December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - Add warning texts when codec driver Kconfig doesn't match
by Takashi Iwai 19 Dec '13

19 Dec '13

When a Kconfig of a codec driver doesn't match with the controller (CONFIG_SND_HDA_INTEL), it'll result in the non-working automatic probing. Unfortunately kbuild can't give such a restriction, but at least, it's possible to show a warning if such a condition is found. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a patch on top of the previous two patches for making HD-audio codec kconfigs to tristate. sound/pci/hda/Kconfig | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index b0c0481c3339..0e53634dbbd8 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -93,6 +93,9 @@ config SND_HDA_CODEC_REALTEK Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_REALTEK=m + config SND_HDA_CODEC_ANALOG tristate "Build Analog Device HD-audio codec support" select SND_HDA_GENERIC @@ -100,6 +103,9 @@ config SND_HDA_CODEC_ANALOG Say Y or M here to include Analog Device HD-audio codec support in snd-hda-intel driver, such as AD1986A. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_ANALOG=m + config SND_HDA_CODEC_SIGMATEL tristate "Build IDT/Sigmatel HD-audio codec support" select SND_HDA_GENERIC @@ -107,6 +113,9 @@ config SND_HDA_CODEC_SIGMATEL Say Y or M here to include IDT (Sigmatel) HD-audio codec support in snd-hda-intel driver, such as STAC9200. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_SIGMATEL=m + config SND_HDA_CODEC_VIA tristate "Build VIA HD-audio codec support" select SND_HDA_GENERIC @@ -114,6 +123,9 @@ config SND_HDA_CODEC_VIA Say Y or M here to include VIA HD-audio codec support in snd-hda-intel driver, such as VT1708. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_VIA=m + config SND_HDA_CODEC_HDMI tristate "Build HDMI/DisplayPort HD-audio codec support" help @@ -121,6 +133,9 @@ config SND_HDA_CODEC_HDMI support in snd-hda-intel driver. This includes all AMD/ATI, Intel and Nvidia HDMI/DisplayPort codecs. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_HDMI=m + config SND_HDA_I915 bool default y @@ -133,6 +148,9 @@ config SND_HDA_CODEC_CIRRUS Say Y or M here to include Cirrus Logic codec support in snd-hda-intel driver, such as CS4206. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CIRRUS=m + config SND_HDA_CODEC_CONEXANT tristate "Build Conexant HD-audio codec support" select SND_HDA_GENERIC @@ -140,6 +158,9 @@ config SND_HDA_CODEC_CONEXANT Say Y or M here to include Conexant HD-audio codec support in snd-hda-intel driver, such as CX20549. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CONEXANT=m + config SND_HDA_CODEC_CA0110 tristate "Build Creative CA0110-IBG codec support" select SND_HDA_GENERIC @@ -147,12 +168,18 @@ config SND_HDA_CODEC_CA0110 Say Y or M here to include Creative CA0110-IBG codec support in snd-hda-intel driver, found on some Creative X-Fi cards. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CA0110=m + config SND_HDA_CODEC_CA0132 tristate "Build Creative CA0132 codec support" help Say Y or M here to include Creative CA0132 codec support in snd-hda-intel driver. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CA0132=m + config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 @@ -172,18 +199,27 @@ config SND_HDA_CODEC_CMEDIA Say Y or M here to include C-Media HD-audio codec support in snd-hda-intel driver, such as CMI9880. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CMEDIA=m + config SND_HDA_CODEC_SI3054 tristate "Build Silicon Labs 3054 HD-modem codec support" help Say Y or M here to include Silicon Labs 3054 HD-modem codec (and compatibles) support in snd-hda-intel driver. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_CODEC_SI3054=m + config SND_HDA_GENERIC tristate "Enable generic HD-audio codec parser" help Say Y or M here to enable the generic HD-audio codec parser in snd-hda-intel driver. +comment "Set to Y if you want auto-loading the codec driver" + depends on SND_HDA_INTEL=y && SND_HDA_GENERIC=m + config SND_HDA_POWER_SAVE_DEFAULT int "Default time-out for HD-audio power-save mode" depends on PM -- 1.8.5

