soc_dai_driver sunxi_spdif_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SUNXI_RATES, + .formats = SUNXI_FORMATS, + }, + .probe = sunxi_spdif_soc_dai_probe, + .ops = &sunxi_spdif_dai_ops, + .name = "spdif", +}; + +static const struct snd_soc_dapm_widget dit_widgets[] = { + SND_SOC_DAPM_OUTPUT("spdif-out"), +}; + +static const struct snd_soc_dapm_route dit_routes[] = { + { "spdif-out", NULL, "Playback" }, +}; + +#ifdef CONFIG_OF +static const struct of_device_id sunxi_spdif_of_match[] = { + { .compatible = "allwinner,sun4i-a10-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sunxi_spdif_of_match); +#endif + +static const struct snd_soc_component_driver sunxi_spdif_component = { + .name = "sunxi-spdif", +}; + +static int sunxi_spdif_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct sunxi_spdif_dev *host; + struct resource *res; + int ret; + const struct of_device_id *of_id; + void __iomem *base; + + dev_dbg(&pdev->dev, "Entered %s\n", __func__); + + if (!np) + return -ENODEV; + + of_id = of_match_device(sunxi_spdif_of_match, &pdev->dev); + if (!of_id) { + dev_err(&pdev->dev, "No matched devices found.\n"); + return -EINVAL; + } + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + host->pdev = pdev; + + /* Initialize this copy of the CPU DAI driver structure */ + memcpy(&host->cpu_dai_drv, &sunxi_spdif_dai, sizeof(sunxi_spdif_dai)); + host->cpu_dai_drv.name = dev_name(&pdev->dev); + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sunxi_spdif_regmap_config); + + /* Clocks */ + host->apb_clk = devm_clk_get(&pdev->dev, "apb"); + if (IS_ERR(host->apb_clk)) { + dev_err(&pdev->dev, "failed to get a apb clock.\n"); + return PTR_ERR(host->apb_clk); + } + + if (clk_prepare_enable(host->apb_clk)) { + dev_err(&pdev->dev, "try to enable apb_spdif_clk failed\n"); + return -EINVAL; + } + + host->audio_clk = devm_clk_get(&pdev->dev, "audio"); + if (IS_ERR(host->audio_clk)) { + dev_err(&pdev->dev, "failed to get an audio clock.\n"); + return PTR_ERR(host->audio_clk); + } + + host->clk = devm_clk_get(&pdev->dev, "spdif"); + if (IS_ERR(host->clk)) { + dev_err(&pdev->dev, "failed to get a spdif clock.\n"); + return PTR_ERR(host->clk); + } + + ret = clk_set_rate(host->audio_clk, 24576000); + if (ret) { + dev_err(&pdev->dev, "set spdif base clock failed!\n"); + return ret; + } + + if (clk_prepare_enable(host->audio_clk)) { + dev_err(&pdev->dev, "try to enable audio clk failed\n"); + ret = -EINVAL; + goto exit_clkdisable_apb_clk; + } + + host->playback_supported = false; + host->capture_supported = false; + + if (of_property_read_bool(np, "spdif-out")) + host->playback_supported = true; + + if (!host->playback_supported) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto exit_clkdisable_clk; + } + + host->dma_params_tx.addr = res->start + SUNXI_SPDIF_TXFIFO; + host->dma_params_tx.maxburst = 4; + host->dma_params_tx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + host->dma_params_rx.addr = res->start + SUNXI_SPDIF_RXFIFO; + host->dma_params_rx.maxburst = 4; + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + /* Register with ASoC */ + dev_set_drvdata(&pdev->dev, host); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sunxi_spdif_component, &sunxi_spdif_dai, 1); + if (ret) + goto exit_clkdisable_clk; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto exit_clkdisable_clk; + return 0; + +exit_clkdisable_clk: + clk_disable_unprepare(host->clk); +exit_clkdisable_apb_clk: + clk_disable_unprepare(host->apb_clk); + return ret; +} + +static int sunxi_spdif_remove(struct platform_device *pdev) +{ + struct sunxi_spdif_dev *host = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + if (!IS_ERR(host->clk)) { + clk_disable_unprepare(host->clk); + clk_disable_unprepare(host->apb_clk); + } + + return 0; +} + +#ifdef CONFIG_PM +static int sunxi_spdif_suspend(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct sunxi_spdif_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) + clk_disable(host->clk); + + return 0; +} + +static int sunxi_spdif_resume(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct sunxi_spdif_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) { + clk_enable(host->clk); + sunxi_spdif_configure(host); + } + return 0; +} + +static SIMPLE_DEV_PM_OPS(spdif_dev_pm_ops, sunxi_spdif_suspend, + sunxi_spdif_resume); + +#define SPDIF_DEV_PM_OPS (&spdif_dev_pm_ops) +#else +#define SPDIF_DEV_PM_OPS NULL + +#endif + +static struct platform_driver sunxi_spdif_driver = { + .driver = { + .name = "sunxi-spdif", + .owner = THIS_MODULE, + .pm = SPDIF_DEV_PM_OPS, + .of_match_table = of_match_ptr(sunxi_spdif_of_match), + }, + .probe = sunxi_spdif_probe, + .remove = sunxi_spdif_remove, +}; + +module_platform_driver(sunxi_spdif_driver); + +MODULE_AUTHOR("Marcus Cooper <codekipper(a)gmail.com>"); +MODULE_AUTHOR("Andrea Venturi <be17068(a)iperbole.bo.it>"); +MODULE_DESCRIPTION("Allwinner Sunxi SPDIF SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sunxi-spdif"); -- 1.9.1

4 5

[alsa-devel] [sound:topic/firewire-update 15/15] sound/firewire/amdtp-am824.c:402:14: sparse: symbol 'process_rx_data_blocks' was not declared. Should it be static?
by kbuild test robot 29 Sep '15

29 Sep '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/firewire-update head: df075feefbd347f13fba5198294cda619532c237 commit: df075feefbd347f13fba5198294cda619532c237 [15/15] ALSA: firewire-lib: complete AM824 data block processing layer reproduce: # apt-get install sparse git checkout df075feefbd347f13fba5198294cda619532c237 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/firewire/amdtp-am824.c:402:14: sparse: symbol 'process_rx_data_blocks' was not declared. Should it be static? >> sound/firewire/amdtp-am824.c:423:14: sparse: symbol 'process_tx_data_blocks' was not declared. Should it be static? Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [RFC 0/5] another generic audio hdmi codec proposal
by Arnaud Pouliquen 29 Sep '15

29 Sep '15

I 'm working on HDMI implementation for sti platform. As some discussions are on going for audio on HDMI. Here is the begin of my work... This patch set is a tentative to implement a generic code for the HDMI audio. Main concept are aligned with solution proposeded for TI platform. - ASoC codec driver registered by DRM driver - ASOC driver is a generic driver. - compatible with simple card Difference is that i propose a DRM generic interface based on bridge structure. Advantage is that all data exchanges are done through the DRM API. I think also that some helper functions could been used for N and CTS parameters calculation, as suggested by Russell King in a previous mail. I full aware that some features (like ELD and info frame) are partially or not implemented in my patches. This patch set is more a skeleton than a full implementation... I just post it to suggest a possible DRM API. Arnaud Pouliquen (5): video: hdmi: add help function for N and cts drm: add helper functionto add audio capabilities for bridge ASoC: codec: hdmi drm codec driver drm: sti: connect audio driver DT: sti: add audio HDMI dai link in audio card arch/arm/boot/dts/stih410.dtsi | 4 +- arch/arm/boot/dts/stihxxx-b2120.dtsi | 21 +++++ arch/arm/configs/multi_v7_defconfig | 1 + drivers/gpu/drm/drm_bridge.c | 114 +++++++++++++++++++++++++++ drivers/gpu/drm/sti/sti_hdmi.c | 146 +++++++++++++++++++++++++++++++--- drivers/gpu/drm/sti/sti_hdmi.h | 3 + drivers/video/hdmi.c | 148 +++++++++++++++++++++++++++++++++++ include/drm/drm_crtc.h | 31 ++++++++ include/linux/hdmi.h | 24 ++++++ include/sound/hdmi_drm.h | 16 ++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi_drm.c | 125 +++++++++++++++++++++++++++++ 13 files changed, 625 insertions(+), 14 deletions(-) create mode 100644 include/sound/hdmi_drm.h create mode 100644 sound/soc/codecs/hdmi_drm.c -- 1.9.1

2 8

[alsa-devel] [PATCH 0/2] ASoc: add driver for Atmel I2S controller
by Cyrille Pitchen 29 Sep '15

29 Sep '15

This series of patches add support to the new Atmel I2S controller embedded on sama5d2 SoCs. Cyrille Pitchen (2): ASoC: atmel-i2s: add DT bindings for I2S controller ASoC: atmel-i2s: add driver for the new Atmel I2S controller .../devicetree/bindings/sound/atmel-i2s.txt | 43 ++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-i2s.c | 760 +++++++++++++++++++++ 4 files changed, 815 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-i2s.txt create mode 100644 sound/soc/atmel/atmel-i2s.c -- 1.8.2.2

1 3

[alsa-devel] [PATCH 00/15 v3] ALSA: firewire-lib: separate to packet streaming layer and data block processing layer
by Takashi Sakamoto 29 Sep '15

29 Sep '15

This patchset updates a part of my previous post (patch 01 and 02): [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html The aim of this patchset is to allow current amdtp implementation to handle several data block processing backends. For this aim, this patchset separates the implementation to packet streaming layer and AM824 data block processing layer. The communication between these two layers is done by callback function. This patchset adds an overhead to call the functions as the same frequently as 8,000 times per second. Takashi Sakamoto (15): ALSA: firewire-lib: arrange structure for AMDTP stream ALSA: firewire-lib: return error code when amdtp_stream_set_parameters() detects error ALSA: firewire-lib: add an argument for Dice's dual wire mode ALSA: firewire-lib: add a member of frame_multiplier instead of double_pcm_frames ALSA: firewire-lib: add helper functions as interfaces between packet streaming layer and data block processing layer ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' to prepare for functional separation ALSA: firewire-lib: add data block processing layer for AM824 format ALSA: firewire-lib: rename parameter setting function for AM824 with FDF field ALSA: firewire-lib: move PCM substream constraint to AM824 layer ALSA: firewire-lib: add helper functions to set positions of data channels ALSA: firewire-lib: move MIDI trigger helper function to AM824 layer ALSA: firewire-lib: rename PCM format helper function ALSA: firewire-lib: rename macros with AM824 prefix ALSA: firewire-lib: complete AM824 data block processing layer sound/firewire/Makefile | 2 +- sound/firewire/amdtp-am824.c | 465 ++++++++++++++++++++++++++++ sound/firewire/amdtp-am824.h | 52 ++++ sound/firewire/{amdtp.c => amdtp-stream.c} | 377 ++++------------------ sound/firewire/{amdtp.h => amdtp-stream.h} | 116 +++---- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_midi.c | 16 +- sound/firewire/bebob/bebob_pcm.c | 14 +- sound/firewire/bebob/bebob_stream.c | 34 +- sound/firewire/dice/dice-midi.c | 12 +- sound/firewire/dice/dice-pcm.c | 12 +- sound/firewire/dice/dice-stream.c | 22 +- sound/firewire/dice/dice.h | 2 +- sound/firewire/fcp.c | 2 +- sound/firewire/fireworks/fireworks.c | 12 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_midi.c | 12 +- sound/firewire/fireworks/fireworks_pcm.c | 12 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/oxfw/oxfw-midi.c | 16 +- sound/firewire/oxfw/oxfw-pcm.c | 10 +- sound/firewire/oxfw/oxfw-stream.c | 11 +- sound/firewire/oxfw/oxfw.h | 2 +- 23 files changed, 730 insertions(+), 483 deletions(-) create mode 100644 sound/firewire/amdtp-am824.c create mode 100644 sound/firewire/amdtp-am824.h rename sound/firewire/{amdtp.c => amdtp-stream.c} (70%) rename sound/firewire/{amdtp.h => amdtp-stream.h} (78%) -- 2.1.4

2 19

[alsa-devel] [PATCH 1/1] ASoC: ad193x: add support to ad1934
by Cyrille Pitchen 29 Sep '15

29 Sep '15

The AD1934 codec has no DAC feature. Hence it register mapping is slightly different from the register mapping of other members of the AD193x family. Some ASoC controls and widgets are related to the DAC feature so are not relevant in the case of an AD1934 codec. Signed-off-by: Cyrille Pitchen <cyrille.pitchen(a)atmel.com> --- Hi all, This patch adds support to the AD1934 codec, which is DAC only. Best Regards, Cyrille --- sound/soc/codecs/ad193x-i2c.c | 8 +-- sound/soc/codecs/ad193x-spi.c | 16 +++++- sound/soc/codecs/ad193x.c | 121 ++++++++++++++++++++++++++++++++---------- sound/soc/codecs/ad193x.h | 9 +++- 4 files changed, 120 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c index df3a1a415825..8aa50d35662f 100644 --- a/sound/soc/codecs/ad193x-i2c.c +++ b/sound/soc/codecs/ad193x-i2c.c @@ -15,8 +15,8 @@ #include "ad193x.h" static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, + { "ad1936", AD193x }, + { "ad1937", AD193x }, { } }; MODULE_DEVICE_TABLE(i2c, ad193x_id); @@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client, config.val_bits = 8; config.reg_bits = 8; - return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); + return ad193x_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 390cef9b9dc2..364e82112e8c 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -16,6 +16,7 @@ static int ad193x_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct regmap_config config; config = ad193x_regmap_config; @@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi) config.read_flag_mask = 0x09; config.write_flag_mask = 0x08; - return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_spi_remove(struct spi_device *spi) @@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id ad193x_spi_id[] = { + { "ad193x", AD193X }, + { "ad1933", AD1933 }, + { "ad1934", AD1934 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, + { "ad1938", AD193X }, + { } +}; +MODULE_DEVICE_TABLE(spi, ad193x_spi_id); + static struct spi_driver ad193x_spi_driver = { .driver = { .name = "ad193x", @@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = { }, .probe = ad193x_spi_probe, .remove = ad193x_spi_remove, + .id_table = ad193x_spi_id, }; module_spi_driver(ad193x_spi_driver); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 17c953595660..80ffecf98673 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,6 +23,7 @@ /* codec private data */ struct ad193x_priv { struct regmap *regmap; + enum ad193x_type type; int sysclk; }; @@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, - AD193X_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, - AD193X_ADCR2_MUTE, 1, 1), - /* DAC switch control */ SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, AD193X_DACR1_MUTE, 1, 1), @@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, AD193X_DACR4_MUTE, 1, 1), + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = { + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, AD193X_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_INPUT("ADC1IN"), SND_SOC_DAPM_INPUT("ADC2IN"), }; @@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, { "DAC Output", NULL, "DAC" }, { "DAC Output", NULL, "VMID" }, - { "ADC", NULL, "SYSCLK" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC Output" }, { "DAC2OUT", NULL, "DAC Output" }, { "DAC3OUT", NULL, "DAC Output" }, { "DAC4OUT", NULL, "DAC Output" }, + { "SYSCLK", NULL, "PLL_PWR" }, +}; + +static const struct snd_soc_dapm_route adc_audio_paths[] = { + { "ADC", NULL, "SYSCLK" }, + { "ADC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, - { "SYSCLK", NULL, "PLL_PWR" }, }; +static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) +{ + switch (ad193x->type) { + case AD1933: + case AD1934: + return false; + default: + break; + } + + return true; +} + /* * DAI ops entries */ @@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: - return -EINVAL; + if (ad193x_has_adc(ad193x)) + return -EINVAL; + break; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_SERFMT_MASK, adc_serfmt); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_FMT_MASK, adc_fmt); + if (ad193x_has_adc(ad193x)) { + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + } regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, dac_fmt); @@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_WORD_LEN_MASK, word_len); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -326,6 +354,7 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + int num, ret; /* default setting for ad193x */ @@ -335,14 +364,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* dac in tdm mode */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* high-pass filter enable */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + } + /* pll input: mclki/xi */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* add adc controls */ + num = ARRAY_SIZE(ad193x_adc_snd_controls); + ret = snd_soc_add_codec_controls(codec, + ad193x_adc_snd_controls, + num); + if (ret) + return ret; + + /* add adc widgets */ + num = ARRAY_SIZE(ad193x_adc_widgets); + ret = snd_soc_dapm_new_controls(&codec->dapm, + ad193x_adc_widgets, + num); + if (ret) + return ret; + + /* add adc routes */ + num = ARRAY_SIZE(adc_audio_paths); + ret = snd_soc_dapm_add_routes(&codec->dapm, + adc_audio_paths, + num); + if (ret) + return ret; + } + return 0; } @@ -367,7 +428,8 @@ const struct regmap_config ad193x_regmap_config = { }; EXPORT_SYMBOL_GPL(ad193x_regmap_config); -int ad193x_probe(struct device *dev, struct regmap *regmap) +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type) { struct ad193x_priv *ad193x; @@ -379,6 +441,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; ad193x->regmap = regmap; + ad193x->type = type; dev_set_drvdata(dev, ad193x); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index ab9a998f15be..8b1e65f928d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -13,8 +13,15 @@ struct device; +enum ad193x_type { + AD193X, + AD1933, + AD1934, +}; + extern const struct regmap_config ad193x_regmap_config; -int ad193x_probe(struct device *dev, struct regmap *regmap); +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type); #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -- 1.8.2.2

2 2

[alsa-devel] [PATCH v2 1/1] ASoC: ad193x: add support to ad1934
by Cyrille Pitchen 29 Sep '15

29 Sep '15

The AD1934 codec has no DAC feature. Hence it register mapping is slightly different from the register mapping of other members of the AD193x family. Some ASoC controls and widgets are related to the DAC feature so are not relevant in the case of an AD1934 codec. Signed-off-by: Cyrille Pitchen <cyrille.pitchen(a)atmel.com> --- Hi all, This patch adds support to the AD1934 codec, which is DAC only. Best Regards, Cyrille ChangeLog v2: - replace AD193x by AD193X in ad193x-i2c.c - add ad193x_ prefix to adc_audio_paths --- sound/soc/codecs/ad193x-i2c.c | 8 +-- sound/soc/codecs/ad193x-spi.c | 16 +++++- sound/soc/codecs/ad193x.c | 121 ++++++++++++++++++++++++++++++++---------- sound/soc/codecs/ad193x.h | 9 +++- 4 files changed, 120 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c index df3a1a415825..171313664bc8 100644 --- a/sound/soc/codecs/ad193x-i2c.c +++ b/sound/soc/codecs/ad193x-i2c.c @@ -15,8 +15,8 @@ #include "ad193x.h" static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, { } }; MODULE_DEVICE_TABLE(i2c, ad193x_id); @@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client, config.val_bits = 8; config.reg_bits = 8; - return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); + return ad193x_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 390cef9b9dc2..364e82112e8c 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -16,6 +16,7 @@ static int ad193x_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct regmap_config config; config = ad193x_regmap_config; @@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi) config.read_flag_mask = 0x09; config.write_flag_mask = 0x08; - return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_spi_remove(struct spi_device *spi) @@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id ad193x_spi_id[] = { + { "ad193x", AD193X }, + { "ad1933", AD1933 }, + { "ad1934", AD1934 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, + { "ad1938", AD193X }, + { } +}; +MODULE_DEVICE_TABLE(spi, ad193x_spi_id); + static struct spi_driver ad193x_spi_driver = { .driver = { .name = "ad193x", @@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = { }, .probe = ad193x_spi_probe, .remove = ad193x_spi_remove, + .id_table = ad193x_spi_id, }; module_spi_driver(ad193x_spi_driver); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 17c953595660..fa0a053bda58 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,6 +23,7 @@ /* codec private data */ struct ad193x_priv { struct regmap *regmap; + enum ad193x_type type; int sysclk; }; @@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, - AD193X_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, - AD193X_ADCR2_MUTE, 1, 1), - /* DAC switch control */ SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, AD193X_DACR1_MUTE, 1, 1), @@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, AD193X_DACR4_MUTE, 1, 1), + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = { + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, AD193X_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_INPUT("ADC1IN"), SND_SOC_DAPM_INPUT("ADC2IN"), }; @@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, { "DAC Output", NULL, "DAC" }, { "DAC Output", NULL, "VMID" }, - { "ADC", NULL, "SYSCLK" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC Output" }, { "DAC2OUT", NULL, "DAC Output" }, { "DAC3OUT", NULL, "DAC Output" }, { "DAC4OUT", NULL, "DAC Output" }, + { "SYSCLK", NULL, "PLL_PWR" }, +}; + +static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = { + { "ADC", NULL, "SYSCLK" }, + { "ADC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, - { "SYSCLK", NULL, "PLL_PWR" }, }; +static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) +{ + switch (ad193x->type) { + case AD1933: + case AD1934: + return false; + default: + break; + } + + return true; +} + /* * DAI ops entries */ @@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: - return -EINVAL; + if (ad193x_has_adc(ad193x)) + return -EINVAL; + break; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_SERFMT_MASK, adc_serfmt); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_FMT_MASK, adc_fmt); + if (ad193x_has_adc(ad193x)) { + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + } regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, dac_fmt); @@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_WORD_LEN_MASK, word_len); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -326,6 +354,7 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + int num, ret; /* default setting for ad193x */ @@ -335,14 +364,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* dac in tdm mode */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* high-pass filter enable */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + } + /* pll input: mclki/xi */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* add adc controls */ + num = ARRAY_SIZE(ad193x_adc_snd_controls); + ret = snd_soc_add_codec_controls(codec, + ad193x_adc_snd_controls, + num); + if (ret) + return ret; + + /* add adc widgets */ + num = ARRAY_SIZE(ad193x_adc_widgets); + ret = snd_soc_dapm_new_controls(&codec->dapm, + ad193x_adc_widgets, + num); + if (ret) + return ret; + + /* add adc routes */ + num = ARRAY_SIZE(ad193x_adc_audio_paths); + ret = snd_soc_dapm_add_routes(&codec->dapm, + ad193x_adc_audio_paths, + num); + if (ret) + return ret; + } + return 0; } @@ -367,7 +428,8 @@ const struct regmap_config ad193x_regmap_config = { }; EXPORT_SYMBOL_GPL(ad193x_regmap_config); -int ad193x_probe(struct device *dev, struct regmap *regmap) +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type) { struct ad193x_priv *ad193x; @@ -379,6 +441,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; ad193x->regmap = regmap; + ad193x->type = type; dev_set_drvdata(dev, ad193x); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index ab9a998f15be..8b1e65f928d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -13,8 +13,15 @@ struct device; +enum ad193x_type { + AD193X, + AD1933, + AD1934, +}; + extern const struct regmap_config ad193x_regmap_config; -int ad193x_probe(struct device *dev, struct regmap *regmap); +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type); #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -- 1.8.2.2

1 0

[alsa-devel] [PATCH] ALSA: pcm: Avoid double hw_free calls at releasing a stream
by Takashi Iwai 29 Sep '15

29 Sep '15

snd_pcm_release_substream() always calls hw_free op when the stream was opened. This is superfluous in most cases because it's been already released via explicit hw_free ioctl. Although this double call is usually OK as this callback should be written to be called multiple times, it's better to avoid superfluous calls. Reported-by: Vinod Koul <vinod.koul(a)intel.com> Tested-by: Jeeja Kp <jeeja.kp(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4863af5e7b26..139887011ba2 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2227,7 +2227,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->ops->hw_free != NULL) + if (substream->ops->hw_free && + substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) substream->ops->hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; -- 2.5.3

1 0

[alsa-devel] [RESEND PATCH v2 0/2] ASoC: atmel-classd: add driver for Atmel CLASSD
by Songjun Wu 29 Sep '15

29 Sep '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Changes in v2: - Change the "Mono", "Swap" and "Deemphasis" controls to "Mono Switch", "Swap Switch" and "Deemphasis Switch". - Merge "EQ Bass", "EQ Medium" and "EQ Treble" controls into one "EQ" control. - Change the "single-ended" and "differential" to "Single ended", "Differential". - Merge separate left and right controls into one single stereo control. - Add a warning if the user trys to specify an invalid value in function "atmel_classd_codec_probe". - Use the "dev_get_regmap()" in function "atmel_classd_codec_get_remap". - Check for errors when invoking the "clk_prepare_enable". - Remove the "dev_info" in function "atmel_classd_probe". - Add some code to create a sound card in function "atmel_classd_probe". - Remove the DT node "Sound" and the related code. - Delete the device node "Sound" in dts. Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier SoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 52 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 680 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 ++++ 5 files changed, 863 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

4 5

[alsa-devel] [PATCH V4 1/2] ACPI / EC: Fix broken 64bit big-endian users of 'global_lock'
by Viresh Kumar 28 Sep '15

28 Sep '15

global_lock is defined as an unsigned long and accessing only its lower 32 bits from sysfs is incorrect, as we need to consider other 32 bits for big endian 64 bit systems. There are no such platforms yet, but the code needs to be robust for such a case. Fix that by passing a local variable to debugfs_create_bool() and assigning its value to global_lock later. Signed-off-by: Viresh Kumar <viresh.kumar(a)linaro.org> --- V3->V4: - Create a local variable instead of changing type of global_lock (Rafael) - Drop the stable tag - BCC'd a lot of people (rather than cc'ing them) to make sure - the series reaches them - mailing lists do not block the patchset due to long cc list - and we don't spam the BCC'd people for every reply --- drivers/acpi/ec_sys.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/drivers/acpi/ec_sys.c b/drivers/acpi/ec_sys.c index b4c216bab22b..b44b91331a56 100644 --- a/drivers/acpi/ec_sys.c +++ b/drivers/acpi/ec_sys.c @@ -110,6 +110,7 @@ static int acpi_ec_add_debugfs(struct acpi_ec *ec, unsigned int ec_device_count) struct dentry *dev_dir; char name[64]; umode_t mode = 0400; + u32 val; if (ec_device_count == 0) { acpi_ec_debugfs_dir = debugfs_create_dir("ec", NULL); @@ -127,10 +128,11 @@ static int acpi_ec_add_debugfs(struct acpi_ec *ec, unsigned int ec_device_count) if (!debugfs_create_x32("gpe", 0444, dev_dir, (u32 *)&first_ec->gpe)) goto error; - if (!debugfs_create_bool("use_global_lock", 0444, dev_dir, - (u32 *)&first_ec->global_lock)) + if (!debugfs_create_bool("use_global_lock", 0444, dev_dir, &val)) goto error; + first_ec->global_lock = val; + if (write_support) mode = 0600; if (!debugfs_create_file("io", mode, dev_dir, ec, &acpi_ec_io_ops)) -- 2.4.0

8 27

[alsa-devel] [patch] ALSA: usb-audio: harmless underflow in snd_audigy2nx_led_put()
by Dan Carpenter 28 Sep '15

28 Sep '15

We want to verify that "value" is either zero or one, so we test if it is greater than one. Unfortunately, this is a signed int so it could also be negative. I think this is harmless but it introduces a static checker warning. Let's make "value" unsigned. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 337c317..4602a78 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -340,7 +340,7 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); struct usb_mixer_interface *mixer = list->mixer; int index = kcontrol->private_value & 0xff; - int value = ucontrol->value.integer.value[0]; + unsigned int value = ucontrol->value.integer.value[0]; int old_value = kcontrol->private_value >> 8; int err;

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix typos in snd_hdac_regmap_*() documents
by Takashi Iwai 28 Sep '15

28 Sep '15

Fixes the wrong reference names to regmap amp functions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_regmap.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index df705908480a..2767c55a641e 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -67,7 +67,7 @@ int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, * @reg: verb to write * @val: value to write * - * For writing an amp value, use snd_hda_regmap_amp_update(). + * For writing an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, @@ -85,7 +85,7 @@ snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, * @mask: bit mask to update * @val: value to update * - * For updating an amp value, use snd_hda_regmap_amp_update(). + * For updating an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid, -- 2.5.3

1 0

Re: [alsa-devel] [PATCH RFC v4 3/8] ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders
by Arnaud Pouliquen 28 Sep '15

28 Sep '15

few questions/remarks BR, Arnaud > +static void hdmi_codec_abort(struct device *dev) > +{ > + struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); > + > + dev_dbg(dev, "%s()\n", __func__); > + > + mutex_lock(&hcp->current_stream_lock); > + if (hcp->current_stream && hcp->current_stream->runtime && > + snd_pcm_running(hcp->current_stream)) { > + dev_info(dev, "HDMI audio playback aborted\n"); > + snd_pcm_stream_lock_irq(hcp->current_stream); > + snd_pcm_stop(hcp->current_stream, SNDRV_PCM_STATE_DISCONNECTED); > + snd_pcm_stream_unlock_irq(hcp->current_stream); > + } > + mutex_unlock(&hcp->current_stream_lock); > +} Does driver should stop the stream in case of unplug? This could generate unexpected behavior at application level. Perhaps should be better if this part is managed in DRM driver. if HDMI master, I2S bus should be maintained ON to consume the audio stream until application action. > + > +static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); > + struct hdmi_codec_params hp = { > + .iec = { > + .status = { 0 }, > + .subcode = { 0 }, > + .pad = 0, > + .dig_subframe = { 0 }, > + } > + }; > + int ret; > + > + dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__, > + params_width(params), params_rate(params), > + params_channels(params)); > + > + ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, > + sizeof(hp.iec.status)); Tell me if i am wrong, but in case of SPDIF format, IEC status is managed by cpu_dai not by the codec. To manage IEC61937 a control should be needed to update IEC status bits... > + if (ret < 0) { > + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", > + ret); > + return ret; > + } > + > + ret = hdmi_codec_new_stream(substream, dai); > + if (ret) > + return ret; > + > + hdmi_audio_infoframe_init(&hp.cea); > + hp.cea.channels = params_channels(params); > + hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; > + hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; > + hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; > + > + hp.sample_width = params_width(params); > + hp.sample_rate = params_rate(params); > + hp.channels = params_channels(params); > + > + return hcp->hcd.ops->hw_params(dai->dev->parent, > &hcp->daifmt[dai->id], > + &hp); > +} > +

4 3

[alsa-devel] [PATCH 0/3] Add support for Wolfson Micro WM8998 codec
by Richard Fitzgerald 28 Sep '15

28 Sep '15

This set of patches is based on broonie/regulator.git/for-next and broonie/sound.git/for-next branches. Richard Fitzgerald (3): regulator: arizona: add support for WM8998 and WM1814 ASoC: arizona: add defines for single-input gain control ASoC: wm8998: Initial WM8998 codec driver drivers/regulator/arizona-ldo1.c | 2 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 33 + sound/soc/codecs/arizona.h | 12 +- sound/soc/codecs/wm8998.c | 1429 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8998.h | 23 + 7 files changed, 1506 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h -- 1.9.1

1 3

[alsa-devel] [PATCH v2 1/1] ASoC: dwc: fix dma stop transferring issue
by yitian 28 Sep '15

28 Sep '15

Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music contintes to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally cause music playing will be stopped once XRUN occurs for the first time. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- changes in V2: - add definition for i and irq --- sound/soc/dwc/designware_i2s.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..76b2e19 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -141,13 +141,22 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { - + u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 1.7.12.4

1 0

[alsa-devel] [PATCH 1/1] ASoC: dwc: fix dma stop transferring issue
by yitian 28 Sep '15

28 Sep '15

Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music contintes to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally cause music playing will be stopped once XRUN occurs for the first time. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- sound/soc/dwc/designware_i2s.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..4d3e1d3 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -144,10 +144,19 @@ static void i2s_start(struct dw_i2s_dev *dev, i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 1.7.12.4

1 0

[alsa-devel] [PATCH v2 0/2] ASoC: atmel-classd: add driver for Atmel CLASSD
by Songjun Wu 28 Sep '15

28 Sep '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Changes in v2: - Change the "Mono", "Swap" and "Deemphasis" controls to "Mono Switch", "Swap Switch" and "Deemphasis Switch". - Merge "EQ Bass", "EQ Medium" and "EQ Treble" controls into one "EQ" control. - Change the "single-ended" and "differential" to "Single ended", "Differential". - Merge separate left and right controls into one single stereo control. - Add a warning if the user trys to specify an invalid value in function "atmel_classd_codec_probe". - Use the "dev_get_regmap()" in function "atmel_classd_codec_get_remap". - Check for errors when invoking the "clk_prepare_enable". - Remove the "dev_info" in function "atmel_classd_probe". - Add some code to create a sound card in function "atmel_classd_probe". - Remove the DT node "Sound" and the related code. - Delete the device node "Sound". Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier ASoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 52 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 680 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 ++++ 5 files changed, 863 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

3 3

[alsa-devel] [PATCH] mfd: arizona: Call the runtime PM function if the state is runtime resumed
by Inha Song 27 Sep '15

27 Sep '15

This patch call runtiem PM function manually if the state is runtime resumed before enter suspend. This case has occurred when wake-up by the resume after entering to suspend during the playback. $ aplay -> runtime_resume() -> suspend() -> resume() In this case, Can occurred an error when enter suspend. -- [ 82.559234] PM: suspend of devices complete after 57.252 msecs [ 82.567978] arizona spi1.0: Failed to read IRQ status: -108 [ 82.567989] arizona spi1.0: Failed to read main IRQ status: -108 [ 82.568027] arizona spi1.0: Failed to read IRQ status: -108 [ 82.568036] arizona spi1.0: Failed to read main IRQ status: -108 .... So, we should call runtime_suspend() manually when enter suspend if the state is runtime resume. Also should call runtime_resume() when wake-up by resume if the state was runtime resume before entering suspend. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/arizona-core.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index 908c69b..de55706 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -683,6 +683,10 @@ static int arizona_suspend(struct device *dev) struct arizona *arizona = dev_get_drvdata(dev); dev_dbg(arizona->dev, "Suspend, disabling IRQ\n"); + + if (!pm_runtime_status_suspended(dev)) + arizona_runtime_suspend(dev); + disable_irq(arizona->irq); return 0; @@ -713,6 +717,10 @@ static int arizona_resume(struct device *dev) struct arizona *arizona = dev_get_drvdata(dev); dev_dbg(arizona->dev, "Late resume, reenabling IRQ\n"); + + if (!pm_runtime_status_suspended(dev)) + arizona_runtime_resume(dev); + enable_irq(arizona->irq); return 0; -- 2.0.0.390.gcb682f8

2 13

[alsa-devel] Baytrail based HDMI Stick Orbsmart AW-01: No sound
by Michael Leun 26 Sep '15

26 Sep '15

Hi, I've an baytrail / Z3735F based HDMI Stick "Orbsmart AW-01", but sound does not work. I'm a little lost finding out what is missing / required to make it work. I compiled an 4.3-rc2 vanilla kernel with hopefully each and everything turned on relating soc/sound. In dmesg I get: [...] [ 4.252522] sst-acpi 80860F28:00: No matching ASoC machine driver found [ 5.863177] intel_sst_acpi 80860F28:00: No matching machine driver found [...] It looks like there is rt5640 based sound recognized: [ml@orb ~]$ find /sys -name '*5640*' -print /sys/bus/i2c/drivers/rt5640 /sys/bus/acpi/devices/10EC5640:00 /sys/bus/acpi/devices/10EC5640:01 /sys/devices/LNXSYSTM:00/LNXSYBUS:00/80860F41:01/10EC5640:00 /sys/devices/LNXSYSTM:00/LNXSYBUS:00/80860F41:01/10EC5640:01 find: "/sys/kernel/debug": Keine Berechtigung /sys/module/i2c_core/holders/snd_soc_rt5640 /sys/module/snd_soc_rl6231/holders/snd_soc_rt5640 /sys/module/snd_soc_rt5640 /sys/module/snd_soc_rt5640/drivers/i2c:rt5640 /sys/module/snd_soc_core/holders/snd_soc_rt5640 /sys/module/regmap_i2c/holders/snd_soc_rt5640 /sys/module/snd_pcm/holders/snd_soc_rt5640 [ml@orb ~]$ This are the sound related modules loaded: [ml@orb ~]$ lsmod|grep snd snd_intel_sst_acpi 16384 0 snd_intel_sst_core 65536 1 snd_intel_sst_acpi snd_soc_rt5640 106496 0 snd_soc_sst_mfld_platform 73728 1 snd_intel_sst_core snd_soc_rl6231 16384 1 snd_soc_rt5640 snd_soc_core 155648 2 snd_soc_rt5640,snd_soc_sst_mfld_platform snd_compress 20480 1 snd_soc_core regmap_i2c 16384 1 snd_soc_rt5640 snd_pcm_dmaengine 16384 1 snd_soc_core ac97_bus 16384 1 snd_soc_core snd_pcm 86016 4 snd_soc_rt5640,snd_soc_core,snd_soc_sst_mfld_platform,snd_pcm_dmaengine snd_timer 28672 1 snd_pcm snd 65536 5 snd_soc_core,snd_timer,snd_pcm,snd_soc_sst_mfld_platform,snd_compress soundcore 16384 1 snd i2c_core 49152 7 drm,i915,snd_soc_rt5640,i2c_designware_platform,regmap_i2c,drm_kms_helper,i2c_algo_bit snd_soc_sst_acpi 16384 0 [ml@orb ~]$ -- Best Regards, Michael Leun

