February 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: cosmetic: make hda intel interrupt name consistent with others
by Maxim Levitsky 04 Feb '10

04 Feb '10

This renames the interrupt name in /proc/interrupt. HDA Intel -> hda_intel This also eliminates space from the name, probably helping some parsers. Don't think anybody depends on this name in userspace --- sound/pci/hda/hda_intel.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 11ce655..0adea96 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2058,7 +2058,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect) { if (request_irq(chip->pci->irq, azx_interrupt, chip->msi ? 0 : IRQF_SHARED, - "HDA Intel", chip)) { + "hda_intel", chip)) { printk(KERN_ERR "hda-intel: unable to grab IRQ %d, " "disabling device\n", chip->pci->irq); if (do_disconnect) -- 1.6.3.3

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[alsa-devel] [PATCH] hda-analyzer: Avoid AttributeError when codec has no nodes
by Ozan Çağlayan 04 Feb '10

04 Feb '10

When traversing card/codec/nodes during treewidget creation, it's quite possible that codec.nodes raises an AttributeError if the current codec doesn't seem to have a nodes dictionary. Simply initialize it to an empty dictionary in the beginning of analyze(). Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- hda-analyzer/hda_codec.py | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/hda-analyzer/hda_codec.py b/hda-analyzer/hda_codec.py index cbea3f8..d587a28 100644 --- a/hda-analyzer/hda_codec.py +++ b/hda-analyzer/hda_codec.py @@ -818,6 +818,7 @@ class HDACodec: def analyze(self): self.afg = None self.mfg = None + self.nodes = {} self.vendor_id = self.param_read(AC_NODE_ROOT, PARAMS['VENDOR_ID']) self.subsystem_id = self.param_read(AC_NODE_ROOT, PARAMS['SUBSYSTEM_ID']) self.revision_id = self.param_read(AC_NODE_ROOT, PARAMS['REV_ID']) @@ -866,7 +867,6 @@ class HDACodec: nodes_count, nid = self.get_sub_nodes(self.afg) self.base_nid = nid - self.nodes = {} for i in range(nodes_count): self.nodes[nid] = HDANode(self, nid) nid += 1 -- 1.6.5.6

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[alsa-devel] ALC268 - Driving mute led with GPIO
by Ozan Çağlayan 04 Feb '10

04 Feb '10

Hi, We have a laptop here (a local brand) which has a mute button with a LED. Currently the mute button works correctly but the led is non-functional. I've played through hda-analyzer and found out that it's enabled when you enable gpio3 and click out_dir. Clicking out_dir again and again switches it on and off. There's a mute led stuff in patch_sigmatel, sigmatel_spec has gpio_led and gpio_polarity information but alc_spec doesn't have. How should we hook/patch that information into the codec? I'll try to send the alsa-info output tomorrow. Thanks. Ozan

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 04 Feb '10

04 Feb '10

The following changes since commit 026384d614b827f368834860c9b623ce08439b7e: Daniel Mack (1): ASoC: fix PXA SSP port resume are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Guennadi Liakhovetski (1): ASoC: fix compilation breakage in sound/soc/sh/fsi.c Mark Brown (8): ASoC: Allow CODECs to ask soc-cache to suppress physical writes ASoC: Add a cache_sync bit to the CODEC structure ASoC: Convert WM8993 to use shared cache I/O code ASoC: Initial WM8993 regulator API hookup ASoC: Disable WM8993 regulators when turning bias off ASoC: Add support for BIAS_OFF when idle to WM8904 ASoC: Optimise WM8904 output stage power control ASoC: Add WM8912 DAC support Peter Ujfalusi (1): ASoC: TWL4030: Module unloading fix include/sound/soc.h | 2 + sound/soc/codecs/twl4030.c | 5 +- sound/soc/codecs/wm8904.c | 176 ++++++++++++++++++++++++------- sound/soc/codecs/wm8993.c | 247 ++++++++++++++++++++++--------------------- sound/soc/sh/fsi.c | 40 +++---- sound/soc/soc-cache.c | 36 ++++++- 6 files changed, 320 insertions(+), 186 deletions(-)

