September 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/11] ASoC: OMAP: Convert to use dmaengine
by Peter Ujfalusi 13 Sep '12

13 Sep '12

Hello, This series will switch the OMAP audio to use dmaengine. The final patch which does the switch was based on Russell King's earlier patch. The first 10 patch is to prepare the OMAP audio drivers for a smooth change to dmaengine: - sDMA FRAME sync mode is removed and replaced with PACKET mode - dai drivers no longer need to configure sDMA sync mode - dai drivers does not need to specify the DMA word length - with the exception of the omap-hdmi driver which requires 32bit word length regardless of the audio format in use - the McPDM driver used (to my surprise) hackish way of getting the DMA channel and address - via defines from some header files I have tested the series on: - BeagleBoard (audio via McBSP) with aplay/arecord. In element mode and in threshold mode with different period sizes - OMAP4 Blaze (audio via McPDM and DMIC) With this conversion the NO_PERIOD_WAKEUP is not supported at the moment. I'll be looking at the dmaengine core and omap parts to add this feature back. The patches has been generated against: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 Janusz: Can you retest this series on OMAP1 to be sure I have not broken it? Ricardo: Can you test the omap-hmdi if it is still working? Regards, Peter --- Peter Ujfalusi (11): dmaengine: omap: Support for element mode in cyclic DMA ASoC: omap-mcbsp: Use sDMA packet mode instead of frame mode ASoC: omap-pcm: Select sDMA synchronization based on packet_size ASoC: OMAP: Remove sync_mode from omap_pcm_dma_data struct ASoC: omap-pcm: Prepare to configure the DMA data_type based on stream properties ARM: OMAP4: hwmod_data: Add resource names to McPDM memory ranges ASoC: omap-mcpdm: Use platform_get_resource_* to get resources ASoC: OMAP: mcbsp, mcpdm, dmic: Let omap-pcm to pick the dma_type ASoC: omap-pcm, omap-hdmi: Change the use of omap_pcm_dma_data->data_type ASoC: OMAP: mcbsp, mcpdm, dmic, hdmi: Set dma_data at startup time ASoC: omap-pcm: Convert to use dmaengine arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + drivers/dma/omap-dma.c | 5 +- sound/soc/omap/Kconfig | 3 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-hdmi.c | 17 +- sound/soc/omap/omap-mcbsp.c | 60 +++---- sound/soc/omap/omap-mcpdm.c | 41 +++-- sound/soc/omap/omap-pcm.c | 246 ++++++++--------------------- sound/soc/omap/omap-pcm.h | 4 +- 9 files changed, 136 insertions(+), 251 deletions(-) -- 1.7.12

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[alsa-devel] Invitation to connect on LinkedIn
by Vitaly Chernooky via LinkedIn 13 Sep '12

13 Sep '12

LinkedIn ------------ Vitaly Chernooky requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Vitaly Accept invitation from Vitaly Chernooky http://www.linkedin.com/e/jpxkk3-h71n762j-6i/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Vitaly Chernooky http://www.linkedin.com/e/jpxkk3-h71n762j-6i/rso/18329483/HTJa/name/1380363… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-h71n762j-6i/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [sound-unstable:master 118/143] sound/pci/hda/patch_cirrus.c:1414:10: sparse: Initializer entry defined twice
by Fengguang Wu 13 Sep '12

13 Sep '12

Hi Takashi, FYI, there are new sparse warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 82cb521360dfd32f4f03ceaa74ef6b83ae983593 commit: ef596a57b4d7d8b258beb570ed309ef85bf24dd1 [118/143] ALSA: hda - Add support for MacBook Pro 10,1 All sparse warnings: + sound/pci/hda/patch_cirrus.c:1414:10: sparse: Initializer entry defined twice sound/pci/hda/patch_cirrus.c:1418:10: also defined here The root cause is, CS420X_MBP101 == CS420X_GPIO_23: @@ -85,6 +85,8 @@ enum { => CS420X_GPIO_23, CS420X_IMAC27_122 = CS420X_GPIO_23, CS420X_APPLE = CS420X_GPIO_13, => + CS420X_MBP101, + CS420X_MBP101_COEF, CS420X_AUTO, }; vim +1414 sound/pci/hda/patch_cirrus.c 1404 [CS420X_IMAC27] = { 1405 .type = HDA_FIXUP_PINS, 1406 .v.pins = imac27_pincfgs, 1407 .chained = true, 1408 .chain_id = CS420X_GPIO_13, 1409 }, 1410 [CS420X_GPIO_13] = { 1411 .type = HDA_FIXUP_FUNC, 1412 .v.func = cs420x_fixup_gpio_13, 1413 }, > 1414 [CS420X_GPIO_23] = { 1415 .type = HDA_FIXUP_FUNC, 1416 .v.func = cs420x_fixup_gpio_23, 1417 }, 1418 [CS420X_MBP101] = { 1419 .type = HDA_FIXUP_PINS, 1420 .v.pins = mbp101_pincfgs, 1421 .chained = true, 1422 .chain_id = CS420X_MBP101_COEF, 1423 }, 1424 [CS420X_MBP101_COEF] = { --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8961: Convert to direct regmap API usage
by a＠opensource.wolfsonmicro.com 13 Sep '12