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[alsa-devel] [PATCH] ALSA: usb-audio: Add a quirk for Plantronics Gamecom 780
by Takashi Iwai 19 Dec '13

19 Dec '13

Plantronics Gamecom 780 headset has a firmware problem, and when the FU 0x09 volume is changed, it results in either too loud or silence except for a very narrow range. This patch provides a workaround, ignoring the node, initialize the volume in a sane value and keep untouched. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65251 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 10 ++++++++++ sound/usb/quirks.c | 14 ++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index cc2dd1f0decb..32af6b741ef5 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -322,6 +322,12 @@ static struct usbmix_name_map hercules_usb51_map[] = { { 0 } /* terminator */ }; +/* Plantronics Gamecom 780 has a broken volume control, better to disable it */ +static struct usbmix_name_map gamecom780_map[] = { + { 9, NULL }, /* FU, speaker out */ + {} +}; + /* * Control map entries */ @@ -358,6 +364,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x046d, 0x09a4), .ignore_ctl_error = 1, }, + { /* Plantronics GameCom 780 */ + .id = USB_ID(0x047f, 0xc010), + .map = gamecom780_map, + }, { /* Hercules DJ Console (Windows Edition) */ .id = USB_ID(0x06f8, 0xb000), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede99dfd..ede753685491 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -660,6 +660,18 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) return err; } +/* quirk for Plantronics GameCom 780 with CM6302 chip */ +static int snd_usb_gamecon780_boot_quirk(struct usb_device *dev) +{ + /* set the initial volume and don't change; other values are either + * too loud or silent due to firmware bug (bko#65251) + */ + u8 buf[2] = { 0x74, 0xdc }; + return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + UAC_FU_VOLUME << 8, 9 << 8, buf, 2); +} + /* * Novation Twitch DJ controller */ @@ -986,6 +998,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return snd_usb_nativeinstruments_boot_quirk(dev); case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ return snd_usb_fasttrackpro_boot_quirk(dev); + case USB_ID(0x047f, 0xc010): /* Plantronics Gamecom 780 */ + return snd_usb_gamecon780_boot_quirk(dev); } return 0; -- 1.8.5

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[alsa-devel] [PATCH] ALSA: usb-audio: Add support for Focusrite Saffire 6 USB
by Eduard Gilmutdinov 19 Dec '13

19 Dec '13

This patch add support for playback through external audio card Focusrite Saffire 6 USB. Capture and MIDI is not tested. Signed-off-by: Eduard Gilmutdinov <edgilmutdinov(a)gmail.com> --- diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f5f0595..158beb4 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2569,6 +2569,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede..3ea9627 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -660,10 +660,11 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) return err; } -/* +/* * Novation Twitch DJ controller + * Focusrite(Novation) Saffire 6 USB audio card */ -static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +static int snd_usb_novation_boot_quirk(struct usb_device *dev) { /* preemptively set up the device because otherwise the * raw MIDI endpoints are not active */ @@ -972,9 +973,9 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); - case USB_ID(0x1235, 0x0018): - /* Focusrite Novation Twitch */ - return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ + case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ + return snd_usb_novation_boot_quirk(dev); case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */

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[alsa-devel] ASoC fixes for v3.13
by Mark Brown 19 Dec '13