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: fix bad register MIC_BIAS_VOLTAGE setup on probe
by Gianluca Renzi 25 Sep '15

25 Sep '15

Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Gianluca Renzi <gianlucarenzi(a)eurekelettronica.it> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff -up a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c --- a/sound/soc/codecs/sgtl5000.c 2015-09-25 15:24:10.597051191 +0200 +++ b/sound/soc/codecs/sgtl5000.c 2015-09-25 15:24:21.000000000 +0200 @@ -1377,7 +1377,7 @@ static int sgtl5000_probe(struct snd_soc snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, SGTL5000_BIAS_R_MASK, - sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT); + sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT); /* * disable DAP * TODO: @@ -1549,7 +1549,7 @@ static int sgtl5000_i2c_probe(struct i2c else { sgtl5000->micbias_voltage = 0; dev_err(&client->dev, - "Unsuitable MicBias resistor\n"); + "Unsuitable MicBias voltage\n"); } } else { sgtl5000->micbias_voltage = 0;

3 2

[alsa-devel] ASoC: please revert "ASoC: db1200: Use static DAI format setup"
by Manuel Lauss 25 Sep '15

25 Sep '15

Hi Lars, commit b4508d0f95fa4aaed889549e31391641d675d4bb ("ASoC: db1200: Use static DAI format setup") breaks I2S in that the set_dai_fmt() callbacks of neither the codec (wm8731) nor the I2S block (psc-i2s) are called. Please revert this change or fix it up so that the set_dai_fmt() callbacks are at least called once. (This was broken since 4.0, but I just now got around to test linux on the hardware again). Thanks! Manuel -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

3 3

[alsa-devel] [GIT PULL] ASoC updates for v4.4
by Mark Brown 25 Sep '15

25 Sep '15

The following changes since commit 1f93e4a96c9109378204c147b3eec0d0e8100fde: Linux 4.3-rc2 (2015-09-20 14:32:34 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.3-rc2 for you to fetch changes up to 4922e7a1b767fcf6f29ce121ad74e424f4045eff: Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next (2015-09-23 11:01:30 -0700) ---------------------------------------------------------------- ASoC: Updates for v4.4 A first batch of updates targetted at v4.4. There are no substantial core fixes here, the biggest block of changes is updates to the rcar drivers and the addition of a CODEC driver for the AK4613. ---------------------------------------------------------------- Adam Thomson (1): ASoC: fsl_ssi: Fix checking of dai format for AC97 mode Anatol Pomozov (1): ASoC: Document snd-soc-dummy-dai purpose Arnaud Pouliquen (1): ASoC: sti: check return of of_property_read Axel Lin (11): ASoC: mediatek: Convert to devm_snd_soc_register_card ASoC: au1x: Convert to devm_snd_soc_register_card ASoC: blackfin: Convert to devm_snd_soc_register_card ASoC: mxs-sgtl5000: Convert to devm_snd_soc_register_card ASoC: pxa: Convert to devm_snd_soc_register_card ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning ASoC: sh: siu_dai: Convert to use resource managed APIs ASoC: intel: broadwell: Convert to devm_snd_soc_register_card ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM ASoC: rockchip: Fix missing CLKDEV_LOOKUP dependency Charles Keepax (4): ASoC: arizona: Add default cases for event switches ASoC: arizona: Add utility function to check if an input is analog ASoC: Add SOC_SINGLE_RANGE_EXT_TLV macro ASoC: wm5110: Add additional analogue input enable for early revs Emilio López (1): ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs Jarkko Nikula (1): spi: atmel: Remove needless bits_per_word and speed_hz tests Jeeja KP (1): ASoC: Intel: Skylake: Remove unused CPU dais John Lin (3): ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome platform ASoC: rt5645: Add dmi for Broadwell ASoC: rt5645: Add jd_invert for Broadwell Julia Lawall (1): ASoC: qcom: change PTR_ERR argument Jyri Sarha (7): ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver ASoC: tlv320aic3x: Improve tdm support ASoC: davinci-mcasp: Add set_tdm_slots() support ASoC: simple-card: Add tdm slot mask support to simple-card ASoC: soc-core: Fix sparse warning in be32_to_cpup() call ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data ASoC: hdmi: Remove obsolete dummy HDMI codec Koro Chen (1): ASoC: mediatek: Increase periods_min in capture Kuninori Morimoto (20): ASoC: ak4642: use *dev on ak4642_i2c_probe() ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT ASoC: rsnd: tidyup rsnd_dma_to_xxx() position ASoC: rsnd: rename rsnd_src_pcm_new() to rsnd_src_pcm_new_gen2() ASoC: rsnd: remove unneeded sh_clk header ASoC: rsnd: add missing #sound-dai-cells explain on Document ASoC: rsnd: Add Gen3 initial support ASoC: rsnd: add rsnd_mod_get() macro and use it ASoC: rsnd: ADG uses mod base common method ASoC: rsnd: add common mod confirm method ASoC: rsnd: ADG initial setup on rsnd_adg_ssi_clk_init() ASoC: rsnd: tidyup ADG clock calculate method ASoC: rsnd: tidyup ADG debug message for clock selection ASoC: rsnd: adg: ignore undefined clock error ASoC: rsnd: add AUDIO_CLKOUT support ASoC: rsnd: SND_SOC_RCAR depends on COMMON_CLK ASoC: ak4642: exchange macro name to avoid redefinition ASoC: rsnd: fixup clkout_name[] index error for single AUDIO_CLKOUT SoC ASoC: add ak4613 support ASoC: ak4613: add Digital Playback Volume support Liam Girdwood (1): ASoC: intel: Fix SSP port configuration after RTD3 resume. Luis de Bethencourt (8): ASoC: atmel_wm8904: Fix module autoload for OF platform driver ASoC: fsl-asoc-card: Fix module autoload for OF platform driver ASoC: fsl_sai: Fix module autoload for OF platform driver ASoC: jz4740: Fix module autoload for OF platform driver ASoC: kirkwood: Fix module autoload for OF platform driver ASoC: pxa: Fix module autoload for OF platform drivers ASoC: mop500: Fix module autoload for OF platform driver ASoC: ux500: Fix module autoload for OF platform driver Maciej S. Szmigiero (1): ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device Mark Brown (14): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-fix-rt5645 Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/doc', 'asoc/fix/fsl-card', 'asoc/fix/fsl-ssi', 'asoc/fix/intel' and 'asoc/fix/maintainers' into asoc-linus Merge remote-tracking branches 'asoc/fix/mtk', 'asoc/fix/psc', 'asoc/fix/pxa', 'asoc/fix/spear', 'asoc/fix/sti' and 'asoc/fix/wm0010' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4613', 'asoc/topic/ak4642', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next Merge remote-tracking branches 'asoc/topic/blackfin', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/hdmi' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/jz4740', 'asoc/topic/kirkwood', 'asoc/topic/mtk', 'asoc/topic/multi' and 'asoc/topic/mxs' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom', 'asoc/topic/rcar' and 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branches 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/spear', 'asoc/topic/sunxi' and 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Maxime Ripard (1): ASoC: sunxi: Add the Allwinner A10 codec bindings Oder Chiou (5): ASoC: rt5645: Remove incorrect settings ASoC: rt5645: Remove the repeated definitions ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up ASoC: rt5645: Increase the delay time to remove the pop sound ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging Peter Ujfalusi (2): ASoC: davinci-mcasp: Revise the FIFO threshold calculation ASoC: davinci-mcasp: Fix devm_kasprintf format string Ricard Wanderlof (1): ASoC: Handle multiple codecs with split playback / capture Richard Fitzgerald (1): MAINTAINERS: Update website and git repo for Wolfson Microelectronics Robert Jarzmik (2): ASoC: fix broken pxa SoC support ASoC: pxa: pxa2xx-ac97: fix dma requestor lines Sudip Mukherjee (5): ASoC: wm0010: fix memory leak ASoC: dapm: fix memory leak ASoC: Intel: remove unused function ASoC: wm0010: fix memory leak ASoC: wm0010: fix error path Zidan Wang (4): ASoC: wm8960: correct the min gain value of some PGA ASoC: wm8960: correct gain value for input PGA and add microphone PGA ASoC: wm8962: remove 64k sample rate support ASoC: wm8960: correct the max register value of mic boost pga kbuild test robot (1): ASoC: sunxi: fix platform_no_drv_owner.cocci warnings Documentation/devicetree/bindings/sound/ak4613.txt | 17 + Documentation/devicetree/bindings/sound/ak4642.txt | 22 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 7 + .../devicetree/bindings/sound/sun4i-codec.txt | 33 + .../devicetree/bindings/sound/tdm-slot.txt | 11 +- MAINTAINERS | 9 +- drivers/spi/spi-atmel.c | 21 +- include/sound/rcar_snd.h | 1 - include/sound/rt5645.h | 2 + include/sound/simple_card.h | 2 + include/sound/soc.h | 14 + sound/arm/Kconfig | 15 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/atmel_wm8904.c | 1 + sound/soc/au1x/db1000.c | 10 +- sound/soc/au1x/db1200.c | 10 +- sound/soc/au1x/psc-i2s.c | 1 - sound/soc/blackfin/bf5xx-ad1836.c | 11 +- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +- sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/ak4613.c | 497 ++++++++++++++ sound/soc/codecs/ak4642.c | 153 +++-- sound/soc/codecs/arizona.c | 16 + sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/hdmi.c | 109 ---- sound/soc/codecs/rt5645.c | 57 +- sound/soc/codecs/rt5645.h | 6 +- sound/soc/codecs/tlv320aic3x.c | 30 +- sound/soc/codecs/wm0010.c | 23 +- sound/soc/codecs/wm5110.c | 187 +++++- sound/soc/codecs/wm8960.c | 26 +- sound/soc/codecs/wm8962.c | 3 +- sound/soc/davinci/davinci-mcasp.c | 319 +++++---- sound/soc/fsl/fsl-asoc-card.c | 4 +- sound/soc/fsl/fsl_sai.c | 1 + sound/soc/fsl/fsl_ssi.c | 5 +- sound/soc/generic/simple-card.c | 8 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 17 - sound/soc/intel/boards/broadwell.c | 9 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 20 +- sound/soc/intel/skylake/skl-pcm.c | 33 - sound/soc/jz4740/jz4740-i2s.c | 1 + sound/soc/kirkwood/armada-370-db.c | 1 + sound/soc/mediatek/mt8173-max98090.c | 11 +- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 11 +- sound/soc/mediatek/mtk-afe-pcm.c | 17 + sound/soc/mxs/mxs-sgtl5000.c | 6 +- sound/soc/pxa/Kconfig | 2 - sound/soc/pxa/brownstone.c | 9 +- sound/soc/pxa/corgi.c | 11 +- sound/soc/pxa/e740_wm9705.c | 5 +- sound/soc/pxa/e750_wm9705.c | 5 +- sound/soc/pxa/e800_wm9712.c | 5 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 11 +- sound/soc/pxa/mioa701_wm9713.c | 11 +- sound/soc/pxa/palm27x.c | 9 +- sound/soc/pxa/poodle.c | 11 +- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 4 +- sound/soc/pxa/pxa2xx-pcm.c | 1 + sound/soc/pxa/spitz.c | 5 +- sound/soc/pxa/tosa.c | 5 +- sound/soc/pxa/ttc-dkb.c | 12 +- sound/soc/qcom/lpass-cpu.c | 3 +- sound/soc/rockchip/Kconfig | 4 +- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/adg.c | 303 +++++++-- sound/soc/sh/rcar/core.c | 12 + sound/soc/sh/rcar/ctu.c | 6 +- sound/soc/sh/rcar/dvc.c | 6 +- sound/soc/sh/rcar/mix.c | 6 +- sound/soc/sh/rcar/rsnd.h | 15 +- sound/soc/sh/rcar/src.c | 17 +- sound/soc/sh/rcar/ssi.c | 98 +-- sound/soc/sh/siu_dai.c | 85 +-- sound/soc/soc-core.c | 25 + sound/soc/soc-dapm.c | 2 +- sound/soc/soc-pcm.c | 49 ++ sound/soc/soc-utils.c | 9 + sound/soc/spear/Kconfig | 2 +- sound/soc/sti/uniperif_player.c | 14 +- sound/soc/sti/uniperif_reader.c | 6 +- sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-codec.c | 719 +++++++++++++++++++++ sound/soc/ux500/mop500.c | 1 + sound/soc/ux500/ux500_msp_dai.c | 1 + 92 files changed, 2484 insertions(+), 834 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt create mode 100644 sound/soc/codecs/ak4613.c delete mode 100644 sound/soc/codecs/hdmi.c create mode 100644 sound/soc/sunxi/Kconfig create mode 100644 sound/soc/sunxi/Makefile create mode 100644 sound/soc/sunxi/sun4i-codec.c

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[alsa-devel] [PATCH v2] sound: oss: ad1848: Fix returned errno code in ad1848_init()
by Luis de Bethencourt 25 Sep '15

25 Sep '15

The driver is using -1 instead of the -ENOMEM defined macro to specify that a buffer allocation failed. Since the error number is propagated, the caller will get a -EPERM which is the wrong error condition. Changed the checks of the return of ad1848_init() to be for >= 0, instead of != -1. Smatch tool warning: ad1848_init() warn: returning -1 instead of -ENOMEM is sloppy Signed-off-by: Luis de Bethencourt <luisbg(a)osg.samsung.com> --- sound/oss/ad1848.c | 4 ++-- sound/oss/dev_table.c | 2 +- sound/oss/pss.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 10c8de1..58c6d31 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -1992,7 +1992,7 @@ int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback, portc = kmalloc(sizeof(ad1848_port_info), GFP_KERNEL); if(portc==NULL) { release_region(devc->base, 4); - return -1; + return -ENOMEM; } if ((my_dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, @@ -2007,7 +2007,7 @@ int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback, { release_region(devc->base, 4); kfree(portc); - return -1; + return my_dev; } audio_devs[my_dev]->portc = portc; diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c index 6dad515..abb3db4 100644 --- a/sound/oss/dev_table.c +++ b/sound/oss/dev_table.c @@ -148,7 +148,7 @@ EXPORT_SYMBOL(sound_install_mixer); void sound_unload_audiodev(int dev) { - if (dev != -1) { + if (dev >= 0) { DMAbuf_deinit(dev); audio_devs[dev] = NULL; unregister_sound_dsp((dev<<4)+3); diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 81314f9..e84aafe 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -1091,7 +1091,7 @@ static int __init probe_pss_mss(struct address_info *hw_config) pss_mixer_reset(devc); attach_ms_sound(hw_config, ports, THIS_MODULE); /* Slot 0 */ - if (hw_config->slots[0] != -1) + if (hw_config->slots[0] >= 0) { /* The MSS driver installed itself */ audio_devs[hw_config->slots[0]]->coproc = &pss_coproc_operations; -- 2.5.1

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[alsa-devel] [PATCH] ASoC: db1200: Fix DAI link format for db1300 and db1550
by Lars-Peter Clausen 25 Sep '15

25 Sep '15

Commit b4508d0f95fa ("ASoC: db1200: Use static DAI format setup") switched the db1200 driver over to using static DAI format setup instead of a callback function. But the commit only added the dai_fmt field to one of the three DAI links in the driver. This breaks audio on db1300 and db1550. Add the two missing dai_fmt settings to fix the issue. Fixes: b4508d0f95fa ("ASoC: db1200: Use static DAI format setup") Reported-by: Manuel Lauss <manuel.lauss(a)gmail.com> Tested-by: Manuel Lauss <manuel.lauss(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/au1x/db1200.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 638ca0b..5c73061 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -129,6 +129,8 @@ static struct snd_soc_dai_link db1300_i2s_dai = { .cpu_dai_name = "au1xpsc_i2s.2", .platform_name = "au1xpsc-pcm.2", .codec_name = "wm8731.0-001b", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &db1200_i2s_wm8731_ops, }; @@ -146,6 +148,8 @@ static struct snd_soc_dai_link db1550_i2s_dai = { .cpu_dai_name = "au1xpsc_i2s.3", .platform_name = "au1xpsc-pcm.3", .codec_name = "wm8731.0-001b", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &db1200_i2s_wm8731_ops, }; -- 2.1.4

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[alsa-devel] Missing snd_pcm_link() for the rate plugin
by Christian Gruber 25 Sep '15

25 Sep '15

Dear ALSA community, does anybody know, why the rate plugin does not support the snd_pcm_link() function? As I can see from the ALSA-Lib source code, snd_pcm_link() calls the fast_op "link", which is a function pointer, whose destination depends on the type of pcm device/plugin to be linked. If the device is a HW-device for instance, the function snd_pcm_hw_link() is called. Several other plugins (e.g. mmap_emul, linear, lfloat, mulaw, alaw, route) use the generic function snd_pcm_generic_link(), which forwards the function call to their slave devices/plugins. The rate plugin instead does not support snd_pcm_link(), since the function pointer "link" is not assigned in the definition of snd_pcm_rate_fast_ops. Is their any reason to not use the generic function snd_pcm_generic_link() within the rate plugin? Thanks in advance for answering this question, Christian

2 1

[alsa-devel] sunxi, Allwinner TDM Support?
by Caleb Crome 25 Sep '15

25 Sep '15

Hi, I was wondering if the sunxi parts support TDM mode on their DAI ports? Those parts sure have attractive pricing, and as you may have gathered, I'm all about many microphone channels. :-) Thanks, -Caleb

1 0

[alsa-devel] RT5640 jack detection
by Carlo Caione 25 Sep '15

25 Sep '15

Hi, I'm working with the ALC5640-VB audio codec based on the Realtek RT5640. On my setup the codec is only used for audio output through an headphone jack, otherwise the audio should be routed via HDMI. The problem is that the jack sensing only works when the codec is actively playing something, that is when the bias set for the codec is not SND_SOC_BIAS_OFF. When the codec is suspended and it is powered off I'm not able to detect anymore the presence of the jack. Because of this issue actively switching the audio from HDMI to HP when the jack is inserted is not working. The driver I'm using for the RT5640 in the .set_bias_level hook function among other things is gating the I2S input clock and this seems to prevent the generation of the IRQ signal I'm using to detect the presence/absence of the jack (the culprit here is MX-FA[0] bit for those interested). I can easily modify the codec driver to avoid gating the I2S clock when the bias is off but this seems in contrast with the whole DAPM principle. How this is usually supposed to be working? -- Carlo Caione | +39.340.80.30.096 | Endless

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[alsa-devel] [PATCH 0/5] Add ASoC support for AMD APUs [v3]
by Alex Deucher 25 Sep '15

25 Sep '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create mfd cells which we hang the i2s codec and DMA engine on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review V3 changes: - Update the designware driver to handle slave mode, amd specific features - Use the designware driver directly for i2s - Move the DMA handling from the GPU driver into the AMD ASoC driver - Change the license on the ASoC driver to GPL Patch 4 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream3 Thanks, Alex Maruthi Bayyavarapu (1): drm/amd: add ACP driver support Maruthi Srinivas Bayyavarapu (4): ASoC : dwc : support dw i2s in slave mode ASoC : dwc : support dw i2s in AMD platform ASoC : AMD : add ACP 2.2 register headers ASoC: AMD: add AMD ASoC ACP-I2S driver drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 10 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 127 ++ drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 269 +++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 41 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + include/linux/mfd/amd_acp.h | 43 + include/sound/designware_i2s.h | 4 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 4 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 518 ++++++ sound/soc/amd/acp.c | 736 +++++++++ sound/soc/amd/acp.h | 147 ++ sound/soc/amd/include/acp_2_2_d.h | 609 +++++++ sound/soc/amd/include/acp_2_2_enum.h | 1068 ++++++++++++ sound/soc/amd/include/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++++++++ sound/soc/dwc/designware_i2s.c | 270 +-- 24 files changed, 6141 insertions(+), 100 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 include/linux/mfd/amd_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp.c create mode 100644 sound/soc/amd/acp.h create mode 100644 sound/soc/amd/include/acp_2_2_d.h create mode 100644 sound/soc/amd/include/acp_2_2_enum.h create mode 100644 sound/soc/amd/include/acp_2_2_sh_mask.h -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: rl6347a: Clean up unneeded inclusion of header files
by Axel Lin 24 Sep '15

24 Sep '15

Also move the include of sound/hda_verbs.h to rl6347a.h because it is used in rl6347a.h. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rl6347a.c | 19 +------------------ sound/soc/codecs/rl6347a.h | 2 ++ sound/soc/codecs/rt286.c | 1 - sound/soc/codecs/rt298.c | 1 - 4 files changed, 3 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/rl6347a.c b/sound/soc/codecs/rl6347a.c index 91d5166..a4b910e 100644 --- a/sound/soc/codecs/rl6347a.c +++ b/sound/soc/codecs/rl6347a.c @@ -11,25 +11,8 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/delay.h> -#include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> -#include <linux/spi/spi.h> -#include <linux/dmi.h> -#include <linux/acpi.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/initval.h> -#include <sound/tlv.h> -#include <sound/jack.h> -#include <linux/workqueue.h> -#include <sound/hda_verbs.h> +#include <linux/regmap.h> #include "rl6347a.h" diff --git a/sound/soc/codecs/rl6347a.h b/sound/soc/codecs/rl6347a.h index 1cb56e5..e127919 100644 --- a/sound/soc/codecs/rl6347a.h +++ b/sound/soc/codecs/rl6347a.h @@ -12,6 +12,8 @@ #ifndef __RL6347A_H__ #define __RL6347A_H__ +#include <sound/hda_verbs.h> + #define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) #define RL6347A_VENDOR_REGISTERS 0x20 diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index bd93658..1fbdb4f 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -29,7 +29,6 @@ #include <sound/jack.h> #include <linux/workqueue.h> #include <sound/rt286.h> -#include <sound/hda_verbs.h> #include "rl6347a.h" #include "rt286.h" diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 3c2f0f8..2a6c2ba 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -28,7 +28,6 @@ #include <sound/jack.h> #include <linux/workqueue.h> #include <sound/rt298.h> -#include <sound/hda_verbs.h> #include "rl6347a.h" #include "rt298.h" -- 2.1.4

2 1

Re: [alsa-devel] ALSA: pcm_direct.c SND_PCM_DIRECT_MAGIC
by Takashi Iwai 24 Sep '15

24 Sep '15

On Wed, 23 Sep 2015 18:07:46 +0200, Mike Bjorge wrote: > > I see, interesting. > > I am running into an issue where pcm_dmix.c is occasionally failing with an > "unable to create IPC shm instance" (pcm_dmix.c@1004) because in > pcm_direct.c@134, (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) when > buf.shm_nattch == 2. > > Do you know if there are any likely causes that would result in > dmix->shmptr->magic not being set correctly? What value is read there? If the value is completely different, it means that some other programs use the same shmem for other purpose. It'd be just a conflict of shmem key. Takashi > > Thanks for your help, > Mike > > > > On Wed, Sep 23, 2015 at 6:11 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 19 Sep 2015 01:13:42 +0200, > > Mike Bjorge wrote: > > > > > > Hi Takashi, > > > > > > I know this is a bit out of the blue, but I was wondering if you could > > > explain the point of SND_PCM_DIRECT_MAGIC and dmix->shmptr->magic? > > > > > > Specifically, what it's doing in pcm_direct.c from this commit? > > > > > http://git.alsa-project.org/?p=alsa-lib.git;a=blobdiff;f=src/pcm/pcm_direct… > > > > It's to check the consistency of shared data via shmem. > > If we change the data structure in an incompatible way, we'd change > > the magic key so that each instance can detect it. > > > > > > Takashi > > > [2 <text/html; UTF-8 (quoted-printable)>] >

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.3
by Mark Brown 24 Sep '15

24 Sep '15

The following changes since commit 1f93e4a96c9109378204c147b3eec0d0e8100fde: Linux 4.3-rc2 (2015-09-20 14:32:34 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.3-rc2 for you to fetch changes up to ed14ee0eea8b6808025356cecc87a8007885263f: Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus (2015-09-23 11:01:12 -0700) ---------------------------------------------------------------- ASoC: Fixes for v4.3 A disappointingly large set of fixes, though none of them very big and very widely spread over many different drivers. Nothing especially stands out, it's mostly all device specific and relatively minor. ---------------------------------------------------------------- Adam Thomson (1): ASoC: fsl_ssi: Fix checking of dai format for AC97 mode Anatol Pomozov (1): ASoC: Document snd-soc-dummy-dai purpose Arnaud Pouliquen (1): ASoC: sti: check return of of_property_read Axel Lin (2): ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning John Lin (1): ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome platform Jyri Sarha (1): ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver Koro Chen (1): ASoC: mediatek: Increase periods_min in capture Liam Girdwood (1): ASoC: intel: Fix SSP port configuration after RTD3 resume. Maciej S. Szmigiero (1): ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device Mark Brown (6): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-fix-rt5645 Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/doc', 'asoc/fix/fsl-card', 'asoc/fix/fsl-ssi', 'asoc/fix/intel' and 'asoc/fix/maintainers' into asoc-linus Merge remote-tracking branches 'asoc/fix/mtk', 'asoc/fix/psc', 'asoc/fix/pxa', 'asoc/fix/spear', 'asoc/fix/sti' and 'asoc/fix/wm0010' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus Oder Chiou (4): ASoC: rt5645: Remove incorrect settings ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up ASoC: rt5645: Increase the delay time to remove the pop sound ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging Peter Ujfalusi (2): ASoC: davinci-mcasp: Revise the FIFO threshold calculation ASoC: davinci-mcasp: Fix devm_kasprintf format string Richard Fitzgerald (1): MAINTAINERS: Update website and git repo for Wolfson Microelectronics Robert Jarzmik (2): ASoC: fix broken pxa SoC support ASoC: pxa: pxa2xx-ac97: fix dma requestor lines Sudip Mukherjee (4): ASoC: wm0010: fix memory leak ASoC: dapm: fix memory leak ASoC: wm0010: fix memory leak ASoC: wm0010: fix error path Zidan Wang (4): ASoC: wm8960: correct the min gain value of some PGA ASoC: wm8960: correct gain value for input PGA and add microphone PGA ASoC: wm8962: remove 64k sample rate support ASoC: wm8960: correct the max register value of mic boost pga MAINTAINERS | 9 +++------ sound/arm/Kconfig | 15 ++++++++------- sound/soc/au1x/psc-i2s.c | 1 - sound/soc/codecs/rt5645.c | 22 +++++++++++++--------- sound/soc/codecs/wm0010.c | 23 +++++++++-------------- sound/soc/codecs/wm8960.c | 26 ++++++++++++++++++-------- sound/soc/codecs/wm8962.c | 3 ++- sound/soc/davinci/davinci-mcasp.c | 14 +++++++------- sound/soc/fsl/fsl-asoc-card.c | 3 ++- sound/soc/fsl/fsl_ssi.c | 5 +++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 20 ++++++++++++-------- sound/soc/mediatek/mtk-afe-pcm.c | 17 +++++++++++++++++ sound/soc/pxa/Kconfig | 2 -- sound/soc/pxa/pxa2xx-ac97.c | 4 ++-- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-utils.c | 9 +++++++++ sound/soc/spear/Kconfig | 2 +- sound/soc/sti/uniperif_player.c | 14 ++++++++++---- sound/soc/sti/uniperif_reader.c | 6 +++++- 19 files changed, 122 insertions(+), 75 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: mediatek: Increase periods_min in capture
by Koro Chen 24 Sep '15

24 Sep '15

In capture, there is chance that hw_ptr reported at IRQ is a little smaller than period_size due to internal AFE buffer. In the case of ping-pong buffer: |xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------| hw_ptr < period_size This available buffer will not be read since its size is smaller than avail_min (which is period_size by default), and read thread continues to sleep. If the next hw_ptr is just a little larger than buffer_size, overrun occurs. One more period can hold the possible unread buffer. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index d190fe0..f5baf3c 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -549,6 +549,23 @@ static int mtk_afe_dais_startup(struct snd_pcm_substream *substream, memif->substream = substream; snd_soc_set_runtime_hwparams(substream, &mtk_afe_hardware); + + /* + * Capture cannot use ping-pong buffer since hw_ptr at IRQ may be + * smaller than period_size due to AFE's internal buffer. + * This easily leads to overrun when avail_min is period_size. + * One more period can hold the possible unread buffer. + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + ret = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIODS, + 3, + mtk_afe_hardware.periods_max); + if (ret < 0) { + dev_err(afe->dev, "hw_constraint_minmax failed\n"); + return ret; + } + } ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) -- 1.7.9.5

3 6

[alsa-devel] Applied "spi: atmel: Remove needless bits_per_word and speed_hz tests" to the asoc tree
by Mark Brown 24 Sep '15

24 Sep '15

The patch spi: atmel: Remove needless bits_per_word and speed_hz tests 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 3f8958cf9aaaac8a2bfc7417cb87db757eeb4465 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Date: Tue, 15 Sep 2015 16:26:22 +0300 Subject: [PATCH] spi: atmel: Remove needless bits_per_word and speed_hz tests SPI core validates both bits_per_word and speed_hz transfer parameters and defaults to spi->bits_per_word and spi->max_speed_hz in case these per transfer parameters are not set. This makes possible to remove two if statements and remove one code block that is never executed. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/spi/spi-atmel.c | 21 ++++++--------------- 1 file changed, 6 insertions(+), 15 deletions(-) diff --git a/drivers/spi/spi-atmel.c b/drivers/spi/spi-atmel.c index bf9ed38..1cc1f1e 100644 --- a/drivers/spi/spi-atmel.c +++ b/drivers/spi/spi-atmel.c @@ -871,14 +871,7 @@ static int atmel_spi_set_xfer_speed(struct atmel_spi *as, * Calculate the lowest divider that satisfies the * constraint, assuming div32/fdiv/mbz == 0. */ - if (xfer->speed_hz) - scbr = DIV_ROUND_UP(bus_hz, xfer->speed_hz); - else - /* - * This can happend if max_speed is null. - * In this case, we set the lowest possible speed - */ - scbr = 0xff; + scbr = DIV_ROUND_UP(bus_hz, xfer->speed_hz); /* * If the resulting divider doesn't fit into the @@ -1300,14 +1293,12 @@ static int atmel_spi_one_transfer(struct spi_master *master, return -EINVAL; } - if (xfer->bits_per_word) { - asd = spi->controller_state; - bits = (asd->csr >> 4) & 0xf; - if (bits != xfer->bits_per_word - 8) { - dev_dbg(&spi->dev, + asd = spi->controller_state; + bits = (asd->csr >> 4) & 0xf; + if (bits != xfer->bits_per_word - 8) { + dev_dbg(&spi->dev, "you can't yet change bits_per_word in transfers\n"); - return -ENOPROTOOPT; - } + return -ENOPROTOOPT; } /* -- 2.5.0

2 2

[alsa-devel] [PATCH RFC v4 0/8] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 24 Sep '15

24 Sep '15

Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 The binding for tda998x is taken from Jean Francois' patch series[1] on the same subject. The implementation of the of-node parsing has some minor changes from my self. Here is what I think at least could or should still be done, but non of that stuff does not sounds critical right now. Missing from tda998x driver side - hdmi_codec_ops audio_startup() implementation for audio abort support - multi channel audio support (I would need specs and preferably some HW to test for this). Missing from ASoC side generic implementation: - channel_allocation handling is completely left for the video side driver, see if ASoC side could help in any way - snd_soc_jack functionality to handle hdmi cable plug/unplug events [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095596.html Best regards, Jyri Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports Jyri Sarha (7): ASoC: hdmi: Remove obsolete dummy HDMI codec ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders drm/i2c: tda998x: Remove include/sound/tda998x.h and fix graph parsing drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++ arch/arm/boot/dts/am335x-boneblack.dts | 90 ++++- drivers/gpu/drm/armada/armada_drv.c | 25 +- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 319 +++++++++++++--- include/drm/i2c/tda998x.h | 24 +- include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 52 ++- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/hdmi-codec.c | 404 +++++++++++++++++++++ sound/soc/codecs/hdmi.c | 109 ------ 13 files changed, 985 insertions(+), 204 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c delete mode 100644 sound/soc/codecs/hdmi.c -- 1.9.1

6 20

[alsa-devel] [PATCH] ALSA: hda - Disable power_save_node for Thinkpads
by Takashi Iwai 24 Sep '15

24 Sep '15

Lenovo Thinkpads with recent Realtek codecs seem suffering from click noises at power transition since the introduction of widget power saving in 4.1 kernel. Although this might be solved by some delays in appropriate points, as a quick workaround, just disable the power_save_node feature for now. The gain it gives is relatively small, and this makes the situation back to pre 4.1 time. This patch ended up with a bit more code changes than usual because the existing fixup for Thinkpads is highly chained. Instead of adding yet another chain, combine a few of them into a single fixup entry, as a gratis cleanup. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=943982 Cc: <stable(a)vger.kernel.org> # v4.1+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a75b5611d1e4..afec6dc9f91f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4188,6 +4188,24 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec, } } +/* fixup for Thinkpad docks: add dock pins, avoid HP parser fixup */ +static void alc_fixup_tpt440_dock(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x16, 0x21211010 }, /* dock headphone */ + { 0x19, 0x21a11010 }, /* dock mic */ + { } + }; + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; + codec->power_save_node = 0; /* avoid click noises */ + snd_hda_apply_pincfgs(codec, pincfgs); + } +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4562,7 +4580,6 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, - ALC292_FIXUP_TPT440_DOCK2, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -5029,17 +5046,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC292_FIXUP_TPT440_DOCK] = { .type = HDA_FIXUP_FUNC, - .v.func = alc269_fixup_pincfg_no_hp_to_lineout, - .chained = true, - .chain_id = ALC292_FIXUP_TPT440_DOCK2 - }, - [ALC292_FIXUP_TPT440_DOCK2] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x16, 0x21211010 }, /* dock headphone */ - { 0x19, 0x21a11010 }, /* dock mic */ - { } - }, + .v.func = alc_fixup_tpt440_dock, .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, -- 2.5.3

1 0

[alsa-devel] [linux-sunxi][PATCH 0/3]Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 24 Sep '15

24 Sep '15

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver as present on the A10, A20 and A31 SoC boards. For now just the SPDIF transmitter has been tested on a Mele A2000. In order for this patch set to be functional we require audio clock patches 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 Marcus Cooper (3): dt-bindings: add sunxi SPDIF transceiver bindings dt-binding: Add sunxi SPDIF machine driver ASOC: sunxi: Add support for the spdif block .../devicetree/bindings/sound/sunxi,spdif.txt | 49 ++ .../bindings/sound/sunxi-audio-spdif.txt | 36 + sound/soc/sunxi/Kconfig | 10 + sound/soc/sunxi/Makefile | 4 + sound/soc/sunxi/sunxi-machine-spdif.c | 110 +++ sound/soc/sunxi/sunxi-spdif.c | 801 +++++++++++++++++++++ 6 files changed, 1010 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/sunxi-audio-spdif.txt create mode 100644 sound/soc/sunxi/sunxi-machine-spdif.c create mode 100644 sound/soc/sunxi/sunxi-spdif.c -- 2.5.3