13 Sep '12

From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8961.c | 426 +++++++++++++++++---------------------------- 1 file changed, 161 insertions(+), 265 deletions(-) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 719fb69..4ea64d6e 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -19,6 +19,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -31,277 +32,158 @@ #define WM8961_MAX_REGISTER 0xFC -static u16 wm8961_reg_defaults[] = { - 0x009F, /* R0 - Left Input volume */ - 0x009F, /* R1 - Right Input volume */ - 0x0000, /* R2 - LOUT1 volume */ - 0x0000, /* R3 - ROUT1 volume */ - 0x0020, /* R4 - Clocking1 */ - 0x0008, /* R5 - ADC & DAC Control 1 */ - 0x0000, /* R6 - ADC & DAC Control 2 */ - 0x000A, /* R7 - Audio Interface 0 */ - 0x01F4, /* R8 - Clocking2 */ - 0x0000, /* R9 - Audio Interface 1 */ - 0x00FF, /* R10 - Left DAC volume */ - 0x00FF, /* R11 - Right DAC volume */ - 0x0000, /* R12 */ - 0x0000, /* R13 */ - 0x0040, /* R14 - Audio Interface 2 */ - 0x0000, /* R15 - Software Reset */ - 0x0000, /* R16 */ - 0x007B, /* R17 - ALC1 */ - 0x0000, /* R18 - ALC2 */ - 0x0032, /* R19 - ALC3 */ - 0x0000, /* R20 - Noise Gate */ - 0x00C0, /* R21 - Left ADC volume */ - 0x00C0, /* R22 - Right ADC volume */ - 0x0120, /* R23 - Additional control(1) */ - 0x0000, /* R24 - Additional control(2) */ - 0x0000, /* R25 - Pwr Mgmt (1) */ - 0x0000, /* R26 - Pwr Mgmt (2) */ - 0x0000, /* R27 - Additional Control (3) */ - 0x0000, /* R28 - Anti-pop */ - 0x0000, /* R29 */ - 0x005F, /* R30 - Clocking 3 */ - 0x0000, /* R31 */ - 0x0000, /* R32 - ADCL signal path */ - 0x0000, /* R33 - ADCR signal path */ - 0x0000, /* R34 */ - 0x0000, /* R35 */ - 0x0000, /* R36 */ - 0x0000, /* R37 */ - 0x0000, /* R38 */ - 0x0000, /* R39 */ - 0x0000, /* R40 - LOUT2 volume */ - 0x0000, /* R41 - ROUT2 volume */ - 0x0000, /* R42 */ - 0x0000, /* R43 */ - 0x0000, /* R44 */ - 0x0000, /* R45 */ - 0x0000, /* R46 */ - 0x0000, /* R47 - Pwr Mgmt (3) */ - 0x0023, /* R48 - Additional Control (4) */ - 0x0000, /* R49 - Class D Control 1 */ - 0x0000, /* R50 */ - 0x0003, /* R51 - Class D Control 2 */ - 0x0000, /* R52 */ - 0x0000, /* R53 */ - 0x0000, /* R54 */ - 0x0000, /* R55 */ - 0x0106, /* R56 - Clocking 4 */ - 0x0000, /* R57 - DSP Sidetone 0 */ - 0x0000, /* R58 - DSP Sidetone 1 */ - 0x0000, /* R59 */ - 0x0000, /* R60 - DC Servo 0 */ - 0x0000, /* R61 - DC Servo 1 */ - 0x0000, /* R62 */ - 0x015E, /* R63 - DC Servo 3 */ - 0x0010, /* R64 */ - 0x0010, /* R65 - DC Servo 5 */ - 0x0000, /* R66 */ - 0x0001, /* R67 */ - 0x0003, /* R68 - Analogue PGA Bias */ - 0x0000, /* R69 - Analogue HP 0 */ - 0x0060, /* R70 */ - 0x01FB, /* R71 - Analogue HP 2 */ - 0x0000, /* R72 - Charge Pump 1 */ - 0x0065, /* R73 */ - 0x005F, /* R74 */ - 0x0059, /* R75 */ - 0x006B, /* R76 */ - 0x0038, /* R77 */ - 0x000C, /* R78 */ - 0x000A, /* R79 */ - 0x006B, /* R80 */ - 0x0000, /* R81 */ - 0x0000, /* R82 - Charge Pump B */ - 0x0087, /* R83 */ - 0x0000, /* R84 */ - 0x005C, /* R85 */ - 0x0000, /* R86 */ - 0x0000, /* R87 - Write Sequencer 1 */ - 0x0000, /* R88 - Write Sequencer 2 */ - 0x0000, /* R89 - Write Sequencer 3 */ - 0x0000, /* R90 - Write Sequencer 4 */ - 0x0000, /* R91 - Write Sequencer 5 */ - 0x0000, /* R92 - Write Sequencer 6 */ - 