June 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] sound: codec: tlv320aic3x: remove mono controls
by Jan Weitzel 17 Jun '11

17 Jun '11

if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite registers for class-d amplifier. classd amplifier controls are only used for AIC3X_MODEL_3007. Removing all mono snd_kcontrol_new, snd_soc_dapm_widget, snd_soc_dapm_route and aic3x_init stuff from common code and call only for not AIC3X_MODEL_3007 codecs. Testet only with AIC3X_MODEL_3007 Signed-off-by: Jan Weitzel <j.weitzel(a)phytec.de> --- sound/soc/codecs/tlv320aic3x.c | 106 ++++++++++++++++++++++++--------------- 1 files changed, 65 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c3d96fc..7146832 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -324,16 +324,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", - LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", - PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", - DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), @@ -357,7 +347,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output pin mute controls */ SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, 0x01, 0), - SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, @@ -377,6 +366,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; +static const struct snd_kcontrol_new aic3x_mono_controls[] = { + SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", + LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", + PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", + DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), +}; + /* * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps */ @@ -526,9 +528,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0), - /* Mono Output */ - SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), - /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, @@ -590,9 +589,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, &aic3x_right_line_mixer_controls[0], ARRAY_SIZE(aic3x_right_line_mixer_controls)), - SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_mono_mixer_controls[0], - ARRAY_SIZE(aic3x_mono_mixer_controls)), SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_hp_mixer_controls[0], ARRAY_SIZE(aic3x_left_hp_mixer_controls)), @@ -608,7 +604,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LLOUT"), SND_SOC_DAPM_OUTPUT("RLOUT"), - SND_SOC_DAPM_OUTPUT("MONO_LOUT"), SND_SOC_DAPM_OUTPUT("HPLOUT"), SND_SOC_DAPM_OUTPUT("HPROUT"), SND_SOC_DAPM_OUTPUT("HPLCOM"), @@ -630,6 +625,18 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Detection"), }; +static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = { + /* Mono Output */ + SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_mono_mixer_controls[0], + ARRAY_SIZE(aic3x_mono_mixer_controls)), + + SND_SOC_DAPM_OUTPUT("MONO_LOUT"), +}; + static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = { /* Class-D outputs */ SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0), @@ -714,17 +721,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line Out", NULL, "Right DAC Mux"}, {"RLOUT", NULL, "Right Line Out"}, - /* Mono Output */ - {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, - {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, - {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, - {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, - {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, - - {"Mono Out", NULL, "Mono Mixer"}, - {"MONO_LOUT", NULL, "Mono Out"}, - /* Left HP Output */ {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, @@ -780,6 +776,19 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPRCOM", NULL, "Right HP Com"}, }; +static const struct snd_soc_dapm_route intercon_mono[] = { + /* Mono Output */ + {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Mono Out", NULL, "Mono Mixer"}, + {"MONO_LOUT", NULL, "Mono Out"}, +}; + static const struct snd_soc_dapm_route intercon_3007[] = { /* Class-D outputs */ {"Left Class-D Out", NULL, "Left Line Out"}, @@ -804,6 +813,11 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) ARRAY_SIZE(aic3007_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_3007, ARRAY_SIZE(intercon_3007)); + } else { + snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets, + ARRAY_SIZE(aic3x_dapm_mono_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_mono, + ARRAY_SIZE(intercon_mono)); } return 0; @@ -1290,17 +1304,12 @@ static int aic3x_init(struct snd_soc_codec *codec) /* DAC to Line Out default volume and route to Output mixer */ snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - /* DAC to Mono Line Out default volume and route to Output mixer */ - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ reg = snd_soc_read(codec, LLOPM_CTRL); snd_soc_write(codec, LLOPM_CTRL, reg | UNMUTE); reg = snd_soc_read(codec, RLOPM_CTRL); snd_soc_write(codec, RLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, MONOLOPM_CTRL); - snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); reg = snd_soc_read(codec, HPLOUT_CTRL); snd_soc_write(codec, HPLOUT_CTRL, reg | UNMUTE); reg = snd_soc_read(codec, HPROUT_CTRL); @@ -1325,9 +1334,6 @@ static int aic3x_init(struct snd_soc_codec *codec) /* PGA to Line Out default volume, disconnect from Output Mixer */ snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); @@ -1337,14 +1343,30 @@ static int aic3x_init(struct snd_soc_codec *codec) /* 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); - /* Line2 to Mono Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); if (aic3x->model == AIC3X_MODEL_3007) { aic3x_init_3007(codec); snd_soc_write(codec, CLASSD_CTRL, 0); + + return 0; } + /* Not on AIC3X_MODEL_3007 init */ + + /* DAC to Mono Line Out default volume and route to Output mixer */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + + /* unmute outputs */ + reg = snd_soc_read(codec, MONOLOPM_CTRL); + snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); + + /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); + + /* Line2 to Mono Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); return 0; } @@ -1423,7 +1445,9 @@ static int aic3x_probe(struct snd_soc_codec *codec) ARRAY_SIZE(aic3x_snd_controls)); if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); - + else + snd_soc_add_controls(codec, aic3x_mono_controls, + ARRAY_SIZE(aic3x_mono_controls)); aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); -- 1.7.0.4