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[alsa-devel] [PATCH] [ALSA] es1968: remove unused code and macros
by Jiri Kosina 04 Feb '10

04 Feb '10

Remove code that has been commented out since beginning of git history. Also remove macros that are not used throughout the driver code at all. Signed-off-by: Jiri Kosina <jkosina(a)suse.cz> --- sound/pci/es1968.c | 181 ---------------------------------------------------- 1 files changed, 0 insertions(+), 181 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a11f453..5c35075 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -173,19 +173,10 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define ESS_FMT_STEREO 0x01 #define ESS_FMT_16BIT 0x02 -#define DAC_RUNNING 1 -#define ADC_RUNNING 2 - /* Values for the ESM_LEGACY_AUDIO_CONTROL */ #define ESS_DISABLE_AUDIO 0x8000 #define ESS_ENABLE_SERIAL_IRQ 0x4000 -#define IO_ADRESS_ALIAS 0x0020 -#define MPU401_IRQ_ENABLE 0x0010 -#define MPU401_IO_ENABLE 0x0008 -#define GAME_IO_ENABLE 0x0004 -#define FM_IO_ENABLE 0x0002 -#define SB_IO_ENABLE 0x0001 /* Values for the ESM_CONFIG_A */ @@ -193,9 +184,6 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define PIC_SNOOP2 0x2000 #define SAFEGUARD 0x0800 #define DMA_CLEAR 0x0700 -#define DMA_DDMA 0x0000 -#define DMA_TDMA 0x0100 -#define DMA_PCPCI 0x0200 #define POST_WRITE 0x0080 #define PCI_TIMING 0x0040 #define SWAP_LR 0x0020 @@ -217,7 +205,6 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define RINGB_DIS_VALIDATION 0x0008 #define RINGB_EN_SPDIF 0x0010 #define RINGB_EN_2CODEC 0x0020 -#define RINGB_SING_BIT_DUAL 0x0040 /* ****Port Adresses**** */ @@ -231,7 +218,6 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define ESM_RING_BUS_DEST 0x34 #define ESM_RING_BUS_CONTR_A 0x36 #define ESM_RING_BUS_CONTR_B 0x38 -#define ESM_RING_BUS_SDO 0x3A /* WaveCache*/ #define WC_INDEX 0x10 @@ -239,15 +225,9 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define WC_CONTROL 0x14 /* ASSP*/ -#define ASSP_INDEX 0x80 -#define ASSP_MEMORY 0x82 -#define ASSP_DATA 0x84 #define ASSP_CONTROL_A 0xA2 #define ASSP_CONTROL_B 0xA4 #define ASSP_CONTROL_C 0xA6 -#define ASSP_HOSTW_INDEX 0xA8 -#define ASSP_HOSTW_DATA 0xAA -#define ASSP_HOSTW_IRQ 0xAC /* Midi */ #define ESM_MPU401_PORT 0x98 /* Others */ @@ -256,19 +236,11 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define IDR0_DATA_PORT 0x00 #define IDR1_CRAM_POINTER 0x01 #define IDR2_CRAM_DATA 0x02 -#define IDR3_WAVE_DATA 0x03 -#define IDR4_WAVE_PTR_LOW 0x04 -#define IDR5_WAVE_PTR_HI 0x05 -#define IDR6_TIMER_CTRL 0x06 #define IDR7_WAVE_ROMRAM 0x07 -#define WRITEABLE_MAP 0xEFFFFF -#define READABLE_MAP 0x64003F - /* PCI Register */ #define ESM_LEGACY_AUDIO_CONTROL 0x40 -#define ESM_ACPI_COMMAND 0x54 #define ESM_CONFIG_A 0x50 #define ESM_CONFIG_B 0x52 #define ESM_DDMA 0x60 @@ -280,149 +252,23 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); /* Host