0x0000, /* R93 - Write Sequencer 7 */ - 0x0000, /* R94 */ - 0x0000, /* R95 */ - 0x0000, /* R96 */ - 0x0000, /* R97 */ - 0x0000, /* R98 */ - 0x0000, /* R99 */ - 0x0000, /* R100 */ - 0x0000, /* R101 */ - 0x0000, /* R102 */ - 0x0000, /* R103 */ - 0x0000, /* R104 */ - 0x0000, /* R105 */ - 0x0000, /* R106 */ - 0x0000, /* R107 */ - 0x0000, /* R108 */ - 0x0000, /* R109 */ - 0x0000, /* R110 */ - 0x0000, /* R111 */ - 0x0000, /* R112 */ - 0x0000, /* R113 */ - 0x0000, /* R114 */ - 0x0000, /* R115 */ - 0x0000, /* R116 */ - 0x0000, /* R117 */ - 0x0000, /* R118 */ - 0x0000, /* R119 */ - 0x0000, /* R120 */ - 0x0000, /* R121 */ - 0x0000, /* R122 */ - 0x0000, /* R123 */ - 0x0000, /* R124 */ - 0x0000, /* R125 */ - 0x0000, /* R126 */ - 0x0000, /* R127 */ - 0x0000, /* R128 */ - 0x0000, /* R129 */ - 0x0000, /* R130 */ - 0x0000, /* R131 */ - 0x0000, /* R132 */ - 0x0000, /* R133 */ - 0x0000, /* R134 */ - 0x0000, /* R135 */ - 0x0000, /* R136 */ - 0x0000, /* R137 */ - 0x0000, /* R138 */ - 0x0000, /* R139 */ - 0x0000, /* R140 */ - 0x0000, /* R141 */ - 0x0000, /* R142 */ - 0x0000, /* R143 */ - 0x0000, /* R144 */ - 0x0000, /* R145 */ - 0x0000, /* R146 */ - 0x0000, /* R147 */ - 0x0000, /* R148 */ - 0x0000, /* R149 */ - 0x0000, /* R150 */ - 0x0000, /* R151 */ - 0x0000, /* R152 */ - 0x0000, /* R153 */ - 0x0000, /* R154 */ - 0x0000, /* R155 */ - 0x0000, /* R156 */ - 0x0000, /* R157 */ - 0x0000, /* R158 */ - 0x0000, /* R159 */ - 0x0000, /* R160 */ - 0x0000, /* R161 */ - 0x0000, /* R162 */ - 0x0000, /* R163 */ - 0x0000, /* R164 */ - 0x0000, /* R165 */ - 0x0000, /* R166 */ - 0x0000, /* R167 */ - 0x0000, /* R168 */ - 0x0000, /* R169 */ - 0x0000, /* R170 */ - 0x0000, /* R171 */ - 0x0000, /* R172 */ - 0x0000, /* R173 */ - 0x0000, /* R174 */ - 0x0000, /* R175 */ - 0x0000, /* R176 */ - 0x0000, /* R177 */ - 0x0000, /* R178 */ - 0x0000, /* R179 */ - 0x0000, /* R180 */ - 0x0000, /* R181 */ - 0x0000, /* R182 */ - 0x0000, /* R183 */ - 0x0000, /* R184 */ - 0x0000, /* R185 */ - 0x0000, /* R186 */ - 0x0000, /* R187 */ - 0x0000, /* R188 */ - 0x0000, /* R189 */ - 0x0000, /* R190 */ - 0x0000, /* R191 */ - 0x0000, /* R192 */ - 0x0000, /* R193 */ - 0x0000, /* R194 */ - 0x0000, /* R195 */ - 0x0030, /* R196 */ - 0x0006, /* R197 */ - 0x0000, /* R198 */ - 0x0060, /* R199 */ - 0x0000, /* R200 */ - 0x003F, /* R201 */ - 0x0000, /* R202 */ - 0x0000, /* R203 */ - 0x0000, /* R204 */ - 0x0001, /* R205 */ - 0x0000, /* R206 */ - 0x0181, /* R207 */ - 0x0005, /* R208 */ - 0x0008, /* R209 */ - 0x0008, /* R210 */ - 0x0000, /* R211 */ - 0x013B, /* R212 */ - 0x0000, /* R213 */ - 0x0000, /* R214 */ - 0x0000, /* R215 */ - 0x0000, /* R216 */ - 0x0070, /* R217 */ - 0x0000, /* R218 */ - 0x0000, /* R219 */ - 0x0000, /* R220 */ - 0x0000, /* R221 */ - 0x0000, /* R222 */ - 0x0003, /* R223 */ - 0x0000, /* R224 */ - 0x0000, /* R225 */ - 0x0001, /* R226 */ - 0x0008, /* R227 */ - 0x0000, /* R228 */ - 0x0000, /* R229 */ - 0x0000, /* R230 */ - 0x0000, /* R231 */ - 0x0004, /* R232 */ - 0x0000, /* R233 */ - 0x0000, /* R234 */ - 0x0000, /* R235 */ - 0x0000, /* R236 */ - 0x0000, /* R237 */ - 0x0080, /* R238 */ - 0x0000, /* R239 */ - 0x0000, /* R240 */ - 0x0000, /* R241 */ - 0x0000, /* R242 */ - 0x0000, /* R243 */ - 0x0000, /* R244 */ - 0x0052, /* R245 */ - 0x0110, /* R246 */ - 0x0040, /* R247 */ - 0x0000, /* R248 */ - 0x0030, /* R249 */ - 0x0000, /* R250 */ - 0x0000, /* R251 */ - 0x0001, /* R252 - General test 1 */ +static const struct reg_default wm8961_reg_defaults[] = { + { 0, 0x009F }, /* R0 - Left Input volume */ + { 1, 0x009F }, /* R1 - Right Input volume */ + { 2, 0x0000 }, /* R2 - LOUT1 volume */ + { 3, 0x0000 }, /* R3 - ROUT1 volume */ + { 4, 0x0020 }, /* R4 - Clocking1 */ + { 5, 0x0008 }, /* R5 - ADC & DAC Control 1 */ + { 6, 0x0000 }, /* R6 - ADC & DAC Control 2 */ + { 7, 0x000A }, /* R7 - Audio Interface 0 */ + { 8, 0x01F4 }, /* R8 - Clocking2 */ + { 9, 0x0000 }, /* R9 - Audio Interface 1 */ + { 10, 0x00FF }, /* R10 - Left DAC volume */ + { 11, 0x00FF }, /* R11 - Right DAC volume */ + + { 14, 0x0040 }, /* R14 - Audio Interface 2 */ + + { 17, 0x007B }, /* R17 - ALC1 */ + { 18, 0x0000 }, /* R18 - ALC2 */ + { 19, 0x0032 }, /* R19 - ALC3 */ + { 20, 0x0000 }, /* R20 - Noise Gate */ + { 21, 0x00C0 }, /* R21 - Left ADC volume */ + { 22, 0x00C0 }, /* R22 - Right ADC volume */ + { 23, 0x0120 }, /* R23 - Additional control(1) */ + { 24, 0x0000 }, /* R24 - Additional control(2) */ + { 25, 0x0000 }, /* R25 - Pwr Mgmt (1) */ + { 26, 0x0000 }, /* R26 - Pwr Mgmt (2) */ + { 27, 0x0000 }, /* R27 - Additional Control (3) */ + { 28, 0x0000 }, /* R28 - Anti-pop */ + + { 30, 0x005F }, /* R30 - Clocking 3 */ + + { 32, 0x0000 }, /* R32 - ADCL signal path */ + { 33, 0x0000 }, /* R33 - ADCR signal path */ + + { 40, 0x0000 }, /* R40 - LOUT2 volume */ + { 41, 0x0000 }, /* R41 - ROUT2 volume */ + + { 47, 0x0000 }, /* R47 - Pwr Mgmt (3) */ + { 48, 0x0023 }, /* R48 - Additional Control (4) */ + { 49, 0x0000 }, /* R49 - Class D Control 1 */ + + { 51, 0x0003 }, /* R51 - Class D Control 2 */ + + { 56, 0x0106 }, /* R56 - Clocking 4 */ + { 57, 0x0000 }, /* R57 - DSP Sidetone 0 */ + { 58, 0x0000 }, /* R58 - DSP Sidetone 1 */ + + { 60, 0x0000 }, /* R60 - DC Servo 0 */ + { 61, 0x0000 }, /* R61 - DC Servo 1 */ + + { 63, 0x015E }, /* R63 - DC Servo 3 */ + + { 65, 0x0010 }, /* R65 - DC Servo 5 */ + + { 68, 0x0003 }, /* R68 - Analogue PGA Bias */ + { 69, 0x0000 }, /* R69 - Analogue HP 0 */ + + { 71, 0x01FB }, /* R71 - Analogue HP 2 */ + { 72, 0x0000 }, /* R72 - Charge Pump 1 */ + + { 82, 0x0000 }, /* R82 - Charge Pump B */ + + { 87, 0x0000 }, /* R87 - Write Sequencer 1 */ + { 88, 0x0000 }, /* R88 - Write Sequencer 2 */ + { 89, 0x0000 }, /* R89 - Write Sequencer 3 */ + { 90, 0x0000 }, /* R90 - Write Sequencer 4 */ + { 91, 0x0000 }, /* R91 - Write Sequencer 5 */ + { 92, 0x0000 }, /* R92 - Write Sequencer 6 */ + { 93, 0x0000 }, /* R93 - Write Sequencer 7 */ + + { 252, 0x0001 }, /* R252 - General test 1 */ }; struct wm8961_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk; }; -static int wm8961_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8961_volatile(struct device *dev, unsigned int reg) { switch (reg) { case WM8961_SOFTWARE_RESET: case WM8961_WRITE_SEQUENCER_7: case WM8961_DC_SERVO_1: - return 1; + return true; default: - return 0; + return false; + } +} + +static bool wm8961_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8961_LEFT_INPUT_VOLUME: + case WM8961_RIGHT_INPUT_VOLUME: + case WM8961_LOUT1_VOLUME: + case