19 Dec '13

The following changes since commit 319e2e3f63c348a9b66db4667efa73178e18b17d: Linux 3.13-rc4 (2013-12-15 12:31:33 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc4 for you to fetch changes up to 96b7fe0119b932ad25451d2b6357e727bbe6a309: Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/tegra', 'asoc/fix/wm8904' and 'asoc/fix/wm8962' into asoc-linus (2013-12-19 10:25:27 +0000) The diffstat is larger than the description since the pull request is against Linus' tree due the scheduling of your pull request last week compared to mine to you. ---------------------------------------------------------------- ASoC: Fixes for v3.13 The fixes here are all driver specific ones, none of which particularly stand out but all of which are useful to users of those drivers. ---------------------------------------------------------------- Bo Shen (3): ASoC: atmel_ssc_dai: add dai trigger ops ASoC: sam9x5_wm8731: change to work in DSP A mode ASoC: wm8904: fix DSP mode B configuration Charles Keepax (2): ASoC: wm5110: Correct HPOUT3 DAPM route typo ASoC: wm_adsp: Add small delay while polling DSP RAM start Jean-Francois Moine (1): ASoC: kirkwood: Fix the CPU DAI rates Mark Brown (3): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dma' into asoc-linus Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/tegra', 'asoc/fix/wm8904' and 'asoc/fix/wm8962' into asoc-linus Nicolin Chen (3): ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free() ASoC: wm8962: Enable SYSCLK provisonally before fetching generated DSPCLK_DIV ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver Stephen Warren (2): ASoC: tegra: fix uninitialized variables in set_fmt ASoC: don't leak on error in snd_dmaengine_pcm_register sound/soc/atmel/atmel_ssc_dai.c | 30 ++++++++++++++++++++++++++- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8962.c | 13 ++++++++++++ sound/soc/codecs/wm_adsp.c | 10 ++++++--- sound/soc/fsl/imx-wm8962.c | 2 -- sound/soc/kirkwood/kirkwood-i2s.c | 24 +++++++++++----------- sound/soc/soc-generic-dmaengine-pcm.c | 38 +++++++++++++++++++++++++---------- sound/soc/soc-pcm.c | 5 +++-- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 10 ++++----- sound/soc/tegra/tegra30_i2s.c | 6 +++--- 13 files changed, 105 insertions(+), 45 deletions(-)

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[alsa-devel] [PATCH 1/2] ASoC: wm5110: Add FSH for ISRCs
by Charles Keepax 19 Dec '13

19 Dec '13

Currently, the driver only supports configuration of the lower sample rate (FSL) on the ISRCs. With the higher rate being fixed a SYSCLK, this patch adds support for configuring the higher sample rate (FSH). Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 16 ++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 3 +++ 3 files changed, 20 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6bfd803..56d3ff5 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -498,6 +498,22 @@ const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { EXPORT_SYMBOL_GPL(arizona_rate_val); +const struct soc_enum arizona_isrc_fsh[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_1, + ARIZONA_ISRC1_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_2_CTRL_1, + ARIZONA_ISRC2_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_3_CTRL_1, + ARIZONA_ISRC3_FSH_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; +EXPORT_SYMBOL_GPL(arizona_isrc_fsh); + const struct soc_enum arizona_isrc_fsl[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_2, ARIZONA_ISRC1_FSL_SHIFT, 0xf, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 6641f3d..99a97c0 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -186,6 +186,7 @@ extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; extern const struct soc_enum arizona_isrc_fsl[]; +extern const struct soc_enum arizona_isrc_fsh[]; extern const struct soc_enum arizona_in_vi_ramp; extern const struct soc_enum arizona_in_vd_ramp; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 7fb8599..18c7e1f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -280,6 +280,9 @@ SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]), +SOC_VALUE_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]), +SOC_VALUE_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]), +SOC_VALUE_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]), ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: simple-card: Use devm_snd_soc_register_card()
by Xiubo Li 19 Dec '13

19 Dec '13

Makes the code slightly shorter. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7a9b6b4..3d190d0 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -234,14 +234,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.num_links = 1; cinfo->snd_card.dev = &pdev->dev; - return snd_soc_register_card(&cinfo->snd_card); -} - -static int asoc_simple_card_remove(struct platform_device *pdev) -{ - struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; - - return snd_soc_unregister_card(&cinfo->snd_card); + return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { @@ -257,7 +250,6 @@ static struct platform_driver asoc_simple_card = { .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, - .remove = asoc_simple_card_remove, }; module_platform_driver(asoc_simple_card); -- 1.8.4