3 4

[alsa-devel] [PATCH v5 00/18] MFD/ASoC/Input: TWL4030/TWL60X0 changes
by Peter Ujfalusi 17 Jun '11

17 Jun '11

Hello, Changes since v4: - Comments from Dmitry Torokhov addressed for the twl6030-vibra driver. Intro message from the v4 series: The series will do five major things, and they are in one series, because they pretty much depending on each other, so it is easier to handle them together. 1. Move the common pmic related configurations out from the board files. New twl-common.c/h file has been created for this purpose. OMAP3 board files received two patches (pdata, and regualtor changes), OMAP4 board files has been changed with one patch. 2. Rename the twl4030-codec MFD driver to twl4030-audio Since the ASoC multicomponent introduction (2.6.37) the twl4030-codec MFD dirver had been using twl4030-audio to register (twl4030-codec is the ASoC codec driver). The documentation refers to this part of twl as audio block, so it is better to rename the driver as well to avoid confusion. 3. Introduction of MFD driver for twl6040. The TWL6040 audio IC has codec and vibra functionality. Convert the existing ASoC codec driver to use the new MFD driver's interface. 4. Vibrator driver for TWL6040 (Input/ForceFeedback driver) The driver itself, and support for the vibrators in SDP4430 board. 5. Updates for twl6040 ASoC codec driver HS ramp configuration, new ABE dais, updated sample rate constraints, SDP4430 configuration. Regards, Peter --- Axel Castaneda Gonzalez (1): ASoC: twl6040: Configure ramp step based on platform Liam Girdwood (3): ASoC: twl6040: add all ABE DAIs ASoC: twl6040: Support other sample rates in constraints. ASoC: twl6040: set default constraints. Misael Lopez Cruz (5): mfd: twl6040: Add initial support ASoC: twl6040: Convert into TWL6040 MFD child input: Add initial support for TWL6040 vibrator ASoC: twl6040: Remove pll and headset mode dependency OMAP4: SDP4430: Add twl6040 codec platform support Peter Ujfalusi (9): OMAP: New twl-common for common TWL configuration OMAP4: Move common twl6030 configuration to twl-common OMAP3: Move common twl configuration to twl-common OMAP3: Move common regulator configuration to twl-common MFD: twl4030-codec: Rename internals from codec to audio MFD: twl4030-codec -> twl4030-audio: Rename the driver MFD: twl4030-audio: Rename platform data MFD: twl6040: Change platform data for soc codec driver OMAP4: SDP4430: Add twl6040 vibrator platform support arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/board-3430sdp.c | 89 +--- arch/arm/mach-omap2/board-4430sdp.c | 187 ++---- arch/arm/mach-omap2/board-cm-t35.c | 48 +-- arch/arm/mach-omap2/board-devkit8000.c | 37 +- arch/arm/mach-omap2/board-igep0020.c | 44 +-- arch/arm/mach-omap2/board-ldp.c | 15 +- arch/arm/mach-omap2/board-omap3beagle.c | 60 +-- arch/arm/mach-omap2/board-omap3evm.c | 67 +-- arch/arm/mach-omap2/board-omap3pandora.c | 59 +-- arch/arm/mach-omap2/board-omap3stalker.c | 70 +-- arch/arm/mach-omap2/board-omap3touchbook.c | 57 +-- arch/arm/mach-omap2/board-omap4panda.c | 149 +----- arch/arm/mach-omap2/board-overo.c | 59 +-- arch/arm/mach-omap2/board-rm680.c | 8 +- arch/arm/mach-omap2/board-rx51-peripherals.c | 44 +-- arch/arm/mach-omap2/board-zoom-peripherals.c | 88 +--- arch/arm/mach-omap2/common-board-devices.c | 21 - arch/arm/mach-omap2/common-board-devices.h | 26 +- arch/arm/mach-omap2/twl-common.c | 304 ++++++++++ arch/arm/mach-omap2/twl-common.h | 59 ++ arch/arm/plat-omap/include/plat/irqs.h | 12 +- drivers/input/misc/Kconfig | 13 +- drivers/input/misc/Makefile | 1 + drivers/input/misc/twl4030-vibra.c | 12 +- drivers/input/misc/twl6040-vibra.c | 425 +++++++++++++ drivers/mfd/Kconfig | 8 +- drivers/mfd/Makefile | 3 +- drivers/mfd/twl-core.c | 13 +- drivers/mfd/twl4030-audio.c | 277 +++++++++ drivers/mfd/twl4030-codec.c | 277 --------- drivers/mfd/twl6040-core.c | 601 +++++++++++++++++++ drivers/mfd/twl6040-irq.c | 205 +++++++ include/linux/i2c/twl.h | 25 +- .../linux/mfd/{twl4030-codec.h => twl4030-audio.h} | 16 +- include/linux/mfd/twl6040.h | 261 ++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl4030.c | 22 +- sound/soc/codecs/twl6040.c | 630 ++++++++------------ sound/soc/codecs/twl6040.h | 118 ---- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/sdp4430.c | 2 + sound/soc/omap/zoom2.c | 2 +- 43 files changed, 2648 insertions(+), 1773 deletions(-) create mode 100644 arch/arm/mach-omap2/twl-common.c create mode 100644 arch/arm/mach-omap2/twl-common.h create mode 100644 drivers/input/misc/twl6040-vibra.c create mode 100644 drivers/mfd/twl4030-audio.c delete mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 drivers/mfd/twl6040-core.c create mode 100644 drivers/mfd/twl6040-irq.c rename include/linux/mfd/{twl4030-codec.h => twl4030-audio.h} (96%) create mode 100644 include/linux/mfd/twl6040.h -- 1.7.5.3