IRQ Control Bits */ #define ESM_RESET_MAESTRO 0x8000 #define ESM_RESET_DIRECTSOUND 0x4000 -#define ESM_HIRQ_ClkRun 0x0100 #define ESM_HIRQ_HW_VOLUME 0x0040 -#define ESM_HIRQ_HARPO 0x0030 /* What's that? */ -#define ESM_HIRQ_ASSP 0x0010 #define ESM_HIRQ_DSIE 0x0004 #define ESM_HIRQ_MPU401 0x0002 -#define ESM_HIRQ_SB 0x0001 /* Host IRQ Status Bits */ #define ESM_MPU401_IRQ 0x02 -#define ESM_SB_IRQ 0x01 #define ESM_SOUND_IRQ 0x04 -#define ESM_ASSP_IRQ 0x10 #define ESM_HWVOL_IRQ 0x40 #define ESS_SYSCLK 50000000 #define ESM_BOB_FREQ 200 #define ESM_BOB_FREQ_MAX 800 -#define ESM_FREQ_ESM1 (49152000L / 1024L) /* default rate 48000 */ -#define ESM_FREQ_ESM2 (50000000L / 1024L) - /* APU Modes: reg 0x00, bit 4-7 */ -#define ESM_APU_MODE_SHIFT 4 -#define ESM_APU_MODE_MASK (0xf << 4) -#define ESM_APU_OFF 0x00 -#define ESM_APU_16BITLINEAR 0x01 /* 16-Bit Linear Sample Player */ -#define ESM_APU_16BITSTEREO 0x02 /* 16-Bit Stereo Sample Player */ #define ESM_APU_8BITLINEAR 0x03 /* 8-Bit Linear Sample Player */ -#define ESM_APU_8BITSTEREO 0x04 /* 8-Bit Stereo Sample Player */ -#define ESM_APU_8BITDIFF 0x05 /* 8-Bit Differential Sample Playrer */ -#define ESM_APU_DIGITALDELAY 0x06 /* Digital Delay Line */ -#define ESM_APU_DUALTAP 0x07 /* Dual Tap Reader */ -#define ESM_APU_CORRELATOR 0x08 /* Correlator */ #define ESM_APU_INPUTMIXER 0x09 /* Input Mixer */ -#define ESM_APU_WAVETABLE 0x0A /* Wave Table Mode */ #define ESM_APU_SRCONVERTOR 0x0B /* Sample Rate Convertor */ -#define ESM_APU_16BITPINGPONG 0x0C /* 16-Bit Ping-Pong Sample Player */ -#define ESM_APU_RESERVED1 0x0D /* Reserved 1 */ -#define ESM_APU_RESERVED2 0x0E /* Reserved 2 */ -#define ESM_APU_RESERVED3 0x0F /* Reserved 3 */ - -/* reg 0x00 */ -#define ESM_APU_FILTER_Q_SHIFT 0 -#define ESM_APU_FILTER_Q_MASK (3 << 0) -/* APU Filtey Q Control */ -#define ESM_APU_FILTER_LESSQ 0x00 -#define ESM_APU_FILTER_MOREQ 0x03 - -#define ESM_APU_FILTER_TYPE_SHIFT 2 -#define ESM_APU_FILTER_TYPE_MASK (3 << 2) -#define ESM_APU_ENV_TYPE_SHIFT 8 -#define ESM_APU_ENV_TYPE_MASK (3 << 8) -#define ESM_APU_ENV_STATE_SHIFT 10 -#define ESM_APU_ENV_STATE_MASK (3 << 10) -#define ESM_APU_END_CURVE (1 << 12) -#define ESM_APU_INT_ON_LOOP (1 << 13) -#define ESM_APU_DMA_ENABLE (1 << 14) - -/* reg 0x02 */ -#define ESM_APU_SUBMIX_GROUP_SHIRT 0 -#define ESM_APU_SUBMIX_GROUP_MASK (7 << 0) -#define ESM_APU_SUBMIX_MODE (1 << 3) -#define ESM_APU_6dB (1 << 4) -#define ESM_APU_DUAL_EFFECT (1 << 5) -#define ESM_APU_EFFECT_CHANNELS_SHIFT 6 -#define ESM_APU_EFFECT_CHANNELS_MASK (3 << 6) - -/* reg 0x03 */ -#define ESM_APU_STEP_SIZE_MASK 0x0fff - -/* reg 0x04 */ -#define ESM_APU_PHASE_SHIFT 0 -#define ESM_APU_PHASE_MASK (0xff << 0) -#define ESM_APU_WAVE64K_PAGE_SHIFT 8 /* most 8bit of wave start offset */ -#define ESM_APU_WAVE64K_PAGE_MASK (0xff << 8) - -/* reg 0x05 - wave start offset */ -/* reg 0x06 - wave