2 4

[alsa-devel] [PATCH] drm/i915: set proper N/CTS in modeset
by libin.yang＠intel.com 24 Sep '15

24 Sep '15

From: Libin Yang <libin.yang(a)intel.com> When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- drivers/gpu/drm/i915/intel_audio.c | 56 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 10 +++++++ 2 files changed, 57 insertions(+), 9 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index e3c32ce..a12bf57 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -128,6 +128,20 @@ static int audio_config_get_n(const struct drm_display_mode *mode, int rate) return 0; } +static uint32_t audio_config_setup_n_reg(int n, uint32_t val) +{ + int n_low, n_up; + uint32_t tmp = val; + + n_low = n & 0xfff; + n_up = (n >> 12) & 0xff; + tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); + tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | + (n_low << AUD_CONFIG_LOWER_N_SHIFT) | + AUD_CONFIG_N_PROG_ENABLE); + return tmp; +} + /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, struct drm_display_mode *mode) @@ -262,9 +276,14 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, struct drm_i915_private *dev_priv = connector->dev->dev_private; struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc); enum pipe pipe = intel_crtc->pipe; + struct i915_audio_component *acomp = dev_priv->audio_component; const uint8_t *eld = connector->eld; + struct intel_digital_port *intel_dig_port = + enc_to_dig_port(&encoder->base); + enum port port = intel_dig_port->port; uint32_t tmp; int len, i; + int n, rate; DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n", pipe_name(pipe), drm_eld_size(eld)); @@ -302,12 +321,29 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, /* Enable timestamps */ tmp = I915_READ(HSW_AUD_CFG(pipe)); tmp &= ~AUD_CONFIG_N_VALUE_INDEX; - tmp &= ~AUD_CONFIG_N_PROG_ENABLE; tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) tmp |= AUD_CONFIG_N_VALUE_INDEX; else tmp |= audio_config_hdmi_pixel_clock(mode); + + tmp &= ~AUD_CONFIG_N_PROG_ENABLE; + if (audio_rate_need_prog(intel_crtc, mode)) { + if (!acomp) + rate = 0; + else if (port >= PORT_A && port <= PORT_E) + rate = acomp->aud_sample_rate[port]; + else { + DRM_ERROR("invalid port: %d\n", port); + rate = 0; + } + n = audio_config_get_n(mode, rate); + if (n != 0) + tmp = audio_config_setup_n_reg(n, tmp); + else + DRM_DEBUG_KMS("no suitable N value is found\n"); + } + I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -594,9 +630,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, struct intel_digital_port *intel_dig_port; struct intel_crtc *crtc; struct drm_display_mode *mode; + struct i915_audio_component *acomp = dev_priv->audio_component; enum pipe pipe = -1; u32 tmp; - int n_low, n_up, n; + int n; /* HSW, BDW SKL need this fix */ if (!IS_SKYLAKE(dev_priv) && @@ -630,6 +667,9 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; + /* port must be valid now, otherwise the pipe will be invalid */ + acomp->aud_sample_rate[port] = rate; + /* 2. check whether to set the N/CTS/M manually or not */ if (!audio_rate_need_prog(crtc, mode)) { tmp = I915_READ(HSW_AUD_CFG(pipe)); @@ -649,15 +689,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, mutex_unlock(&dev_priv->av_mutex); return 0; } - n_low = n & 0xfff; - n_up = (n >> 12) & 0xff; - /* 4. set the N/CTS/M */ + /* 3. set the N/CTS/M */ tmp = I915_READ(HSW_AUD_CFG(pipe)); - tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); - tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | - (n_low << AUD_CONFIG_LOWER_N_SHIFT) | - AUD_CONFIG_N_PROG_ENABLE); + tmp = audio_config_setup_n_reg(n, tmp); I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -678,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev, { struct i915_audio_component *acomp = data; struct drm_i915_private *dev_priv = dev_to_i915(i915_dev); + int i; if (WARN_ON(acomp->ops || acomp->dev)) return -EEXIST; @@ -685,6 +721,8 @@ static int i915_audio_component_bind(struct device *i915_dev, drm_modeset_lock_all(dev_priv->dev); acomp->ops = &i915_audio_component_ops; acomp->dev = i915_dev; + for (i = 0; i < I915_MAX_PORTS; i++) + acomp->aud_sample_rate[i] = 0; dev_priv->audio_component = acomp; drm_modeset_unlock_all(dev_priv->dev); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7..89dc7d6 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,18 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- 1.9.1

4 4

[alsa-devel] Bug: LADSPA implementation - no up/downmix handling
by Andreas Hartmann 24 Sep '15

24 Sep '15

Hi, Using the surround_encoder of the swh-plugins I noticed, that the alsa implementation only assumes an equal number of in- and output channels for ladspa plugins. This occurs from Line 752 onwards in pcm_ladspa.c and leads to a Segfault shortly after. There needs to be a check and correct handling for up/downmix scenarios. Andreas

2 4

[alsa-devel] No stream timing info with alsa pulse plugin?
by sqweek 24 Sep '15

24 Sep '15

Hi guys, I discovered on my machine that alsa_snd_pcm_status_get_tstamp (and also get_htstamp) stop working when my asoundrc invokes the pulse plugin. That is, normally they work fine but once the sound is going via pulse the returned time stamp is always zero. I'm trying to understand why that is. I couldn't really find any discussion of the pulse plugin on the archives so I'm not entirely sure I've found the right list; I was directed here via the pulseaudio bugzilla[1]. [1] https://bugs.freedesktop.org/show_bug.cgi?id=88503 Thanks for any assistance! -sqweek

4 3

[alsa-devel] [PATCH 0/3] ALSA: hda - Avoid potential deadlock
by Thierry Reding 24 Sep '15

24 Sep '15

From: Thierry Reding <treding(a)nvidia.com> The Tegra HDA controller driver committed in v3.16 causes deadlocks when loaded as a module. The reason is that the driver core will lock the HDA controller device upon calling its probe callback and the probe callback then goes on to create child devices for detected codecs and loads their modules via a request_module() call. This is problematic because the new driver will immediately be bound to the device, which will in turn cause the parent of the codec device (the HDA controller device) to be locked again, causing a deadlock. This problem seems to have been present since the modularization of the HD-audio driver in commit 1289e9e8b42f ("ALSA: hda - Modularize HD-audio driver"). On Intel platforms this has been worked around by splitting up the probe sequence into a synchronous and an asynchronous part where the request_module() calls are asynchronous and hence avoid the deadlock. An alternative proposal is provided in this series of patches. Rather than relying on explicit request_module() calls to load kernel modules for HDA codec drivers, this implements a uevent callback for the HDA bus to advertises the MODALIAS information to the userspace helper. Effectively this results in the same modules being loaded, but it uses the more canonical infrastructure to perform this. Deferring the module loading to userspace removes the need for the explicit request_module() calls and works around the recursive locking issue because both drivers will be bound from separate contexts. Thierry Thierry Reding (3): ALSA: hda/hdmi - Add missing MODALIAS information ALSA: hda - Advertise MODALIAS in uevent ALSA: hda: Do not rely on explicit module loading sound/hda/hda_bus_type.c | 12 +++++++ sound/pci/hda/hda_bind.c | 80 ---------------------------------------------- sound/pci/hda/patch_hdmi.c | 3 ++ 3 files changed, 15 insertions(+), 80 deletions(-) -- 2.5.0

3 9

[alsa-devel] ALSA: core: check for underflow in snd_pcm_sw_params()
by Dan Carpenter 24 Sep '15

24 Sep '15

As far as I can see, having an invalid ->tstamp_mode is harmless, but adding a check silences a static checker warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 75888dd..4863af5 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -650,7 +650,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, } snd_pcm_stream_unlock_irq(substream); - if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) + if (params->tstamp_mode < 0 || + params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) return -EINVAL; if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) && params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST)

2 1

[alsa-devel] [PATCH] ALSA: oxfw: add Mackie Onyx Satellite quirk to inform wrong value in 'dbs' field in tx CIP header
by Takashi Sakamoto 24 Sep '15

24 Sep '15

Mackie Onyx Satellite has two usage; standalone and with base station. These two modes has different stream formats. In standalone mode, rx data block includes 2 Multi Bit Linear Audio (MBLA) data channels and tx data block includes 2. With base station, they're 6 and 2. Although, with base station, the actual tx packet include wrong value in 'dbs' field in its CIP header. This quirk causes packet streaming layer to detect packet discontinuity and to stop PCM substream. This is a response of 'single' subfunction 'extended stream format information' command in AV/C Stream Format Information Specification 1.1. It means that a data block in tx packets includes 2 MBLA data channels. $ ./firewire-request /dev/fw1 fcp 0x01ffbfc001000000ffffffff response: 000: 0c ff bf c0 01 00 00 00 ff 00 90 40 03 02 01 02 response: 010: 06 Current OXFW driver parses the response and detects stream formats correctly. $ cat /proc/asound/card1/firewire/formation ... Output Stream from device: Rate PCM MIDI * 48000 2 0 44100 2 0 88200 2 0 96000 2 0 On the other hand, in actual tx CIP, the 'dbs' field has 6. But the number of quadlets in CIP payload is not multiple of 6. Seeing the subtraction of two successive payload quadlets, it should be multiple of 2. payload CIP CIP quadlets header0 header1 24 00060052 9002ffff 24 0006005e 9002ffff 26 0006006a 9002ffff 24 00060077 9002ffff 24 00060083 9002ffff 26 0006008f 9002ffff 24 0006009c 9002ffff 24 000600a8 9002ffff 26 000600b4 9002ffff 24 000600c1 9002ffff This commit adds support for this quirk to OXFW driver, by using CIP_WRONG_DBS flag. When this flag is set, packet streaming layer uses the value of its 'data_block_quadlets' member instead of the value in CIP header. This value is already set by OXFW driver and no discontinuity is detected. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-stream.c | 5 ++++- sound/firewire/oxfw/oxfw.c | 29 +++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw.h | 1 + 3 files changed, 34 insertions(+), 1 deletion(-) diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 77ad5b9..863eb4a 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -238,9 +238,12 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, * packets. As a result, next isochronous packet includes more data * blocks than IEC 61883-6 defines. */ - if (stream == &oxfw->tx_stream) + if (stream == &oxfw->tx_stream) { oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK | CIP_JUMBO_PAYLOAD; + if (oxfw->wrong_dbs) + oxfw->tx_stream.flags |= CIP_WRONG_DBS; + } end: return err; } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 8c6ce01..d606e3a 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -19,6 +19,8 @@ #define VENDOR_BEHRINGER 0x001564 #define VENDOR_LACIE 0x00d04b +#define MODEL_SATELLITE 0x00200f + #define SPECIFIER_1394TA 0x00a02d #define VERSION_AVC 0x010001 @@ -129,6 +131,31 @@ static void oxfw_card_free(struct snd_card *card) mutex_destroy(&oxfw->mutex); } +static void detect_quirks(struct snd_oxfw *oxfw) +{ + struct fw_device *fw_dev = fw_parent_device(oxfw->unit); + struct fw_csr_iterator it; + int key, val; + int vendor, model; + + /* Seek from Root Directory of Config ROM. */ + vendor = model = 0; + fw_csr_iterator_init(&it, fw_dev->config_rom + 5); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_VENDOR) + vendor = val; + else if (key == CSR_MODEL) + model = val; + } + + /* + * Mackie Onyx Satellite with base station has a quirk to report a wrong + * value in 'dbs' field of CIP header against its format information. + */ + if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + oxfw->wrong_dbs = true; +} + static int oxfw_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { @@ -157,6 +184,8 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; + detect_quirks(oxfw); + err = name_card(oxfw); if (err < 0) goto error; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index cace5ad..7b5dad8 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -49,6 +49,7 @@ struct snd_oxfw { struct mutex mutex; spinlock_t lock; + bool wrong_dbs; bool has_output; u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; -- 2.1.4

2 1

Re: [alsa-devel] [Alsa-user] snd-usb-caiaq (NI Maschine Mikro) device not recognised by alsa
by sypwex 24 Sep '15

24 Sep '15

ow, sorry! I not mentioned that there is no PID of Maschine Mikro (0x1110) in caiaq device list *https://github.com/torvalds/linux/blob/v4.2/sound/usb/caiaq/device.h <https://github.com/torvalds/linux/blob/v4.2/sound/usb/caiaq/device.h>* yes there is some endless dump from usbhid but I don't know how to deal with it $sudo usbhid-dump --entity=all --address=1:15 http://pastebin.com/raw.php?i=KC96ERSV (300KiB) ok, I've tried to extend the driver and end up with following doodad https://github.com/krasnobaev/linux/commits/feature/caiaq-mmikro-1 for every commit I've generated dmesg log *http://pastebin.com/raw.php?i=pJzdAyf4 <http://pastebin.com/raw.php?i=pJzdAyf4>* $amidi -l have no devices $cat /proc/asound/cards 4 [MaschineMikro ]: snd-usb-caiaq - Maschine Mikro Native Instruments Maschine Mikro (usb-0000:00:02.1-8) $cat /proc/asound/devices 24: [ 4] : control $cat /proc/asound/oss/sndstat Sound Driver:3.8.1a-980706 (ALSA emulation code) Kernel: Linux ***** 3.19.0+ #1 SMP Sun Sep 13 10:44:23 MSK 2015 x86_64 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: X-Fi Extreme Audio [SB0790] at 0xdc00 irq 18 HDA NVidia at 0xfcf7c000 irq 19 Aureal Vortex au8810 at 0xfbfc0000 irq 17 Webcam C170 at usb-0000:00:02.1-4, high speed Native Instruments Maschine Mikro (usb-0000:00:02.1-8) Audio devices: 0: CA0106 (DUPLEX) 2: au8810 adb (DUPLEX) 3: USB Audio Synth devices: NOT ENABLED IN CONFIG Midi devices: 0: CA0106 MPU-401 (UART) 2: au8810 MIDI 2 Timers: 31: system timer Mixers: 0: CA0106 1: Nvidia GPU 1c HDMI/DP 2: Wolfson WM9701,WM9701A 3: USB Mixer 4: Maschine Mikro $cat /proc/asound/oss/devices 64: [4- 0]: mixer So, currently customized driver fails (all info in dmesg log) on every usb_submit_urb and usb_set_interface functions, so I've decided to watch wireshark usb logs of windows driver filtered windows log (usb.transfer_type != 0x01) && (usb.device_address == 3) http://filebin.ca/2G6kerv8x9TD/usbpcap-maschine-mikro-win7-x86-3maschinesof… RAW http://filebin.ca/2G6g6civjKfI/usbpcap-maschine-mikro-win7-x86-3maschinesof… filtered log for custom driver http://filebin.ca/2G6kFZxgTTKR/usbpcap-maschine-mikro-ubuntu15.04-linux3.19… RAW http://filebin.ca/2G6oNuU6c6b5/usbpcap-maschine-mikro-ubuntu15.04-linux3.19… the difference is... - windows driver do not send SET INTERFACE request while native caiaq driver is. - windows driver receives Response and Status on each request while caiaq recieves only Response - windows driver send two GET_REPORT with wValues: 0x03f8 and 0x0101 while caiaq driver sends three GET_REPORT requests with wValues: 0x0101, 0x0120, 0x03f8 - windows send 3 SET_REPORT requests with same wValue 0x03f8 while caiaq driver do not send any SET_REPORT request I do not know what means these wValues (0x0101, 0x0120, 0x03f8) and how to find out. Is there probability if linux driver send SET_REPORT requests in the same way as do windows driver - the device will set up correctly? How to send SET REPORT request? I believe that there is no big difference between Maschine Controller and Maschine Mikro. BUT there is no Maschine Mikro on caiaq portfolio http://caiaq.de/references_en.html On Sun, Sep 6, 2015 at 8:02 PM, Clemens Ladisch <cladisch(a)googlemail.com> wrote: > sypwex wrote: > > Native Instruments Mashine Mikro which is snd-usb-caiaq. > > No it's not; the Maschine Mikro is not the same as the Maschine Controller. > > As far as I can see from your lsusb output, the Mikro either works as > a HID input device, or needs a separate driver. To check the first > possibility, try a tool like <https://github.com/DIGImend/usbhid-dump>. > > > Regards, > Clemens > Cheers.

2 6

[alsa-devel] MIDI port using UART interface
by Francesco Lavra 24 Sep '15

24 Sep '15

Hi, I'm looking to use an UART interface as a MIDI in/out port. Since MIDI is an asynchronous serial interface just like UART, and provided a given UART controller is able to operate at 31.25 kbps (in my case, it is) a UART port can be transformed into a MIDI port by adding some hardware components (e.g. an optocopuler) to it, I would have expected to find a generic MIDI<->UART adapter layer in the kernel that bridges the ALSA MIDI interface with UART drivers and allows to operate UARTs as MIDI ports. But apparently there is no such generic adapter, the closest driver I found is for the UART 16550 but this is specific to a type of UART controller different from the one in the platform I'm using (which is i.MX6). So how would you approach the task of writing a MIDI driver that uses UART behind the scenes? Am I correct in assuming that I have to use the RawMIDI interface (as the UART 16550 driver does) to make my MIDI port available to ALSA applications? Or is there a better alternative? Thanks, Francesco

2 2

[alsa-devel] [PATCH] ASOC: fsl_spdif: fix bit order for spdif RX channel status
by Kevin Groeneveld 24 Sep '15

24 Sep '15

The bits within each byte of the hardware registers for the channel status bits are reversed. Therefore when the channel status is read from the hardware we need to reverse the bits. The code which writes the channel status bits for the TX already correctly reverses the bits. This patch implements similar functionality for the RX. I have tested this patch running on imx6 using several different SPDIF source devices. Signed-off-by: Kevin Groeneveld <kgroeneveld(a)lenbrook.com> --- sound/soc/fsl/fsl_spdif.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index ab729f2..e84e839 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -676,14 +676,14 @@ static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol, return -EAGAIN; regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus); - ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF; - ucontrol->value.iec958.status[1] = (cstatus >> 8) & 0xFF; - ucontrol->value.iec958.status[2] = cstatus & 0xFF; + ucontrol->value.iec958.status[0] = bitrev8((cstatus >> 16) & 0xFF); + ucontrol->value.iec958.status[1] = bitrev8((cstatus >> 8) & 0xFF); + ucontrol->value.iec958.status[2] = bitrev8(cstatus & 0xFF); regmap_read(regmap, REG_SPDIF_SRCSL, &cstatus); - ucontrol->value.iec958.status[3] = (cstatus >> 16) & 0xFF; - ucontrol->value.iec958.status[4] = (cstatus >> 8) & 0xFF; - ucontrol->value.iec958.status[5] = cstatus & 0xFF; + ucontrol->value.iec958.status[3] = bitrev8((cstatus >> 16) & 0xFF); + ucontrol->value.iec958.status[4] = bitrev8((cstatus >> 8) & 0xFF); + ucontrol->value.iec958.status[5] = bitrev8(cstatus & 0xFF); /* Clear intr */ regmap_write(regmap, REG_SPDIF_SIC, INT_CNEW); -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/4] ASoC: rt5645: Remove the repeated definitions
by Oder Chiou 23 Sep '15

23 Sep '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 90325d9..f45861c 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -777,8 +777,6 @@ #define RT5645_PWR_CLS_D_R_BIT 9 #define RT5645_PWR_CLS_D_L (0x1 << 8) #define RT5645_PWR_CLS_D_L_BIT 8 -#define RT5645_PWR_ADC_R (0x1 << 1) -#define RT5645_PWR_ADC_R_BIT 1 #define RT5645_PWR_DAC_L2 (0x1 << 7) #define RT5645_PWR_DAC_L2_BIT 7 #define RT5645_PWR_DAC_R2 (0x1 << 6) -- 1.8.1.1.439.g50a6b54

2 7

[alsa-devel] [PATCH] topology: Add API to set a vendor specific version number
by mengdong.lin＠intel.com 23 Sep '15

23 Sep '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This vendor-specific version number is optional. It will be written to the 'version' field of each block header of the binary toplogy data file. The vendor driver can check this number for further processing in kernel. The topology ABI version number is still stored in the 'abi' field of block headers. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/topology.h | 8 ++++++++ src/topology/builder.c | 6 +++--- src/topology/parser.c | 7 +++++++ 3 files changed, 18 insertions(+), 3 deletions(-) diff --git a/include/topology.h b/include/topology.h index 6ff8c5f..9b84bd9 100644 --- a/include/topology.h +++ b/include/topology.h @@ -690,6 +690,14 @@ int snd_tplg_build(snd_tplg_t *tplg, const char *outfile); */ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len); +/** + * \brief Set an optional vendor specific version number. + * \param tplg Topology instance. + * \param version Vendor specific version number. + * \return Zero on success, otherwise a negative error code + */ +int snd_tplg_set_version(snd_tplg_t *tplg, unsigned int version); + /* \} */ #ifdef __cplusplus diff --git a/src/topology/builder.c b/src/topology/builder.c index 3bccd44..2f01137 100644 --- a/src/topology/builder.c +++ b/src/topology/builder.c @@ -89,7 +89,7 @@ static int write_data_block(snd_tplg_t *tplg, int size, int tplg_type, /* write the header for this block */ ret = write_block_header(tplg, tplg_type, 0, - SND_SOC_TPLG_ABI_VERSION, 0, size, 1); + tplg->version, 0, size, 1); if (ret < 0) { SNDERR("error: failed to write %s block %d\n", obj_name, ret); return ret; @@ -125,7 +125,7 @@ static int write_elem_block(snd_tplg_t *tplg, } ret = write_block_header(tplg, tplg_type, vendor_type, - SND_SOC_TPLG_ABI_VERSION, 0, size, count); + tplg->version, 0, size, count); if (ret < 0) { SNDERR("error: failed to write %s block %d\n", obj_name, ret); @@ -243,7 +243,7 @@ static int write_manifest_data(snd_tplg_t *tplg) /* write the header for this block */ ret = write_block_header(tplg, SND_SOC_TPLG_TYPE_MANIFEST, 0, - SND_SOC_TPLG_ABI_VERSION, 0, + tplg->version, 0, sizeof(tplg->manifest) + tplg->manifest.priv.size, 1); if (ret < 0) { SNDERR("error: failed to write manifest block %d\n", ret); diff --git a/src/topology/parser.c b/src/topology/parser.c index 7ca244c..c0ab5e4 100644 --- a/src/topology/parser.c +++ b/src/topology/parser.c @@ -362,6 +362,13 @@ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len) return 0; } +int snd_tplg_set_version(snd_tplg_t *tplg, unsigned int version) +{ + tplg->version = version; + + return 0; +} + void snd_tplg_verbose(snd_tplg_t *tplg, int verbose) { tplg->verbose = verbose; -- 1.9.1

5 4

[alsa-devel] [PATCH - 1/2] HDA - Add SND_JACK_HEADSET and Headset jack kctl
by Raymond Yau 23 Sep '15

23 Sep '15

>From b3ed187912eccf922000aee09c04f9546b068925 Mon Sep 17 00:00:00 2001 From: Raymond Yau <superquad.vortex2(a)gmail.com> Date: Sun, 20 Sep 2015 22:10:18 +0800 Subject: [PATCH - 1/2] HDA - Add SND_JACK_HEADSET and Headset jack kctl add headset jack kctl for alsa jack type SND_JACK_HEADSET skip creation of Headset Mic Phantom Jack use Headset Playback Volume and Headset Playback Switch allow auto mic selection by using HP pin for jack sense Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 24f9111..ce6da6f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -2140,7 +2140,8 @@ static int create_hp_out_ctls(struct hda_codec *codec) struct hda_gen_spec *spec = codec->spec; return create_extra_outs(codec, spec->autocfg.hp_outs, spec->hp_paths, - "Headphone"); + spec->hs_mic_use_hp_sense ? + "Headset" : "Headphone"); } static int create_speaker_out_ctls(struct hda_codec *codec) @@ -4362,6 +4363,16 @@ void snd_hda_gen_line_automute(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_gen_line_automute); +bool is_headset_mic(struct auto_pin_cfg *cfg, hda_nid_t pin) +{ + int i; + + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].pin == pin) + return cfg->inputs[i].is_headset_mic; + return 0; +} + /** * snd_hda_gen_mic_autoswitch - standard mic auto-switch helper * @codec: the HDA codec @@ -4381,6 +4392,9 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; + if (spec->hs_mic_use_hp_sense && + is_headset_mic(&spec->autocfg, pin)) + pin = auto_cfg_hp_pins(&spec->autocfg)[0]; if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return; @@ -4662,7 +4676,9 @@ static int check_auto_mic_availability(struct hda_codec *codec) if (!spec->line_in_auto_switch && cfg->inputs[i].type != AUTO_PIN_MIC) return 0; /* only mic is allowed */ - if (!is_jack_detectable(codec, nid)) + if (!is_jack_detectable(codec, nid) && + !(cfg->inputs[i].is_headset_mic && + spec->hs_mic_use_hp_sense)) return 0; /* no unsol support */ break; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139..96f8214 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -236,6 +236,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int hs_mic_use_hp_sense:1; /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 366efbf..5fa8ace 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -19,6 +19,7 @@ #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" +#include "hda_generic.h" /** * is_jack_detectable - Check whether the given pin is jack-detectable @@ -350,11 +351,14 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_report_sync); static int get_input_jack_type(struct hda_codec *codec, hda_nid_t nid) { unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + struct hda_gen_spec *spec = codec->spec; switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: case AC_JACK_SPEAKER: return SND_JACK_LINEOUT; case AC_JACK_HP_OUT: + if (spec->hs_mic_use_hp_sense) + return SND_JACK_HEADSET; return SND_JACK_HEADPHONE; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: @@ -434,6 +438,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int err; bool phantom_jack; + struct hda_gen_spec *spec = codec->spec; if (!nid) return 0; @@ -446,8 +451,13 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, if (base_name) strlcpy(name, base_name, sizeof(name)); - else + else { snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), NULL); + if ((strcmp(name, "Headphone") == 0) && + (nid == auto_cfg_hp_pins(cfg)[0]) && + spec->hs_mic_use_hp_sense) + strlcpy(name, "Headset", sizeof(name)); + } if (phantom_jack) /* Example final name: "Internal Mic Phantom Jack" */ strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); @@ -469,6 +479,7 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; + struct hda_gen_spec *spec = codec->spec; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -482,6 +493,10 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, "Headphone Mic"); } else + if (cfg->inputs[i].is_headset_mic && + spec->hs_mic_use_hp_sense) + err = 0; + else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, NULL); if (err < 0) -- 2.5.0

2 2

[alsa-devel] [PATCH utils] aplay: fix VU meter for S24_LE etc formats
by Ricard Wanderlof 23 Sep '15

23 Sep '15

When recording or playing back audio in a format where the number of significant bits is less than the physical width (e.g. S24_LE), the VU meter code needs to consider the number of significant bits in the samples rather than the physical sample width (e.g. 24 vs 32 bits). Otherwise the resulting VU meter display will be far too low and it will just indicate 0% all the time. Tested with a device supporting the S24_LE format. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- alsa-utils/aplay/aplay.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index e58e1bc..19b6a92 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -124,7 +124,7 @@ static int fatal_errors = 0; static int verbose = 0; static int vumeter = VUMETER_NONE; static int buffer_pos = 0; -static size_t bits_per_sample, bits_per_frame; +static size_t significant_bits_per_sample, bits_per_sample, bits_per_frame; static size_t chunk_bytes; static int test_position = 0; static int test_coef = 8; @@ -1344,6 +1344,7 @@ static void set_params(void) snd_pcm_dump(handle, log); bits_per_sample = snd_pcm_format_physical_width(hwparams.format); + significant_bits_per_sample = snd_pcm_format_width(hwparams.format); bits_per_frame = bits_per_sample * hwparams.channels; chunk_bytes = chunk_size * bits_per_frame / 8; audiobuf = realloc(audiobuf, chunk_bytes); @@ -1745,7 +1746,7 @@ static void compute_max_peak(u_char *data, size_t count) } return; } - max = 1 << (bits_per_sample-1); + max = 1 << (significant_bits_per_sample-1); if (max <= 0) max = 0x7fffffff; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH RESEND 1/4] sound/onyx: remove redundant case entry
by Yaowei Bai 23 Sep '15

23 Sep '15

Case ti->tag == 0 can be included in the last return, remove it. No functional change. Signed-off-by: Yaowei Bai <bywxiaobai(a)163.com> --- sound/aoa/codecs/onyx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index a04edff..a407e1f 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -683,7 +683,6 @@ static int onyx_usable(struct codec_info_item *cii, mutex_unlock(&onyx->mutex); switch (ti->tag) { - case 0: return 1; case 1: return analog_enabled; case 2: return spdif_enabled; } -- 1.9.1

1 3

[alsa-devel] [PATCH 1/4] sound/onyx: remove redundant case entry
by Yaowei Bai 23 Sep '15

23 Sep '15

Case ti->tag == 0 can be included in the last return, remove it. No functional change. Signed-off-by: Yaowei Bai <bywxiaobai(a)163.com> --- sound/aoa/codecs/onyx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index a04edff..a407e1f 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -683,7 +683,6 @@ static int onyx_usable(struct codec_info_item *cii, mutex_unlock(&onyx->mutex); switch (ti->tag) { - case 0: return 1; case 1: return analog_enabled; case 2: return spdif_enabled; } -- 1.9.1

2 7

[alsa-devel] [PATCH - alsa-utils 1/1] arecord: Remove only regular files
by Alexander Volkov 23 Sep '15

23 Sep '15

arecord removes a file before writing into it. It's not appropriate in some cases. For example, if you a pass a symlink to a file, then the symlink will be removed while the user expects to record into the symlink's target. Another case is recording into the device file. Some modems provide a tty device file as a voice device. And it's not possible to write into it under root with arecord, because it removes the device file. So check the type of a file before writing into it and remove only regular files. Signed-off-by: Alexander Volkov <a.volkov(a)rusbitech.ru> diff --git a/aplay/aplay.c b/aplay/aplay.c index 459f7dd..1b2cdfc 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2929,6 +2929,7 @@ static void capture(char *orig_name) char *name = orig_name; /* current filename */ char namebuf[PATH_MAX+1]; off64_t count, rest; /* number of bytes to capture */ + struct stat statbuf; /* get number of bytes to capture */ count = calc_count(); @@ -2973,7 +2974,10 @@ static void capture(char *orig_name) } /* open a new file */ - remove(name); + if (!lstat(name, &statbuf)) { + if (S_ISREG(statbuf.st_mode)) + remove(name); + } fd = safe_open(name); if (fd < 0) { perror(name); -- 2.4.0

4 3

[alsa-devel] [PATCH alsa-lib] build: Do not try to detect cross-compiler
by Tanu Kaskinen 23 Sep '15

23 Sep '15

From: Khem Raj <raj.khem(a)gmail.com> cross compilers are passed via path may not be a gcc based cross compiler in such cases this check fails and try's to force gcc based cross compiler detection, This code is a convenience that limits the build system Signed-off-by: Khem Raj <raj.khem(a)gmail.com> --- configure.ac | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/configure.ac b/configure.ac index a482b3e..a14e52d 100644 --- a/configure.ac +++ b/configure.ac @@ -27,20 +27,6 @@ AC_PREFIX_DEFAULT(/usr) dnl Checks for programs. -dnl try to gues cross-compiler if not set -if test "x$host" != "x$build" -a -z "`echo $CC | grep -e '-gcc'`"; -then - AC_MSG_CHECKING(for cross-compiler) - - which ${program_prefix}gcc >/dev/null 2>&1 && CC=${program_prefix}gcc - which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 \ - && CC=${host_cpu}-${host_os}-gcc - which ${host_cpu}-${host_vendor}-${host_os}-gcc >/dev/null 2>&1 \ - && CC=${host_cpu}-${host_vendor}-${host_os}-gcc - - AC_MSG_RESULT($CC) -fi - CFLAGS="$CFLAGS -D_GNU_SOURCE" -- 2.5.1

2 1

[alsa-devel] [PATCH] sound: oss: ad1848: Fix returned errno code in ad1848_init()
by Luis de Bethencourt 23 Sep '15

23 Sep '15

The driver is using -1 instead of the -ENOMEM defined macro to specify that a buffer allocation failed. Since the error number is propagated, the caller will get a -EPERM which is the wrong error condition. Smatch tool warning: ad1848_init() warn: returning -1 instead of -ENOMEM is sloppy Signed-off-by: Luis de Bethencourt <luisbg(a)osg.samsung.com> --- sound/oss/ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 10c8de1..6b35656 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -1992,7 +1992,7 @@ int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback, portc = kmalloc(sizeof(ad1848_port_info), GFP_KERNEL); if(portc==NULL) { release_region(devc->base, 4); - return -1; + return -ENOMEM; } if ((my_dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, -- 2.5.3

2 3

[alsa-devel] [PATCH v2- ALSA 3/3] HDA - add headset-sonicmaster and headset-multi-sonicmaster models
by Raymond Yau 23 Sep '15

23 Sep '15

add headset only model for ALC668 with Sonic Master Subwoofer add 625ms delay after headset is plugged for correct detection of headset type for ALC668 add headset multi model with Sonice Master Subwoofer update HD-Audio-Models.txt Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ec099d4..8baddac 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -55,7 +55,12 @@ ALC66x/67x/892 asus-mode7 ASUS asus-mode8 ASUS inv-dmic Inverted internal mic workaround + +ALC668 +====== dell-headset-multi Headset jack, which can also be used as mic-in + headset-sonicmaster Headset jack and SonicMaster Subwoofer + headset-multi-sonicmaster Headset, HP or Mic and SonicMaster Subwoofer ALC680 ====== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a75b561..91de00c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4029,7 +4029,13 @@ static void alc_update_headset_mode(struct hda_codec *codec) PIN_VREFHIZ); } spec->current_headset_mode = new_headset_mode; - + if (codec->core.vendor_id == 0x10ec0668) { + if (spec->gen.hs_mic_use_hp_sense && spec->gen.auto_mic) { + if (new_headset_mode == ALC_HEADSET_MODE_HEADPHONE) + msleep(625); + snd_hda_gen_mic_autoswitch(codec, NULL); + }; + } snd_hda_gen_update_outputs(codec); } @@ -6071,6 +6077,42 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, } } +static void alc668_fixup_asus_headset_only(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl hs_mic_pin[] = { + { 0x1b, 0x03a1113c }, /* headset mic */ + { } + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, hs_mic_pin); + spec->gen.hs_mic_use_hp_sense = 1; + alc_fixup_disable_aamix(codec, fix, action); + break; + } +} + +static void alc668_fixup_sonicmaster(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl bass_pin[] = { + { 0x1a, 0x04110011 }, /* external subwoofer */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, bass_pin); + break; + case HDA_FIXUP_ACT_BUILD: + alc_fixup_bass_chmap(codec, fix, action); + break; + } +} + static struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -6130,6 +6172,9 @@ enum { ALC668_FIXUP_AUTO_MUTE, ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, + ALC668_FIXUP_HEADSET_ONLY, + ALC668_FIXUP_HEADSET_SONICMASTER, + ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, }; static const struct hda_fixup alc662_fixups[] = { @@ -6370,6 +6415,24 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC668_FIXUP_HEADSET_ONLY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_asus_headset_only, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE, + }, + [ALC668_FIXUP_HEADSET_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_ONLY, + }, + [ALC668_FIXUP_HEADSET_MULTI_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_DISABLE_AAMIX, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6476,6 +6539,9 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, + {.id = ALC668_FIXUP_HEADSET_SONICMASTER, .name = "headset-sonicmaster"}, + {.id = ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, + .name = "headset-multi-sonicmaster"}, {} }; -- 2.5.0