2 25

[alsa-devel] Can I get audio via HDMI with radeon mobility HD 5400?
by Tom Horsley 17 Jun '11

17 Jun '11

If I install the binary ATI catalyst driver, I can get audio out the HDMI connection, and alsa-info says it is using the snd-hda-intel driver. If I use the open source radeon driver, I cannot get audio out the HDMI connection, yet alsa-info says it is using the same snd-hda-intel driver. Does anyone know how to persuade snd-hda-intel to work with the open source radeon driver? (Because the video quality is about 1000% better with the open source driver). With binary catalyst driver, I see this from alsa-info: http://home.comcast.net/~tomhorsley/hardware/zino/catalyst-alsa-info.txt With open source radeon driver, I see this: http://home.comcast.net/~tomhorsley/hardware/zino/radeon-alsa-info.txt

1 0

[alsa-devel] unsigned int 'return -1' in /sound/soc/codecs/tlv320aic23.c
by Lambrecht Jürgen 17 Jun '11

17 Jun '11

Hello, I see this error in /sound/soc/codecs/tlv320aic23.c: static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec ^^^^^^^^ *codec, unsigned int reg) { u16 *cache = codec->reg_cache; if (reg >= ARRAY_SIZE(tlv320aic23_reg)) return -1; ^^ return cache[reg]; } Maybe better return 0? Best read the codec datasheet to find a safe or non-existing value? And also a "printk(KERN_ERR " then? Actually, better just remove the check, because if you want to write a codec register, you must know what you are doing? I am busy with the codec driver for mc13783. Currently, the mc13783 (from Sascha Hauer) code does not check 'reg'. (I want to supply a patch finally, but as newbie: from which GIT?, because mine is a merge of 2 branches of the Pengutronix GIT) Regards, Jürgen -- Jürgen Lambrecht R&D Associate Tel: +32 (0)51 303045 Fax: +32 (0)51 310670 http://www.televic-rail.com Televic Rail NV - Leo Bekaertlaan 1 - 8870 Izegem - Belgium Company number 0825.539.581 - RPR Kortrijk

1 0

[alsa-devel] [PATCH v4 00/18] MFD/ASoC/Input: TWL4030/TWL60X0 changes
by Peter Ujfalusi 17 Jun '11