WM8961_ROUT1_VOLUME: + case WM8961_CLOCKING1: + case WM8961_ADC_DAC_CONTROL_1: + case WM8961_ADC_DAC_CONTROL_2: + case WM8961_AUDIO_INTERFACE_0: + case WM8961_CLOCKING2: + case WM8961_AUDIO_INTERFACE_1: + case WM8961_LEFT_DAC_VOLUME: + case WM8961_RIGHT_DAC_VOLUME: + case WM8961_AUDIO_INTERFACE_2: + case WM8961_SOFTWARE_RESET: + case WM8961_ALC1: + case WM8961_ALC2: + case WM8961_ALC3: + case WM8961_NOISE_GATE: + case WM8961_LEFT_ADC_VOLUME: + case WM8961_RIGHT_ADC_VOLUME: + case WM8961_ADDITIONAL_CONTROL_1: + case WM8961_ADDITIONAL_CONTROL_2: + case WM8961_PWR_MGMT_1: + case WM8961_PWR_MGMT_2: + case WM8961_ADDITIONAL_CONTROL_3: + case WM8961_ANTI_POP: + case WM8961_CLOCKING_3: + case WM8961_ADCL_SIGNAL_PATH: + case WM8961_ADCR_SIGNAL_PATH: + case WM8961_LOUT2_VOLUME: + case WM8961_ROUT2_VOLUME: + case WM8961_PWR_MGMT_3: + case WM8961_ADDITIONAL_CONTROL_4: + case WM8961_CLASS_D_CONTROL_1: + case WM8961_CLASS_D_CONTROL_2: + case WM8961_CLOCKING_4: + case WM8961_DSP_SIDETONE_0: + case WM8961_DSP_SIDETONE_1: + case WM8961_DC_SERVO_0: + case WM8961_DC_SERVO_1: + case WM8961_DC_SERVO_3: + case WM8961_DC_SERVO_5: + case WM8961_ANALOGUE_PGA_BIAS: + case WM8961_ANALOGUE_HP_0: + case WM8961_ANALOGUE_HP_2: + case WM8961_CHARGE_PUMP_1: + case WM8961_CHARGE_PUMP_B: + case WM8961_WRITE_SEQUENCER_1: + case WM8961_WRITE_SEQUENCER_2: + case WM8961_WRITE_SEQUENCER_3: + case WM8961_WRITE_SEQUENCER_4: + case WM8961_WRITE_SEQUENCER_5: + case WM8961_WRITE_SEQUENCER_6: + case WM8961_WRITE_SEQUENCER_7: + case WM8961_GENERAL_TEST_1: + return true; + default: + return false; } } @@ -958,11 +840,12 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { + struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -975,9 +858,9 @@ static int wm8961_probe(struct snd_soc_codec *codec) } /* This isn't volatile - readback doesn't correspond to write */ - codec->cache_bypass = 1; + regcache_cache_bypass(wm8961->regmap, true); reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME); - codec->cache_bypass = 0; + regcache_cache_bypass(wm8961->regmap, false); dev_info(codec->dev, "WM8961 family %d revision %c\n", (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) @@ -1066,10 +949,19 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8961 = { .suspend = wm8961_suspend, .resume = wm8961_resume, .set_bias_level = wm8961_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8961_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8961_reg_defaults, - .volatile_register = wm8961_volatile_register, +}; + +static const struct regmap_config wm8961_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = WM8961_MAX_REGISTER, + + .reg_defaults = wm8961_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8961_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8961_volatile, + .readable_reg = wm8961_readable, }; static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, @@ -1083,6 +975,10 @@ static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, if (wm8961 == NULL) return -ENOMEM; + wm8961->regmap = devm_regmap_init_i2c(i2c, &wm8961_regmap); + if (IS_ERR(wm8961->regmap)) + return PTR_ERR(wm8961->regmap); + i2c_set_clientdata(i2c, wm8961); ret = snd_soc_register_codec(&i2c->dev, -- 1.7.10.4