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[alsa-devel] [PATCH 1/2] mfd: wm5110: Expose DRE control registers
by Charles Keepax 19 Dec '13

19 Dec '13

Certain use-cases require the DRE to be disabled so expose registers necessary to control the DRE enables. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 2 ++ include/linux/mfd/arizona/registers.h | 4 ++++ 2 files changed, 6 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 338cfbe..abd6713 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -601,6 +601,7 @@ static const struct reg_default wm5110_reg_default[] = { { 0x0000043D, 0x0180 }, /* R1085 - DAC Digital Volume 6R */ { 0x0000043E, 0x0080 }, /* R1086 - DAC Volume Limit 6R */ { 0x0000043F, 0x0800 }, /* R1087 - Noise Gate Select 6R */ + { 0x00000440, 0x8FFF }, /* R1088 - DRE Enable */ { 0x00000450, 0x0000 }, /* R1104 - DAC AEC Control 1 */ { 0x00000458, 0x0000 }, /* R1112 - Noise Gate Control */ { 0x00000480, 0x0040 }, /* R1152 - Class W ANC Threshold 1 */ @@ -1631,6 +1632,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_DAC_DIGITAL_VOLUME_6R: case ARIZONA_DAC_VOLUME_LIMIT_6R: case ARIZONA_NOISE_GATE_SELECT_6R: + case ARIZONA_DRE_ENABLE: case ARIZONA_DAC_AEC_CONTROL_1: case ARIZONA_NOISE_GATE_CONTROL: case ARIZONA_PDM_SPK1_CTRL_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 8987814..22916c0 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -3207,6 +3207,10 @@ /* * R1088 (0x440) - DRE Enable */ +#define ARIZONA_DRE3R_ENA 0x0020 /* DRE3R_ENA */ +#define ARIZONA_DRE3R_ENA_MASK 0x0020 /* DRE3R_ENA */ +#define ARIZONA_DRE3R_ENA_SHIFT 5 /* DRE3R_ENA */ +#define ARIZONA_DRE3R_ENA_WIDTH 1 /* DRE3R_ENA */ #define ARIZONA_DRE3L_ENA 0x0010 /* DRE3L_ENA */ #define ARIZONA_DRE3L_ENA_MASK 0x0010 /* DRE3L_ENA */ #define ARIZONA_DRE3L_ENA_SHIFT 4 /* DRE3L_ENA */ -- 1.7.2.5

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[alsa-devel] [PATCH V2 1/2] ASoC: SPEAr: get rid of spear-pcm-audio struct device
by Stephen Warren 18 Dec '13