1 0

[alsa-devel] [PATCH v2- ALSA 2/3] HDA - create headset mic jack which use headphone pin sense
by Raymond Yau 23 Sep '15

23 Sep '15

- use __snd_hda_jack_add_kctl to create headset mic jack which use headphone pin sense - set headset mic jack as slave of headphone jack - allow non detectable headset mic pin and hs_mic_use_hp_sense pass check auto mic availability Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 24f9111..e8e4e34 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4662,7 +4662,9 @@ static int check_auto_mic_availability(struct hda_codec *codec) if (!spec->line_in_auto_switch && cfg->inputs[i].type != AUTO_PIN_MIC) return 0; /* only mic is allowed */ - if (!is_jack_detectable(codec, nid)) + if (!is_jack_detectable(codec, nid) && + !(spec->hs_mic_use_hp_sense && + cfg->inputs[i].is_headset_mic)) return 0; /* no unsol support */ break; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139..96f8214 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -236,6 +236,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int hs_mic_use_hp_sense:1; /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d81d65f..c29f05e 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -480,6 +480,10 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; + struct hda_jack_tbl *jack; + struct hda_gen_spec *spec = codec->spec; + hda_nid_t hp_pin = auto_cfg_hp_pins(cfg)[0]; + hda_nid_t hs_mic_pin = 0; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -493,6 +497,13 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, "Headphone Mic"); } else + if (cfg->inputs[i].is_headset_mic && + !is_jack_detectable(codec, cfg->inputs[i].pin) && + spec->hs_mic_use_hp_sense) { + hs_mic_pin = cfg->inputs[i].pin; + err = __snd_hda_jack_add_kctl(codec, cfg->inputs[i].pin, + "Headset Mic", false, hp_pin); + } else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, NULL); if (err < 0) @@ -510,6 +521,11 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; + if (*p == hp_pin && spec->hs_mic_use_hp_sense && hs_mic_pin) { + jack = snd_hda_jack_tbl_get(codec, *p); + if (jack) + jack->slave_nid = hs_mic_pin; + } } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ -- 2.5.0

1 0

[alsa-devel] [PATCH v2] ASoC: pxa: pxa2xx-ac97: fix dma requestor lines
by Robert Jarzmik 23 Sep '15

23 Sep '15

PCM receive and transmit DMA requestor lines were reverted, breaking the PCM playback interface for PXA platforms using the sound/soc/ variant instead of the sound/arm variant. The commit below shows the inversion in the requestor lines. Fixes: d65a14587a9b ("ASoC: pxa: use snd_dmaengine_dai_dma_data") Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: rebase on Linus's master, not my dmaengine tip branch. --- sound/soc/pxa/pxa2xx-ac97.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1f6054650991..9e4b04e0fbd1 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -49,7 +49,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12; +static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -57,7 +57,7 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, }; -static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11; +static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, -- 2.1.4

2 1

[alsa-devel] [PATCH v3 00/21] Update ALSA, and au0828 drivers to use Managed Media Controller API
by Shuah Khan 22 Sep '15

22 Sep '15

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API to share tuner. Please note that Managed Media Controller API and DVB and V4L2 drivers updates to use Media Controller API have been added in a prior patch series. Media Controller API is enhanced with two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. Changes since v2: http://www.spinics.net/lists/linux-media/msg91926.html 1. An important change in this patch series is made to ALSA to check for resources and hold in snd_usb_hw_params(), and release from snd_usb_hw_free(). This change fixed the lockdep warnings seen when resources were held in TRIGGER_START and released from TRIGGER_STOP which could run in IRQ context. I acknowledge Clemens Ladisch for suggesting the correct places to hold/free resources to avoid IRQ path complications. 2. With the above change, the patch series is simpler without the need to change the graph_mutex into a spinlock. 3. I split the patches up differently for easy reviews - no code bloat from v2. 4. A second important change is now the Bridge driver (au0828) owns and drives the graph creation as well as enabling and disabling tuner. It also keeps state information to avoid graph walks in enable_source and disable_source handler. I acknowledge Hans Verkuil for his suggestions and ideas for this change. History: This patch series has been updated to address comments from 3 previous versions of this series. Links to v3 version for reference are: https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89491.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg89492.html https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89493.html Shuah Khan (21): Revert "[media] media: media controller entity framework enhancements for ALSA" media: Media Controller register/unregister entity_notify API media: Add ALSA Media Controller devnodes media: Media Controller enable/disable source handler API media: Media Controller fix to not let stream_count go negative media: Media Controller export non locking __media_entity_setup_link() media: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add v4l_enable/disable_media_tuner() helper functions media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: au0828 Use au8522_media_pads enum for pad defines media: au0828 fix au0828_create_media_graph() entity checks media: Change v4l-core to check for tuner availability media: dvb-frontend invoke enable/disable_source handlers media: au0828 video remove au0828_enable_analog_tuner() media: au0828 video change to use v4l_enable_media_tuner() media: au0828 change to use Managed Media Controller API media: au0828 change to register/unregister entity_notify hook media: au0828 implement enable_source and disable_source handlers media: media: dvb-frontend fix enable_source error legs sound/usb: Update ALSA driver to use Managed Media Controller API drivers/media/dvb-core/dvb_frontend.c | 142 ++----------- drivers/media/dvb-core/dvb_frontend.h | 3 + drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 46 +++- drivers/media/media-entity.c | 64 ++++-- drivers/media/usb/au0828/au0828-core.c | 305 ++++++++++++++++++++++----- drivers/media/usb/au0828/au0828-video.c | 76 ++----- drivers/media/usb/au0828/au0828.h | 8 + drivers/media/v4l2-core/v4l2-dev.c | 27 +++ drivers/media/v4l2-core/v4l2-fh.c | 1 + drivers/media/v4l2-core/v4l2-ioctl.c | 29 +++ drivers/media/v4l2-core/videobuf2-core.c | 3 + include/media/media-device.h | 41 ++++ include/media/media-entity.h | 7 +- include/media/v4l2-dev.h | 4 + include/uapi/linux/media.h | 5 + sound/usb/Makefile | 15 +- sound/usb/card.c | 5 + sound/usb/card.h | 1 + sound/usb/media.c | 178 ++++++++++++++++ sound/usb/media.h | 51 +++++ sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 1 + 28 files changed, 790 insertions(+), 280 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

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[alsa-devel] [PATCH v3 1/2] ASoC: rt5645: Add dmi for Broadwell
by John Lin 22 Sep '15

22 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index dbc1d76..e830688 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3209,6 +3209,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3241,7 +3266,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwell)) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add dmi for Broadwell
by Bard Liao 22 Sep '15

22 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 66e9ce6..2fc437f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3210,6 +3210,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3242,7 +3267,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwel) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] Applied "MAINTAINERS: Update website and git repo for Wolfson Microelectronics" to the asoc tree
by Mark Brown 22 Sep '15

22 Sep '15

The patch MAINTAINERS: Update website and git repo for Wolfson Microelectronics 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 f0e03dbd2d61d991bdd2d76b4e84681fe3077176 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Date: Tue, 22 Sep 2015 15:30:33 +0100 Subject: [PATCH] MAINTAINERS: Update website and git repo for Wolfson Microelectronics Support for Wolfson Microelectronics devices is now part of Cirrus Logic and the relevant parts of the old opensource.wolfsonmicro.com site have moved to the Cirrus Logic GitHub area. This patch updates the website and git repo links, and also removes an obsolete website link for the voltage and current drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- MAINTAINERS | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 7ba7ab7..dffea21 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -11239,7 +11239,6 @@ VOLTAGE AND CURRENT REGULATOR FRAMEWORK M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)kernel.org> L: linux-kernel(a)vger.kernel.org -W: http://opensource.wolfsonmicro.com/node/15 W: http://www.slimlogic.co.uk/?p=48 T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator.git S: Supported @@ -11368,17 +11367,15 @@ WM97XX TOUCHSCREEN DRIVERS M: Mark Brown <broonie(a)kernel.org> M: Liam Girdwood <lrg(a)slimlogic.co.uk> L: linux-input(a)vger.kernel.org -T: git git://opensource.wolfsonmicro.com/linux-2.6-touch -W: http://opensource.wolfsonmicro.com/node/7 +W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported F: drivers/input/touchscreen/*wm97* F: include/linux/wm97xx.h WOLFSON MICROELECTRONICS DRIVERS L: patches(a)opensource.wolfsonmicro.com -T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc -T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus -W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices +T: git https://github.com/CirrusLogic/linux-drivers.git +W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported F: Documentation/hwmon/wm83?? F: arch/arm/mach-s3c64xx/mach-crag6410* -- 2.5.0

1 0

Re: [alsa-devel] [PATCH 1/3] ASoC: codecs: Add da7219 codec driver
by Mark Brown 22 Sep '15

22 Sep '15

On Thu, Sep 17, 2015 at 05:01:14PM +0100, Adam Thomson wrote: > + do { > + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); > + if (statusa & DA7219_MICBIAS_UP_STS_MASK) > + micbias_up = true; > + } while (!micbias_up); This could go into an inifinite loop. > +static void da7219_aad_hptest_work(struct work_struct *work) > +{ > + struct da7219_aad_priv *da7219_aad = > + container_of(work, struct da7219_aad_priv, hptest_work); > + struct snd_soc_codec *codec = da7219_aad->codec; > + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); > + > + u8 tonegen_cfg1, tonegen_cfg2, tonegen_onper; > + u16 tonegen_freq1, tonegen_freq_hptest; > + u8 hpl_gain, hpr_gain, dacl_gain, dacr_gain, dac_filters1, dac_filters4; > + u8 dac_filters5, cp_ctrl, routing_dac, dacl_ctrl, dacr_ctrl; > + u8 mixoutl_sel, mixoutr_sel, st_outfilt_1l, st_outfilt_1r; > + u8 mixoutl_ctrl, mixoutr_ctrl, hpl_ctrl, hpr_ctrl, accdet_cfg8; > + int report = 0; > + > + /* Save current settings */ This is obviously a massive reconfiguration of the device. I'm not seeing anything here which prevents userspace coming in and change the configuration while we're in this function, that would obviously have serious issues. I'm also wondering if this might be more elegantly implemented by going into cache bypass mode, doing the test and then using a cache resync to restore the initial configuration. That will at least avoid issues with updates adding a new register but not modifying it. > + if (statusa & DA7219_JACK_TYPE_STS_MASK) { > + report |= SND_JACK_HEADSET; > + mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT; > + schedule_work(&da7219_aad->btn_det_work); > + } else { > + schedule_work(&da7219_aad->hptest_work); > + } Why are we scheduling work - we're already in thread context? > + /* Un-drive headphones/lineout */ > + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, > + DA7219_HP_R_AMP_OE_MASK, 0); > + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, > + DA7219_HP_L_AMP_OE_MASK, 0); This looks like DAPM? > +static enum da7219_aad_jack_ins_deb da7219_aad_of_jack_ins_deb(u32 val) > +{ > + switch (val) { > + case 5: > + return DA7219_AAD_JACK_INS_DEB_5MS; > + case 10: > + return DA7219_AAD_JACK_INS_DEB_10MS; > + case 20: > + return DA7219_AAD_JACK_INS_DEB_20MS; > + case 50: > + return DA7219_AAD_JACK_INS_DEB_50MS; > + case 100: > + return DA7219_AAD_JACK_INS_DEB_100MS; > + case 200: > + return DA7219_AAD_JACK_INS_DEB_200MS; > + case 500: > + return DA7219_AAD_JACK_INS_DEB_500MS; > + case 1000: > + return DA7219_AAD_JACK_INS_DEB_1S; > + default: > + return DA7219_AAD_JACK_INS_DEB_20MS; This isn't an error? > +/* Input/Output Enums */ > +static const char * const da7219_gain_ramp_rate_txt[] = { > + "nominal rate * 8", "nominal rate", "nominal rate / 8", > + "nominal rate / 16" > +}; The ALSA ABI generally capitalises words. > +/* ToneGen */ > +static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); > + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); > + struct soc_mixer_control *mixer_ctrl = > + (struct soc_mixer_control *) kcontrol->private_value; > + unsigned int reg = mixer_ctrl->reg; > + u16 val; > + int ret; > + > + ret = regmap_bulk_read(da7219->regmap, reg, &val, sizeof(val)); > + if (ret) > + return ret; > + > + ucontrol->value.integer.value[0] = le16_to_cpu(val); This is *weird*. We do a bulk read for a single register using an API that returns CPU endian data then make it CPU endian (without any annotations on variables...). Why not use regmap_read()? Why swap? Why not use raw I/O? > +static int da7219_hpf_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + 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 bits; > + > + switch (sel) { > + case DA7219_HPF_MODE_DISABLED: > + bits = DA7219_HPF_DISABLED; > + break; > + case DA7219_HPF_MODE_AUDIO: > + bits = DA7219_HPF_AUDIO_EN; > + break; > + case DA7219_HPF_MODE_VOICE: > + bits = DA7219_HPF_VOICE_EN; > + break; > + default: > + return -EINVAL; > + } > + > + snd_soc_update_bits(codec, reg, DA7219_HPF_MODE_MASK, bits); > + > + return 0; > +} This looks like a standard enumeration with a simple mapping to the register map? > + > + /* ADCs */ > + SOC_SINGLE_TLV("ADC Volume", DA7219_ADC_L_GAIN, > + DA7219_ADC_L_DIGITAL_GAIN_SHIFT, > + DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, > + da7219_adc_dig_gain_tlv), > + SOC_SINGLE("ADC Switch", DA7219_ADC_L_CTRL, > + DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, > + DA7219_INVERT), > + SOC_SINGLE("ADC Gain Ramp Switch", DA7219_ADC_L_CTRL, > + DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, > + DA7219_NO_INVERT), Capture Digital rather than ADC. > + SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS, > + DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_gain_tlv), > + SOC_SINGLE_TLV("ALC Max Gain", DA7219_ALC_GAIN_LIMITS, > + DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_gain_tlv), > + SOC_SINGLE_RANGE_TLV("ALC Min Analog Gain", DA7219_ALC_ANA_GAIN_LIMITS, > + DA7219_ALC_ANA_GAIN_MIN_SHIFT, > + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), > + SOC_SINGLE_RANGE_TLV("ALC Max Analog Gain", DA7219_ALC_ANA_GAIN_LIMITS, > + DA7219_ALC_ANA_GAIN_MAX_SHIFT, > + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), Volume not Gain. > + SOC_SINGLE("ToneGen DTMF Key", DA7219_TONE_GEN_CFG1, > + DA7219_DTMF_REG_SHIFT, DA7219_DTMF_REG_MAX, > + DA7219_NO_INVERT), Should this be an enumeration with the DTMF digits in it (# and * aren't numbers)? > + /* DACs */ > + SOC_DOUBLE_R_TLV("DAC Volume", DA7219_DAC_L_GAIN, DA7219_DAC_R_GAIN, > + DA7219_DAC_L_DIGITAL_GAIN_SHIFT, > + DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, > + da7219_dac_dig_gain_tlv), > + SOC_DOUBLE_R("DAC Switch", DA7219_DAC_L_CTRL, DA7219_DAC_R_CTRL, > + DA7219_DAC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, > + DA7219_INVERT), > + SOC_DOUBLE_R("DAC Gain Ramp Switch", DA7219_DAC_L_CTRL, > + DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT, > + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), All the DAC controls should probably be Playback Digital instead. > + > +static int da7219_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 da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); > + > + if (da7219->mclk_rate == freq) > + return 0; Given that we also have source selection is this safe - should we also be checking the source? > + if (da7219->mclk) { > + freq = clk_round_rate(da7219->mclk, freq); > + clk_set_rate(da7219->mclk, freq); > + } Missing error checking. > + /* Internal LDO */ > + if (da7219->use_int_ldo) > + snd_soc_update_bits(codec, DA7219_LDO_CTRL, > + DA7219_LDO_EN_MASK, > + DA7219_LDO_EN_MASK); If there is an option to use an external supply I would expect to see the regulator API used to discover the external LDO (and ideally also to configure the integrated LDO). If the driver works outside the frameworks then it is likely this will lead to integration issues later on. > + /* Internal LDO */ > + da7219->use_int_ldo = pdata->use_internal_ldo; This is likely to lead to surprises for example... > + /* Check if MCLK provided, if not the clock is NULL */ > + da7219->mclk = devm_clk_get(codec->dev, "mclk"); > + if (IS_ERR(da7219->mclk)) { > + if (PTR_ERR(da7219->mclk) == -EPROBE_DEFER) > + return -EPROBE_DEFER; > + da7219->mclk = NULL; > + } This should specifically look for the error code that corresponds to a clock not being mapped and only continue silently in that case rather than silently accepting all failures to obtain a clock - if we silently accept all failures that will mean that actual errors will be ignored. > + /* > + * There are multiple control bits for the input mixer. > + * The following can be enabled now as it's not power related. > + */ > + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, > + DA7219_MIXIN_L_MIX_EN_MASK, > + DA7219_MIXIN_L_MIX_EN_MASK); So the chip designers just put these in for randomness? Fun. It'd be more idiomatic to do something like making these supply widgets so they're controlled via DAPM even if they don't matter much. > + else if (device_may_wakeup(codec->dev)) > + disable_irq_wake(da7219->aad->irq); You can use dev_pm_set_wake_irq() and skip having to manage this stuff explicitly in the driver.

2 3

[alsa-devel] unable to create IPC shm instance; SND_PCM_DIRECT_MAGIC
by Mike Bjorge 22 Sep '15

22 Sep '15

I was wondering if anyone could explain the point of SND_PCM_DIRECT_MAGIC and dmix->shmptr->magic? Specifically, what it's doing in pcm_direct.c from this commit? http://git.alsa-project.org/?p=alsa-lib.git;a=blobdiff;f=src/pcm/pcm_direct… I am trying to use pcm_dmix.c, but this is occasionally failing first_instance = ret = snd_pcm_direct_shm_create_or_connect(dmix); if (ret < 0) { SNDERR("unable to create IPC shm instance"); goto _err; } because dmix->shmptr->magic is not correctly set to SND_PCM_DIRECT_MAGIC. Thanks for any information! -Mike

1 0

[alsa-devel] [PATCH v2 0/9] Add DSP topology management for SKL
by Vinod Koul 21 Sep '15

21 Sep '15

The SKL driver does not code DSP topology in driver. With this series the ASoC topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, compute resources for modules, initialize and bind the pipe modules. It uses the SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules. Last patch in this series removes the unused dais. changes in v2: - add more explanation in changelog and code - add explanation of MCPS - fix whitespace issues - remove skl widget check on event handler as it is redundant - add switch for depth - add else for direction based code - remove cast for NHLT blob query - make debug prints to error - remove dump info - clean some debug prints Jeeja KP (9): ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Remove unused CPU dai's sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-pcm.c | 171 ++-- sound/soc/intel/skylake/skl-topology.c | 1262 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 21 + sound/soc/intel/skylake/skl-tplg-interface.h | 78 ++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 7 files changed, 1507 insertions(+), 68 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c -- 1.9.1

3 34

[alsa-devel] Intel SST on a Bay Trail tablet
by Michele Curti 21 Sep '15

21 Sep '15

> On Thu, 25 Jun 2015 22:17:17 +0530 Vinod Koul wrote: > > > On Thu, Jun 25, 2015 at 12:21:24PM +0200, Antonio Ospite wrote: > > > On Thu, 25 Jun 2015 11:20:46 +0530 > > > Vinod Koul <vinod.koul at intel.com> wrote: > > > > > > > On Wed, Jun 24, 2015 at 03:46:13PM +0530, Vinod Koul wrote: > > > > > On Tue, Apr 14, 2015 at 05:06:27PM +0300, Jarkko Nikula wrote: > > > [...] > > > > > > > > > > > > Which makes me thinking how does those strings describe the SSP port > > > > > > setup? E.g. do they reflect what port is actually used or could it > > > > > > be possible that those are just driver strings but firmware could > > > > > > have been tuned for SSP0? If I looked at earlier right, Teclast has > > > > > > the low pin-count Baytrail without SSP2 but I'm not sure about that. > > > > > The FW doesnt provide way to changes ports from driver in this, so this > > > > > doesnt mean much. If we are sure it is not ssp2, I can provide ssp1 fw for > > > > > test > > > > I ahve pushed latest BYT version we have as well as added binary for second > > > > SSP port. Please give it a try: > > > > > > > > https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/commit/?h=b… > > > > > > > > > > Thanks Vinod, but I see these are in the ELF format, and the mainline > > > driver does not support this format, at least the last time I checked it > > > didn't. > > My bad, updated the patch now with right format > > > > https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/commit/?h=b… > > > > (Adding Michele to CC) > > OK, so Jarkko was right: after a quick test I confirm that with > fw_sst_0f28_ssp0.bin I get sound; this is quite an improvement, > thanks! Damn, completely missed.. Hi Vinod, if it's possible, could you please generate a fw_sst_0f28_ssp1.bin also, so I can test my driver (mine was for the Asus X205TA laptop, where the codec is on SSP1 port)? Thanks, Michele

4 9

[alsa-devel] [PATCH BAT V1 0/7] BAT: Add Basic Audio Tester command line tool
by han.lu＠intel.com 21 Sep '15

21 Sep '15

From: "Lu, Han" <han.lu(a)intel.com> BAT (Basic Audio Tester) is a simple command line utility intended to automate audio driver and sound server QA testing with no human interaction. BAT works by playing an audio stream and capturing the same stream in either a digital or analog loopback. It then compares the captured stream to the original to determine if the testcase passes or fails. The main idea of frequency detecting is: The analysis function reads data from wav file, run fft against the data to get magnitude of frequency vectors, and then calculates the average value and standard deviation of frequency vectors. After that, we define a threshold: threshold = 3 * standard_deviation + average_value Frequencies with amplitude larger than threshold will be recognized as a peak, and the frequency with largest peak value will be recognized as a detected frequency. BAT then compares the detected frequency to target frequency, to decide if the detecting passes or fails. BAT supports 4 working modes: 1. single line playback; 2. single line capture and analysis; 3. playback and capture in loop, and analyze captured data; 4. local analyze without actual playback or capture. BAT will check devices input by user to decide which mode to use. BAT will create threads for playback and record, and will run spectrum analysis for mode 2, 3 or 4. Lu, Han (7): BAT: Add initial functions BAT: Add common definitions and functions BAT: Add playback and record functions BAT: Add signal generator BAT: Add converting functions BAT: Add spectrum analysis functions BAT: Add Makefile and configures Makefile.am | 3 + bat/Makefile.am | 14 ++ bat/alsa.c | 618 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/alsa.h | 20 ++ bat/analyze.c | 314 ++++++++++++++++++++++++++++ bat/analyze.h | 16 ++ bat/bat.c | 608 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/common.c | 198 ++++++++++++++++++ bat/common.h | 169 ++++++++++++++++ bat/convert.c | 113 +++++++++++ bat/convert.h | 23 +++ bat/signal.c | 88 ++++++++ configure.ac | 19 +- 13 files changed, 2202 insertions(+), 1 deletion(-) create mode 100644 bat/Makefile.am create mode 100644 bat/alsa.c create mode 100644 bat/alsa.h create mode 100644 bat/analyze.c create mode 100644 bat/analyze.h create mode 100644 bat/bat.c create mode 100644 bat/common.c create mode 100644 bat/common.h create mode 100644 bat/convert.c create mode 100644 bat/convert.h create mode 100644 bat/signal.c -- 1.9.1

7 25

[alsa-devel] [PATCH - 2/2] HDA - add headset-sonicmaster and headset-multi-sonicmaster models
by Raymond Yau 21 Sep '15

21 Sep '15

>From e97383136359c5d3ef93282888540bbb20939a9e Mon Sep 17 00:00:00 2001 From: Raymond Yau <superquad.vortex2(a)gmail.com> Date: Sun, 20 Sep 2015 22:18:56 +0800 Subject: [PATCH - 2/2] HDA - add headset-sonicmaster and headset-multi-sonicmaster models add headset only model for ALC668 with Sonic Master Subwoofer add 625ms delay after headset is plugged for correct detection of headset type for ALC668 add headset multi model with Sonice Master Subwoofer update HD-Audio-Models.txt Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ec099d4..8baddac 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -55,7 +55,12 @@ ALC66x/67x/892 asus-mode7 ASUS asus-mode8 ASUS inv-dmic Inverted internal mic workaround + +ALC668 +====== dell-headset-multi Headset jack, which can also be used as mic-in + headset-sonicmaster Headset jack and SonicMaster Subwoofer + headset-multi-sonicmaster Headset, HP or Mic and SonicMaster Subwoofer ALC680 ====== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a75b561..91de00c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4029,7 +4029,13 @@ static void alc_update_headset_mode(struct hda_codec *codec) PIN_VREFHIZ); } spec->current_headset_mode = new_headset_mode; - + if (codec->core.vendor_id == 0x10ec0668) { + if (spec->gen.hs_mic_use_hp_sense && spec->gen.auto_mic) { + if (new_headset_mode == ALC_HEADSET_MODE_HEADPHONE) + msleep(625); + snd_hda_gen_mic_autoswitch(codec, NULL); + }; + } snd_hda_gen_update_outputs(codec); } @@ -6071,6 +6077,42 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, } } +static void alc668_fixup_asus_headset_only(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl hs_mic_pin[] = { + { 0x1b, 0x03a1113c }, /* headset mic */ + { } + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, hs_mic_pin); + spec->gen.hs_mic_use_hp_sense = 1; + alc_fixup_disable_aamix(codec, fix, action); + break; + } +} + +static void alc668_fixup_sonicmaster(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl bass_pin[] = { + { 0x1a, 0x04110011 }, /* external subwoofer */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, bass_pin); + break; + case HDA_FIXUP_ACT_BUILD: + alc_fixup_bass_chmap(codec, fix, action); + break; + } +} + static struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -6130,6 +6172,9 @@ enum { ALC668_FIXUP_AUTO_MUTE, ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, + ALC668_FIXUP_HEADSET_ONLY, + ALC668_FIXUP_HEADSET_SONICMASTER, + ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, }; static const struct hda_fixup alc662_fixups[] = { @@ -6370,6 +6415,24 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC668_FIXUP_HEADSET_ONLY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_asus_headset_only, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE, + }, + [ALC668_FIXUP_HEADSET_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_ONLY, + }, + [ALC668_FIXUP_HEADSET_MULTI_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_DISABLE_AAMIX, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6476,6 +6539,9 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, + {.id = ALC668_FIXUP_HEADSET_SONICMASTER, .name = "headset-sonicmaster"}, + {.id = ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, + .name = "headset-multi-sonicmaster"}, {} }; -- 2.5.0

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: rt5645: Add dmi for Broadwell
by Bard Liao 20 Sep '15

20 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 66e9ce6..2fc437f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3210,6 +3210,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3242,7 +3267,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwel) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH v3 1/2] mfd: arizona: Add register bits to support the ANC block
by Charles Keepax 20 Sep '15