1 1

Re: [alsa-devel] [pulseaudio-discuss] [PATCH] alsa-mixer: Don't use the mono path in the analog-stereo mapping.
by Raymond Yau 13 Sep '12

13 Sep '12

2012-9-12 下午9:04 於 "Tanu Kaskinen" <tanuk(a)iki.fi> 寫道： > > As far as I can see, having a mono path in a stereo mapping doesn't > make any sense. It also causes breakage: if the Master Mono mixer > element has two volume channels, Because ac97 codecs have an analog mixer which can downmix stereo to the mono out pin. Some hda codecs also have the mono out pin which connected to a mixer widget for downmixing stereo from stereo DAC to mono The master mono playback volume control of ac97 has only one channel , so it is a bug of vm 's ac97 codec register emulation if the control has two volume channels Either removed the master mono register emulation from vm or pulseaudio ignore the master mono switch of vm's ac97 codec > the analog-output path gets removed > due to being a subset of analog-output-mono, and that in turn causes > the Master element getting muted. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/a… The above patch try to assign the headphone path for the USB headsets which have speaker playback volume control since speaker playback volume control is not in pulseaudio ' s headphone path. >Users generally don't like that. > > BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=54673 It look like a bug in output volume calculation of ac97 codec emulation in those virtual machine if the emulated ac97 codec only has 16 bits DAC resolution or pulseaudio does not unmute pcm playback switch on those ac97 codec after user mute the sound in sound preference and pulseaudio mute both master playback switch and pcm playback switch Some tablet with alc203 use the master mono output for the mono speaker http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=8f4f4ef6fed55… For als4000 the master mono is only enum control to change the channel map of the mono out Simple mixer control 'Master Mono',0 Capabilities: pswitch pswitch-joined cenum Items: 'L chan only' 'R chan only' 'L ch/2 + R ch/2' Item0: 'L chan only' https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/827393