17 Jun '11

Hello, Changes since v3: - patches generated with git format-patch -M to handle renames better - Included depending updates for twl6040 soc codec driver: - HS ramp configuration, new ABE dais, updated sample rate constraints - Configure the twl6040 codec on SDP4430 board Updated intro message from the v3 series: The series will do five major things, and they are in one series, because they pretty much depending on each other, so it is easier to handle them together. 1. Move the common pmic related configurations out from the board files. New twl-common.c/h file has been created for this purpose. OMAP3 board files received two patches (pdata, and regualtor changes), OMAP4 board files has been changed with one patch. 2. Rename the twl4030-codec MFD driver to twl4030-audio Since the ASoC multicomponent introduction (2.6.37) the twl4030-codec MFD dirver had been using twl4030-audio to register (twl4030-codec is the ASoC codec driver). The documentation refers to this part of twl as audio block, so it is better to rename the driver as well to avoid confusion. 3. Introduction of MFD driver for twl6040. The TWL6040 audio IC has codec and vibra functionality. Convert the existing ASoC codec driver to use the new MFD driver's interface. 4. Vibrator driver for TWL6040 (Input/ForceFeedback driver) The driver itself, and support for the vibrators in SDP4430 board. 5. Updates for twl6040 ASoC codec driver HS ramp configuration, new ABE dais, updated sample rate constraints, SDP4430 configuration. Regards, Peter --- Axel Castaneda Gonzalez (1): ASoC: twl6040: Configure ramp step based on platform Liam Girdwood (3): ASoC: twl6040: add all ABE DAIs ASoC: twl6040: Support other sample rates in constraints. ASoC: twl6040: set default constraints. Misael Lopez Cruz (5): mfd: twl6040: Add initial support ASoC: twl6040: Convert into TWL6040 MFD child input: Add initial support for TWL6040 vibrator ASoC: twl6040: Remove pll and headset mode dependency OMAP4: SDP4430: Add twl6040 codec platform support Peter Ujfalusi (9): OMAP: New twl-common for common TWL configuration OMAP4: Move common twl6030 configuration to twl-common OMAP3: Move common twl configuration to twl-common OMAP3: Move common regulator configuration to twl-common MFD: twl4030-codec: Rename internals from codec to audio MFD: twl4030-codec -> twl4030-audio: Rename the driver MFD: twl4030-audio: Rename platform data MFD: twl6040: Change platform data for soc codec driver OMAP4: SDP4430: Add twl6040 vibrator platform support arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/board-3430sdp.c | 89 +--- arch/arm/mach-omap2/board-4430sdp.c | 187 ++---- arch/arm/mach-omap2/board-cm-t35.c | 48 +-- arch/arm/mach-omap2/board-devkit8000.c | 37 +- arch/arm/mach-omap2/board-igep0020.c | 44 +-- arch/arm/mach-omap2/board-ldp.c | 15 +- arch/arm/mach-omap2/board-omap3beagle.c | 60 +-- arch/arm/mach-omap2/board-omap3evm.c | 67 +-- arch/arm/mach-omap2/board-omap3pandora.c | 59 +-- arch/arm/mach-omap2/board-omap3stalker.c | 70 +-- arch/arm/mach-omap2/board-omap3touchbook.c | 57 +-- arch/arm/mach-omap2/board-omap4panda.c | 149 +----- arch/arm/mach-omap2/board-overo.c | 59 +-- arch/arm/mach-omap2/board-rm680.c | 8 +- arch/arm/mach-omap2/board-rx51-peripherals.c | 44 +-- arch/arm/mach-omap2/board-zoom-peripherals.c | 88 +--- arch/arm/mach-omap2/common-board-devices.c | 21 - arch/arm/mach-omap2/common-board-devices.h | 26 +- arch/arm/mach-omap2/twl-common.c | 304 ++++++++++ arch/arm/mach-omap2/twl-common.h | 59 ++ arch/arm/plat-omap/include/plat/irqs.h | 12 +- drivers/input/misc/Kconfig | 13 +- drivers/input/misc/Makefile | 1 + drivers/input/misc/twl4030-vibra.c | 12 +- drivers/input/misc/twl6040-vibra.c | 428 +++++++++++++ drivers/mfd/Kconfig | 8 +- drivers/mfd/Makefile | 3 +- drivers/mfd/twl-core.c | 13 +- drivers/mfd/twl4030-audio.c | 277 +++++++++ drivers/mfd/twl4030-codec.c | 277 --------- drivers/mfd/twl6040-core.c | 601 +++++++++++++++++++ drivers/mfd/twl6040-irq.c | 205 +++++++ include/linux/i2c/twl.h | 25 +- .../linux/mfd/{twl4030-codec.h => twl4030-audio.h} | 16 +- include/linux/mfd/twl6040.h | 261 ++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl4030.c | 22 +- sound/soc/codecs/twl6040.c | 630 ++++++++------------ sound/soc/codecs/twl6040.h | 118 ---- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/sdp4430.c | 2 + sound/soc/omap/zoom2.c | 2 +- 43 files changed, 2651 insertions(+), 1773 deletions(-) create mode 100644 arch/arm/mach-omap2/twl-common.c create mode 100644 arch/arm/mach-omap2/twl-common.h create mode 100644 drivers/input/misc/twl6040-vibra.c create mode 100644 drivers/mfd/twl4030-audio.c delete mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 drivers/mfd/twl6040-core.c create mode 100644 drivers/mfd/twl6040-irq.c rename include/linux/mfd/{twl4030-codec.h => twl4030-audio.h} (96%) create mode 100644 include/linux/mfd/twl6040.h -- 1.7.5.3