20 Sep '15

Some Arizona devices have a hardware ANC block present. This patch adds the registers necessary to configure this hardware block. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Added missing default for 0xF0A Thanks, Charles drivers/mfd/wm5110-tables.c | 186 +++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/registers.h | 70 ++++++++++++ 2 files changed, 256 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index dae04dd..f154991 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -1637,6 +1637,185 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000EF8, 0x0000 }, /* R3832 - ISRC 3 CTRL 3 */ { 0x00000F00, 0x0000 }, /* R3840 - Clock Control */ { 0x00000F01, 0x0000 }, /* R3841 - ANC_SRC */ + { 0x00000F08, 0x001c }, /* R3848 - ANC Coefficient */ + { 0x00000F09, 0x0000 }, /* R3849 - ANC Coefficient */ + { 0x00000F0A, 0x0000 }, /* R3850 - ANC Coefficient */ + { 0x00000F0B, 0x0000 }, /* R3851 - ANC Coefficient */ + { 0x00000F0C, 0x0000 }, /* R3852 - ANC Coefficient */ + { 0x00000F0D, 0x0000 }, /* R3853 - ANC Coefficient */ + { 0x00000F0E, 0x0000 }, /* R3854 - ANC Coefficient */ + { 0x00000F0F, 0x0000 }, /* R3855 - ANC Coefficient */ + { 0x00000F10, 0x0000 }, /* R3856 - ANC Coefficient */ + { 0x00000F11, 0x0000 }, /* R3857 - ANC Coefficient */ + { 0x00000F12, 0x0000 }, /* R3858 - ANC Coefficient */ + { 0x00000F15, 0x0000 }, /* R3861 - FCL Filter Control */ + { 0x00000F17, 0x0004 }, /* R3863 - FCL ADC Reformatter Control */ + { 0x00000F18, 0x0004 }, /* R3864 - ANC Coefficient */ + { 0x00000F19, 0x0002 }, /* R3865 - ANC Coefficient */ + { 0x00000F1A, 0x0000 }, /* R3866 - ANC Coefficient */ + { 0x00000F1B, 0x0010 }, /* R3867 - ANC Coefficient */ + { 0x00000F1C, 0x0000 }, /* R3868 - ANC Coefficient */ + { 0x00000F1D, 0x0000 }, /* R3869 - ANC Coefficient */ + { 0x00000F1E, 0x0000 }, /* R3870 - ANC Coefficient */ + { 0x00000F1F, 0x0000 }, /* R3871 - ANC Coefficient */ + { 0x00000F20, 0x0000 }, /* R3872 - ANC Coefficient */ + { 0x00000F21, 0x0000 }, /* R3873 - ANC Coefficient */ + { 0x00000F22, 0x0000 }, /* R3874 - ANC Coefficient */ + { 0x00000F23, 0x0000 }, /* R3875 - ANC Coefficient */ + { 0x00000F24, 0x0000 }, /* R3876 - ANC Coefficient */ + { 0x00000F25, 0x0000 }, /* R3877 - ANC Coefficient */ + { 0x00000F26, 0x0000 }, /* R3878 - ANC Coefficient */ + { 0x00000F27, 0x0000 }, /* R3879 - ANC Coefficient */ + { 0x00000F28, 0x0000 }, /* R3880 - ANC Coefficient */ + { 0x00000F29, 0x0000 }, /* R3881 - ANC Coefficient */ + { 0x00000F2A, 0x0000 }, /* R3882 - ANC Coefficient */ + { 0x00000F2B, 0x0000 }, /* R3883 - ANC Coefficient */ + { 0x00000F2C, 0x0000 }, /* R3884 - ANC Coefficient */ + { 0x00000F2D, 0x0000 }, /* R3885 - ANC Coefficient */ + { 0x00000F2E, 0x0000 }, /* R3886 - ANC Coefficient */ + { 0x00000F2F, 0x0000 }, /* R3887 - ANC Coefficient */ + { 0x00000F30, 0x0000 }, /* R3888 - ANC Coefficient */ + { 0x00000F31, 0x0000 }, /* R3889 - ANC Coefficient */ + { 0x00000F32, 0x0000 }, /* R3890 - ANC Coefficient */ + { 0x00000F33, 0x0000 }, /* R3891 - ANC Coefficient */ + { 0x00000F34, 0x0000 }, /* R3892 - ANC Coefficient */ + { 0x00000F35, 0x0000 }, /* R3893 - ANC Coefficient */ + { 0x00000F36, 0x0000 }, /* R3894 - ANC Coefficient */ + { 0x00000F37, 0x0000 }, /* R3895 - ANC Coefficient */ + { 0x00000F38, 0x0000 }, /* R3896 - ANC Coefficient */ + { 0x00000F39, 0x0000 }, /* R3897 - ANC Coefficient */ + { 0x00000F3A, 0x0000 }, /* R3898 - ANC Coefficient */ + { 0x00000F3B, 0x0000 }, /* R3899 - ANC Coefficient */ + { 0x00000F3C, 0x0000 }, /* R3900 - ANC Coefficient */ + { 0x00000F3D, 0x0000 }, /* R3901 - ANC Coefficient */ + { 0x00000F3E, 0x0000 }, /* R3902 - ANC Coefficient */ + { 0x00000F3F, 0x0000 }, /* R3903 - ANC Coefficient */ + { 0x00000F40, 0x0000 }, /* R3904 - ANC Coefficient */ + { 0x00000F41, 0x0000 }, /* R3905 - ANC Coefficient */ + { 0x00000F42, 0x0000 }, /* R3906 - ANC Coefficient */ + { 0x00000F43, 0x0000 }, /* R3907 - ANC Coefficient */ + { 0x00000F44, 0x0000 }, /* R3908 - ANC Coefficient */ + { 0x00000F45, 0x0000 }, /* R3909 - ANC Coefficient */ + { 0x00000F46, 0x0000 }, /* R3910 - ANC Coefficient */ + { 0x00000F47, 0x0000 }, /* R3911 - ANC Coefficient */ + { 0x00000F48, 0x0000 }, /* R3912 - ANC Coefficient */ + { 0x00000F49, 0x0000 }, /* R3913 - ANC Coefficient */ + { 0x00000F4A, 0x0000 }, /* R3914 - ANC Coefficient */ + { 0x00000F4B, 0x0000 }, /* R3915 - ANC Coefficient */ + { 0x00000F4C, 0x0000 }, /* R3916 - ANC Coefficient */ + { 0x00000F4D, 0x0000 }, /* R3917 - ANC Coefficient */ + { 0x00000F4E, 0x0000 }, /* R3918 - ANC Coefficient */ + { 0x00000F4F, 0x0000 }, /* R3919 - ANC Coefficient */ + { 0x00000F50, 0x0000 }, /* R3920 - ANC Coefficient */ + { 0x00000F51, 0x0000 }, /* R3921 - ANC Coefficient */ + { 0x00000F52, 0x0000 }, /* R3922 - ANC Coefficient */ + { 0x00000F53, 0x0000 }, /* R3923 - ANC Coefficient */ + { 0x00000F54, 0x0000 }, /* R3924 - ANC Coefficient */ + { 0x00000F55, 0x0000 }, /* R3925 - ANC Coefficient */ + { 0x00000F56, 0x0000 }, /* R3926 - ANC Coefficient */ + { 0x00000F57, 0x0000 }, /* R3927 - ANC Coefficient */ + { 0x00000F58, 0x0000 }, /* R3928 - ANC Coefficient */ + { 0x00000F59, 0x0000 }, /* R3929 - ANC Coefficient */ + { 0x00000F5A, 0x0000 }, /* R3930 - ANC Coefficient */ + { 0x00000F5B, 0x0000 }, /* R3931 - ANC Coefficient */ + { 0x00000F5C, 0x0000 }, /* R3932 - ANC Coefficient */ + { 0x00000F5D, 0x0000 }, /* R3933 - ANC Coefficient */ + { 0x00000F5E, 0x0000 }, /* R3934 - ANC Coefficient */ + { 0x00000F5F, 0x0000 }, /* R3935 - ANC Coefficient */ + { 0x00000F60, 0x0000 }, /* R3936 - ANC Coefficient */ + { 0x00000F61, 0x0000 }, /* R3937 - ANC Coefficient */ + { 0x00000F62, 0x0000 }, /* R3938 - ANC Coefficient */ + { 0x00000F63, 0x0000 }, /* R3939 - ANC Coefficient */ + { 0x00000F64, 0x0000 }, /* R3940 - ANC Coefficient */ + { 0x00000F65, 0x0000 }, /* R3941 - ANC Coefficient */ + { 0x00000F66, 0x0000 }, /* R3942 - ANC Coefficient */ + { 0x00000F67, 0x0000 }, /* R3943 - ANC Coefficient */ + { 0x00000F68, 0x0000 }, /* R3944 - ANC Coefficient */ + { 0x00000F69, 0x0000 }, /* R3945 - ANC Coefficient */ + { 0x00000F70, 0x0000 }, /* R3952 - FCR Filter Control */ + { 0x00000F72, 0x0004 }, /* R3954 - FCR ADC Reformatter Control */ + { 0x00000F73, 0x0004 }, /* R3955 - ANC Coefficient */ + { 0x00000F74, 0x0002 }, /* R3956 - ANC Coefficient */ + { 0x00000F75, 0x0000 }, /* R3957 - ANC Coefficient */ + { 0x00000F76, 0x0010 }, /* R3958 - ANC Coefficient */ + { 0x00000F77, 0x0000 }, /* R3959 - ANC Coefficient */ + { 0x00000F78, 0x0000 }, /* R3960 - ANC Coefficient */ + { 0x00000F79, 0x0000 }, /* R3961 - ANC Coefficient */ + { 0x00000F7A, 0x0000 }, /* R3962 - ANC Coefficient */ + { 0x00000F7B, 0x0000 }, /* R3963 - ANC Coefficient */ + { 0x00000F7C, 0x0000 }, /* R3964 - ANC Coefficient */ + { 0x00000F7D, 0x0000 }, /* R3965 - ANC Coefficient */ + { 0x00000F7E, 0x0000 }, /* R3966 - ANC Coefficient */ + { 0x00000F7F, 0x0000 }, /* R3967 - ANC Coefficient */ + { 0x00000F80, 0x0000 }, /* R3968 - ANC Coefficient */ + { 0x00000F81, 0x0000 }, /* R3969 - ANC Coefficient */ + { 0x00000F82, 0x0000 }, /* R3970 - ANC Coefficient */ + { 0x00000F83, 0x0000 }, /* R3971 - ANC Coefficient */ + { 0x00000F84, 0x0000 }, /* R3972 - ANC Coefficient */ + { 0x00000F85, 0x0000 }, /* R3973 - ANC Coefficient */ + { 0x00000F86, 0x0000 }, /* R3974 - ANC Coefficient */ + { 0x00000F87, 0x0000 }, /* R3975 - ANC Coefficient */ + { 0x00000F88, 0x0000 }, /* R3976 - ANC Coefficient */ + { 0x00000F89, 0x0000 }, /* R3977 - ANC Coefficient */ + { 0x00000F8A, 0x0000 }, /* R3978 - ANC Coefficient */ + { 0x00000F8B, 0x0000 }, /* R3979 - ANC Coefficient */ + { 0x00000F8C, 0x0000 }, /* R3980 - ANC Coefficient */ + { 0x00000F8D, 0x0000 }, /* R3981 - ANC Coefficient */ + { 0x00000F8E, 0x0000 }, /* R3982 - ANC Coefficient */ + { 0x00000F8F, 0x0000 }, /* R3983 - ANC Coefficient */ + { 0x00000F90, 0x0000 }, /* R3984 - ANC Coefficient */ + { 0x00000F91, 0x0000 }, /* R3985 - ANC Coefficient */ + { 0x00000F92, 0x0000 }, /* R3986 - ANC Coefficient */ + { 0x00000F93, 0x0000 }, /* R3987 - ANC Coefficient */ + { 0x00000F94, 0x0000 }, /* R3988 - ANC Coefficient */ + { 0x00000F95, 0x0000 }, /* R3989 - ANC Coefficient */ + { 0x00000F96, 0x0000 }, /* R3990 - ANC Coefficient */ + { 0x00000F97, 0x0000 }, /* R3991 - ANC Coefficient */ + { 0x00000F98, 0x0000 }, /* R3992 - ANC Coefficient */ + { 0x00000F99, 0x0000 }, /* R3993 - ANC Coefficient */ + { 0x00000F9A, 0x0000 }, /* R3994 - ANC Coefficient */ + { 0x00000F9B, 0x0000 }, /* R3995 - ANC Coefficient */ + { 0x00000F9C, 0x0000 }, /* R3996 - ANC Coefficient */ + { 0x00000F9D, 0x0000 }, /* R3997 - ANC Coefficient */ + { 0x00000F9E, 0x0000 }, /* R3998 - ANC Coefficient */ + { 0x00000F9F, 0x0000 }, /* R3999 - ANC Coefficient */ + { 0x00000FA0, 0x0000 }, /* R4000 - ANC Coefficient */ + { 0x00000FA1, 0x0000 }, /* R4001 - ANC Coefficient */ + { 0x00000FA2, 0x0000 }, /* R4002 - ANC Coefficient */ + { 0x00000FA3, 0x0000 }, /* R4003 - ANC Coefficient */ + { 0x00000FA4, 0x0000 }, /* R4004 - ANC Coefficient */ + { 0x00000FA5, 0x0000 }, /* R4005 - ANC Coefficient */ + { 0x00000FA6, 0x0000 }, /* R4006 - ANC Coefficient */ + { 0x00000FA7, 0x0000 }, /* R4007 - ANC Coefficient */ + { 0x00000FA8, 0x0000 }, /* R4008 - ANC Coefficient */ + { 0x00000FA9, 0x0000 }, /* R4009 - ANC Coefficient */ + { 0x00000FAA, 0x0000 }, /* R4010 - ANC Coefficient */ + { 0x00000FAB, 0x0000 }, /* R4011 - ANC Coefficient */ + { 0x00000FAC, 0x0000 }, /* R4012 - ANC Coefficient */ + { 0x00000FAD, 0x0000 }, /* R4013 - ANC Coefficient */ + { 0x00000FAE, 0x0000 }, /* R4014 - ANC Coefficient */ + { 0x00000FAF, 0x0000 }, /* R4015 - ANC Coefficient */ + { 0x00000FB0, 0x0000 }, /* R4016 - ANC Coefficient */ + { 0x00000FB1, 0x0000 }, /* R4017 - ANC Coefficient */ + { 0x00000FB2, 0x0000 }, /* R4018 - ANC Coefficient */ + { 0x00000FB3, 0x0000 }, /* R4019 - ANC Coefficient */ + { 0x00000FB4, 0x0000 }, /* R4020 - ANC Coefficient */ + { 0x00000FB5, 0x0000 }, /* R4021 - ANC Coefficient */ + { 0x00000FB6, 0x0000 }, /* R4022 - ANC Coefficient */ + { 0x00000FB7, 0x0000 }, /* R4023 - ANC Coefficient */ + { 0x00000FB8, 0x0000 }, /* R4024 - ANC Coefficient */ + { 0x00000FB9, 0x0000 }, /* R4025 - ANC Coefficient */ + { 0x00000FBA, 0x0000 }, /* R4026 - ANC Coefficient */ + { 0x00000FBB, 0x0000 }, /* R4027 - ANC Coefficient */ + { 0x00000FBC, 0x0000 }, /* R4028 - ANC Coefficient */ + { 0x00000FBD, 0x0000 }, /* R4029 - ANC Coefficient */ + { 0x00000FBE, 0x0000 }, /* R4030 - ANC Coefficient */ + { 0x00000FBF, 0x0000 }, /* R4031 - ANC Coefficient */ + { 0x00000FC0, 0x0000 }, /* R4032 - ANC Coefficient */ + { 0x00000FC1, 0x0000 }, /* R4033 - ANC Coefficient */ + { 0x00000FC2, 0x0000 }, /* R4034 - ANC Coefficient */ + { 0x00000FC3, 0x0000 }, /* R4035 - ANC Coefficient */ + { 0x00000FC4, 0x0000 }, /* R4036 - ANC Coefficient */ { 0x00001100, 0x0010 }, /* R4352 - DSP1 Control 1 */ { 0x00001101, 0x0000 }, /* R4353 - DSP1 Clocking 1 */ { 0x00001200, 0x0010 }, /* R4608 - DSP2 Control 1 */ @@ -2716,6 +2895,13 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_CLOCK_CONTROL: case ARIZONA_ANC_SRC: case ARIZONA_DSP_STATUS: + case ARIZONA_ANC_COEFF_START ... ARIZONA_ANC_COEFF_END: + case ARIZONA_FCL_FILTER_CONTROL: + case ARIZONA_FCL_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCL_COEFF_START ... ARIZONA_FCL_COEFF_END: + case ARIZONA_FCR_FILTER_CONTROL: + case ARIZONA_FCR_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCR_COEFF_START ... ARIZONA_FCR_COEFF_END: case ARIZONA_DSP1_CONTROL_1: case ARIZONA_DSP1_CLOCKING_1: case ARIZONA_DSP1_STATUS_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 11affb3..fdb12bf 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -1050,6 +1050,16 @@ #define ARIZONA_CLOCK_CONTROL 0xF00 #define ARIZONA_ANC_SRC 0xF01 #define ARIZONA_DSP_STATUS 0xF02 +#define ARIZONA_ANC_COEFF_START 0xF08 +#define ARIZONA_ANC_COEFF_END 0xF12 +#define ARIZONA_FCL_FILTER_CONTROL 0xF15 +#define ARIZONA_FCL_ADC_REFORMATTER_CONTROL 0xF17 +#define ARIZONA_FCL_COEFF_START 0xF18 +#define ARIZONA_FCL_COEFF_END 0xF69 +#define ARIZONA_FCR_FILTER_CONTROL 0xF70 +#define ARIZONA_FCR_ADC_REFORMATTER_CONTROL 0xF72 +#define ARIZONA_FCR_COEFF_START 0xF73 +#define ARIZONA_FCR_COEFF_END 0xFC4 #define ARIZONA_DSP1_CONTROL_1 0x1100 #define ARIZONA_DSP1_CLOCKING_1 0x1101 #define ARIZONA_DSP1_STATUS_1 0x1104 @@ -7823,6 +7833,66 @@ #define ARIZONA_ISRC3_NOTCH_ENA_WIDTH 1 /* ISRC3_NOTCH_ENA */ /* + * R3840 (0xF00) - Clock Control + */ +#define ARIZONA_EXT_NG_SEL_CLR 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_MASK 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_SHIFT 7 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_WIDTH 1 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_SET 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_MASK 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_SHIFT 6 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_WIDTH 1 /* EXT_NG_SEL_SET */ +#define ARIZONA_CLK_R_ENA_CLR 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_MASK 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_SHIFT 5 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_WIDTH 1 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_SET 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_MASK 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_SHIFT 4 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_WIDTH 1 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_CLR 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_MASK 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_SHIFT 3 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_WIDTH 1 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_SET 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_MASK 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_SHIFT 2 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_WIDTH 1 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_L_ENA_CLR 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_MASK 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_SHIFT 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_WIDTH 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_SET 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_MASK 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_SHIFT 0 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_WIDTH 1 /* CLK_L_ENA_SET */ + +/* + * R3841 (0xF01) - ANC SRC + */ +#define ARIZONA_IN_RXANCR_SEL_MASK 0x0070 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_SHIFT 4 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_WIDTH 3 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCL_SEL_MASK 0x0007 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_SHIFT 0 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_WIDTH 3 /* IN_RXANCL_SEL - [0:2] */ + +/* + * R3863 (0xF17) - FCL ADC Reformatter Control + */ +#define ARIZONA_FCL_MIC_MODE_SEL 0x000C /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_SHIFT 2 /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_WIDTH 2 /* FCL_MIC_MODE_SEL - [2:3] */ + +/* + * R3954 (0xF72) - FCR ADC Reformatter Control + */ +#define ARIZONA_FCR_MIC_MODE_SEL 0x000C /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_SHIFT 2 /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_WIDTH 2 /* FCR_MIC_MODE_SEL - [2:3] */ + +/* * R4352 (0x1100) - DSP1 Control 1 */ #define ARIZONA_DSP1_RATE_MASK 0x7800 /* DSP1_RATE - [14:11] */ -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: ak4613: add Digital Playback Volume support
by Kuninori Morimoto 20 Sep '15

20 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> For L/ROUT1 to L/ROUT6 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index fd96a8f..07a2664 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -22,6 +22,7 @@ #include <linux/regmap.h> #include <sound/soc.h> #include <sound/pcm_params.h> +#include <sound/tlv.h> #define PW_MGMT1 0x00 /* Power Management 1 */ #define PW_MGMT2 0x01 /* Power Management 2 */ @@ -91,6 +92,31 @@ struct ak4613_interface { struct ak4613_formats playback; }; +/* + * Playback Volume + * + * max : 0x00 : 0 dB + * ( 0.5 dB step ) + * min : 0xFE : -127.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new ak4613_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6, + 0, 0xFF, 1, out_tlv), +}; + static const struct reg_default ak4613_reg[] = { { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, @@ -397,6 +423,8 @@ static int ak4613_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { .resume = ak4613_resume, .set_bias_level = ak4613_set_bias_level, + .controls = ak4613_snd_controls, + .num_controls = ARRAY_SIZE(ak4613_snd_controls), .dapm_widgets = ak4613_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), .dapm_routes = ak4613_intercon, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix devm_kasprintf format string
by Peter Ujfalusi 20 Sep '15

20 Sep '15

The '\n' at the end of the format string is not needed. It adds an extra line break when doing cat /proc/interrupts Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1260b315a96c..e631f96f9cf7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1684,7 +1684,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "common"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_common_irq_handler, @@ -1701,7 +1701,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "rx"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_rx_irq_handler, @@ -1716,7 +1716,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "tx"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_tx_irq_handler, -- 2.5.2

2 1

[alsa-devel] [PATCH] ASoC: rockchip: Fix missing CLKDEV_LOOKUP dependency
by Axel Lin 20 Sep '15

20 Sep '15

Make SND_SOC_ROCKCHIP_MAX98090 and SND_SOC_ROCKCHIP_RT5645 depend on CLKDEV_LOOKUP to fix below build warning: warning: (SND_SOC_ROCKCHIP_MAX98090 && SND_SOC_ROCKCHIP_RT5645) selects SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && CLKDEV_LOOKUP && SND_SOC_ROCKCHIP) Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/rockchip/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 58bae8e..5709057 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -17,7 +17,7 @@ config SND_SOC_ROCKCHIP_I2S config SND_SOC_ROCKCHIP_MAX98090 tristate "ASoC support for Rockchip boards using a MAX98090 codec" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98090 select SND_SOC_TS3A227E @@ -27,7 +27,7 @@ config SND_SOC_ROCKCHIP_MAX98090 config SND_SOC_ROCKCHIP_RT5645 tristate "ASoC support for Rockchip boards using a RT5645/RT5650 codec" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP select SND_SOC_ROCKCHIP_I2S select SND_SOC_RT5645 help -- 2.1.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm0010: disable regulator on error
by Sudip Mukherjee 20 Sep '15

20 Sep '15

We have done regulator_bulk_enable() while booting the DSP but on the error exit path we have not disbled it. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/wm0010.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 8434d45..79a7cd3 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -739,8 +739,6 @@ static int wm0010_boot(struct snd_soc_codec *codec) abort: /* Put the chip back into reset */ wm0010_halt(codec); - mutex_unlock(&wm0010->lock); - return ret; err_core: mutex_unlock(&wm0010->lock); -- 1.9.1

3 9

[alsa-devel] [PATCH RFC v2 0/7] Implement generic ASoC HDMI codec
by Jyri Sarha 20 Sep '15

20 Sep '15

Only the first three patches are meant for serious review. The ASoC part should be ready for review in other respects but the EDID SADs handling is waiting for Russell King's DRM ELD helper patches. There is a copy-pasted not-to-be-merged patch with the same functionality in the patch series. The tda998x part is not complete and I do not think Jean-Francois' binding are ready as such either. At least the usage of reg property in a port node for something that is not an address or even guaranteed to be unique is not a good idea. Some documentation of how to configure AP0-AP4 pins (REG_ENA_AP?) would help in making a proper binding for the tda998x audio configuration. These patches, my tilcdc refactoring[1], and my latest BCLK fixes for davinci-mcasp driver [2], can be found in a branch that produces working HDMI audio on Beaglebone-Black here: https://github.com/jsarha/linux.git linux-master-bbb-hdmi-20150512 [1] http://lists.freedesktop.org/archives/dri-devel/2015-May/082537.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-April/090974.html Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports DO NOT MERGE Jyri Sarha (6): ASoC: core: If component doesn't have of_node use parent's node instead ASoC: hdmi: Remove obsolete dummy HDMI codec ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add ELD based audio pcm rules DO NOT MERGE drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality DO NOT MERGE ARM: dts: am335x-boneblack: Add HDMI audio support DO NOT MERGE .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++ arch/arm/boot/dts/am335x-boneblack.dts | 78 ++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 328 +++++++++++- include/sound/hdmi-codec.h | 99 ++++ include/sound/tda998x.h | 8 + sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/hdmi-codec.c | 560 +++++++++++++++++++++ sound/soc/codecs/hdmi.c | 109 ---- sound/soc/soc-core.c | 14 +- 11 files changed, 1129 insertions(+), 125 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/hdmi-codec.c delete mode 100644 sound/soc/codecs/hdmi.c -- 1.9.1

2 12

[alsa-devel] [PATCH v2] ASoC: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 20 Sep '15

20 Sep '15

Add the capability to use multiple codecs on the same DAI linke where one codec is used for playback and another one is used for capture. Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS microphone for capture. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- The patch was created after a discussion on the alsa-devel mailing list as to how best to implement this functionality. (http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html). V2: Minor code change, otherwise the differences compared to V1 are solely related to comments, in particular considerations given to the potential consequences of the patch. After consideration, it seems to me that the patch as it stands augments the current framework with the functionality indicated in the commit message above, without restricting other current uses. Further functionality could be added, but IMHO that should be done as the need arises, when it can be properly tested in a real-world setup. sound/soc/soc-pcm.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4..08407ba 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -34,6 +34,24 @@ #define DPCM_MAX_BE_USERS 8 +/* + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * Returns true if the DAI supports the indicated stream type. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return codec_stream->rates; +} + /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* + * Skip CODECs which don't support the current stream type. + * Otherwise, since the rate, channel, and format values will + * zero in that case, we would have no usable settings left, + * causing the resulting setup to fail. + * At least one CODEC should match, otherwise we should have + * bailed out on a higher level, since there would be no + * CODEC to support the transfer direction in that case. + */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +859,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* + * Skip CODECs which don't support the current stream type, + * the idea being that if a CODEC is not used for the currently + * set up transfer direction, it should not need to be + * configured, especially since the configuration used might + * not even be supported by that CODEC. There may be cases + * however where a CODEC needs to be set up although it is + * actually not being used for the transfer, e.g. if a + * capture-only CODEC is acting as an LRCLK and/or BCLK master + * for the DAI link including a playback-only CODEC. + * If this becomes necessary, we will have to augment the + * machine driver setup with information on how to act, so + * we can do the right thing here. + */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 8

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fixing invalid fmt SND_SOC_DAIFMT_AC97 test
by Arnaud Mouiche 20 Sep '15

20 Sep '15

SND_SOC_DAIFMT_AC97 (6) has similar bits than SND_SOC_DAIFMT_DSP_A (4) and can't be tested with a simple AND operation Signed-off-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b0c8f7..c25a1e8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -945,7 +946,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: add wm8960 support
by Zidan Wang 19 Sep '15

19 Sep '15

add wm8960 support for fsl-asoc-card Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ sound/soc/fsl/fsl-asoc-card.c | 37 ++++++++++++++++++++-- 2 files changed, 36 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..fa0a10d 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -59,6 +59,8 @@ Optional properties: - audio-asrc : The phandle of ASRC. It can be absent if there's no need to add ASRC support via DPCM. + - gpr : The gpr handle + Example: sound-cs42888 { compatible = "fsl,imx-audio-cs42888"; diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..c058e68 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -16,6 +16,7 @@ #include <linux/of_platform.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <linux/mfd/syscon.h> #include "fsl_esai.h" #include "fsl_sai.h" @@ -23,6 +24,7 @@ #include "../codecs/sgtl5000.h" #include "../codecs/wm8962.h" +#include "../codecs/wm8960.h" #define RX 0 #define TX 1 @@ -90,6 +92,7 @@ struct fsl_asoc_card_priv { u32 asrc_format; u32 dai_fmt; char name[32]; + struct regmap *gpr; }; /** @@ -400,7 +403,7 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) static int fsl_asoc_card_probe(struct platform_device *pdev) { - struct device_node *cpu_np, *codec_np, *asrc_np; + struct device_node *cpu_np, *codec_np, *asrc_np, *gpr_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; struct platform_device *cpu_pdev; @@ -443,6 +446,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); + gpr_np = of_parse_phandle(np, "gpr", 0); + if (gpr_np) { + priv->gpr = syscon_node_to_regmap(gpr_np); + if (IS_ERR(priv->gpr)) { + ret = PTR_ERR(priv->gpr); + dev_err(&pdev->dev, "failed to get gpr regmap\n"); + goto fail; + } + /* + * for imx6ul we should set SAI2_MCLK_DIR to enable + * codec MCLK + */ + if (of_device_is_compatible(np, "fsl,imx6ul-evk-wm8960")) + regmap_update_bits(priv->gpr, 4, 1 << 20, 1 << 20); + } + /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ codec_clk = clk_get(&codec_dev->dev, NULL); if (!IS_ERR(codec_clk)) { @@ -475,6 +494,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; + priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; @@ -521,7 +545,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; - priv->dai_link[0].codec_dai_name = codec_dev->name; + if (strstr(codec_dev->name, "wm8960")) + priv->dai_link[0].codec_dai_name = "wm8960-hifi"; + else + priv->dai_link[0].codec_dai_name = codec_dev->name; priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -530,7 +557,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exsits */ priv->dai_link[1].cpu_of_node = asrc_np; priv->dai_link[1].platform_of_node = asrc_np; - priv->dai_link[2].codec_dai_name = codec_dev->name; + if (strstr(codec_dev->name, "wm8960")) + priv->dai_link[2].codec_dai_name = "wm8960-hifi"; + else + priv->dai_link[2].codec_dai_name = codec_dev->name; priv->dai_link[2].codec_of_node = codec_np; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; @@ -578,6 +608,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, + { .compatible = "fsl,imx-audio-wm8960", }, {} }; -- 1.9.1

4 15

[alsa-devel] [RESEND PATCH 1/1] ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
by Adam Thomson 19 Sep '15

19 Sep '15

Current code incorrectly treats dai format for AC97 as bit mask whereas it's actually an integer value. This causes DAI formats other than AC97 (e.g. DSP_B) to trigger AC97 related code, which is incorrect and breaks functionality. This patch fixes the code to correctly compare values to determine AC97 or not. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Patch based on v4.3-rc1. sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8ec6fb2..37c5cd4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -947,7 +948,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 1.9.3

3 2

[alsa-devel] [PATCH v2] ASoC: fsl-asoc-card: add AC'97 support
by Maciej S. Szmigiero 19 Sep '15

19 Sep '15

Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: don't use of_find_device_by_node() to get pointer to I2S/PCM CODEC device as this only matches platform bus devices while I2S/PCM CODECs supported by this driver sit on I2C bus. .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 140 ++++++++++++++++----- 2 files changed, 116 insertions(+), 34 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 0901d5e..1b05d1c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -407,7 +454,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; - struct clk *codec_clk; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,22 +479,24 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) + codec_dev = of_find_i2c_device_by_node(codec_np); + else + codec_dev = NULL; asrc_np = of_parse_phandle(np, "audio-asrc", 0); if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_dev) { + struct clk *codec_clk = clk_get(&codec_dev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,12 +533,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; goto asrc_fail; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { + dev_err(&pdev->dev, "failed to find codec device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -508,7 +565,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_dev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -532,8 +591,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -544,6 +621,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -579,14 +658,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", },

1 0

Re: [alsa-devel] Asus N751JK - push/merge patches on mailinglist
by Raymond Yau 18 Sep '15

18 Sep '15

2015-09-13 16:13 GMT+08:00 Takashi Iwai <tiwai(a)suse.de>: > On Sat, 12 Sep 2015 23:09:43 +0200, > Arthur Borsboom wrote: > > > > Hi Takashi, > > > > Previous month Raymond Yau has done great work to get the audio working > for > > the Asus N751JK laptop. > > He developed 7 patches, which I have tested and confirmed working on this > > laptop. > > > > Raymond is not responding for about two weeks (maybe on holiday?). > > Would you mind to post these patches on the mailinglist (the correct way) > > and/or merge them in a way the get into the mainline kernel a.s.a.p? > > Sorry, it can't be done like that. The patches aren't yet in > acceptable forms, and it needs more changes to be merged to mainline. > It must be done by someone who did it originally or who has the > hardware and test. > I don't have the hardware, the alternate approach is to create headset jack (SND_JACK_HEADSET) and skip the creation of headset mic phanton jack instead of headphone jack and headset mic jack since implement sense_nid and slave_nid is not easy to debug What you need is adding "Jack Headset" to pulseaudio analog-input-headset-mic.conf since "Headset Jack" kctl already in analog-out-headphones.conf static int get_input_jack_type(struct hda_codec *codec, hda_nid_t nid) { + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: case AC_JACK_SPEAKER: return SND_JACK_LINEOUT; case AC_JACK_HP_OUT: + if (spec->hs_mic_use_hp_sense && + (nid == auto_cfg_hp_pins(cfg)[0])) + return SND_JACK_HEADSET; return SND_JACK_HEADPHONE; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: const struct auto_pin_cfg *cfg) { const hda_nid_t *p; + struct hda_gen_spec *spec = codec->spec; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -482,6 +499,12 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, "Headphone Mic"); } else + if (cfg->inputs[i].is_headset_mic && + spec->hs_mic_use_hp_sense && + !is_jack_detectable(codec, cfg->inputs[i].pin)) { + err = 0; + + } else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, NULL); if (err < 0) @@ -489,14 +512,21 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, } for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg, NULL); + if (is_subwoofer(codec, *p, i, cfg->line_outs) && + is_external_pin(codec, *p)) + err = add_jack_kctl(codec, *p, cfg, "Ext Subwoofer"); + else + err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg, NULL); + if (i == 0 && spec->hs_mic_use_hp_sense) + err = add_jack_kctl(codec, *p, cfg, "Headset"); + else + err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; } change the pin to hp pin when mic pin is headset mic +bool is_headset_mic(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) + if (nid == cfg->inputs[i].pin) + return cfg->inputs[i].is_headset_mic; + return false; +} + @@ -4384,6 +4399,8 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; + if (is_headset_mic(codec, pin) && spec->hs_mic_use_hp_sense) + pin = auto_cfg_hp_pins(cfg)[0]; if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return;

2 5

[alsa-devel] [PATCH 1/4] ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 18 Sep '15

18 Sep '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..5c737f1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -802,10 +805,40 @@ static const struct of_device_id fsl_sai_ids[] = { { /* sentinel */ } }; +#if CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; -- 1.9.1

1 3

[alsa-devel] Intel SST on a Bay Trail tablet
by Antonio Ospite 18 Sep '15

18 Sep '15

Hi, I am trying to get the Intel SST driver working on a Teclast X98 Air 3G [1], it's a Bay Trail tablet. The tests below have been made with 4.0.0-rc1 and the firmware files from https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/tr… The DSDT on this device is quite broken, and I am patching it to get stuff working. The audio device works just fine under Android, I don't know if that's because the Android kernel is ignoring the DSDT and is hard-coding some platform data. Anyhow, with the untouched DSDT[2] I was getting this message in dmesg[3]: [ 11.344095] platform byt-rt5640: Driver byt-rt5640 requests probe deferral [ 11.404831] resource sanity check: requesting [mem 0x91000000-0x911fffff], which spans more than ADMA0F28:00 [mem 0x91000000-0x91000fff] [ 11.404833] ------------[ cut here ]------------ [ 11.404849] WARNING: CPU: 0 PID: 213 at arch/x86/mm/ioremap.c:197 __ioremap_caller+0x22c/0x350() ... I guessed this was because the DSDT defined an ADMA0F28 device with a memory resource overlapping with the LPMA device. So I removed the overlapping resources (ADMA0F28 and SSPX0000) in the DSDT and then the driver attempted to boot the audio device but failed with this message in dmesg[4]: [ 11.602291] baytrail-pcm-audio baytrail-pcm-audio: ipc: error DSP boot timeout ... Then I noticed that the Interrupt resources of the LPMA device were out of order in the DSDT, so I put the Interrupt 0x18 as the first one[5] and now the driver can boot the device and upload the firmware to it[6]: [ 11.433460] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 11.433468] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 11.433473] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 11.448257] byt-rt5640 byt-rt5640: rt5640-aif1 <-> baytrail-pcm-audio mapping ok ... But still no sound. I can see the mixer controls in alsamixer and I tried to initialize them copying from the alsactl output from the Android kernel[7], but no, still no sound. I do not know if that matters, but the driver still emits this message: [ 11.398304] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 11.398314] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 ... I noticed that on the Android kernel[8] the DMA channels for the ADMA0F28 device are actually configured, the intel_mid_dma driver sets up 4 channels for it along with 2 x 6 channels for the INTL9C60 device. On the mainline kernel only the INTL9C60 device gets configured, by the dw_dmac driver and with 2 x 8 channels. I do not know much about these details tho. Also, the codec driver on the Android kernel (byt_rt5642) seems to use one GPIO as an interrupt[9]: 390: 0 1 0 0 VLV-GPIO-gpio byt-codec-int Which is pin 4 on GPO2, but the codec driver in the mainline kernel does not seem to be using this; however AFAICS in sound/soc/codecs/rt5640.c a gpio pin in only used for suspend/resume, so it is not strictly needed if it's already in the correct state, right? Last question, what is the difference between having the device detected by sound/soc/intel/sst-acpi.c (like in my case) opposed to sound/soc/intel/sst/sst_acpi.c? I can see the 80860F28 id in both the files but they seem to load different firmwares. Any idea of what is missing? It's the DSDT which still needs adjusting, or are some changes needed (also) in the kernel driver? Thanks, Antonio [1] http://ao2.it/110 [2] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_original.dsl [3] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_ori… [4] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [5] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_without_ADMA0… [6] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [7] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_asound.state [8] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_Android_Teclast_… [9] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_proc_interrupt… -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

6 23

[alsa-devel] Applied "ASoC: fsl-asoc-card: add AC'97 support" to the asoc tree
by Mark Brown 17 Sep '15

17 Sep '15

The patch ASoC: fsl-asoc-card: add AC'97 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 e06b508481e1916b25038289b945104009d774c9 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" <mail(a)maciej.szmigiero.name> Date: Mon, 31 Aug 2015 17:11:35 +0200 Subject: [PATCH] ASoC: fsl-asoc-card: add AC'97 support Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 145 ++++++++++++++++----- 2 files changed, 120 insertions(+), 35 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed..86aa498 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -404,10 +451,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct device_node *cpu_np, *codec_np, *asrc_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; - struct platform_device *cpu_pdev; + struct platform_device *cpu_pdev, *codec_pdev; struct fsl_asoc_card_priv *priv; - struct i2c_client *codec_dev; - struct clk *codec_clk; + struct i2c_client *codec_i2c_dev; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,11 +479,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) { + codec_pdev = of_find_device_by_node(codec_np); + codec_i2c_dev = of_find_i2c_device_by_node(codec_np); + } else { + codec_pdev = NULL; + codec_i2c_dev = NULL; } asrc_np = of_parse_phandle(np, "audio-asrc", 0); @@ -446,10 +493,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_pdev) { + struct clk *codec_clk = clk_get(&codec_pdev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,11 +536,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_pdev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -507,7 +567,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_i2c_dev ? codec_i2c_dev->name : codec_pdev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -531,8 +593,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -543,6 +623,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -578,14 +660,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, -- 2.5.0

3 4

[alsa-devel] [PATCH v2] audio: sai: Add Power Management support
by Alison Wang 17 Sep '15

17 Sep '15

This patch adds Power Management support for SAI. Activate regmap cache with REGCACHE_FLAT, and use regmap cache code to save and restore registers in suspend and resume. The Transmit Control Register (TCSR) and Receive Control Register(RCSR) should be volatile registers. Signed-off-by: Alison Wang <alison.wang(a)freescale.com> --- Changes since v1: - Use REGCACHE_FLAT instead of REGCACHE_RBTREE. - Use SIMPLE_DEV_PM_OPS to simplify the code. sound/soc/fsl/fsl_sai.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..030f71e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,9 +637,11 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: case FSL_SAI_TFR: - case FSL_SAI_RFR: case FSL_SAI_TDR: + case FSL_SAI_RCSR: + case FSL_SAI_RFR: case FSL_SAI_RDR: return true; default: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -796,6 +799,31 @@ static int fsl_sai_probe(struct platform_device *pdev) return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } +#ifdef CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + /* Restore all registers */ + regcache_cache_only(sai->regmap, false); + regcache_sync(sai->regmap); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static SIMPLE_DEV_PM_OPS(fsl_sai_pm, fsl_sai_suspend, fsl_sai_resume); + static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, @@ -807,6 +835,7 @@ static struct platform_driver fsl_sai_driver = { .driver = { .name = "fsl-sai", .of_match_table = fsl_sai_ids, + .pm = &fsl_sai_pm, }, }; module_platform_driver(fsl_sai_driver); -- 2.1.0.27.g96db324

4 4

[alsa-devel] Multiple codecs on one sound card for multi-channel sound card
by Caleb Crome 17 Sep '15

17 Sep '15

Hi all, What's the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus? I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved... I currently have something like this in my am335x-boneblack.dts file: (i.e. attempting to set tdm slot width and num in the i2c codec) but it doesn't seem to be pickin up the fact that I want all the codecs linked together as one. (FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves). Thanks for any help! BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort! Sincerely, -Caleb &i2c1 { clock-frequency = <100000>; status = "okay"; pinctrl-names = "default"; pinctrl-0 = <&i2c1_pins_default>; status="okay"; tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x18>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <0>; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x19>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <2>; }; tlv320aic3x_c: tlv320aic3x@1a { compatible = "ti,tlv320aic3x"; reg = <0x1a>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <4>; }; tlv320aic3x_d: tlv320aic3x@1b { compatible = "ti,tlv320aic3x"; reg = <0x1b>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <6>; }; }; sound { compatible = "simple-audio-card"; simple-audio-card,name = "puck audio"; // simple-audio-card,widgets= // simple-audio-card,routing= simple-audio-card,mclk-fs = <256>; status="okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0 0>; }; // *** Here's where I'm really confused -- this part doesn't seem to be supported to // have all the codecs on a single DAI. // The examples given seem to be for different codec DAIs // sharing a single CPU DAI, but not used at the same time. codec { #sound-dai-cells = <0>; sound-dai = < &tlv320aic3x_a &tlv320aic3x_b &tlv320aic3x_c &tlv320aic3x_d >; }; }; };

2 10

Re: [alsa-devel] [linux-sunxi] Re: [PATCH 2/2] ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs
by Mark Brown 17 Sep '15

17 Sep '15

On Thu, Sep 17, 2015 at 11:10:41PM +0800, Chen-Yu Tsai wrote: > On Thu, Sep 17, 2015 at 9:31 PM, Maxime Ripard > >> > + .formats = SNDRV_PCM_FMTBIT_S16_LE | > >> > + SNDRV_PCM_FMTBIT_S32_LE, > >> > + .sig_bits = 24, > >> So presumably also S24_LE (ie, 24 bits packed into a 32 bit word)? > > Hmm, probably yes, I'll test that. > IIRC when Emilio first wrote the driver, we tried 24 bit and no sound > came out. Turns out it's an alignment issue. The codec's FIFO register > is 32 bits wide, and takes the higher 24 bits as input when set to 24 > bit mode. The internal FIFO is only 24 bits wide. A20 user manual P174 > describes how the bits are copied. > So for 24 bit audio, you would actually send it 32 bit audio samples, > and let it truncate or drop the least significant 8 bits. This is why > we have SNDRV_PCM_FMTBIT_S32_LE with .sig_bits = 24. > I don't know if this is just a workaround, but a few other drivers do > this as well, for example twl3040 and omap-mcpdm. That's very common - it's essentially what S24_LE mode is.