end offset */ -/* reg 0x07 - wave loop length */ - -/* reg 0x08 */ -#define ESM_APU_EFFECT_GAIN_SHIFT 0 -#define ESM_APU_EFFECT_GAIN_MASK (0xff << 0) -#define ESM_APU_TREMOLO_DEPTH_SHIFT 8 -#define ESM_APU_TREMOLO_DEPTH_MASK (0xf << 8) -#define ESM_APU_TREMOLO_RATE_SHIFT 12 -#define ESM_APU_TREMOLO_RATE_MASK (0xf << 12) - -/* reg 0x09 */ -/* bit 0-7 amplitude dest? */ -#define ESM_APU_AMPLITUDE_NOW_SHIFT 8 -#define ESM_APU_AMPLITUDE_NOW_MASK (0xff << 8) - -/* reg 0x0a */ -#define ESM_APU_POLAR_PAN_SHIFT 0 -#define ESM_APU_POLAR_PAN_MASK (0x3f << 0) -/* Polar Pan Control */ -#define ESM_APU_PAN_CENTER_CIRCLE 0x00 -#define ESM_APU_PAN_MIDDLE_RADIUS 0x01 -#define ESM_APU_PAN_OUTSIDE_RADIUS 0x02 - -#define ESM_APU_FILTER_TUNING_SHIFT 8 -#define ESM_APU_FILTER_TUNING_MASK (0xff << 8) - -/* reg 0x0b */ -#define ESM_APU_DATA_SRC_A_SHIFT 0 -#define ESM_APU_DATA_SRC_A_MASK (0x7f << 0) -#define ESM_APU_INV_POL_A (1 << 7) -#define ESM_APU_DATA_SRC_B_SHIFT 8 -#define ESM_APU_DATA_SRC_B_MASK (0x7f << 8) -#define ESM_APU_INV_POL_B (1 << 15) - -#define ESM_APU_VIBRATO_RATE_SHIFT 0 -#define ESM_APU_VIBRATO_RATE_MASK (0xf << 0) -#define ESM_APU_VIBRATO_DEPTH_SHIFT 4 -#define ESM_APU_VIBRATO_DEPTH_MASK (0xf << 4) -#define ESM_APU_VIBRATO_PHASE_SHIFT 8 -#define ESM_APU_VIBRATO_PHASE_MASK (0xff << 8) - -/* reg 0x0c */ -#define ESM_APU_RADIUS_SELECT (1 << 6) - -/* APU Filter Control */ -#define ESM_APU_FILTER_2POLE_LOPASS 0x00 -#define ESM_APU_FILTER_2POLE_BANDPASS 0x01 -#define ESM_APU_FILTER_2POLE_HIPASS 0x02 -#define ESM_APU_FILTER_1POLE_LOPASS 0x03 -#define ESM_APU_FILTER_1POLE_HIPASS 0x04 -#define ESM_APU_FILTER_OFF 0x05 - -/* APU ATFP Type */ -#define ESM_APU_ATFP_AMPLITUDE 0x00 -#define ESM_APU_ATFP_TREMELO 0x01 -#define ESM_APU_ATFP_FILTER 0x02 -#define ESM_APU_ATFP_PAN 0x03 - -/* APU ATFP Flags */ -#define ESM_APU_ATFP_FLG_OFF 0x00 -#define ESM_APU_ATFP_FLG_WAIT 0x01 -#define ESM_APU_ATFP_FLG_DONE 0x02 -#define ESM_APU_ATFP_FLG_INPROCESS 0x03 - /* capture mixing buffer size */ #define ESM_MEM_ALIGN 0x1000 @@ -729,33 +575,6 @@ static u16 apu_get_register(struct es1968 *chip, u16 channel, u8 reg) return v; } -#if 0 /* ASSP is not supported */ - -static void assp_set_register(struct es1968 *chip, u32 reg, u32 value) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->reg_lock, flags); - outl(reg, chip->io_port + ASSP_INDEX); - outl(value, chip->io_port + ASSP_DATA); - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -static u32 assp_get_register(struct es1968 *chip, u32 reg) -{ - unsigned long flags; - u32 value; - - spin_lock_irqsave(&chip->reg_lock, flags); - outl(reg, chip->io_port + ASSP_INDEX); - value = inl(chip->io_port + ASSP_DATA); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - return value; -} - -#endif - static void wave_set_register(struct es1968 *chip, u16 reg, u16 value) { unsigned long flags; -- 1.6.4.2