4 32

[alsa-devel] WM8731 set sample rate
by MIroslav Dušek 17 Jun '11

17 Jun '11

Hi all, I am using processor S3C2416 as master and WM8731 as slave and I am trying to set right sample rate of ADC DAC data between them. I set I2S clock to EPLL output (96MHz) , and I mainly need calculate 8KHz sample rate. In user manual of S3C2416 is poor explanation of clocking I2S bus. What is BCLKMaster and where can I set 1/M ratio and (Figure 23-2 in UM rev 1.3 S3C2416) ?? What exactly means values in registers (BFS, RFS) ??Is Re-scaler set by IIS_PSR really 1 / N rescaller in Figure 23-2 in UM rev 1.3 S3C2416. Thanks for any help. Mirek

1 0

[alsa-devel] [PATCH V2 0/2] Add WM8994 PCM Machine driver for Exynos4210
by Sangbeom Kim 17 Jun '11

17 Jun '11

Hi, This is 2nd patchset for WM8994 pcm machine driver that supports PCM audio on SMDKV310, SMDKC210 and test is done. This is minor change version. AS jassi comment, I added new pcm structure for quirk handling. And added Liam's acked. Based on these patches WM8994 pcm machine driver supports followings: o Playback supports 8kHz sampling rates. o Capture supports 8kHz sampling rates. This patchset contains followings o To Kukjin Kim and Ben Dooks, [PATCH 1/2] ARM: EXYNOS4: Add PCM audio support for WM8994 o To Jassi Brar, Mark Brown and Liam Girdwood, [PATCH 2/2] ASoC: SAMSUNG: Add WM8580 PCM Machine driver Best Regards, SB Kim (Sangbeom Kim)

5 10

[alsa-devel] [PATCH 1/3 v2] ASoC: Add ADAU1701 codec driver
by Lars-Peter Clausen 17 Jun '11