1 0

[alsa-devel] [PATCH v2] ASoC: wm5110: Add additional analogue input enable for early revs
by Charles Keepax 17 Sep '15

17 Sep '15

Earlier revisions of the wm5110/8280 silicon require a slightly more complex procedure to enable analogue inputs. This patch adds this into the driver. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Added a comment to explain why we need to hold the DAPM lock when setting the PGA volume for the inputs Thanks, Charles sound/soc/codecs/wm5110.c | 187 +++++++++++++++++++++++++++++++++++++++------ 1 files changed, 163 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 9756578..c04c0bc 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -38,6 +38,12 @@ struct wm5110_priv { struct arizona_priv core; struct arizona_fll fll[2]; + + unsigned int in_value; + int in_pre_pending; + int in_post_pending; + + unsigned int in_pga_cache[6]; }; static const struct wm_adsp_region wm5110_dsp1_regions[] = { @@ -428,6 +434,127 @@ err: return ret; } +static int wm5110_in_pga_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct snd_soc_card *card = dapm->card; + int ret; + + /* + * PGA Volume is also used as part of the enable sequence, so + * usage of it should be avoided whilst that is running. + */ + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_get_volsw_range(kcontrol, ucontrol); + + mutex_unlock(&card->dapm_mutex); + + return ret; +} + +static int wm5110_in_pga_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct snd_soc_card *card = dapm->card; + int ret; + + /* + * PGA Volume is also used as part of the enable sequence, so + * usage of it should be avoided whilst that is running. + */ + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_put_volsw_range(kcontrol, ucontrol); + + mutex_unlock(&card->dapm_mutex); + + return ret; +} + +static int wm5110_in_analog_ev(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 arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + unsigned int reg, mask; + struct reg_sequence analog_seq[] = { + { 0x80, 0x3 }, + { 0x35d, 0 }, + { 0x80, 0x0 }, + }; + + reg = ARIZONA_IN1L_CONTROL + ((w->shift ^ 0x1) * 4); + mask = ARIZONA_IN1L_PGA_VOL_MASK; + + switch (event) { + case SND_SOC_DAPM_WILL_PMU: + wm5110->in_value |= 0x3 << ((w->shift ^ 0x1) * 2); + wm5110->in_pre_pending++; + wm5110->in_post_pending++; + return 0; + case SND_SOC_DAPM_PRE_PMU: + wm5110->in_pga_cache[w->shift] = snd_soc_read(codec, reg); + + snd_soc_update_bits(codec, reg, mask, + 0x40 << ARIZONA_IN1L_PGA_VOL_SHIFT); + + wm5110->in_pre_pending--; + if (wm5110->in_pre_pending == 0) { + analog_seq[1].def = wm5110->in_value; + regmap_multi_reg_write_bypassed(arizona->regmap, + analog_seq, + ARRAY_SIZE(analog_seq)); + + msleep(55); + + wm5110->in_value = 0; + } + + break; + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, reg, mask, + wm5110->in_pga_cache[w->shift]); + + wm5110->in_post_pending--; + if (wm5110->in_post_pending == 0) + regmap_multi_reg_write_bypassed(arizona->regmap, + analog_seq, + ARRAY_SIZE(analog_seq)); + break; + default: + break; + } + + return 0; +} + +static int wm5110_in_ev(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 arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + + switch (arizona->rev) { + case 0 ... 4: + if (arizona_input_analog(codec, w->shift)) + wm5110_in_analog_ev(w, kcontrol, event); + + break; + default: + break; + } + + return arizona_in_ev(w, kcontrol, event); +} + static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); @@ -454,18 +581,24 @@ SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]), SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]), -SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, - ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, - ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, - ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum), @@ -896,29 +1029,35 @@ SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | -- 1.7.2.5

2 1

[alsa-devel] [PATCH v2] ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
by Jyri Sarha 17 Sep '15

17 Sep '15

Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1929f0e..af4999c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3294,7 +3294,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, unsigned int *mask) { u32 val; - const u32 *of_slot_mask = of_get_property(np, prop_name, &val); + const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); int i; if (!of_slot_mask) -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] change PTR_ERR argument
by Julia Lawall 17 Sep '15

17 Sep '15

Apply PTR_ERR to the value that was recently assigned. --- var/julia/linuxcopy/drivers/spi/spi-ep93xx.c | 3 ++- var/julia/linuxcopy/sound/soc/qcom/lpass-cpu.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-)

4 11

[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data
by Jyri Sarha 17 Sep '15

17 Sep '15

The slot_width is for essentially same thing. Instead of storing bclk_lrclk_ratio, just store the slot_width. Comments has been updated accordingly and some variable names changed to more descriptive. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Added the missing __func__ to dev_warn() warn call in __davinci_mcasp_set_clkdiv(). sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 25 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de1e3a8..9a9d633 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -87,7 +87,6 @@ struct davinci_mcasp { u8 *serial_dir; u8 version; u8 bclk_div; - u16 bclk_lrclk_ratio; int streams; u32 irq_request[2]; int dma_request[2]; @@ -558,8 +557,21 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, mcasp->bclk_div = div; break; - case 2: /* BCLK/LRCLK ratio */ - mcasp->bclk_lrclk_ratio = div; + case 2: /* + * BCLK/LRCLK ratio descries how many bit-clock cycles + * fit into one frame. The clock ratio is given for a + * full period of data (for I2S format both left and + * right channels), so it has to be divided by number + * of tdm-slots (for I2S - divided by 2). + * Instead of storing this ratio, we calculate a new + * tdm_slot width by dividing the the ratio by the + * number of configured tdm slots. + */ + mcasp->slot_width = div / mcasp->tdm_slots; + if (div % mcasp->tdm_slots) + dev_warn(mcasp->dev, + "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots", + __func__, div, mcasp->tdm_slots); break; default: @@ -682,11 +694,13 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, } static int davinci_config_channel_size(struct davinci_mcasp *mcasp, - int word_length) + int sample_width) { u32 fmt; - u32 tx_rotate = (word_length / 4) & 0x7; - u32 mask = (1ULL << word_length) - 1; + u32 tx_rotate = (sample_width / 4) & 0x7; + u32 mask = (1ULL << sample_width) - 1; + u32 slot_width = sample_width; + /* * For captured data we should not rotate, inversion and masking is * enoguh to get the data to the right position: @@ -699,31 +713,23 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, u32 rx_rotate = 0; /* - * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() - * callback, take it into account here. That allows us to for example - * send 32 bits per channel to the codec, while only 16 of them carry - * audio payload. - * The clock ratio is given for a full period of data (for I2S format - * both left and right channels), so it has to be divided by number of - * tdm-slots (for I2S - divided by 2). + * Setting the tdm slot width either with set_clkdiv() or + * set_tdm_slot() allows us to for example send 32 bits per + * channel to the codec, while only 16 of them carry audio + * payload. */ - if (mcasp->bclk_lrclk_ratio) { - u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots; - + if (mcasp->slot_width) { /* - * When we have more bclk then it is needed for the data, we - * need to use the rotation to move the received samples to have - * correct alignment. + * When we have more bclk then it is needed for the + * data, we need to use the rotation to move the + * received samples to have correct alignment. */ - rx_rotate = (slot_length - word_length) / 4; - word_length = slot_length; - } else if (mcasp->slot_width) { - rx_rotate = (mcasp->slot_width - word_length) / 4; - word_length = mcasp->slot_width; + slot_width = mcasp->slot_width; + rx_rotate = (slot_width - sample_width) / 4; } /* mapping of the XSSZ bit-field as described in the datasheet */ - fmt = (word_length >> 1) - 1; + fmt = (slot_width >> 1) - 1; if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt), -- 1.9.1

2 1

[alsa-devel] [asoc:topic/sunxi 2/2] sound/soc/sunxi/sun4i-codec.c:708:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 17 Sep '15

17 Sep '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/sunxi head: 45fb6b6f2aa3f6b22b81078db0dba4b26c9d0bdb commit: 45fb6b6f2aa3f6b22b81078db0dba4b26c9d0bdb [2/2] ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs coccinelle warnings: (new ones prefixed by >>) >> sound/soc/sunxi/sun4i-codec.c:708:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 3

[alsa-devel] [PATCH v2] ASoC: add ak4613 support
by Kuninori Morimoto 17 Sep '15

17 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - fix for checkpatch - adds ak4613 in SND_SOC_ALL_CODECS - ak4613_write() -> snd_soc_write() - ak4613_read() was removed - spin lock -> mutex - ak4613_resume() returns regcache_sync()'s result Documentation/devicetree/bindings/sound/ak4613.txt | 17 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4613.c | 469 +++++++++++++++++++++ 4 files changed, 493 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 sound/soc/codecs/ak4613.c diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt new file mode 100644 index 0000000..15a9195 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.txt @@ -0,0 +1,17 @@ +AK4613 I2C transmitter + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak4613" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4613: ak4613@0x10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..a92e4d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 + select SND_SOC_AK4613 if I2C select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C @@ -319,6 +320,10 @@ config SND_SOC_AK4535 config SND_SOC_AK4554 tristate "AKM AK4554 CODEC" +config SND_SOC_AK4613 + tristate "AKM AK4613 CODEC" + depends on I2C + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077..5b6c8af 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o +snd-soc-ak4613-objs := ak4613.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -216,6 +217,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o +obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c new file mode 100644 index 0000000..fd96a8f --- /dev/null +++ b/sound/soc/codecs/ak4613.c @@ -0,0 +1,469 @@ +/* + * ak4613.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright (C) 2015 Renesas Electronics Corporation + * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + * + * Based on ak4642.c by Kuninori Morimoto + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#define PW_MGMT1 0x00 /* Power Management 1 */ +#define PW_MGMT2 0x01 /* Power Management 2 */ +#define PW_MGMT3 0x02 /* Power Management 3 */ +#define CTRL1 0x03 /* Control 1 */ +#define CTRL2 0x04 /* Control 2 */ +#define DEMP1 0x05 /* De-emphasis1 */ +#define DEMP2 0x06 /* De-emphasis2 */ +#define OFD 0x07 /* Overflow Detect */ +#define ZRD 0x08 /* Zero Detect */ +#define ICTRL 0x09 /* Input Control */ +#define OCTRL 0x0a /* Output Control */ +#define LOUT1 0x0b /* LOUT1 Volume Control */ +#define ROUT1 0x0c /* ROUT1 Volume Control */ +#define LOUT2 0x0d /* LOUT2 Volume Control */ +#define ROUT2 0x0e /* ROUT2 Volume Control */ +#define LOUT3 0x0f /* LOUT3 Volume Control */ +#define ROUT3 0x10 /* ROUT3 Volume Control */ +#define LOUT4 0x11 /* LOUT4 Volume Control */ +#define ROUT4 0x12 /* ROUT4 Volume Control */ +#define LOUT5 0x13 /* LOUT5 Volume Control */ +#define ROUT5 0x14 /* ROUT5 Volume Control */ +#define LOUT6 0x15 /* LOUT6 Volume Control */ +#define ROUT6 0x16 /* ROUT6 Volume Control */ + +/* PW_MGMT1 */ +#define RSTN BIT(0) +#define PMDAC BIT(1) +#define PMADC BIT(2) +#define PMVR BIT(3) + +/* PW_MGMT2 */ +#define PMAD_ALL 0x7 + +/* PW_MGMT3 */ +#define PMDA_ALL 0x3f + +/* CTRL1 */ +#define DIF0 BIT(3) +#define DIF1 BIT(4) +#define DIF2 BIT(5) +#define TDM0 BIT(6) +#define TDM1 BIT(7) +#define NO_FMT (0xff) +#define FMT_MASK (0xf8) + +/* CTRL2 */ +#define DFS_NORMAL_SPEED (0 << 2) +#define DFS_DOUBLE_SPEED (1 << 2) +#define DFS_QUAD_SPEED (2 << 2) + +struct ak4613_priv { + struct mutex lock; + + unsigned int fmt; + u8 fmt_ctrl; + int cnt; +}; + +struct ak4613_formats { + unsigned int width; + unsigned int fmt; +}; + +struct ak4613_interface { + struct ak4613_formats capture; + struct ak4613_formats playback; +}; + +static const struct reg_default ak4613_reg[] = { + { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, + { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, + { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 }, + { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 }, + { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 }, + { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, +}; + +#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3) +#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } +static const struct ak4613_interface ak4613_iface[] = { + /* capture */ /* playback */ + [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, + [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, + [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, + [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, +}; + +static const struct regmap_config ak4613_regmap_cfg = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x16, + .reg_defaults = ak4613_reg, + .num_reg_defaults = ARRAY_SIZE(ak4613_reg), +}; + +static const struct of_device_id ak4613_of_match[] = { + { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4613_of_match); + +static const struct i2c_device_id ak4613_i2c_id[] = { + { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id); + +static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = { + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("LOUT4"), + SND_SOC_DAPM_OUTPUT("LOUT5"), + SND_SOC_DAPM_OUTPUT("LOUT6"), + + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + SND_SOC_DAPM_OUTPUT("ROUT4"), + SND_SOC_DAPM_OUTPUT("ROUT5"), + SND_SOC_DAPM_OUTPUT("ROUT6"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0), + SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0), + SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0), + SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0), + SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0), + SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0), + SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0), +}; + +static const struct snd_soc_dapm_route ak4613_intercon[] = { + {"LOUT1", NULL, "DAC1"}, + {"LOUT2", NULL, "DAC2"}, + {"LOUT3", NULL, "DAC3"}, + {"LOUT4", NULL, "DAC4"}, + {"LOUT5", NULL, "DAC5"}, + {"LOUT6", NULL, "DAC6"}, + + {"ROUT1", NULL, "DAC1"}, + {"ROUT2", NULL, "DAC2"}, + {"ROUT3", NULL, "DAC3"}, + {"ROUT4", NULL, "DAC4"}, + {"ROUT5", NULL, "DAC5"}, + {"ROUT6", NULL, "DAC6"}, + + {"DAC1", NULL, "Playback"}, + {"DAC2", NULL, "Playback"}, + {"DAC3", NULL, "Playback"}, + {"DAC4", NULL, "Playback"}, + {"DAC5", NULL, "Playback"}, + {"DAC6", NULL, "Playback"}, + + {"Capture", NULL, "ADC1"}, + {"Capture", NULL, "ADC2"}, + + {"ADC1", NULL, "LIN1"}, + {"ADC2", NULL, "LIN2"}, + + {"ADC1", NULL, "RIN1"}, + {"ADC2", NULL, "RIN2"}, +}; + +static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + + mutex_lock(&priv->lock); + priv->cnt--; + if (priv->cnt < 0) { + dev_err(dev, "unexpected counter error\n"); + priv->cnt = 0; + } + if (!priv->cnt) + priv->fmt_ctrl = NO_FMT; + mutex_unlock(&priv->lock); +} + +static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + fmt &= SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: + priv->fmt = fmt; + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ak4613_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 ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + const struct ak4613_formats *fmts; + struct device *dev = codec->dev; + unsigned int width = params_width(params); + unsigned int fmt = priv->fmt; + unsigned int rate; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i, ret; + u8 fmt_ctrl, ctrl2; + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + ctrl2 = DFS_NORMAL_SPEED; + break; + case 88200: + case 96000: + ctrl2 = DFS_DOUBLE_SPEED; + break; + case 176400: + case 192000: + ctrl2 = DFS_QUAD_SPEED; + break; + default: + return -EINVAL; + } + + /* + * FIXME + * + * It doesn't support TDM at this point + */ + fmt_ctrl = NO_FMT; + for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) { + fmts = (is_play) ? &ak4613_iface[i].playback : + &ak4613_iface[i].capture; + + if (fmts->fmt != fmt) + continue; + + if (fmt == SND_SOC_DAIFMT_RIGHT_J) { + if (fmts->width != width) + continue; + } else { + if (fmts->width < width) + continue; + } + + fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i); + break; + } + + ret = -EINVAL; + if (fmt_ctrl == NO_FMT) + goto hw_params_end; + + mutex_lock(&priv->lock); + if ((priv->fmt_ctrl == NO_FMT) || + (priv->fmt_ctrl == fmt_ctrl)) { + priv->fmt_ctrl = fmt_ctrl; + priv->cnt++; + ret = 0; + } + mutex_unlock(&priv->lock); + + if (ret < 0) + goto hw_params_end; + + snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); + snd_soc_write(codec, CTRL2, ctrl2); + +hw_params_end: + if (ret < 0) + dev_warn(dev, "unsupported data width/format combination\n"); + + return ret; +} + +static int ak4613_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 mgmt1 = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + mgmt1 |= RSTN; + /* fall through */ + case SND_SOC_BIAS_PREPARE: + mgmt1 |= PMADC | PMDAC; + /* fall through */ + case SND_SOC_BIAS_STANDBY: + mgmt1 |= PMVR; + /* fall through */ + case SND_SOC_BIAS_OFF: + default: + break; + } + + snd_soc_write(codec, PW_MGMT1, mgmt1); + + return 0; +} + +static const struct snd_soc_dai_ops ak4613_dai_ops = { + .shutdown = ak4613_dai_shutdown, + .set_fmt = ak4613_dai_set_fmt, + .hw_params = ak4613_dai_hw_params, +}; + +#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_64000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver ak4613_dai = { + .name = "ak4613-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .ops = &ak4613_dai_ops, + .symmetric_rates = 1, +}; + +static int ak4613_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + return regcache_sync(regmap); +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { + .resume = ak4613_resume, + .set_bias_level = ak4613_set_bias_level, + .dapm_widgets = ak4613_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), + .dapm_routes = ak4613_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), +}; + +static int ak4613_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + const struct regmap_config *regmap_cfg; + struct regmap *regmap; + struct ak4613_priv *priv; + + regmap_cfg = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4613_of_match, dev); + if (of_id) + regmap_cfg = of_id->data; + } else { + regmap_cfg = (const struct regmap_config *)id->driver_data; + } + + if (!regmap_cfg) + return -EINVAL; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->fmt_ctrl = NO_FMT; + priv->cnt = 0; + + mutex_init(&priv->lock); + + i2c_set_clientdata(i2c, priv); + + regmap = devm_regmap_init_i2c(i2c, regmap_cfg); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(dev, &soc_codec_dev_ak4613, + &ak4613_dai, 1); +} + +static int ak4613_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver ak4613_i2c_driver = { + .driver = { + .name = "ak4613-codec", + .owner = THIS_MODULE, + .of_match_table = ak4613_of_match, + }, + .probe = ak4613_i2c_probe, + .remove = ak4613_i2c_remove, + .id_table = ak4613_i2c_id, +}; + +module_i2c_driver(ak4613_i2c_driver); + +MODULE_DESCRIPTION("Soc AK4613 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); +MODULE_LICENSE("GPL v2"); -- 1.9.1

2 1

[alsa-devel] [PATCH v3] ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
by Jyri Sarha 17 Sep '15

17 Sep '15

Store return value of of_get_property() to a pointer of __be32 type as device tree has big endian type. This fixes a sparse warning couple of lines later when be32_to_cpup() is used to convert from big endian to cpu endian. The whole conversion is not really necessary, as we are only checking if the value is zero or not, but I wanted to add it to remind in the future that the data has to be converted before use. Compiler should optimize the unnecessary operations away. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Damn, I wrote description ready but forgot to produce the patch again. So here is a v3 of the same one liner patch. sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1929f0e..af4999c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3294,7 +3294,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, unsigned int *mask) { u32 val; - const u32 *of_slot_mask = of_get_property(np, prop_name, &val); + const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); int i; if (!of_slot_mask) -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
by Jyri Sarha 17 Sep '15

17 Sep '15

Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1929f0e..614b831 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3301,7 +3301,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, return 0; val /= sizeof(u32); for (i = 0; i < val; i++) - if (be32_to_cpup(&of_slot_mask[i])) + if (be32_to_cpup((__be32 *)&of_slot_mask[i])) *mask |= (1 << i); return val; -- 1.9.1

2 2

[alsa-devel] [PATCH 1/4] ASoC: arizona: Add default cases for event switches
by Charles Keepax 17 Sep '15

17 Sep '15

Since the addition of the WILL_PMU / WILL_PMD several of the switches in arizona.c no longer cover all cases or have a default case. Whilst this isn't causing any problems in the interests of robustness add default cases. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8a2221a..1bc19f1 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -147,6 +147,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, 0x4f5, 0x0da); } break; + default: + break; } return 0; @@ -725,6 +727,9 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES); if (reg == 0) arizona_in_set_vu(codec, 0); + break; + default: + break; } return 0; @@ -806,6 +811,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, break; } break; + default: + break; } return 0; -- 1.7.2.5

2 8

[alsa-devel] Test device for automated testing
by Uli Franke 17 Sep '15

17 Sep '15

Hi everybody, we're currently writing a lot of ALSA user space applications. In order to test this code with our unit testing and other test environments we'd like to know if there's a dummy PCM device which provides the fundamental operations like reading/writing and setting the common hardware parameters. It doesn't even have to supply some useful functionality like loop back or similar. Neither the docs nor the goo led us to something useful. Thanks and regards Uli

2 4

[alsa-devel] [PATCH 0/2] ASoC: atmel-classd: add the Audio Class D Amplifier
by Songjun Wu 17 Sep '15

17 Sep '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier code ASoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 73 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 801 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 +++ 5 files changed, 1005 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

3 21

[alsa-devel] [asoc:topic/simple 1/1] sound/soc/soc-core.c:3306:47: sparse: incorrect type in argument 1 (different base types)
by kbuild test robot 17 Sep '15

17 Sep '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/simple head: 6131084a0bc966107021d8c89489f9cd1663b902 commit: 6131084a0bc966107021d8c89489f9cd1663b902 [1/1] ASoC: simple-card: Add tdm slot mask support to simple-card reproduce: # apt-get install sparse git checkout 6131084a0bc966107021d8c89489f9cd1663b902 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/soc-core.c:3306:47: sparse: incorrect type in argument 1 (different base types) sound/soc/soc-core.c:3306:47: expected restricted __be32 const [usertype] *p sound/soc/soc-core.c:3306:47: got unsigned int const [usertype] * vim +3306 sound/soc/soc-core.c 3290 return 0; 3291 } 3292 EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); 3293 3294 static int snd_soc_of_get_slot_mask(struct device_node *np, 3295 const char *prop_name, 3296 unsigned int *mask) 3297 { 3298 u32 val; 3299 const u32 *of_slot_mask = of_get_property(np, prop_name, &val); 3300 int i; 3301 3302 if (!of_slot_mask) 3303 return 0; 3304 val /= sizeof(u32); 3305 for (i = 0; i < val; i++) > 3306 if (be32_to_cpup(&of_slot_mask[i])) 3307 *mask |= (1 << i); 3308 3309 return val; 3310 } 3311 3312 int snd_soc_of_parse_tdm_slot(struct device_node *np, 3313 unsigned int *tx_mask, 3314 unsigned int *rx_mask, --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM
by Axel Lin 16 Sep '15

16 Sep '15

devm_snd_dmaengine_pcm_register() is guarded by CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/spear/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index 0a53053..4fb9141 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -1,6 +1,6 @@ config SND_SPEAR_SOC tristate - select SND_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SPEAR_SPDIF_OUT tristate -- 2.1.0

2 1

[alsa-devel] [PATCH RFC 0/5] ASoC: TDM stuff for simple-card, mcasp, and tlv320aic3x
by Jyri Sarha 16 Sep '15

16 Sep '15

The first patch is an independent fix. The symmetric_samplebits constraint could probably be removed too (with some changes to mcasp code too), but as tlv320aic3x family codecs won't support it, I could not test it. Jyri Sarha (5): ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver ASoC: simple-card: Add tdm slot mask support to simple-card ASoC: davinci-mcasp: Add set_tdm_slots() support ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data ASoC: tlv320aic3x: Improve tdm support .../devicetree/bindings/sound/tdm-slot.txt | 11 +- include/sound/simple_card.h | 2 + include/sound/soc.h | 2 + sound/soc/codecs/tlv320aic3x.c | 30 +- sound/soc/davinci/davinci-mcasp.c | 306 ++++++++++++++------- sound/soc/generic/simple-card.c | 8 +- sound/soc/soc-core.c | 25 ++ 7 files changed, 274 insertions(+), 110 deletions(-) -- 1.9.1

3 11

[alsa-devel] [PATCH 1/2] ASoC: dapm: fix memory leak
by Sudip Mukherjee 16 Sep '15

16 Sep '15

Incase of an unknown event we were directly returning but we missed freeing params. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f4bf21a..ff8bda4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3501,7 +3501,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, default: WARN(1, "Unknown event %d\n", event); - return -EINVAL; + ret = -EINVAL; } out: -- 1.9.1

2 2

[alsa-devel] Applied "ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs" to the asoc tree
by Mark Brown 16 Sep '15

16 Sep '15

The patch ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs 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 45fb6b6f2aa3f6b22b81078db0dba4b26c9d0bdb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Emilio=20L=C3=B3pez?= <emilio(a)elopez.com.ar> Date: Sat, 12 Sep 2015 15:26:24 +0200 Subject: [PATCH] ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The sun4i, sun5i and sun7i SoC families have a built-in codec, capable of both audio capture and playback. While this is called a codec by Allwinner, it really is an in-SoC combination of a codec and a DAI, with its own DAC/ADC and amplifiers in a single memory-mapped controller. The capture part has been left out for now, and will be added eventually. Signed-off-by: Emilio López <emilio(a)elopez.com.ar> Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-codec.c | 720 ++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 735 insertions(+) create mode 100644 sound/soc/sunxi/Kconfig create mode 100644 sound/soc/sunxi/Makefile create mode 100644 sound/soc/sunxi/sun4i-codec.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 225bfda..7de792b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -58,6 +58,7 @@ source "sound/soc/sh/Kconfig" source "sound/soc/sirf/Kconfig" source "sound/soc/spear/Kconfig" source "sound/soc/sti/Kconfig" +source "sound/soc/sunxi/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 134aca1..af0a571 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -40,6 +40,7 @@ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += sirf/ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += sti/ +obj-$(CONFIG_SND_SOC) += sunxi/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig new file mode 100644 index 0000000..84c72ec --- /dev/null +++ b/sound/soc/sunxi/Kconfig @@ -0,0 +1,11 @@ +menu "Allwinner SoC Audio support" + +config SND_SUN4I_CODEC + tristate "Allwinner A10 Codec Support" + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Select Y or M to add support for the Codec embedded in the Allwinner + A10 and affiliated SoCs. + +endmenu diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile new file mode 100644 index 0000000..ea8a08c --- /dev/null +++ b/sound/soc/sunxi/Makefile @@ -0,0 +1,2 @@ +obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o + diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c new file mode 100644 index 0000000..6e83e62 --- /dev/null +++ b/sound/soc/sunxi/sun4i-codec.c @@ -0,0 +1,720 @@ +/* + * Copyright 2014 Emilio López <emilio(a)elopez.com.ar> + * Copyright 2014 Jon Smirl <jonsmirl(a)gmail.com> + * Copyright 2015 Maxime Ripard <maxime.ripard(a)free-electrons.com> + * + * Based on the Allwinner SDK driver, released under the GPL. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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/init.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_address.h> +#include <linux/clk.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/initval.h> +#include <sound/dmaengine_pcm.h> + +/* Codec DAC register offsets and bit fields */ +#define SUN4I_CODEC_DAC_DPC (0x00) +#define SUN4I_CODEC_DAC_DPC_EN_DA (31) +#define SUN4I_CODEC_DAC_DPC_DVOL (12) +#define SUN4I_CODEC_DAC_FIFOC (0x04) +#define SUN4I_CODEC_DAC_FIFOC_DAC_FS (29) +#define SUN4I_CODEC_DAC_FIFOC_FIR_VERSION (28) +#define SUN4I_CODEC_DAC_FIFOC_SEND_LASAT (26) +#define SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE (24) +#define SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT (21) +#define SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL (8) +#define SUN4I_CODEC_DAC_FIFOC_MONO_EN (6) +#define SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS (5) +#define SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN (4) +#define SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH (0) +#define SUN4I_CODEC_DAC_FIFOS (0x08) +#define SUN4I_CODEC_DAC_TXDATA (0x0c) +#define SUN4I_CODEC_DAC_ACTL (0x10) +#define SUN4I_CODEC_DAC_ACTL_DACAENR (31) +#define SUN4I_CODEC_DAC_ACTL_DACAENL (30) +#define SUN4I_CODEC_DAC_ACTL_MIXEN (29) +#define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) +#define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) +#define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_DACPAS (8) +#define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) +#define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) +#define SUN4I_CODEC_DAC_ACTL_PA_VOL (0) +#define SUN4I_CODEC_DAC_TUNE (0x14) +#define SUN4I_CODEC_DAC_DEBUG (0x18) + +/* Codec ADC register offsets and bit fields */ +#define SUN4I_CODEC_ADC_FIFOC (0x1c) +#define SUN4I_CODEC_ADC_FIFOC_EN_AD (28) +#define SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE (24) +#define SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL (8) +#define SUN4I_CODEC_ADC_FIFOC_MONO_EN (7) +#define SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS (6) +#define SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN (4) +#define SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH (0) +#define SUN4I_CODEC_ADC_FIFOS (0x20) +#define SUN4I_CODEC_ADC_RXDATA (0x24) +#define SUN4I_CODEC_ADC_ACTL (0x28) +#define SUN4I_CODEC_ADC_ACTL_ADC_R_EN (31) +#define SUN4I_CODEC_ADC_ACTL_ADC_L_EN (30) +#define SUN4I_CODEC_ADC_ACTL_PREG1EN (29) +#define SUN4I_CODEC_ADC_ACTL_PREG2EN (28) +#define SUN4I_CODEC_ADC_ACTL_VMICEN (27) +#define SUN4I_CODEC_ADC_ACTL_VADCG (20) +#define SUN4I_CODEC_ADC_ACTL_ADCIS (17) +#define SUN4I_CODEC_ADC_ACTL_PA_EN (4) +#define SUN4I_CODEC_ADC_ACTL_DDE (3) +#define SUN4I_CODEC_ADC_DEBUG (0x2c) + +/* Other various ADC registers */ +#define SUN4I_CODEC_DAC_TXCNT (0x30) +#define SUN4I_CODEC_ADC_RXCNT (0x34) +#define SUN4I_CODEC_AC_SYS_VERI (0x38) +#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) + +struct sun4i_codec { + struct device *dev; + struct regmap *regmap; + struct clk *clk_apb; + struct clk *clk_module; + + struct snd_dmaengine_dai_dma_data playback_dma_data; +}; + +static void sun4i_codec_start_playback(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO high here on some boards + */ + + /* Flush TX FIFO */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH)); + + /* Enable DAC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN)); +} + +static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO low here on some boards + */ + + /* Disable DAC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), + 0); +} + +static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENOTSUPP; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sun4i_codec_start_playback(scodec); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sun4i_codec_stop_playback(scodec); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int sun4i_codec_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + u32 val; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENOTSUPP; + + /* Flush the TX FIFO */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH)); + + /* Set TX FIFO Empty Trigger Level */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + 0x3f << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL, + 0xf << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL); + + if (substream->runtime->rate > 32000) + /* Use 64 bits FIR filter */ + val = 0; + else + /* Use 32 bits FIR filter */ + val = BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION); + + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION), + val); + + /* Send zeros when we have an underrun */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_SEND_LASAT), + 0); + + return 0; +} + +static unsigned long sun4i_codec_get_mod_freq(struct snd_pcm_hw_params *params) +{ + unsigned int rate = params_rate(params); + + switch (rate) { + case 176400: + case 88200: + case 44100: + case 33075: + case 22050: + case 14700: + case 11025: + case 7350: + return 22579200; + + case 192000: + case 96000: + case 48000: + case 32000: + case 24000: + case 16000: + case 12000: + case 8000: + return 24576000; + + default: + return 0; + } +} + +static int sun4i_codec_get_hw_rate(struct snd_pcm_hw_params *params) +{ + unsigned int rate = params_rate(params); + + switch (rate) { + case 192000: + case 176400: + return 6; + + case 96000: + case 88200: + return 7; + + case 48000: + case 44100: + return 0; + + case 32000: + case 33075: + return 1; + + case 24000: + case 22050: + return 2; + + case 16000: + case 14700: + return 3; + + case 12000: + case 11025: + return 4; + + case 8000: + case 7350: + return 5; + + default: + return -EINVAL; + } +} + +static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + unsigned long clk_freq; + int hwrate; + u32 val; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENOTSUPP; + + clk_freq = sun4i_codec_get_mod_freq(params); + if (!clk_freq) + return -EINVAL; + + if (clk_set_rate(scodec->clk_module, clk_freq)) + return -EINVAL; + + hwrate = sun4i_codec_get_hw_rate(params); + if (hwrate < 0) + return hwrate; + + /* Set DAC sample rate */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + 7 << SUN4I_CODEC_DAC_FIFOC_DAC_FS, + hwrate << SUN4I_CODEC_DAC_FIFOC_DAC_FS); + + /* Set the number of channels we want to use */ + if (params_channels(params) == 1) + val = BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN); + else + val = 0; + + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN), + val); + + /* Set the number of sample bits to either 16 or 24 bits */ + if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) { + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS), + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS)); + + /* Set TX FIFO mode to padding the LSBs with 0 */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE), + 0); + + scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + } else { + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS), + 0); + + /* Set TX FIFO mode to repeat the MSB */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE), + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE)); + + scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + } + + return 0; +} + +static int sun4i_codec_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + + /* + * Stop issuing DRQ when we have room for less than 16 samples + * in our TX FIFO + */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT, + 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT); + + return clk_prepare_enable(scodec->clk_module); +} + +static void sun4i_codec_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + + clk_disable_unprepare(scodec->clk_module); +} + +static const struct snd_soc_dai_ops sun4i_codec_dai_ops = { + .startup = sun4i_codec_startup, + .shutdown = sun4i_codec_shutdown, + .trigger = sun4i_codec_trigger, + .hw_params = sun4i_codec_hw_params, + .prepare = sun4i_codec_prepare, +}; + +static struct snd_soc_dai_driver sun4i_codec_dai = { + .name = "Codec", + .ops = &sun4i_codec_dai_ops, + .playback = { + .stream_name = "Codec Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 192000, + .rates = SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000 | + SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, + }, +}; + +/*** Codec ***/ +static const struct snd_kcontrol_new sun4i_codec_pa_mute = + SOC_DAPM_SINGLE("Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); + +static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); + +static const struct snd_kcontrol_new sun4i_codec_widgets[] = { + SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, + sun4i_codec_pa_volume_scale), +}; + +static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), +}; + +static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { + SOC_DAPM_SINGLE("Right DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), + SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), +}; + +static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_DACPAS, 1, 0), + SOC_DAPM_SINGLE("Mixer Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIXPAS, 1, 0), +}; + +static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { + /* Digital parts of the DACs */ + SND_SOC_DAPM_SUPPLY("DAC", SUN4I_CODEC_DAC_DPC, + SUN4I_CODEC_DAC_DPC_EN_DA, 0, + NULL, 0), + + /* Analog parts of the DACs */ + SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_DACAENL, 0), + SND_SOC_DAPM_DAC("Right DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_DACAENR, 0), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + sun4i_codec_left_mixer_controls, + ARRAY_SIZE(sun4i_codec_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + sun4i_codec_right_mixer_controls, + ARRAY_SIZE(sun4i_codec_right_mixer_controls)), + + /* Global Mixer Enable */ + SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0), + + /* Pre-Amplifier */ + SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PA_EN, 0, + sun4i_codec_pa_mixer_controls, + ARRAY_SIZE(sun4i_codec_pa_mixer_controls)), + SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0, + &sun4i_codec_pa_mute), + + SND_SOC_DAPM_OUTPUT("HP Right"), + SND_SOC_DAPM_OUTPUT("HP Left"), +}; + +static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { + /* Left DAC Routes */ + { "Left DAC", NULL, "DAC" }, + + /* Right DAC Routes */ + { "Right DAC", NULL, "DAC" }, + + /* Right Mixer Routes */ + { "Right Mixer", NULL, "Mixer Enable" }, + { "Right Mixer", "Left DAC Playback Switch", "Left DAC" }, + { "Right Mixer", "Right DAC Playback Switch", "Right DAC" }, + + /* Left Mixer Routes */ + { "Left Mixer", NULL, "Mixer Enable" }, + { "Left Mixer", "Left DAC Playback Switch", "Left DAC" }, + + /* Pre-Amplifier Mixer Routes */ + { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" }, + { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" }, + { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" }, + { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" }, + + /* PA -> HP path */ + { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, + { "HP Right", NULL, "Pre-Amplifier Mute" }, + { "HP Left", NULL, "Pre-Amplifier Mute" }, +}; + +static struct snd_soc_codec_driver sun4i_codec_codec = { + .controls = sun4i_codec_widgets, + .num_controls = ARRAY_SIZE(sun4i_codec_widgets), + .dapm_widgets = sun4i_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets), + .dapm_routes = sun4i_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes), +}; + +static const struct snd_soc_component_driver sun4i_codec_component = { + .name = "sun4i-codec", +}; + +#define SUN4I_CODEC_RATES SNDRV_PCM_RATE_8000_192000 +#define SUN4I_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static int sun4i_codec_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai); + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card); + + snd_soc_dai_init_dma_data(dai, &scodec->playback_dma_data, + NULL); + + return 0; +} + +static struct snd_soc_dai_driver dummy_cpu_dai = { + .name = "sun4i-codec-cpu-dai", + .probe = sun4i_codec_dai_probe, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SUN4I_CODEC_RATES, + .formats = SUN4I_CODEC_FORMATS, + .sig_bits = 24, + }, +}; + +static const struct regmap_config sun4i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, +}; + +static const struct of_device_id sun4i_codec_of_match[] = { + { .compatible = "allwinner,sun4i-a10-codec" }, + { .compatible = "allwinner,sun7i-a20-codec" }, + {} +}; +MODULE_DEVICE_TABLE(of, sun4i_codec_of_match); + +static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev, + int *num_links) +{ + struct snd_soc_dai_link *link = devm_kzalloc(dev, sizeof(*link), + GFP_KERNEL); + if (!link) + return NULL; + + link->name = "cdc"; + link->stream_name = "CDC PCM"; + link->codec_dai_name = "Codec"; + link->cpu_dai_name = dev_name(dev); + link->codec_name = dev_name(dev); + link->platform_name = dev_name(dev); + link->dai_fmt = SND_SOC_DAIFMT_I2S; + + *num_links = 1; + + return link; +}; + +static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) +{ + struct snd_soc_card *card; + int ret; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return NULL; + + card->dai_link = sun4i_codec_create_link(dev, &card->num_links); + if (!card->dai_link) + return NULL; + + card->dev = dev; + card->name = "sun4i-codec"; + + ret = snd_soc_of_parse_audio_routing(card, "routing"); + if (ret) { + dev_err(dev, "Failed to create our audio routing\n"); + return NULL; + } + + return card; +}; + +static int sun4i_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct sun4i_codec *scodec; + struct resource *res; + void __iomem *base; + int ret; + + scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL); + if (!scodec) + return -ENOMEM; + + scodec->dev = &pdev->dev; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) { + dev_err(&pdev->dev, "Failed to map the registers\n"); + return PTR_ERR(base); + } + + scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun4i_codec_regmap_config); + if (IS_ERR(scodec->regmap)) { + dev_err(&pdev->dev, "Failed to create our regmap\n"); + return PTR_ERR(scodec->regmap); + } + + /* Get the clocks from the DT */ + scodec->clk_apb = devm_clk_get(&pdev->dev, "apb"); + if (IS_ERR(scodec->clk_apb)) { + dev_err(&pdev->dev, "Failed to get the APB clock\n"); + return PTR_ERR(scodec->clk_apb); + } + + scodec->clk_module = devm_clk_get(&pdev->dev, "codec"); + if (IS_ERR(scodec->clk_module)) { + dev_err(&pdev->dev, "Failed to get the module clock\n"); + return PTR_ERR(scodec->clk_module); + } + + /* Enable the bus clock */ + if (clk_prepare_enable(scodec->clk_apb)) { + dev_err(&pdev->dev, "Failed to enable the APB clock\n"); + return -EINVAL; + } + + /* DMA configuration for TX FIFO */ + scodec->playback_dma_data.addr = res->start + SUN4I_CODEC_DAC_TXDATA; + scodec->playback_dma_data.maxburst = 4; + scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec, + &sun4i_codec_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Failed to register our codec\n"); + goto err_clk_disable; + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun4i_codec_component, + &dummy_cpu_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Failed to register our DAI\n"); + goto err_unregister_codec; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Failed to register against DMAEngine\n"); + goto err_unregister_codec; + } + + card = sun4i_codec_create_card(&pdev->dev); + if (!card) { + dev_err(&pdev->dev, "Failed to create our card\n"); + goto err_unregister_codec; + } + + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, scodec); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "Failed to register our card\n"); + goto err_unregister_codec; + } + + return 0; + +err_unregister_codec: + snd_soc_unregister_codec(&pdev->dev); +err_clk_disable: + clk_disable_unprepare(scodec->clk_apb); + return ret; +} + +static int sun4i_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + snd_soc_unregister_codec(&pdev->dev); + clk_disable_unprepare(scodec->clk_apb); + + return 0; +} + +static struct platform_driver sun4i_codec_driver = { + .driver = { + .name = "sun4i-codec", + .owner = THIS_MODULE, + .of_match_table = sun4i_codec_of_match, + }, + .probe = sun4i_codec_probe, + .remove = sun4i_codec_remove, +}; +module_platform_driver(sun4i_codec_driver); + +MODULE_DESCRIPTION("Allwinner A10 codec driver"); +MODULE_AUTHOR("Emilio López <emilio(a)elopez.com.ar>"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_AUTHOR("Maxime Ripard <maxime.ripard(a)free-electrons.com>"); +MODULE_LICENSE("GPL"); -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: SND_SOC_RCAR depends on COMMON_CLK
by Kuninori Morimoto 16 Sep '15