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[alsa-devel] [PATCH] ASoC: TWL4030: Module unloading fix
by Peter Ujfalusi 04 Feb '10

04 Feb '10

The module unloading path had several problems: - it freed up the private structure twice - it freed up the codec structure, which was allocated as part of the private structure - it did not freed up the reg_cache - it did not unregistered the dais and the codec Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e0106a5..b32aeb3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2152,8 +2152,6 @@ static int twl4030_soc_remove(struct platform_device *pdev) twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - kfree(codec->private_data); - kfree(codec); return 0; } @@ -2237,6 +2235,9 @@ static int __devexit twl4030_codec_remove(struct platform_device *pdev) { struct twl4030_priv *twl4030 = platform_get_drvdata(pdev); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + snd_soc_unregister_codec(&twl4030->codec); + kfree(twl4030->codec.reg_cache); kfree(twl4030); twl4030_codec = NULL; -- 1.6.6.1

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[alsa-devel] [PATCH 1/2] ASoC: Allow CODECs to ask soc-cache to suppress physical writes
by Mark Brown 03 Feb '10

03 Feb '10

Currently the soc-cache code will always write to the device, meaning that we need the device to be powered and active at pretty much all times the system is active. Allowing cache only writes lays some groundwork for future enhancements to allow devices to be put into a full off state when the audio subsystem is idle. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc.h | 1 + sound/soc/soc-cache.c | 26 ++++++++++++++++++++++++-- 2 files changed, 25 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4bbeb9f..4e8f14b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -423,6 +423,7 @@ struct snd_soc_codec { short reg_cache_step; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + unsigned int cache_only:1; /* Suppress writes to hardware */ /* dapm */ u32 pop_time; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 097e335..84b6916 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -38,6 +38,10 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -100,6 +104,10 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -153,6 +161,9 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + if (codec->cache_only) + return 0; + if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -181,6 +192,9 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, if (!snd_soc_codec_volatile_register(codec, reg)) reg_cache[reg] = value; + if (codec->cache_only) + return 0; + if (codec->hw_write(codec->control_data, data, 3) == 3) return 0; else @@ -193,10 +207,14 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; if (reg >= codec->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) + snd_soc_codec_volatile_register(codec, reg)) { + if (codec->cache_only) + return -EINVAL; + return codec->hw_read(codec, reg); - else + } else { return cache[reg]; + } } #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) @@ -294,6 +312,10 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 3); if (ret == 3) return 0; -- 1.6.6