17 Jun '11

This patch adds support for the Analog Devices ADAU1701 SigmaDSP. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: Move the error message for unsupported register addresses to adau1701_register_size. --- sound/soc/codecs/Kconfig | 7 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adau1701.c | 548 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adau1701.h | 17 ++ 4 files changed, 573 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/adau1701.c create mode 100644 sound/soc/codecs/adau1701.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..f745e55 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -130,7 +131,11 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate - + +config SND_SOC_ADAU1701 + select SIGMA + tristate + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..30a4c63 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-adau1701-objs := adau1701.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -95,6 +96,7 @@ obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(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 diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c new file mode 100644 index 0000000..6c01bb6 --- /dev/null +++ b/sound/soc/codecs/adau1701.c @@ -0,0 +1,548 @@ +/* + * Driver for ADAU1701 SigmaDSP processor + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * based on an inital version by Cliff Cai <cliff.cai(a)analog.com> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/sigma.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau1701.h" + +#define ADAU1701_DSPCTRL 0x1c +#define ADAU1701_SEROCTL 0x1e +#define ADAU1701_SERICTL 0x1f + +#define ADAU1701_AUXNPOW 0x22 + +#define ADAU1701_OSCIPOW 0x26 +#define ADAU1701_DACSET 0x27 + +#define ADAU1701_NUM_REGS 0x28 + +#define ADAU1701_DSPCTRL_CR (1 << 2) +#define ADAU1701_DSPCTRL_DAM (1 << 3) +#define ADAU1701_DSPCTRL_ADM (1 << 4) +#define ADAU1701_DSPCTRL_SR_48 0x00 +#define ADAU1701_DSPCTRL_SR_96 0x01 +#define ADAU1701_DSPCTRL_SR_192 0x02 +#define ADAU1701_DSPCTRL_SR_MASK 0x03 + +#define ADAU1701_SEROCTL_INV_LRCLK 0x2000 +#define ADAU1701_SEROCTL_INV_BCLK 0x1000 +#define ADAU1701_SEROCTL_MASTER 0x0800 + +#define ADAU1701_SEROCTL_OBF16 0x0000 +#define ADAU1701_SEROCTL_OBF8 0x0200 +#define ADAU1701_SEROCTL_OBF4 0x0400 +#define ADAU1701_SEROCTL_OBF2 0x0600 +#define ADAU1701_SEROCTL_OBF_MASK 0x0600 + +#define ADAU1701_SEROCTL_OLF1024 0x0000 +#define ADAU1701_SEROCTL_OLF512 0x0080 +#define ADAU1701_SEROCTL_OLF256 0x0100 +#define ADAU1701_SEROCTL_OLF_MASK 0x0180 + +#define ADAU1701_SEROCTL_MSB_DEALY1 0x0000 +#define ADAU1701_SEROCTL_MSB_DEALY0 0x0004 +#define ADAU1701_SEROCTL_MSB_DEALY8 0x0008 +#define ADAU1701_SEROCTL_MSB_DEALY12 0x000c +#define ADAU1701_SEROCTL_MSB_DEALY16 0x0010 +#define ADAU1701_SEROCTL_MSB_DEALY_MASK 0x001c + +#define ADAU1701_SEROCTL_WORD_LEN_24 0x0000 +#define ADAU1701_SEROCTL_WORD_LEN_20 0x0001 +#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010 +#define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003 + +#define ADAU1701_AUXNPOW_VBPD 0x40 +#define ADAU1701_AUXNPOW_VRPD 0x20 + +#define ADAU1701_SERICTL_I2S 0 +#define ADAU1701_SERICTL_LEFTJ 1 +#define ADAU1701_SERICTL_TDM 2 +#define ADAU1701_SERICTL_RIGHTJ_24 3 +#define ADAU1701_SERICTL_RIGHTJ_20 4 +#define ADAU1701_SERICTL_RIGHTJ_18 5 +#define ADAU1701_SERICTL_RIGHTJ_16 6 +#define ADAU1701_SERICTL_MODE_MASK 7 +#define ADAU1701_SERICTL_INV_BCLK BIT(3) +#define ADAU1701_SERICTL_INV_LRCLK BIT(4) + +#define ADAU1701_OSCIPOW_OPD 0x04 +#define ADAU1701_DACSET_DACINIT 1 + +#define ADAU1701_FIRMWARE "adau1701.bin" + +struct adau1701 { + unsigned int dai_fmt; +}; + +static const struct snd_kcontrol_new adau1701_controls[] = { + SOC_SINGLE("Master Capture Switch", ADAU1701_DSPCTRL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau1701_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC0", "Playback", ADAU1701_AUXNPOW, 3, 1), + SND_SOC_DAPM_DAC("DAC1", "Playback", ADAU1701_AUXNPOW, 2, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", ADAU1701_AUXNPOW, 1, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", ADAU1701_AUXNPOW, 0, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", ADAU1701_AUXNPOW, 7, 1), + + SND_SOC_DAPM_OUTPUT("OUT0"), + SND_SOC_DAPM_OUTPUT("OUT1"), + SND_SOC_DAPM_OUTPUT("OUT2"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_INPUT("IN0"), + SND_SOC_DAPM_INPUT("IN1"), +}; + +static const struct snd_soc_dapm_route adau1701_dapm_routes[] = { + { "OUT0", NULL, "DAC0" }, + { "OUT1", NULL, "DAC1" }, + { "OUT2", NULL, "DAC2" }, + { "OUT3", NULL, "DAC3" }, + + { "ADC", NULL, "IN0" }, + { "ADC", NULL, "IN1" }, +}; + +static unsigned int adau1701_register_size(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case ADAU1701_DSPCTRL: + case ADAU1701_SEROCTL: + case ADAU1701_AUXNPOW: + case ADAU1701_OSCIPOW: + case ADAU1701_DACSET: + return 2; + case ADAU1701_SERICTL: + return 1; + } + + dev_err(codec->dev, "Unsupported register address: %d\n", reg); + return 0; +} + +static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + unsigned int i, ret; + unsigned int size; + uint8_t buf[4]; + + size = adau1701_register_size(codec, reg); + if (size == 0) + return -EINVAL; + + snd_soc_cache_write(codec, reg, value); + + buf[0] = 0x08; + buf[1] = reg; + + for (i = size + 1; i >= 2; --i) { + buf[i] = value; + value >>= 8; + } + + ret = i2c_master_send(to_i2c_client(codec->dev), buf, size + 2); + if (ret == size + 2) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int value; + unsigned int ret; + + ret = snd_soc_cache_read(codec, reg, &value); + if (ret) + return ret; + + return value; +} + +static int adau1701_load_firmware(struct snd_soc_codec *codec) +{ + return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE); +} + +static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK; + unsigned int val; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU1701_SEROCTL_WORD_LEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAU1701_SEROCTL_WORD_LEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU1701_SEROCTL_WORD_LEN_24; + break; + default: + return -EINVAL; + } + + if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) { + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val |= ADAU1701_SEROCTL_MSB_DEALY16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= ADAU1701_SEROCTL_MSB_DEALY12; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= ADAU1701_SEROCTL_MSB_DEALY8; + break; + } + mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; + } + + snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + + return 0; +} + +static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU1701_SERICTL_RIGHTJ_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAU1701_SERICTL_RIGHTJ_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU1701_SERICTL_RIGHTJ_24; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_SERICTL, + ADAU1701_SERICTL_MODE_MASK, val); + + return 0; +} + +static int adau1701_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 snd_soc_codec *codec = rtd->codec; + snd_pcm_format_t format; + unsigned int val; + + switch (params_rate(params)) { + case 192000: + val = ADAU1701_DSPCTRL_SR_192; + break; + case 96000: + val = ADAU1701_DSPCTRL_SR_96; + break; + case 48000: + val = ADAU1701_DSPCTRL_SR_48; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + ADAU1701_DSPCTRL_SR_MASK, val); + + format = params_format(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return adau1701_set_playback_pcm_format(codec, format); + else + return adau1701_set_capture_pcm_format(codec, format); +} + +static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int serictl = 0x00, seroctl = 0x00; + bool invert_lrclk; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* master, 64-bits per sample, 1 frame per sample */ + seroctl |= ADAU1701_SEROCTL_MASTER | ADAU1701_SEROCTL_OBF16 + | ADAU1701_SEROCTL_OLF1024; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_lrclk = true; + break; + case SND_SOC_DAIFMT_IB_NF: + invert_lrclk = false; + serictl |= ADAU1701_SERICTL_INV_BCLK; + seroctl |= ADAU1701_SEROCTL_INV_BCLK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert_lrclk = true; + serictl |= ADAU1701_SERICTL_INV_BCLK; + seroctl |= ADAU1701_SEROCTL_INV_BCLK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + serictl |= ADAU1701_SERICTL_LEFTJ; + seroctl |= ADAU1701_SEROCTL_MSB_DEALY0; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serictl |= ADAU1701_SERICTL_RIGHTJ_24; + seroctl |= ADAU1701_SEROCTL_MSB_DEALY8; + invert_lrclk = !invert_lrclk; + break; + default: + return -EINVAL; + } + + if (invert_lrclk) { + seroctl |= ADAU1701_SEROCTL_INV_LRCLK; + serictl |= ADAU1701_SERICTL_INV_LRCLK; + } + + adau1701->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + snd_soc_write(codec, ADAU1701_SERICTL, serictl); + snd_soc_update_bits(codec, ADAU1701_SEROCTL, + ~ADAU1701_SEROCTL_WORD_LEN_MASK, seroctl); + + return 0; +} + +static int adau1701_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* Enable VREF and VREF buffer */ + snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + break; + case SND_SOC_BIAS_OFF: + /* Disable VREF and VREF buffer */ + snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int mask = ADAU1701_DSPCTRL_DAM; + unsigned int val; + + if (mute) + val = 0; + else + val = mask; + + snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + + return 0; +} + +static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir) +{ + unsigned int val; + + switch (clk_id) { + case ADAU1701_CLK_SRC_OSC: + val = 0x0; + break; + case ADAU1701_CLK_SRC_MCLK: + val = ADAU1701_OSCIPOW_OPD; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + + return 0; +} + +#define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) + +#define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops adau1701_dai_ops = { + .set_fmt = adau1701_set_dai_fmt, + .hw_params = adau1701_hw_params, + .digital_mute = adau1701_digital_mute, +}; + +static struct snd_soc_dai_driver adau1701_dai = { + .name = "adau1701", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = ADAU1701_RATES, + .formats = ADAU1701_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 8, + .rates = ADAU1701_RATES, + .formats = ADAU1701_FORMATS, + }, + .ops = &adau1701_dai_ops, + .symmetric_rates = 1, +}; + +static int adau1701_probe(struct snd_soc_codec *codec) +{ + int ret; + + codec->dapm.idle_bias_off = 1; + + ret = adau1701_load_firmware(codec); + if (ret) + dev_warn(codec->dev, "Failed to load firmware\n"); + + snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); + snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); + + return 0; +} + +static struct snd_soc_codec_driver adau1701_codec_drv = { + .probe = adau1701_probe, + .set_bias_level = adau1701_set_bias_level, + + .reg_cache_size = ADAU1701_NUM_REGS, + .reg_word_size = sizeof(u16), + + .controls = adau1701_controls, + .num_controls = ARRAY_SIZE(adau1701_controls), + .dapm_widgets = adau1701_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1701_dapm_widgets), + .dapm_routes = adau1701_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1701_dapm_routes), + + .write = adau1701_write, + .read = adau1701_read, + + .set_sysclk = adau1701_set_sysclk, +}; + +static __devinit int adau1701_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct adau1701 *adau1701; + int ret; + + adau1701 = kzalloc(sizeof(*adau1701), GFP_KERNEL); + if (!adau1701) + return -ENOMEM; + + i2c_set_clientdata(client, adau1701); + ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, + &adau1701_dai, 1); + if (ret < 0) + kfree(adau1701); + + return ret; +} + +static __devexit int adau1701_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id adau1701_i2c_id[] = { + { "adau1701", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); + +static struct i2c_driver adau1701_i2c_driver = { + .driver = { + .name = "adau1701", + .owner = THIS_MODULE, + }, + .probe = adau1701_i2c_probe, + .remove = __devexit_p(adau1701_i2c_remove), + .id_table = adau1701_i2c_id, +}; + +static int __init adau1701_init(void) +{ + return i2c_add_driver(&adau1701_i2c_driver); +} +module_init(adau1701_init); + +static void __exit adau1701_exit(void) +{ + i2c_del_driver(&adau1701_i2c_driver); +} +module_exit(adau1701_exit); + +MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver"); +MODULE_AUTHOR("Cliff Cai <cliff.cai(a)analog.com>"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1701.h b/sound/soc/codecs/adau1701.h new file mode 100644 index 0000000..8d0949a --- /dev/null +++ b/sound/soc/codecs/adau1701.h @@ -0,0 +1,17 @@ +/* + * header file for ADAU1701 SigmaDSP processor + * + * Copyright 2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _ADAU1701_H +#define _ADAU1701_H + +enum adau1701_clk_src { + ADAU1701_CLK_SRC_OSC, + ADAU1701_CLK_SRC_MCLK, +}; + +#endif -- 1.7.2.5