18 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Modify the SPEAr PCM driver so that it's a utility library that can be registered on each DAI, rather than a separate struct device. This is more in line with how many recent DT-converted platforms operate, and avoids the need for yet another struct device. This is also required as a pre-cursor to removing spear_pcm_request_chan(). Cc: Rajeev Kumar <rajeev-dlh.kumar(a)st.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Rebased on a merge of topic/spear and topic/dma --- sound/soc/spear/spdif_in.c | 9 +++++++-- sound/soc/spear/spdif_out.c | 10 ++++++++-- sound/soc/spear/spear_pcm.c | 18 ++++-------------- sound/soc/spear/spear_pcm.h | 22 ++++++++++++++++++++++ 4 files changed, 41 insertions(+), 18 deletions(-) create mode 100644 sound/soc/spear/spear_pcm.h diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 21a8c954af1c..4627110f3441 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -24,6 +24,7 @@ #include <sound/spear_dma.h> #include <sound/spear_spdif.h> #include "spdif_in_regs.h" +#include "spear_pcm.h" struct spdif_in_params { u32 format; @@ -257,8 +258,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, - &spdif_in_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); + if (ret) + return ret; + + return devm_spear_pcm_platform_register(&pdev->dev); } static struct platform_driver spdif_in_driver = { diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index b6ef6f78dc78..731a1e0cfeaa 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -22,6 +22,7 @@ #include <sound/spear_dma.h> #include <sound/spear_spdif.h> #include "spdif_out_regs.h" +#include "spear_pcm.h" struct spdif_out_params { u32 rate; @@ -280,6 +281,7 @@ static int spdif_out_probe(struct platform_device *pdev) struct spdif_out_dev *host; struct spear_spdif_platform_data *pdata; struct resource *res; + int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { @@ -306,8 +308,12 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, - &spdif_out_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); + if (ret) + return ret; + + return devm_spear_pcm_platform_register(&pdev->dev); } #ifdef CONFIG_PM diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9a02141666ea..f288724961da 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -18,6 +18,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/spear_dma.h> +#include "spear_pcm.h" static const struct snd_pcm_hardware spear_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -47,26 +48,15 @@ static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = { .prealloc_buffer_size = 16 * 1024, }; -static int spear_soc_platform_probe(struct platform_device *pdev) +int devm_spear_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(dev, &spear_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } - -static struct platform_driver spear_pcm_driver = { - .driver = { - .name = "spear-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = spear_soc_platform_probe, -}; - -module_platform_driver(spear_pcm_driver); +EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register); MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar(a)st.com>"); MODULE_DESCRIPTION("SPEAr PCM DMA module"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:spear-pcm-audio"); diff --git a/sound/soc/spear/spear_pcm.h b/sound/soc/spear/spear_pcm.h new file mode 100644 index 000000000000..631e2aa1fb33 --- /dev/null +++ b/sound/soc/spear/spear_pcm.h @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#ifndef __SPEAR_PCM_H__ +#define __SPEAR_PCM_H__ + +int devm_spear_pcm_platform_register(struct device *dev); + +#endif -- 1.8.1.5

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[alsa-devel] [PATCH V2 1/2] ASoC: ep93xx: get rid of ep93xx-pcm-audio struct device
by Stephen Warren 18 Dec '13