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[alsa-devel] [PATCH] ASoC: fix compilation breakage in sound/soc/sh/fsi.c
by Guennadi Liakhovetski 03 Feb '10

03 Feb '10

ctrl_outl() has become void at some point, which breaks compilation of fsi.c. Make writing functions void, as their output is anyway not evaluated, and use __raw_writel and __raw_readl instead of deprecated ctrl_outl and ctrl_inl respectively. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- => linux-next / 2.6.34 sound/soc/sh/fsi.c | 40 +++++++++++++++++----------------------- 1 files changed, 17 insertions(+), 23 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ebf3588..3c36d24 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -120,35 +120,35 @@ struct fsi_master { ************************************************************************/ -static int __fsi_reg_write(u32 reg, u32 data) +static void __fsi_reg_write(u32 reg, u32 data) { /* valid data area is 24bit */ data &= 0x00ffffff; - return ctrl_outl(data, reg); + __raw_writel(data, reg); } static u32 __fsi_reg_read(u32 reg) { - return ctrl_inl(reg); + return __raw_readl(reg); } -static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +static void __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) { u32 val = __fsi_reg_read(reg); val &= ~mask; val |= data & mask; - return __fsi_reg_write(reg, val); + __fsi_reg_write(reg, val); } -static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) +static void fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) { if (reg > REG_END) - return -1; + return; - return __fsi_reg_write((u32)(fsi->base + reg), data); + __fsi_reg_write((u32)(fsi->base + reg), data); } static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) @@ -159,28 +159,25 @@ static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) return __fsi_reg_read((u32)(fsi->base + reg)); } -static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) +static void fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) { if (reg > REG_END) - return -1; + return; - return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); + __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); } -static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) +static void fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { - int ret; unsigned long flags; if ((reg < MREG_START) || (reg > MREG_END)) - return -1; + return; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_write((u32)(master->base + reg), data); + __fsi_reg_write((u32)(master->base + reg), data); spin_unlock_irqrestore(&master->lock, flags); - - return ret; } static u32 fsi_master_read(struct fsi_master *master, u32 reg) @@ -199,21 +196,18 @@ static u32 fsi_master_read(struct fsi_master *master, u32 reg) return ret; } -static int fsi_master_mask_set(struct fsi_master *master, +static void fsi_master_mask_set(struct fsi_master *master, u32 reg, u32 mask, u32 data) { - int ret; unsigned long flags; if ((reg < MREG_START) || (reg > MREG_END)) - return -1; + return; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + __fsi_reg_mask_set((u32)(master->base + reg), mask, data); spin_unlock_irqrestore(&master->lock, flags); - - return ret; } /************************************************************************ -- 1.6.2.4

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[alsa-devel] Access to bytes from the raw audio stream? (AC'97)
by John Graham 03 Feb '10

03 Feb '10

Hi there, I'm new to ALSA, and working on a project where I need to find out where I can see/modify the raw bytes coming in from/going out to a device. The device is an AC'97 codec (a philips UCB1400, if it makes any difference...) on an embedded board. I tried looking on the web, and found the tutorial at http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/, but that seems to be more about device management and configuration. So, can anybody tell me what functions and files in the sound/ directory of the linux kernel I need to look at to see/modify the raw data stream coming in? Am I just missing something in the tutorial? I'd appreciate any suggestions very much. Thanks, John Graham

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[alsa-devel] amixer scontents fails, amixer contents succeeds
by Eliot Blennerhassett 03 Feb '10

03 Feb '10

Greetings, Probably a bug in my driver (asihpi), but I don't understand how "amixer controls" can work while "amixer scontrols" fails. (with error code -EINVAL: "Mixer hw:1 load error: Invalid argument") Where should I start looking for why this happens? thanks -- Eliot

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