1 0

Re: [alsa-devel] [PATCH 8/9] sound/soc/soc-core.c: Remove useless kfree
by Mark Brown 13 Sep '12

13 Sep '12

On Wed, Sep 12, 2012 at 05:06:46PM +0200, Peter Senna Tschudin wrote: > From: Peter Senna Tschudin <peter.senna(a)gmail.com> > > Remove useless kfree() and clean up code related to the removal. > > The semantic patch that finds this problem is as follows: > (http://coccinelle.lip6.fr/) Applied, thanks.

1 0

[alsa-devel] [RFC PATCH 0/5] mpc5200 asoc fixups
by Eric Millbrandt 13 Sep '12

13 Sep '12

The mpc5200 asoc code did not run after the multi-codec patches. This series fixes the existing code and adds a generic platform driver that allows the audio layout to be described in the device tree. Eric Millbrandt (5): ASoC: fsl: mpc5200 multi-codec fixups ASoC: fsl: mpc5200 combine psc_dma platform data ASoC: fsl: mpc5200-soc-audio driver arch/powerpc/boot/dts pcm030 add mpc5200-soc-audio node ASoC: fsl: mpc5200 remove pcm030 and efika audio fabric .../devicetree/bindings/powerpc/fsl/mpc5200.txt | 17 ++ arch/powerpc/boot/dts/pcm030.dts | 23 ++- arch/powerpc/platforms/52xx/Kconfig | 1 + sound/soc/fsl/Kconfig | 20 +-- sound/soc/fsl/Makefile | 3 +- sound/soc/fsl/efika-audio-fabric.c | 91 -------- sound/soc/fsl/mpc5200_dma.c | 28 +-- sound/soc/fsl/mpc5200_dma.h | 3 + sound/soc/fsl/mpc5200_psc_ac97.c | 13 +- sound/soc/fsl/mpc5200_psc_i2s.c | 8 + sound/soc/fsl/mpc5200_soc_audio.c | 232 ++++++++++++++++++++ sound/soc/fsl/pcm030-audio-fabric.c | 91 -------- 12 files changed, 306 insertions(+), 224 deletions(-) delete mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_soc_audio.c delete mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- 1.7.2.5

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[alsa-devel] [PATCH 1/2] [v2] ASoC: fsl: use snd_soc_register_card to register the card
by Timur Tabi 13 Sep '12

13 Sep '12

Use snd_soc_register_card() instead of platform_device_alloc("soc-audio") to register the sound card from the machine drivers. Although platform_device_alloc is officially deprecated, it is still used by several other drivers. Unfortunately, something is broken somewhere and the Freescale drivers don't register properly when using it. Since we need to transition to snd_soc_register_card() anyway, this fixes the problem and updates our code at the same time. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 23 +++++------------------ sound/soc/fsl/p1022_ds.c | 22 ++++------------------ 2 files changed, 9 insertions(+), 36 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 60bcba1..5479882 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -192,7 +192,6 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct mpc8610_hpcd_data *machine_data; int ret = -ENODEV; const char *sprop; @@ -341,34 +340,22 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->card.probe = mpc8610_hpcd_machine_probe; machine_data->card.remove = mpc8610_hpcd_machine_remove; machine_data->card.name = pdev->name; /* The platform driver name */ + machine_data->card.owner = THIS_MODULE; + machine_data->card.dev = &pdev->dev; machine_data->card.num_links = 2; machine_data->card.dai_link = machine_data->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &machine_data->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&machine_data->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); - goto error_sound; + dev_err(&pdev->dev, "could not register card\n"); + goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; -error_sound: - platform_device_put(sound_device); error: kfree(machine_data); error_alloc: diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 50adf40..a44a8c3 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -202,7 +202,6 @@ static int p1022_ds_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct machine_data *mdata; int ret = -ENODEV; const char *sprop; @@ -349,36 +348,23 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->card.probe = p1022_ds_machine_probe; mdata->card.remove = p1022_ds_machine_remove; mdata->card.name = pdev->name; /* The platform driver name */ + mdata->card.owner = THIS_MODULE; + mdata->card.dev = &pdev->dev; mdata->card.num_links = 2; mdata->card.dai_link = mdata->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &mdata->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&mdata->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); + dev_err(&pdev->dev, "could not register card\n"); goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; error: - if (sound_device) - platform_device_put(sound_device); - kfree(mdata); error_put: of_node_put(codec_np); -- 1.7.3.4