18 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Modify the ep93xx PCM driver so that it's a utility library that can be registered on each DAI, rather than a separate struct device. This is more in line with how many recent DT-converted platforms operate, and avoids the need for yet another struct device. Cc: H Hartley Sweeten <hsweeten(a)visionengravers.com> Cc: Ryan Mallon <rmallon(a)gmail.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Rebased on a merge of topic/ep93xx and topic/dma --- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 8 ++++++++ sound/soc/cirrus/ep93xx-i2s.c | 8 ++++++++ sound/soc/cirrus/ep93xx-pcm.c | 19 +++++-------------- sound/soc/cirrus/ep93xx-pcm.h | 22 ++++++++++++++++++++++ sound/soc/cirrus/simone.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- 7 files changed, 46 insertions(+), 17 deletions(-) create mode 100644 sound/soc/cirrus/ep93xx-pcm.h diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index c43fb214558a..4f900efc437c 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -63,7 +63,7 @@ static struct snd_soc_ops edb93xx_ops = { static struct snd_soc_dai_link edb93xx_dai = { .name = "CS4271", .stream_name = "CS4271 HiFi", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-i2s", .cpu_dai_name = "ep93xx-i2s", .codec_name = "spi0.0", .codec_dai_name = "cs4271-hifi", diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index efa75b5086a4..cc5583da5380 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -24,6 +24,8 @@ #include <linux/platform_data/dma-ep93xx.h> +#include "ep93xx-pcm.h" + /* * Per channel (1-4) registers. */ @@ -394,8 +396,14 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) if (ret) goto fail; + ret = devm_ep93xx_pcm_platform_register(&pdev->dev); + if (ret) + goto fail_unregister; + return 0; +fail_unregister: + snd_soc_unregister_component(&pdev->dev); fail: ep93xx_ac97_info = NULL; snd_soc_set_ac97_ops(NULL); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index a57643d6402f..167728a73da2 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -30,6 +30,8 @@ #include <mach/ep93xx-regs.h> #include <linux/platform_data/dma-ep93xx.h> +#include "ep93xx-pcm.h" + #define EP93XX_I2S_TXCLKCFG 0x00 #define EP93XX_I2S_RXCLKCFG 0x04 #define EP93XX_I2S_GLCTRL 0x0C @@ -405,8 +407,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) if (err) goto fail_put_lrclk; + err = devm_ep93xx_pcm_platform_register(&pdev->dev); + if (err) + goto fail_unregister; + return 0; +fail_unregister: + snd_soc_unregister_component(&pdev->dev); fail_put_lrclk: clk_put(info->lrclk); fail_put_sclk: diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index fdb8b8feb4ed..198c5405f2ce 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -23,6 +23,8 @@ #include <linux/platform_data/dma-ep93xx.h> +#include "ep93xx-pcm.h" + static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -76,27 +78,16 @@ static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { .prealloc_buffer_size = 131072, }; -static int ep93xx_soc_platform_probe(struct platform_device *pdev) +int devm_ep93xx_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(&pdev->dev, + return devm_snd_dmaengine_pcm_register(dev, &ep93xx_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } - -static struct platform_driver ep93xx_pcm_driver = { - .driver = { - .name = "ep93xx-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = ep93xx_soc_platform_probe, -}; - -module_platform_driver(ep93xx_pcm_driver); +EXPORT_SYMBOL_GPL(devm_ep93xx_pcm_platform_register); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:ep93xx-pcm-audio"); diff --git a/sound/soc/cirrus/ep93xx-pcm.h b/sound/soc/cirrus/ep93xx-pcm.h new file mode 100644 index 000000000000..b7a12a2fae9c --- /dev/null +++ b/sound/soc/cirrus/ep93xx-pcm.h @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#ifndef __EP93XX_PCM_H__ +#define __EP93XX_PCM_H__ + +int devm_ep93xx_pcm_platform_register(struct device *dev); + +#endif diff --git a/sound/soc/cirrus/simone.c b/sound/soc/cirrus/simone.c index 4d094d00c34a..822a19a89e74 100644 --- a/sound/soc/cirrus/simone.c +++ b/sound/soc/cirrus/simone.c @@ -27,7 +27,7 @@ static struct snd_soc_dai_link simone_dai = { .cpu_dai_name = "ep93xx-ac97", .codec_dai_name = "ac97-hifi", .codec_name = "ac97-codec", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-ac97", }; static struct snd_soc_card snd_soc_simone = { diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 69041074f2c1..29238a7476dd 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link snappercl15_dai = { .cpu_dai_name = "ep93xx-i2s", .codec_dai_name = "tlv320aic23-hifi", .codec_name = "tlv320aic23-codec.0-001a", - .platform_name = "ep93xx-pcm-audio", + .platform_name = "ep93xx-i2s", .init = snappercl15_tlv320aic23_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS, -- 1.8.1.5

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[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: read chip revision for once
by Shawn Guo 18 Dec '13

18 Dec '13

Store chip revision in struct sgtl5000_priv when sgtl5000_i2c_probe() reads it out from register, so that we can use it in sgtl5000_enable_regulators() with no need to read register again. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/codecs/sgtl5000.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f4093f..bd291d2 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -115,6 +115,7 @@ struct sgtl5000_priv { struct ldo_regulator *ldo; struct regmap *regmap; struct clk *mclk; + int revision; }; /* @@ -1300,9 +1301,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - int reg; int ret; - int rev; int i; int external_vddd = 0; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); @@ -1332,14 +1331,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) * workaround for revision 0x11 and later, * roll back to use internal LDO */ - - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) - goto err_regulator_disable; - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - - if (external_vddd && rev >= 0x11) { + if (external_vddd && sgtl5000->revision >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1362,9 +1354,6 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return 0; -err_regulator_disable: - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); err_regulator_free: regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1566,6 +1555,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); + sgtl5000->revision = rev; i2c_set_clientdata(client, sgtl5000); -- 1.7.9.5

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