3 5

[alsa-devel] ALSA Fireworks driver
by Doug McLain 17 Jun '11

17 Jun '11

Hi Clemens, I came across your fireworks driver the other day, and would like to get involved. Has there been any discussion on this anywhere or have you been completely on your own? I did a search of alsa-devel and didn't find anything, suprisingly. It seems like native alsa support of what seems to me to be the first FW device would be the talk of the town. :) I built the driver as a module on F15 and loaded it on my AF12, and found it in /proc/asound/cards, and tried to start jack on it and it failed to start for duplex, capture, or playback, just FYI. Doug

2 2

[alsa-devel] [PATCH] ASoC: Clean up some coding style nits in the bf5xx-i2s-pcm driver
by Mark Brown 17 Jun '11

17 Jun '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index a611483..819d3bd 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -169,7 +169,7 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); - ret = snd_pcm_hw_constraint_integer(runtime, \ + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) goto out; @@ -307,8 +307,8 @@ static int __devexit bfin_i2s_soc_platform_remove(struct platform_device *pdev) static struct platform_driver bfin_i2s_pcm_driver = { .driver = { - .name = "bfin-i2s-pcm-audio", - .owner = THIS_MODULE, + .name = "bfin-i2s-pcm-audio", + .owner = THIS_MODULE, }, .probe = bfin_i2s_soc_platform_probe, -- 1.7.5.4

3 2