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[alsa-devel] [PATCH] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug unconditionally
by Wang Xingchao 13 Sep '12

13 Sep '12

Clear Audio Enable bit to trigger unsolicated event to notify Audio Driver part the HDMI hot plug change. The patch fixed the bug when remove HDMI cable the bit was not cleared correctly. In intel_enable_hdmi(), if intel_hdmi->has_audio been true, the "Audio enable bit" will be set to trigger unsolicated event to notify Alsa driver the change. intel_hdmi->has_audio will be reset to false from intel_hdmi_detect() after remove the hdmi cable, here's debug log: [ 187.494153] [drm:output_poll_execute], [CONNECTOR:17:HDMI-A-1] status updated from 1 to 2 [ 187.525349] [drm:intel_hdmi_detect], HDMI: has_audio = 0 so when comes back to intel_disable_hdmi(), the "Audio enable bit" will not be cleared. And this cause the eld infomation and pin presence doesnot update accordingly in alsa driver side. This patch will also trigger unsolicated event to alsa driver to notify the hot plug event: [ 187.853159] ALSA sound/pci/hda/patch_hdmi.c:772 HDMI hot plug event: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=1 [ 187.853268] ALSA sound/pci/hda/patch_hdmi.c:990 HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- drivers/gpu/drm/i915/intel_hdmi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c index 5d02aad..d39ed58 100644 --- a/drivers/gpu/drm/i915/intel_hdmi.c +++ b/drivers/gpu/drm/i915/intel_hdmi.c @@ -676,8 +676,7 @@ static void intel_disable_hdmi(struct intel_encoder *encoder) u32 temp; u32 enable_bits = SDVO_ENABLE; - if (intel_hdmi->has_audio) - enable_bits |= SDVO_AUDIO_ENABLE; + enable_bits |= SDVO_AUDIO_ENABLE; temp = I915_READ(intel_hdmi->sdvox_reg); -- 1.7.9.5

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[alsa-devel] [PATCH] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug
by Wang Xingchao 13 Sep '12

13 Sep '12

Clear Audio Enable bit to trigger unsolicated event to notify Audio Driver part the HDMI hot plug change. The patch fixed the bug when remove HDMI cable the bit was not cleared correctly. In intel_hdmi_dpms(), if intel_hdmi->has_audio been true, the "Audio enable bit" will be set to trigger unsolicated event to notify Alsa driver the change. intel_hdmi->has_audio will be reset to false from intel_hdmi_detect() after remove the hdmi cable, here's debug log: [ 187.494153] [drm:output_poll_execute], [CONNECTOR:17:HDMI-A-1] status updated from 1 to 2 [ 187.525349] [drm:intel_hdmi_detect], HDMI: has_audio = 0 so when comes back to intel_hdmi_dpms(), the "Audio enable bit" will not be cleared. And this cause the eld infomation and pin presence doesnot update accordingly in alsa driver side. This patch will also trigger unsolicated event to alsa driver to notify the hot plug event: [ 187.853159] ALSA sound/pci/hda/patch_hdmi.c:772 HDMI hot plug event: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=1 [ 187.853268] ALSA sound/pci/hda/patch_hdmi.c:990 HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- drivers/gpu/drm/i915/intel_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c index 98f6024..12dc330 100644 --- a/drivers/gpu/drm/i915/intel_hdmi.c +++ b/drivers/gpu/drm/i915/intel_hdmi.c @@ -609,7 +609,7 @@ static void intel_hdmi_dpms(struct drm_encoder *encoder, int mode) u32 temp; u32 enable_bits = SDVO_ENABLE; - if (intel_hdmi->has_audio) + if (intel_hdmi->has_audio || mode != DRM_MODE_DPMS_ON) enable_bits |= SDVO_AUDIO_ENABLE; temp = I915_READ(intel_hdmi->sdvox_reg); -- 1.7.9.5

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