16 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") uses of_clk_add_provider() which is requesting struct clk_onecell_data. But it is COMMON_CLK feature. SND_SOC_RCAR depends on COMMON_CLK This patch also solved compile error of 7486d80f7("ASoC: rsnd: remove unneeded sh_clk header") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 07114b0..6ca90aa 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" depends on DMA_OF + depends on COMMON_CLK select SND_SIMPLE_CARD select REGMAP_MMIO help -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: ak4642: exchange macro name to avoid redefinition
by Kuninori Morimoto 16 Sep '15

16 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 71a0138ab("ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT") added new FS() macro, but x86 already has it in arch/x86/include/uapi/asm/ptrace-abi.h This patch exchange FS() to FSs() to avoid redefinition warning Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index b5c4981..cda27c2 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -130,8 +130,8 @@ #define I2S (3 << 0) /* MD_CTL2 */ -#define FS(val) (((val & 0x7) << 0) | ((val & 0x8) << 2)) -#define PS(val) ((val & 0x3) << 6) +#define FSs(val) (((val & 0x7) << 0) | ((val & 0x8) << 2)) +#define PSs(val) ((val & 0x3) << 6) /* MD_CTL3 */ #define BST1 (1 << 3) @@ -458,7 +458,8 @@ static int ak4642_set_mcko(struct snd_soc_codec *codec, for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) { for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) { if (frequency == ps_list[ps] * fs_list[fs]) { - snd_soc_write(codec, MD_CTL2, PS(ps) | FS(fs)); + snd_soc_write(codec, MD_CTL2, + PSs(ps) | FSs(fs)); return 0; } } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fixup clkout_name[] index error for single AUDIO_CLKOUT SoC
by Kuninori Morimoto 16 Sep '15

16 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") added AUDIO_CLKOUT support for ADG. But single AUDIO_CLKOUT needs clkout_name[CLKOUT], not clkout_name[i]. Kernel will have NULL pointer access without this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 3fecb87..c4ebbb7 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -532,7 +532,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, * for clkout */ if (!count) { - clk = clk_register_fixed_rate(dev, clkout_name[i], + clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], parent_clk_name, (parent_clk_name) ? 0 : CLK_IS_ROOT, req_rate); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fix broken pxa SoC support
by Robert Jarzmik 16 Sep '15

16 Sep '15

The previous fix of pxa library support, which was introduced to fix the library dependency, broke the previous SoC behavior, where a machine code binding pxa2xx-ac97 with a coded relied on : - sound/soc/pxa/pxa2xx-ac97.c - sound/soc/codecs/XXX.c For example, the mioa701_wm9713.c machine code is currently broken. The "select ARM" statement wrongly selects the soc/arm/pxa2xx-ac97 for compilation, as per an unfortunate fate SND_PXA2XX_AC97 is both declared in sound/arm/Kconfig and sound/soc/pxa/Kconfig. Fix this by ensuring that SND_PXA2XX_SOC correctly triggers the correct pxa2xx-ac97 compilation. Fixes: 846172dfe33c ("ASoC: fix SND_PXA2XX_LIB Kconfig warning") Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/arm/Kconfig | 15 ++++++++------- sound/soc/pxa/Kconfig | 2 -- 2 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 885683a3b0bd..e0406211716b 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -9,6 +9,14 @@ menuconfig SND_ARM Drivers that are implemented on ASoC can be found in "ALSA for SoC audio support" section. +config SND_PXA2XX_LIB + tristate + select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 + select SND_DMAENGINE_PCM + +config SND_PXA2XX_LIB_AC97 + bool + if SND_ARM config SND_ARMAACI @@ -21,13 +29,6 @@ config SND_PXA2XX_PCM tristate select SND_PCM -config SND_PXA2XX_LIB - tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 - -config SND_PXA2XX_LIB_AC97 - bool - config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" depends on ARCH_PXA diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 39cea80846c3..f2bf8661dd21 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,7 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA - select SND_ARM select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to @@ -25,7 +24,6 @@ config SND_PXA2XX_AC97 config SND_PXA2XX_SOC_AC97 tristate select AC97_BUS - select SND_ARM select SND_PXA2XX_LIB_AC97 select SND_SOC_AC97_BUS -- 2.1.4

2 1

[alsa-devel] CM6206 S/PDIF trouble
by Linus Walleij 15 Sep '15

15 Sep '15

Hi ALSA folks, feel like an amateur around here. Anyways, I got this USB sound card named "TRIXES External Sound Card USB 6 Channel 5.1 Audio" Looks like so: http://www.amazon.co.uk/External-Sound-Card-Channel-Audio/dp/B003TO3KHY/ref… Here is the alsa-info.sh output (yeah on a machine with hardwired HDA in it too, sorry): http://www.alsa-project.org/db/?f=f52ce9edc11918d70bc0f230f3cbd5f984bea5b5 The lsusb -v output is in the end of the mail, too. This is a 0d8c:0102 "C-Media Electronics, Inc. CM106 Like Sound Device" Actually, and according to the drivers on the CD that comes with it, this is a CM6206-based card. It seems like a nice and cheap 5.1 card actually. But I wanted to use the digital S/PDIF (IEC958) input/output and that does not work at all. It appears very nicely in the pulseaudio UI thing in GNOME, one output/input for analog line and S/PDIF respectively. First problem: running alsamixer -c <card> the "PCM Capture Source" appear in the Playback controls. This seems like a bug, selecting "IEC958 In" with the arrow keys work fine. "Line" and "Mic" work fine too. In the Capture controls, S/PDIF appears as mutable but without any input level setting, I guess this is normal for digital in. Anyways: no samples go out on the digital out and no samples get taken from the digital in. The endpoints are just dead. With an S/PDIF device listinging on the out line, it says "no signal" and I have tested the same equipment with the S/PDIF out on my HDA audio card and that works just fine. Since the CM6206 has that quirky quirk in usb/quirks.c that set up it's internal registers to work as expected there is a CM6206 datasheet on the web these days and it says interesting things like it needs bit 15 set in the first control register to use S/PDIF as master and also has to set the output frequency in the next three bits (etc etc) so unless someone is onto this already I'll just start hacking at it. Since Eric was patching this relating to S/PDIF I was under the impression it was working for him though? Anyways, Yours, Linus Walleij Here is lsusb -v: Bus 003 Device 004: ID 0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x0d8c C-Media Electronics, Inc. idProduct 0x0102 CM106 Like Sound Device bcdDevice 0.10 iManufacturer 0 iProduct 2 USB Sound Device iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 534 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 200 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 8 wChannelConfig 0x063f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) Side Left (SL) Side Right (SR) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 8 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 25 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 12 bNrInPins 3 baSourceID( 0) 1 baSourceID( 1) 9 baSourceID( 2) 11 bNrChannels 8 wChannelConfig 0x063f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) Side Left (SL) Side Right (SR) iChannelNames 0 bmControls 0x00 bmControls 0x00 bmControls 0x00 iMixer 0 junk at descriptor end: 00 00 00 00 00 00 00 00 00 AudioControl Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 13 bSourceID 12 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control bmaControls( 3) 0x02 Volume Control bmaControls( 4) 0x02 Volume Control bmaControls( 5) 0x02 Volume Control bmaControls( 6) 0x02 Volume Control bmaControls( 7) 0x02 Volume Control bmaControls( 8) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 13 iTerminal 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 17 bNrInPins 2 baSourceID( 0) 9 baSourceID( 1) 11 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 bmControls 0x00 iMixer 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 10 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 7 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 7 bNrInPins 4 baSource( 0) 8 baSource( 1) 15 baSource( 2) 16 baSource( 3) 2 iSelector 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 15 bSourceID 6 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 6 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 17 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 5 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 8 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0310 1x 784 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c4 1x 196 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 4 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0188 1x 392 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 6 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0248 1x 584 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 5 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0188 1x 392 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 10 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 50 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 Device Status: 0x0000 (Bus Powered)

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: soc-compress: add a config item for soc-compress
by Jie Yang 15 Sep '15

15 Sep '15

We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, please add 'select SND_SOC_COMPRESS' to driver Kconfig when it is needed. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc.h | 7 +++++++ sound/soc/Kconfig | 5 ++++- sound/soc/Makefile | 3 ++- sound/soc/intel/Kconfig | 1 + 4 files changed, 14 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf..cc1cd4f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -440,7 +440,14 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +#if IS_ENABLED(CONFIG_SND_SOC_COMPRESS) int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#else +static inline int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +{ + return -EPERM; +} +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index e2828e1..a124759 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,6 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +29,10 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + tristate + select SND_COMPRESS_OFFLOAD + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index a0e1ee6..184c1e6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,7 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o snd-soc-core-objs += soc-topology.o +snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 791953f..e559174 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -12,6 +12,7 @@ config SND_MFLD_MACHINE config SND_SST_MFLD_PLATFORM tristate + select SND_SOC_COMPRESS config SND_SST_IPC tristate -- 1.9.1

6 23

[alsa-devel] Occasional channel swapping, rt5631 vs mmp2 sspa
by James Cameron 15 Sep '15

15 Sep '15

G'day, The rt5631 has a single serial data port shared by playback and capture at the same time, using I2S. Sometimes on playback or capture the channels are swapped left for right. The swap starts as a stream starts. The swap continues until the stream stops. The SoC is an Armada 610 with an SSPA peripheral. A workaround is to remove a clock domain reset when a stream is started. This is a reset bit in a register of the SSPA. My theory is that the clock domain reset in the SSPA while I2S is clocking interferes with LRCK (aka fsync) edge detection. I'm wondering how this kind of problem should be solved properly; avoiding the clock domain reset while I2S is clocking seems right, but which codec and SoC driver ops should start I2S and/or reset the host port? My post is probably incomplete and inadequate; please ask any questions. ;-) -- James Cameron http://quozl.linux.org.au/

1 0

[alsa-devel] [PATCH] ASoC: add ak4613 support
by Kuninori Morimoto 15 Sep '15

15 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/ak4613.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4613.c | 493 +++++++++++++++++++++ 4 files changed, 516 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 sound/soc/codecs/ak4613.c diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt new file mode 100644 index 0000000..15a9195 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.txt @@ -0,0 +1,17 @@ +AK4613 I2C transmitter + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak4613" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4613: ak4613@0x10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..06bbe75 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -319,6 +319,10 @@ config SND_SOC_AK4535 config SND_SOC_AK4554 tristate "AKM AK4554 CODEC" +config SND_SOC_AK4613 + tristate "AKM AK4613 CODEC" + depends on I2C + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077..5b6c8af 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o +snd-soc-ak4613-objs := ak4613.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -216,6 +217,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o +obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c new file mode 100644 index 0000000..19e1741 --- /dev/null +++ b/sound/soc/codecs/ak4613.c @@ -0,0 +1,493 @@ +/* + * ak4613.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright (C) 2015 Renesas Electronics Corporation + * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + * + * Based on ak4642.c by Kuninori Morimoto + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#define PW_MGMT1 0x00 /* Power Management 1 */ +#define PW_MGMT2 0x01 /* Power Management 2 */ +#define PW_MGMT3 0x02 /* Power Management 3 */ +#define CTRL1 0x03 /* Control 1 */ +#define CTRL2 0x04 /* Control 2 */ +#define DEMP1 0x05 /* De-emphasis1 */ +#define DEMP2 0x06 /* De-emphasis2 */ +#define OFD 0x07 /* Overflow Detect */ +#define ZRD 0x08 /* Zero Detect */ +#define ICTRL 0x09 /* Input Control */ +#define OCTRL 0x0a /* Output Control */ +#define LOUT1 0x0b /* LOUT1 Volume Control */ +#define ROUT1 0x0c /* ROUT1 Volume Control */ +#define LOUT2 0x0d /* LOUT2 Volume Control */ +#define ROUT2 0x0e /* ROUT2 Volume Control */ +#define LOUT3 0x0f /* LOUT3 Volume Control */ +#define ROUT3 0x10 /* ROUT3 Volume Control */ +#define LOUT4 0x11 /* LOUT4 Volume Control */ +#define ROUT4 0x12 /* ROUT4 Volume Control */ +#define LOUT5 0x13 /* LOUT5 Volume Control */ +#define ROUT5 0x14 /* ROUT5 Volume Control */ +#define LOUT6 0x15 /* LOUT6 Volume Control */ +#define ROUT6 0x16 /* ROUT6 Volume Control */ + +/* PW_MGMT1 */ +#define RSTN BIT(0) +#define PMDAC BIT(1) +#define PMADC BIT(2) +#define PMVR BIT(3) + +/* PW_MGMT2 */ +#define PMAD_ALL 0x7 + +/* PW_MGMT3 */ +#define PMDA_ALL 0x3f + +/* CTRL1 */ +#define DIF0 BIT(3) +#define DIF1 BIT(4) +#define DIF2 BIT(5) +#define TDM0 BIT(6) +#define TDM1 BIT(7) +#define NO_FMT (0xff) +#define FMT_MASK (0xf8) + +/* CTRL2 */ +#define DFS_NORMAL_SPEED (0 << 2) +#define DFS_DOUBLE_SPEED (1 << 2) +#define DFS_QUAD_SPEED (2 << 2) + +struct ak4613_priv { + spinlock_t lock; + + unsigned int fmt; + u8 fmt_ctrl; + int cnt; +}; + +struct ak4613_formats { + unsigned int width; + unsigned int fmt; +}; + +struct ak4613_interface { + struct ak4613_formats capture; + struct ak4613_formats playback; +}; + +static const struct reg_default ak4613_reg[] = { + { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, + { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, + { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 }, + { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 }, + { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 }, + { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, +}; + +#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3) +#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } +static const struct ak4613_interface ak4613_iface[] = { + /* capture */ /* playback */ + [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, + [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, + [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, + [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, +}; + +static const struct regmap_config ak4613_regmap_cfg = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x16, + .reg_defaults = ak4613_reg, + .num_reg_defaults = ARRAY_SIZE(ak4613_reg), +}; + +static const struct of_device_id ak4613_of_match[] = { + { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4613_of_match); + +static const struct i2c_device_id ak4613_i2c_id[] = { + { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id); + +static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = { + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("LOUT4"), + SND_SOC_DAPM_OUTPUT("LOUT5"), + SND_SOC_DAPM_OUTPUT("LOUT6"), + + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + SND_SOC_DAPM_OUTPUT("ROUT4"), + SND_SOC_DAPM_OUTPUT("ROUT5"), + SND_SOC_DAPM_OUTPUT("ROUT6"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0), + SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0), + SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0), + SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0), + SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0), + SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0), + SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0), +}; + +static const struct snd_soc_dapm_route ak4613_intercon[] = { + {"LOUT1", NULL, "DAC1"}, + {"LOUT2", NULL, "DAC2"}, + {"LOUT3", NULL, "DAC3"}, + {"LOUT4", NULL, "DAC4"}, + {"LOUT5", NULL, "DAC5"}, + {"LOUT6", NULL, "DAC6"}, + + {"ROUT1", NULL, "DAC1"}, + {"ROUT2", NULL, "DAC2"}, + {"ROUT3", NULL, "DAC3"}, + {"ROUT4", NULL, "DAC4"}, + {"ROUT5", NULL, "DAC5"}, + {"ROUT6", NULL, "DAC6"}, + + {"DAC1", NULL, "Playback"}, + {"DAC2", NULL, "Playback"}, + {"DAC3", NULL, "Playback"}, + {"DAC4", NULL, "Playback"}, + {"DAC5", NULL, "Playback"}, + {"DAC6", NULL, "Playback"}, + + {"Capture", NULL, "ADC1"}, + {"Capture", NULL, "ADC2"}, + + {"ADC1", NULL, "LIN1"}, + {"ADC2", NULL, "LIN2"}, + + {"ADC1", NULL, "RIN1"}, + {"ADC2", NULL, "RIN2"}, +}; + +static inline void ak4613_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct device *dev = codec->dev; + + dev_dbg(dev, "reg %02x w %02x\n", reg, val); + snd_soc_write(codec, reg, val); +} + +static inline unsigned int ak4613_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct device *dev = codec->dev; + unsigned int val = snd_soc_read(codec, reg); + + dev_dbg(dev, "reg %02x r %02x\n", reg, val); + + return val; +} + +static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + unsigned long flags; + + spin_lock_irqsave(&priv->lock, flags); + priv->cnt--; + if (priv->cnt < 0) { + dev_err(dev, "unexpected counter error\n"); + priv->cnt = 0; + } + if (0 == priv->cnt) + priv->fmt_ctrl = NO_FMT; + spin_unlock_irqrestore(&priv->lock, flags); +} + +static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + fmt &= SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: + priv->fmt = fmt; + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ak4613_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 ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + const struct ak4613_formats *fmts; + struct device *dev = codec->dev; + unsigned int width = params_width(params); + unsigned int fmt = priv->fmt; + unsigned int rate; + unsigned long flags; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i, ret; + u8 fmt_ctrl, ctrl2; + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + ctrl2 = DFS_NORMAL_SPEED; + break; + case 88200: + case 96000: + ctrl2 = DFS_DOUBLE_SPEED; + break; + case 176400: + case 192000: + ctrl2 = DFS_QUAD_SPEED; + break; + default: + return -EINVAL; + } + + /* + * FIXME + * + * It doesn't support TDM at this point + */ + fmt_ctrl = NO_FMT; + for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) { + fmts = (is_play) ? &ak4613_iface[i].playback : + &ak4613_iface[i].capture; + + if (fmts->fmt != fmt) + continue; + + if (SND_SOC_DAIFMT_RIGHT_J == fmt) { + if (fmts->width != width) + continue; + } else { + if (fmts->width < width) + continue; + } + + fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i); + break; + } + + ret = -EINVAL; + if (NO_FMT == fmt_ctrl) + goto hw_params_end; + + spin_lock_irqsave(&priv->lock, flags); + if ((NO_FMT == priv->fmt_ctrl) || + (fmt_ctrl == priv->fmt_ctrl)) { + priv->fmt_ctrl = fmt_ctrl; + priv->cnt++; + ret = 0; + } + spin_unlock_irqrestore(&priv->lock, flags); + + if (ret < 0) + goto hw_params_end; + + snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); + ak4613_write(codec, CTRL2, ctrl2); + +hw_params_end: + if (ret < 0) + dev_warn(dev, "unsupported data width/format combination\n"); + + return ret; +} + +static int ak4613_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 mgmt1 = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + mgmt1 |= RSTN; + /* fall through */ + case SND_SOC_BIAS_PREPARE: + mgmt1 |= PMADC | PMDAC; + /* fall through */ + case SND_SOC_BIAS_STANDBY: + mgmt1 |= PMVR; + /* fall through */ + case SND_SOC_BIAS_OFF: + default: + break; + } + + ak4613_write(codec, PW_MGMT1, mgmt1); + + return 0; +} + +static const struct snd_soc_dai_ops ak4613_dai_ops = { + .shutdown = ak4613_dai_shutdown, + .set_fmt = ak4613_dai_set_fmt, + .hw_params = ak4613_dai_hw_params, +}; + +#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_64000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver ak4613_dai = { + .name = "ak4613-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .ops = &ak4613_dai_ops, + .symmetric_rates = 1, +}; + +static int ak4613_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + regcache_sync(regmap); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { + .resume = ak4613_resume, + .set_bias_level = ak4613_set_bias_level, + .dapm_widgets = ak4613_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), + .dapm_routes = ak4613_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), +}; + +static int ak4613_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + const struct regmap_config *regmap_cfg; + struct regmap *regmap; + struct ak4613_priv *priv; + + regmap_cfg = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4613_of_match, dev); + if (of_id) + regmap_cfg = of_id->data; + } else { + regmap_cfg = (const struct regmap_config *)id->driver_data; + } + + if (!regmap_cfg) + return -EINVAL; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->fmt_ctrl = NO_FMT; + priv->cnt = 0; + + spin_lock_init(&priv->lock); + + i2c_set_clientdata(i2c, priv); + + regmap = devm_regmap_init_i2c(i2c, regmap_cfg); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(dev, &soc_codec_dev_ak4613, + &ak4613_dai, 1); +} + +static int ak4613_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver ak4613_i2c_driver = { + .driver = { + .name = "ak4613-codec", + .owner = THIS_MODULE, + .of_match_table = ak4613_of_match, + }, + .probe = ak4613_i2c_probe, + .remove = ak4613_i2c_remove, + .id_table = ak4613_i2c_id, +}; + +module_i2c_driver(ak4613_i2c_driver); + +MODULE_DESCRIPTION("Soc AK4613 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); +MODULE_LICENSE("GPL v2"); -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: rt5645: Remove incorrect settings
by Oder Chiou 14 Sep '15

14 Sep '15

The patch removes the incorrect settings to avoid the pop sound in the first playback with headphone after boot. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4972bf3..595ace9 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2829,10 +2829,6 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } - - snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); - snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d); - snd_soc_write(codec, RT5645_DEPOP_M1, 0x0001); } else { /* jack out */ rt5645->jack_type = 0; @@ -2880,8 +2876,6 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, rt5645->en_button_func = true; regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); - regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, - RT5645_HP_CB_MASK, RT5645_HP_CB_PU); regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 00/10] Add support for img AXD audio hardware decoder
by Qais Yousef 14 Sep '15

14 Sep '15

This patch series adds AXD Alsa Compress Offload SoC driver. AXD is an audio hardware based on MIPS architecture that supports decoding, encoding, GEQ, resampling, mixing and synchronisation. At the moment only decoding support is added in hope to add the rest of the functionality on top of that once this is accepted. I divided the files into separate patches by functionality in hope it'll make the reviewing process easier. Worth noting that a lot of the cmd interface helper funtions in patch 7 are not used yet but will be as support for more functionality is added later. At the moment this code has been tested on Pistachio SoC using gstreamer patched with the code in this link https://bugzilla.gnome.org/show_bug.cgi?id=743192 Qais Yousef (10): irqchip: irq-mips-gic: export gic_send_ipi dt: add img,axd.txt device tree binding document ALSA: add AXD Audio Processing IP alsa driver ALSA: axd: add fw binary header manipulation files ALSA: axd: add buffers manipulation files ALSA: axd: add basic files for sending/receiving axd cmds ALSA: axd: add cmd interface helper functions ALSA: axd: add low level AXD platform setup files ALSA: axd: add alsa compress offload operations ALSA: axd: add Makefile .../devicetree/bindings/sound/img,axd.txt | 34 + drivers/irqchip/irq-mips-gic.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/img/Kconfig | 11 + sound/soc/img/Makefile | 1 + sound/soc/img/axd/Makefile | 13 + sound/soc/img/axd/axd_alsa_ops.c | 211 ++ sound/soc/img/axd/axd_api.h | 649 ++++ sound/soc/img/axd/axd_buffers.c | 243 ++ sound/soc/img/axd/axd_buffers.h | 74 + sound/soc/img/axd/axd_cmds.c | 102 + sound/soc/img/axd/axd_cmds.h | 532 ++++ sound/soc/img/axd/axd_cmds_config.c | 1235 ++++++++ sound/soc/img/axd/axd_cmds_decoder_config.c | 422 +++ sound/soc/img/axd/axd_cmds_info.c | 1249 ++++++++ sound/soc/img/axd/axd_cmds_internal.c | 3264 ++++++++++++++++++++ sound/soc/img/axd/axd_cmds_internal.h | 317 ++ sound/soc/img/axd/axd_cmds_pipes.c | 1387 +++++++++ sound/soc/img/axd/axd_hdr.c | 64 + sound/soc/img/axd/axd_hdr.h | 24 + sound/soc/img/axd/axd_module.c | 742 +++++ sound/soc/img/axd/axd_module.h | 83 + sound/soc/img/axd/axd_platform.h | 35 + sound/soc/img/axd/axd_platform_mips.c | 416 +++ 25 files changed, 11111 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,axd.txt create mode 100644 sound/soc/img/Kconfig create mode 100644 sound/soc/img/Makefile create mode 100644 sound/soc/img/axd/Makefile create mode 100644 sound/soc/img/axd/axd_alsa_ops.c create mode 100644 sound/soc/img/axd/axd_api.h create mode 100644 sound/soc/img/axd/axd_buffers.c create mode 100644 sound/soc/img/axd/axd_buffers.h create mode 100644 sound/soc/img/axd/axd_cmds.c create mode 100644 sound/soc/img/axd/axd_cmds.h create mode 100644 sound/soc/img/axd/axd_cmds_config.c create mode 100644 sound/soc/img/axd/axd_cmds_decoder_config.c create mode 100644 sound/soc/img/axd/axd_cmds_info.c create mode 100644 sound/soc/img/axd/axd_cmds_internal.c create mode 100644 sound/soc/img/axd/axd_cmds_internal.h create mode 100644 sound/soc/img/axd/axd_cmds_pipes.c create mode 100644 sound/soc/img/axd/axd_hdr.c create mode 100644 sound/soc/img/axd/axd_hdr.h create mode 100644 sound/soc/img/axd/axd_module.c create mode 100644 sound/soc/img/axd/axd_module.h create mode 100644 sound/soc/img/axd/axd_platform.h create mode 100644 sound/soc/img/axd/axd_platform_mips.c Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Jason Cooper <jason(a)lakedaemon.net> Cc: Marc Zyngier <marc.zyngier(a)arm.com> Cc: linux-kernel(a)vger.kernel.org Cc: linux-mips(a)linux-mips.org Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: Kumar Gala <galak(a)codeaurora.org> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> -- 2.1.0

7 57

[alsa-devel] [PATCH] ASoC: nau8825: Add driver for headset chip Nuvoton 8825
by Anatol Pomozov 14 Sep '15

14 Sep '15

Sponsored-by: Google Chromium project Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/nau8825.txt | 88 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 1041 ++++++++++++++++++++ sound/soc/codecs/nau8825.h | 317 ++++++ 5 files changed, 1452 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 0000000..380e820 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,88 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-threshold-num: Number of buttons supported + - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) + where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold calculation. + + - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. + + - nuvoton,sar-voltage: Reference voltage for button impedance measurement. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,short-key-debounce: Button short key press debounce time. + 0 - 30ms + 1 - 50ms + 2 - 100ms + 3 - 30ms + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the + clocks described in clock-names + - clock-names: should include "mclk" for the MCLK master clock + +Example: + + headset: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + + clock-names = "mclk"; + clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..9cabc05 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -896,6 +897,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077..bfbbb2a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -72,6 +72,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -264,6 +265,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..ae36d8b --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,1041 @@ +/* + * Nuvoton NAU8825 audio codec driver + * + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * Copyright 2015 Nuvoton Technology Corp. + * Co-author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/clk.h> + +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + + +#include "nau8825.h" + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static int nau8825_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Prevent startup click by letting charge pump to ramp up */ + msleep(10); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char * const nau8825_adc_decimation[] = { + "32", "64", "128", "256" +}; + +static const struct soc_enum nau8825_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8825_REG_ADC_RATE, NAU8825_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8825_adc_decimation), nau8825_adc_decimation); + +static const char * const nau8825_dac_oversampl[] = { + "64", "256", "128", "", "32" +}; + +static const struct soc_enum nau8825_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8825_dac_oversampl), nau8825_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); + +static const struct snd_kcontrol_new nau8825_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Sidetone Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Gain", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum), +}; + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, + 15, 1), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, + 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_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("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), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_pump_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_PGA("Output Driver R Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver R Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"ADC", NULL, "ADC Power"}, + {"AIFTX", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver L Stage 1", NULL, "ADACL"}, + {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"Output DACL", NULL, "Output Driver L Stage 3"}, + {"Output DACR", NULL, "Output Driver R Stage 3"}, + {"HPOL", NULL, "Output DACL"}, + {"HPOR", NULL, "Output DACR"}, + {"HPOL", NULL, "Charge Pump"}, + {"HPOR", NULL, "Charge Pump"}, +}; + +static int nau8825_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 nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->jack = jack; + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons; + + /* The chip supports up to 8 buttons, but ALSA defines only 6 buttons */ + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + if (value & BIT(4)) + buttons |= SND_JACK_BTN_4; + if (value & BIT(5)) + buttons |= SND_JACK_BTN_5; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type = 0; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + if (type & SND_JACK_HEADPHONE) { + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + lower 8 bits - for long pressed buttons */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_TRACKING_GAIN_MASK, + nau8825->sar_voltage << NAU8825_SAR_TRACKING_GAIN_SFT); + + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_LEVELS_NR_MASK, + (nau8825->sar_threshold_num - 1) << NAU8825_KEYDET_LEVELS_NR_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_HYSTERESIS_MASK, + nau8825->sar_hysteresis << NAU8825_KEYDET_HYSTERESIS_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK, + nau8825->key_debounce << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT); + + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, + (nau8825->sar_threshold[0] << 8) | nau8825->sar_threshold[1]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, + (nau8825->sar_threshold[2] << 8) | nau8825->sar_threshold[3]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_3, + (nau8825->sar_threshold[4] << 8) | nau8825->sar_threshold[5]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_4, + (nau8825->sar_threshold[6] << 8) | nau8825->sar_threshold[7]); + + /* Enable short press and release interruptions */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_KEY_SHORT_PRESS_EN | NAU8825_IRQ_KEY_RELEASE_EN, + 0); +} + +static void nau8825_init_regs(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Bias and clock are needed for button detection */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_GLOBAL_BIAS_EN, NAU8825_GLOBAL_BIAS_EN); + + /* VMID Tieoff */ + regmap_update_bits(regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID_SEL_MASK, + nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT); + /* Disable Boost Driver, Automatic Short circuit protection enable */ + regmap_update_bits(regmap, NAU8825_REG_BOOST, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN); + + /* Ground HP Outputs[1:0], needed for headset auto detection + * Enable Automatic Mic/Gnd switching reading on insert interrupt[6] */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_OUTPUT_EN, + nau8825->jkdet_enable ? 0 : NAU8825_JKDET_OUTPUT_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_EN, + nau8825->jkdet_pull_enable ? 0 : NAU8825_JKDET_PULL_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_UP, + nau8825->jkdet_pull_up ? NAU8825_JKDET_PULL_UP : 0); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_POLARITY, + /* jkdet_polarity - 1 is for active-low */ + nau8825->jkdet_polarity ? 0 : NAU8825_JACK_POLARITY); + + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_INSERT_DEBOUNCE_MASK, + nau8825->jack_insert_debounce << NAU8825_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_EJECT_DEBOUNCE_MASK, + nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT); + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_VOLTAGE_MASK, nau8825->micbias_voltage); + + if (nau8825->sar_threshold_num) + nau8825_setup_buttons(nau8825); + + /* Default oversampling/decimations settings are unusable + * (audible hiss). Set it to something better. */ + regmap_update_bits(regmap, NAU8825_REG_ADC_RATE, + NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128); + regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, + NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->dapm = &codec->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(&codec->dapm, "DDACR"); + snd_soc_dapm_sync(&codec->dapm); + + /* Unmask interruptions. Handler uses dapm object so we can enable + * interruptions only after dapm is fully initialized. */ + regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, + unsigned int freq) +{ + struct regmap *regmap = nau8825->regmap; + + switch (clk_id) { + case NAU8825_CLK_MCLK: + 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) + clk_prepare_enable(nau8825->mclk); + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + clk_set_rate(nau8825->mclk, freq); + } + + break; + case NAU8825_CLK_INTERNAL: + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, + NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + default: + dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8825->dev, "Sysclk is %dHz and clock id is %d\n", freq, + clk_id); + return 0; +} + +static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + return nau8825_configure_sysclk(nau8825, clk_id, freq); +} + +static int nau8825_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (nau8825->mclk_freq) + clk_prepare_enable(nau8825->mclk); + + ret = regcache_sync(nau8825->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + break; + + case SND_SOC_BIAS_OFF: + if (nau8825->mclk_freq) + clk_disable_unprepare(nau8825->mclk); + + regcache_mark_dirty(nau8825->regmap); + break; + } + return 0; +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + .set_sysclk = nau8825_set_sysclk, + .set_bias_level = nau8825_set_bias_level, + .suspend_bias_off = true, + + .controls = nau8825_controls, + .num_controls = ARRAY_SIZE(nau8825_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825; + int ret, value; + + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->irq = i2c->irq; + nau8825->dev = dev; + + nau8825->jkdet_enable = device_property_read_bool(dev, + "nuvoton,jkdet-enable"); + nau8825->jkdet_pull_enable = device_property_read_bool(dev, + "nuvoton,jkdet-pull-enable"); + nau8825->jkdet_pull_up = device_property_read_bool(dev, + "nuvoton,jkdet-pull-up"); + device_property_read_u32(dev, "nuvoton,jkdet-polarity", + &nau8825->jkdet_polarity); + device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8825->micbias_voltage); + device_property_read_u32(dev, "nuvoton,vref-impedance", + &nau8825->vref_impedance); + device_property_read_u32(dev, "nuvoton,sar-threshold-num", + &nau8825->sar_threshold_num); + device_property_read_u32_array(dev, "nuvoton,sar-threshold", + nau8825->sar_threshold, nau8825->sar_threshold_num); + device_property_read_u32(dev, "nuvoton,sar-hysteresis", + &nau8825->sar_hysteresis); + device_property_read_u32(dev, "nuvoton,sar-voltage", + &nau8825->sar_voltage); + device_property_read_u32(dev, "nuvoton,short-key-debounce", + &nau8825->key_debounce); + device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8825->jack_insert_debounce); + device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8825->jack_eject_debounce); + + + nau8825->mclk = devm_clk_get(dev, "mclk"); + if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { + return -EPROBE_DEFER; + } else if (PTR_ERR(nau8825->mclk) == -ENOENT) { + /* The MCLK is managed externally or not used at all */ + nau8825->mclk = NULL; + dev_info(dev, "No 'mclk' clock found, assume MCLK is managed externally"); + } else if (IS_ERR(nau8825->mclk)) { + return -EINVAL; + } + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + nau8825_init_regs(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 */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + if (nau8825->irq) { + int ret = devm_request_threaded_irq(dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + } + + return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..b84148b --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,317 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov(a)chrominium.org> + * + * 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 __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* FLL6 (0x9) */ +#define NAU8825_DCO_EN (1 << 15) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +/* 0 - short to GND, 1 - open */ +#define NAU8825_SPKR_DWN1R (1 << 1) +#define NAU8825_SPKR_DWN1L (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8825_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_EJECT_DEBOUNCE_SFT) +#define NAU8825_JACK_POLARITY (1 << 1) /* 0 - active low, 1 - active high */ + +/* INTERRUPT_MASK (0xf) */ +#define NAU8825_IRQ_OUTPUT_EN (1 << 11) +#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) +#define NAU8825_IRQ_EJECT_EN (1 << 2) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* INTERRUPT_DIS_CTRL (0x12) */ +#define NAU8825_IRQ_HEADSET_COMPLETE_DIS (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) +#define NAU8825_IRQ_EJECT_DIS (1 << 2) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) +#define NAU8825_SAR_TRACKING_GAIN_SFT 8 +#define NAU8825_SAR_TRACKING_GAIN_MASK (0x7 << NAU8825_SAR_TRACKING_GAIN_SFT) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT 12 +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK (0x3 << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT) +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) +#define NAU8825_KEYDET_HYSTERESIS_SFT 0 +#define NAU8825_KEYDET_HYSTERESIS_MASK 0xf + +/* GPIO12_CTRL (0x1a) */ +#define NAU8825_JKDET_PULL_UP (1 << 11) /* 0 - pull down, 1 - pull up */ +#define NAU8825_JKDET_PULL_EN (1 << 9) /* 0 - enable pull, 1 - disable */ +#define NAU8825_JKDET_OUTPUT_EN (1 << 8) /* 0 - enable niput, 1 - enable output */ + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - Normal mode, 1 - Output Hi-Z */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SYNC_DOWN_SFT 0 +#define NAU8825_ADC_SYNC_DOWN_MASK 0x3 +#define NAU8825_ADC_SYNC_DOWN_32 0 +#define NAU8825_ADC_SYNC_DOWN_64 1 +#define NAU8825_ADC_SYNC_DOWN_128 2 +#define NAU8825_ADC_SYNC_DOWN_256 3 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_SFT 0 +#define NAU8825_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8825_DAC_OVERSAMPLE_64 0 +#define NAU8825_DAC_OVERSAMPLE_256 1 +#define NAU8825_DAC_OVERSAMPLE_128 2 +#define NAU8825_DAC_OVERSAMPLE_32 4 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_SFT 4 +#define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 +#define NAU8825_MICBIAS_VOLTAGE_SFT 0 +#define NAU8825_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_G_DIS (1 << 8) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_INTEGR_R (1 << 5) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) + +/* CHARGE_PUMP (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + int irq; + struct snd_soc_jack *jack; + struct clk *mclk; + int mclk_freq; /* 0 - mclk is disabled */ + int button_pressed; + int micbias_voltage; + int vref_impedance; + bool jkdet_enable; + bool jkdet_pull_enable; + bool jkdet_pull_up; + int jkdet_polarity; + int sar_threshold_num; + int sar_threshold[8]; + int sar_hysteresis; + int sar_voltage; + int key_debounce; + int jack_insert_debounce; + int jack_eject_debounce; +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- 2.5.0.457.gab17608

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Revise the FIFO threshold calculation
by Peter Ujfalusi 14 Sep '15

14 Sep '15

The FIFO threshold for McASP should be <=[tx/rx]numevt so the initial value for the refining should meet this requirement as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index add6bb99661d..1260b315a96c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -663,7 +663,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, u8 rx_ser = 0; u8 slots = mcasp->tdm_slots; u8 max_active_serializers = (channels + slots - 1) / slots; - int active_serializers, numevt, n; + int active_serializers, numevt; u32 reg; /* Default configuration */ if (mcasp->version < MCASP_VERSION_3) @@ -745,9 +745,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, * The number of words for numevt need to be in steps of active * serializers. */ - n = numevt % active_serializers; - if (n) - numevt += (active_serializers - n); + numevt = (numevt / active_serializers) * active_serializers; + while (period_words % numevt && numevt > 0) numevt -= active_serializers; if (numevt <= 0) -- 2.5.2

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[alsa-devel] [PATCH] ALSA: pcm: remove structure member of 'struct snd_pcm_hwptr_log *' type because this structure had been removed
by Takashi Sakamoto 13 Sep '15

13 Sep '15

This structure was added by 4d96eb255c53 ('ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions') to store PCM pointers information of latest 10 pointer movements (=XRUN_LOG_CNT). When CONFIG_SND_PCM_XRUN_DEBUG is configured, 'struct snd_pcm_runtime' has 'hwptr_log' member with a pointer to the structure. When calling xrun_log() in pcm_lib.c, the structure was allocated to the pointer. When calling snd_pcm_detach_substream() in pcm.c, the allocated pointer is released. In f5914908a5b7 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints'), the pointer logging is replaced with using Linux Kernel Tracepoints. The structure was also removed, while it's just declared. The member and kfree still remains. This commit removes the member and related codes. I think this was overlooked because it brings no errors/warnings to C compilers. Fixes: f5914908a5b7 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/pcm.h | 6 ------ sound/core/pcm.c | 3 --- 2 files changed, 9 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee..a4fcc94 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -285,8 +285,6 @@ struct snd_pcm_hw_constraint_ranges { unsigned int mask; }; -struct snd_pcm_hwptr_log; - /* * userspace-provided audio timestamp config to kernel, * structure is for internal use only and filled with dedicated unpack routine @@ -428,10 +426,6 @@ struct snd_pcm_runtime { /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif - -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - struct snd_pcm_hwptr_log *hwptr_log; -#endif }; struct snd_pcm_group { /* keep linked substreams */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 02bd969..308c9ec 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1014,9 +1014,6 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) snd_free_pages((void*)runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - kfree(runtime->hwptr_log); -#endif kfree(runtime); substream->runtime = NULL; put_pid(substream->pid); -- 2.1.4

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Re: [alsa-devel] [Alsa-user] 'control ... is already present' message
by Ran Shalit 12 Sep '15

12 Sep '15

On Fri, Sep 11, 2015 at 9:42 AM, Clemens Ladisch <cladisch(a)googlemail.com> wrote: > Ran Shalit wrote: >> asoc: failed to add dapm kcontrol Right HPCOM Mixer Line2L Bypass Switch: -16 >> asoc: failed to add kcontrol Right Line2R Mux >> ... >> What is the reason for the "asoc: failed to add dapm kcontrol" and the >> " 'control ... already present" messeges ? > > That there are two controls with the same name and index. > >> If I move each to be registered to its own card will it resolve this issue ? > > Probably, but a better idea might be to set index=1 for all controls of the > second codec. I don't know how ASoC handles this; ask on the alsa-devel > list. > > > Regards, > Clemens Hi Clemens, Can you please add how to set index=1 to second codec ? Thank you, Ran

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[alsa-devel] What is exact definition of "normal/inverted" FSYNC signal polarity?
by Anatol Pomozov 12 Sep '15

12 Sep '15

Hi I am trying to configure TegraX1 SoC and Realtek5677 codec. These two drivers have different meaning of FSYNC polarity in TDM mode A format. * TegraX1 driver thinks that "normal" polarity is when a frame starts with FSYNC falling edge. TegraX1 applies it to all formats (I2S, PCM/TDM mode A). And inverted FSYNC means that frame starts with rising edge of FSYNC. * RT5677 thinks that "normal" for I2S is "frame starts with falling edge". And for TMD mode A "normal" means "frame starts with rising FSYNC edge". Thus normal/inverted FSYNC in TDM mode A are opposite at these 2 chips. Which one is correct? What is exact definition of SND_SOC_DAIFMT_xB_yF constants in soc-dai.h? I tried to google if I2S spec [1] defines FSYNC polarity but no luck. It worth to clarify definition of BCLK polarity as well. I believe "normal" BCLK is "line sensing happens at rising edge", "inverted" BCLK - "line sensing happens at falling edge". Is it correct? [1] https://web.archive.org/web/20060821114007/http://www.semiconductors.philip…

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[alsa-devel] [PATCH] ALSA: hdsp: fix memory leak
by Sudip Mukherjee 11 Sep '15

11 Sep '15

If the size of the firmware is less than expected size then we are exiting with the error code but we missed releasing the firmware. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/rme9652/hdsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 9bba275..2875b4f 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5112,6 +5112,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) dev_err(hdsp->card->dev, "too short firmware size %d (expected %d)\n", (int)fw->size, HDSP_FIRMWARE_SIZE); + release_firmware(fw); return -EINVAL; } -- 1.9.1

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[alsa-devel] [PATCH v4 0/10] Fix broken DMAFLUSHP on Rockchips platform
by Shawn Lin 11 Sep '15

11 Sep '15

The purpose of the DMAFLUSHP instruction: - Tell the peripheral to clear its status and control registers. - Send a message to the peripheral to resend its level status. There are 3 timings described in PL330 Technical Reference Manual: - Timing 1: Burst request, can work well without DMAFLUSHP. - Timing 2: Single and burst request, DMAC will ignore the single transfer request. This timing happens if there are single and burst request. - Timing 3: Single transfers for a burst request, DMAC should signals datype to request the peripheral to flush the contents of any control registers. This timing happens if there is not enough MFIFO to places the burst data. A peripheral may signal a DMA request during the execution of DMAFLUSHP instruction, that cause DMA request being ignored by DMAC. But DMAC and all peripherals on RK3X SoCs DO NOT support DMAFLUSHP. It can't send a message to the peripheral to resend DMA request, and the peripheral can't acknowledge a flush request from DMAC. So all DMA requests should NOT be ignored by DMAC, and DMAC will not notify the peripheral to flush. To fix this problem, we need: - Do NOT execute DMAFLUSHP instruction. - Timing 2 and timing 3 should not happen. Because on RK3X SoCs, there are 6 or below channels and 32 MFIFO depth for DMAC_BUS, and 8 channels and 64 MFIFO depth for DMAC_PERI, it is impossible to hit the timing 3 if burst length is equal or less than 4. Since the request type signal by the peripheral can only be set by software. We can set Rockchip Soc's GRF_PERIDMAC_CON0[2:1] to select single or burst request, if it is set b01, all of the peripharals will signal a brust request. So the timing 2 will not happen, too. So DMAC on RK3X can support single or burst transfer, but can't support mixed transfer. Because burst transfer is more efficient than single transfer, this is confirmed by our ASIC team, who strongly suggest to use burst transfer. And this is confirmed by Addy's test on RK3288-Pink2 board, the speed of spi flash burst transfer will increase about two times than single transfer. Also, I have tested dw_mmc with pl330 on RK3188 platform to double confirm the result. That means burst transfer is reansonable. So we need a quirk not to execute DMAFLUSHP instruction and to use burst transfer. Note: - The Rockchip Soc default value of GRF_PERIDMAC_CON0[2:1] is b01. To support brust transfer, these bits should not be changed in bootloader. Changes in v4: - remove spi & i2s dts changes and query quirk from dmaengine API suggeseted by Mark Brown. - fix typo - Add dmaengine_get_quirk hook and implement it for pl330 Changes in v3: - add more rockchip drivers' changes in this patchset - add Reviewed-by: Sonny Rao <sonnyrao(a)chromium.org> Changes in v2: - amend the author - reorder the patches suggested by Doug - add Reviewed-by: Doug Anderson <dianders(a)chromium.org> for rk3288.dtsi patch and arm-pl330.txt patch - amend Olof's mail address Changes in v1: - rename broken-no-flushp to "arm,pl330-broken-no-flushp" suggested by Krzysztof. - add From original author. - remove Sunny's tag Addy Ke (3): DMA: pl330: add quirk for broken no flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3288 platform spi: rockchip: modify DMA max burst to 1 Boojin Kim (1): DMA: pl330: support burst mode for dev-to-mem and mem-to-dev transmit Shawn Lin (5): Documentation: arm-pl330: add description of arm,pl330-broken-no-flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3xxx platform dmaengine: add API for getting dma controller's quirk DMA: pl330: implement dmaengine_get_quirks hook snd: dmaengine-pcm: add snd_dmaengine_pcm_get_quirks interface Yiwei Cai (1): ASoC: rockchip_i2s: modify DMA max burst to 1 .../devicetree/bindings/dma/arm-pl330.txt | 1 + arch/arm/boot/dts/rk3288.dtsi | 3 + arch/arm/boot/dts/rk3xxx.dtsi | 3 + drivers/dma/pl330.c | 110 +++++++++++++++------ drivers/spi/spi-rockchip.c | 13 ++- include/linux/dmaengine.h | 9 ++ sound/soc/rockchip/rockchip_i2s.c | 16 ++- sound/soc/soc-generic-dmaengine-pcm.c | 24 +++++ 8 files changed, 147 insertions(+), 32 deletions(-) -- 2.3.7

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[alsa-devel] [PATCH] ASoC: Intel: Bsw: Read sst DSP DMA pointer after period elapse occurs
by yang.a.fang＠intel.com 11 Sep '15

11 Sep '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> We should not read sst hw_ptr and pcm_delay in sst_platform_pcm_pointer Since it will be ricky if sst_platform_pcm_pointer is called while dsp is updating the timestamp.Now read sst hw_ptr after period elapse interrupt occurs. Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 683e501..d58230b 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -263,17 +263,27 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, static void sst_period_elapsed(void *arg) { struct snd_pcm_substream *substream = arg; + struct snd_soc_pcm_runtime *rtd; struct sst_runtime_stream *stream; + struct pcm_stream_info *str_info; int status; + int ret_val; if (!substream || !substream->runtime) return; + rtd = substream->private_data; stream = substream->runtime->private_data; if (!stream) return; status = sst_get_stream_status(stream); if (status != SST_PLATFORM_RUNNING) return; + str_info = &stream->stream_info; + ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); + if (ret_val) { + dev_err(rtd->dev, "sst: err code = %d\n", ret_val); + return; + } snd_pcm_period_elapsed(substream); } @@ -660,20 +670,14 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer (struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream; - int ret_val, status; + int status; struct pcm_stream_info *str_info; - struct snd_soc_pcm_runtime *rtd = substream->private_data; stream = substream->runtime->private_data; status = sst_get_stream_status(stream); if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); - if (ret_val) { - dev_err(rtd->dev, "sst: error code = %d\n", ret_val); - return ret_val; - } substream->runtime->delay = str_info->pcm_delay; return str_info->buffer_ptr; } -- 1.7.9.5

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Re: [alsa-devel] [PATCH 1/1] sound/usb: Add Asus Xonar U7 Mixer - input support
by Johan Aires Rastén 11 Sep '15

11 Sep '15

Hi Piotr, great to see someone else also working on getting the U7 working! Unfortunately I don't have the answer you're looking for as I don't know that much about ALSA actually. Most of what I've done was just comparing the UAC2 specification with what the driver does, and some light USB protocol sniffing. I'm not actively working on the driver since I bought a new sound card for my PC and am using the U7 with my Windows laptop, but send me a message on Google Hangouts if you think I can help with something. When I connect the U7 it starts with "mic" as input, but I have to switch it to line and back to mic before it works. Does this happen to you too? //Johan On 10 September 2015 at 16:47, Piotr SQ9FK <sq9fk(a)wp.pl> wrote: > Hello Takashi,Johan > My patches are unessesary since following last patch was applied to the git: > ALSA: usb-audio: Change internal PCM order > > Capture working but there are no volume sliders for Line-In Capture or > Mic-In Capture (I can only toggle on off). Probably the RANGES are > broken?. From ASUS Windows driver I can see following ranges: > > Master Volume: (Phisycal Knob on Asus) > from 100 to 1 correspond 0dB to -70dB, 0 is -127dB > Range from 1 to 100 -> 0dB to -70dB (logarythmic) > > Mic-In: > from 100 to 1 correspond 30dB to -10dB, 0 is -192dB > Range from 1 to 10 -> -10dB to 10dB (linear?) > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Line-In: > from 100 to 1 correspond 0dB to -40dB, 0 is -192dB > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Actual Situation: > Input Gain Pad Control - 3 steps [0 - 50 - 100] > Double Slider. In Capture and Playback Categories, both of them are > controlled in the same time. > Controls Playback Volume only, Capture not. > > "Mic" and "Line" Switches [on/off] > PCM Cature Source [Line/Mic] > Electret Mic doesn't work. With all possible configurations - Silent. > When I pluged Line level signal, I can record the signal without problems. > PCM Capture Source [Line] - Line [on] Mic [off] - Sound > PCM Capture Source [Line] - Line [on] Mic [on] - Sound > PCM Capture Source [Line] - Line [off] Mic [off] - No Sound > PCM Capture Source [Line] - Line [off] Mic [on] - No Sound > The same with selected [Mic]. There is no Capture Volume control. > > PCM 1 [mute/on] > I think this is SPDIF switch. It doesn't work. When I switch it on -> > Blue Led on Xonar doesn't light up. > > I attached screen of alsamixer, usbmix from proc and my mixer_map from > my old patch. > I think there are is still bugs in parsing. Please point me how I can > add manually Volume control for Capture? > > Piotr Gaska > > W dniu 2015-07-27 o 14:10, Takashi Iwai pisze: > >> On Sun, 26 Jul 2015 10:00:12 +0200, >> Piotr G. wrote: >>> >>> Hello, >>> This is my first patch to alsa-devel, and I'm not very skilled >>> programmer. From lsusb -vvv, I've draw Xonar U7 map with all IT OT FU CS >>> in mixer_map.c I don't know why the driver by default doesn't parse all >>> info correctly. With this patch you can see Line Switch and Mic Switch >>> in alsamixer. I can record my voice in Audacity with all supported bit >>> and sampling rates in Mono or Stereo >>> This is temporary solution. Please correct me if the patch is not >>> properly made. >>> >>> What doesn't work: >>> - Cannot change Capture Volume for Line/Mic. No mixer control in >>> alsamixer >>> - Switching between Line/Mic doesn't work (Mute doesn't work) >>> - SPDIF not tested >>> As what I wrote at beginning I'm not skilled programmer so any help >>> is >>> aprreciated >>> >>> >>> proc-usbmixer -> cat /proc/asound/*/usbmix >>> usb.debug -> lsusb -vv (only Asus Xonar U7) >> >> The changes in mixer_maps.c look good, but what's the reason for >> uac2_audio_feature_info[]? It seems adding "Control" suffix to each >> entry, which doesn't look correct. >> >> >> thanks, >> >> Takashi >> >> >>> --- >>> sound/usb/mixer.c | 23 +++++++++++++++++++++-- >>> sound/usb/mixer_maps.c | 48 >>> ++++++++++++++++++++++++++++++++++++++++++++++++ >>> 2 files changed, 69 insertions(+), 2 deletions(-) >>> >>> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c >>> index 6b3acba..027d160 100644 >>> --- a/sound/usb/mixer.c >>> +++ b/sound/usb/mixer.c >>> @@ -818,6 +818,25 @@ static struct usb_feature_control_info >>> audio_feature_info[] = { >>> { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> }; >>> >>> +static struct usb_feature_control_info uac2_audio_feature_info[] = { >>> + /* UAC2 specific */ >>> + { "Mute Control", USB_MIXER_INV_BOOLEAN }, >>> + { "Volume Control", USB_MIXER_S16 }, >>> + { "Bass Control", USB_MIXER_S8 }, >>> + { "Mid Control", USB_MIXER_S8 }, >>> + { "Treble Control", USB_MIXER_S8 }, >>> + { "Graphic Equalizer Control", USB_MIXER_S8 }, >>> + { "Automatic Gain Control", USB_MIXER_BOOLEAN }, >>> + { "Delay Control", USB_MIXER_S16 }, /* FIXME: U32 in >>> UAC2 */ >>> + { "Bass Boost Control", USB_MIXER_BOOLEAN }, >>> + { "Loudness Control", USB_MIXER_BOOLEAN }, >>> + { "Input Gain Control", USB_MIXER_S16 }, >>> + { "Input Gain Pad Control", USB_MIXER_S16 }, >>> + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> + { "Underflow Control", USB_MIXER_BOOLEAN }, >>> + { "Overflow Control", USB_MIXER_BOOLEAN }, >>> +}; >>> + >>> /* private_free callback */ >>> void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) >>> { >>> @@ -1240,7 +1259,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); >>> cval->control = control; >>> cval->cmask = ctl_mask; >>> - cval->val_type = audio_feature_info[control-1].type; >>> + cval->val_type = uac2_audio_feature_info[control-1].type; >>> if (ctl_mask == 0) { >>> cval->channels = 1; /* master channel */ >>> cval->master_readonly = readonly_mask; >>> @@ -1318,7 +1337,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> break; >>> default: >>> if (!len) >>> - strlcpy(kctl->id.name, >>> audio_feature_info[control-1].name, >>> + strlcpy(kctl->id.name, >>> uac2_audio_feature_info[control-1].name, >>> sizeof(kctl->id.name)); >>> break; >>> } >>> diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c >>> index e5000da..07f510e 100644 >>> --- a/sound/usb/mixer_maps.c >>> +++ b/sound/usb/mixer_maps.c >>> @@ -341,6 +341,48 @@ static const struct usbmix_name_map >>> scms_usb3318_map[] = { >>> { 0 } >>> }; >>> >>> +/* ASUS Xonar U7 - USB CM6632A (ADC CS5361 // DAC Dolby CS4362A // DAC >>> Stereo CS4398 */ >>> +/* Topology: >>> + >>> +USB_IN[1]------->FU[13]-----------CS[18]--->Spk_OUT[7] >>> + >>> +Dig_IN[2]------->FU[14]-----------CS[19]--->Dig_OUT[8] >>> + >>> +Mic_IN[4]------->FU[16]---+ >>> + SU[20]--CS[22]--->USB_OUT[11] >>> +Line_IN[5]------>FU[17]---+ >>> + >>> +*/ >>> + >>> +static const struct usbmix_name_map xonar_u7_map[] = { >>> + /* 1: IT PCM Analog (USB Streaming) */ >>> + /* 2: IT PCM2 Digital (USB Streaming) */ >>> + /* 4: IT Microphone */ >>> + /* 5: IT Line IN */ >>> + /* 7: OT Speaker */ >>> + /* 8: OT SPDIF */ >>> + /* 11 OT Capture Playback (USB Streaming) */ >>> + { 13, "Speaker Playback" }, /* 13 FU Speaker Out Mute/Volume >>> */ >>> + { 14, "IEC958 Playback" }, /* 14 FU SPDIF Out Mute/Volume */ >>> + { 16, "Mic Capture" }, /* 16 FU Capture Mic-in >>> Mute/Volume */ >>> + { 17, "Line Capture" }, /* 17 FU Capture Line-in >>> Mute/Volume */ >>> + /* 18 CS for 1 PCM (USB Streaming) Clock Source */ >>> + /* 19 CS for 2 SPDIF (USB Streaming) Clock Source */ >>> + { 20, "Capture Source" }, /* 20 SU Capture Source Selector >>> */ >>> + /* 22 CS Line-in and Mic-in Clock Source */ >>> + { 0 } >>> +}; >>> + >>> +static struct usbmix_selector_map xonar_u7_selectors[] = { >>> +{ >>> + .id = 20, >>> + .count = 2, >>> + .names = (const char*[]) {"Mic", "Line"} >>> + }, >>> + { 0 } /* terminator */ >>> + >>> +}; >>> + >>> /* >>> * Control map entries >>> */ >>> @@ -451,6 +493,12 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { >>> .id = USB_ID(0x25c4, 0x0003), >>> .map = scms_usb3318_map, >>> }, >>> + { >>> + /* Asus Xonar U7 */ >>> + .id = USB_ID(0x1043, 0x857c), >>> + .map = xonar_u7_map, >>> + .selector_map = xonar_u7_selectors, >>> + }, >>> { 0 } /* terminator */ >>> }; >>> >>> -- >>> 2.3.6 >>> >

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[alsa-devel] Mixer control appears twice in alsamixer
by James Cameron 11 Sep '15

11 Sep '15

G'day, I've a mixer control for "Analog Mic Boost" which appears twice in alsamixer --view all, and once in amixer. How do I cause it to appear only once in alsamixer? amixer shows both Playback and Capture channels for the control: -- Simple mixer control 'Analog Mic Boost',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 8 Mono: 8 [100%] -- The control is declared with: /* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */ static unsigned int mic_bst_tlv[] = { TLV_DB_RANGE_HEAD(7), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), }; static const struct snd_kcontrol_new rt5631_snd_controls[] = { ... SOC_SINGLE_TLV("Analog Mic Boost", RT5631_MIC_CTRL_2, RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv), ... } I'm working on alc5631/rt5631 in a local branch of either 3.0 or 3.5 for the OLPC XO-1.75 and XO-4. But the declaration seems no different in 4.2, apart from the name of the control. https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/… https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/… -- James Cameron http://quozl.linux.org.au/

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[alsa-devel] 'control ... is already present' message
by Ran Shalit 11 Sep '15

11 Sep '15

Hello, I'm using 2 codecs (tlv320aic3x-codec) each which both are registered into same sound card, each as a single device. After boot I have the following devices: root@dm814x:~# cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture But I get the following error messeges during boot: usbcore: registered new interface driver snd-usb-audio Registered tvp5158 audio codec PM: Adding info for platform:soc-audio.0 PM: Adding info for No Bus:TLV320AIC3X_2 asoc: tlv320aic3x-hifi <-> davinci-mcasp.2 mapping ok snd_soc_instantiate_card 7 tlv320aic3x-codec 2-0019: tlv320aic3x-codec.2-0019: Failed to add PCM Playback Volume: -16 asoc: failed to add dapm kcontrol Right HPCOM Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Left HPCOM Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Right HP Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Left HP Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Mono Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Right Line Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Left Line Mixer Line2L Bypass Switch: -16 asoc: failed to add kcontrol Right Line2R Mux asoc: failed to add kcontrol Right Line1R Mux asoc: failed to add kcontrol Right Line1L Mux asoc: failed to add dapm kcontrol Right PGA Mixer Line1R Switch: -16 asoc: failed to add kcontrol Left Line2L Mux asoc: failed to add kcontrol Left Line1R Mux asoc: failed to add kcontrol Left Line1L Mux asoc: failed to add dapm kcontrol Left PGA Mixer Line1L Switch: -16 asoc: failed to add kcontrol Right HPCOM Mux asoc: failed to add kcontrol Right DAC Mux asoc: failed to add kcontrol Left HPCOM Mux asoc: failed to add kcontrol Left DAC Mux PM: Adding info for No Bus:TLV320AIC3X_3 asoc: tlv320aic3x-hifi <-> davinci-mcasp.3 mapping ok PM: Adding info for No Bus:card0 PM: Adding info for No Bus:pcmC0D1p PM: Adding info for No Bus:pcmC0D1c PM: Adding info for No Bus:pcmC0D0p PM: Adding info for No Bus:pcmC0D0c PM: Adding info for No Bus:controlC0 ALSA device list: #0: TI81XX SOUND0 What is the reason for the "asoc: failed to add dapm kcontrol" and the " 'control ... already present" messeges ? Is it becuase both codec are registered on same card (single card) ? 1. If I move each to be registered to its own card, will it resolve this issue ? 2. Should the following patch resolve this issue: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-November/033593.h… Regards, Ran

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[alsa-devel] [PATCH 1/1] sound/usb: Add Asus Xonar U7 Mixer - input support
by Piotr G. 10 Sep '15

10 Sep '15

Hello, This is my first patch to alsa-devel, and I'm not very skilled programmer. From lsusb -vvv, I've draw Xonar U7 map with all IT OT FU CS in mixer_map.c I don't know why the driver by default doesn't parse all info correctly. With this patch you can see Line Switch and Mic Switch in alsamixer. I can record my voice in Audacity with all supported bit and sampling rates in Mono or Stereo This is temporary solution. Please correct me if the patch is not properly made. What doesn't work: - Cannot change Capture Volume for Line/Mic. No mixer control in alsamixer - Switching between Line/Mic doesn't work (Mute doesn't work) - SPDIF not tested As what I wrote at beginning I'm not skilled programmer so any help is aprreciated proc-usbmixer -> cat /proc/asound/*/usbmix usb.debug -> lsusb -vv (only Asus Xonar U7) --- sound/usb/mixer.c | 23 +++++++++++++++++++++-- sound/usb/mixer_maps.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6b3acba..027d160 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -818,6 +818,25 @@ static struct usb_feature_control_info audio_feature_info[] = { { "Phase Inverter Control", USB_MIXER_BOOLEAN }, }; +static struct usb_feature_control_info uac2_audio_feature_info[] = { + /* UAC2 specific */ + { "Mute Control", USB_MIXER_INV_BOOLEAN }, + { "Volume Control", USB_MIXER_S16 }, + { "Bass Control", USB_MIXER_S8 }, + { "Mid Control", USB_MIXER_S8 }, + { "Treble Control", USB_MIXER_S8 }, + { "Graphic Equalizer Control", USB_MIXER_S8 }, + { "Automatic Gain Control", USB_MIXER_BOOLEAN }, + { "Delay Control", USB_MIXER_S16 }, /* FIXME: U32 in UAC2 */ + { "Bass Boost Control", USB_MIXER_BOOLEAN }, + { "Loudness Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16 }, + { "Input Gain Pad Control", USB_MIXER_S16 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Underflow Control", USB_MIXER_BOOLEAN }, + { "Overflow Control", USB_MIXER_BOOLEAN }, +}; + /* private_free callback */ void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) { @@ -1240,7 +1259,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + cval->val_type = uac2_audio_feature_info[control-1].type; if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; @@ -1318,7 +1337,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, break; default: if (!len) - strlcpy(kctl->id.name, audio_feature_info[control-1].name, + strlcpy(kctl->id.name, uac2_audio_feature_info[control-1].name, sizeof(kctl->id.name)); break; } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index e5000da..07f510e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -341,6 +341,48 @@ static const struct usbmix_name_map scms_usb3318_map[] = { { 0 } }; +/* ASUS Xonar U7 - USB CM6632A (ADC CS5361 // DAC Dolby CS4362A // DAC Stereo CS4398 */ +/* Topology: + +USB_IN[1]------->FU[13]-----------CS[18]--->Spk_OUT[7] + +Dig_IN[2]------->FU[14]-----------CS[19]--->Dig_OUT[8] + +Mic_IN[4]------->FU[16]---+ + SU[20]--CS[22]--->USB_OUT[11] +Line_IN[5]------>FU[17]---+ + +*/ + +static const struct usbmix_name_map xonar_u7_map[] = { + /* 1: IT PCM Analog (USB Streaming) */ + /* 2: IT PCM2 Digital (USB Streaming) */ + /* 4: IT Microphone */ + /* 5: IT Line IN */ + /* 7: OT Speaker */ + /* 8: OT SPDIF */ + /* 11 OT Capture Playback (USB Streaming) */ + { 13, "Speaker Playback" }, /* 13 FU Speaker Out Mute/Volume */ + { 14, "IEC958 Playback" }, /* 14 FU SPDIF Out Mute/Volume */ + { 16, "Mic Capture" }, /* 16 FU Capture Mic-in Mute/Volume */ + { 17, "Line Capture" }, /* 17 FU Capture Line-in Mute/Volume */ + /* 18 CS for 1 PCM (USB Streaming) Clock Source */ + /* 19 CS for 2 SPDIF (USB Streaming) Clock Source */ + { 20, "Capture Source" }, /* 20 SU Capture Source Selector */ + /* 22 CS Line-in and Mic-in Clock Source */ + { 0 } +}; + +static struct usbmix_selector_map xonar_u7_selectors[] = { +{ + .id = 20, + .count = 2, + .names = (const char*[]) {"Mic", "Line"} + }, + { 0 } /* terminator */ + +}; + /* * Control map entries */ @@ -451,6 +493,12 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x25c4, 0x0003), .map = scms_usb3318_map, }, + { + /* Asus Xonar U7 */ + .id = USB_ID(0x1043, 0x857c), + .map = xonar_u7_map, + .selector_map = xonar_u7_selectors, + }, { 0 } /* terminator */ }; -- 2.3.6

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