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[alsa-devel] [PATCH 2/6] s3c2443-ac97: convert semaphore to mutex
by Uwe Kleine-König 08 Aug '09

08 Aug '09

This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig): CC [M] sound/soc/s3c24xx/s3c2443-ac97.o sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX' sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read': sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function) sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.) sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write': sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function) Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Timur Tabi <timur(a)freescale.com> Cc: Eric Miao <eric.miao(a)marvell.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 3f03d5d..bf16f20 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -47,7 +47,7 @@ static struct s3c24xx_ac97_info s3c24xx_ac97; static DECLARE_COMPLETION(ac97_completion); static u32 codec_ready; -static DECLARE_MUTEX(ac97_mutex); +static DEFINE_MUTEX(ac97_mutex); static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, unsigned short reg) @@ -56,7 +56,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, u32 ac_codec_cmd; u32 stat, addr, data; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -79,7 +79,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," " rep addr = %02x\n", reg, addr); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); return (unsigned short)data; } @@ -90,7 +90,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, u32 ac_glbctrl; u32 ac_codec_cmd; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -109,7 +109,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); } -- 1.6.3.3

2 1

[alsa-devel] [PATCH] ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6
by Candelaria Villareal, Jorge 07 Aug '09

07 Aug '09

Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6 line, controlled by register INTBR_PMBR1. Machine driver takes care of enabling gpio line through i2c and codec driver manipulates the line during headset ramp up/down sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index c51594d..f7e5b74 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -24,6 +24,7 @@ #include <linux/clk.h> #include <linux/platform_device.h> +#include <linux/i2c/twl4030.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -39,6 +40,9 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" +#define TWL4030_INTBR_PMBR1 0x0D +#define EXTMUTE(value) (value << 2) + static struct snd_soc_card snd_soc_sdp3430; static int sdp3430_hw_params(struct snd_pcm_substream *substream, @@ -280,6 +284,7 @@ static struct snd_soc_card snd_soc_sdp3430 = { static struct twl4030_setup_data twl4030_setup = { .ramp_delay_value = 3, .sysclk = 26000, + .hs_extmute = 1, }; /* Audio subsystem */ @@ -312,6 +317,10 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + /* Set TWL4030 GPIO6 as EXTMUTE signal */ + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), + TWL4030_MODULE_INTBR); + ret = platform_device_add(sdp3430_snd_device); if (ret) goto err1; -- 1.6.0.4

3 6

[alsa-devel] [PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 07 Aug '09

07 Aug '09

We've encountered strange behavior in the alsamixer settings using the wm9712 codec. If we unmute the headphone output and then unmute the PCM output, the headphone output gets reset to mute in the hardware register. At this point the hardware register does not match the value in the register cache. I've spent some time debugging this, and the headphone setting is set outside of any code path that would call the ac97_write() routine. As best as I can tell, there is something strange going on in hardware. I've provided this patch that works around the problem. Have any of you seen this before? Is this patch the right approach? - John The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task))

5 14

[alsa-devel] Fixed module builds of soc/fsl/*.c
by Takashi Iwai 07 Aug '09

07 Aug '09

JFYI, I merged the patch below now. Takashi --- From afc5e65245255a268ab22a20477ed2c9f2cdfcd3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Fri, 7 Aug 2009 16:33:53 +0200 Subject: ASoC: Add missing DRV_NAME definitions for fsl/* drivers Module builds are broken due to missing DRV_NAME for efika-audio-fabric and pcm030-audio-fabric. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/fsl/efika-audio-fabric.c | 2 ++ sound/soc/fsl/pcm030-audio-fabric.c | 2 ++ 2 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index 85b0e75..3326e2a 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -30,6 +30,8 @@ #include "mpc5200_psc_ac97.h" #include "../codecs/stac9766.h" +#define DRV_NAME "efika-audio-fabric" + static struct snd_soc_device device; static struct snd_soc_card card; diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 8766f7a..b928ef7 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -30,6 +30,8 @@ #include "mpc5200_psc_ac97.h" #include "../codecs/wm9712.h" +#define DRV_NAME "pcm030-audio-fabric" + static struct snd_soc_device device; static struct snd_soc_card card; -- 1.6.3.3

3 2

[alsa-devel] [PATCH] ARM: DaVinci: Audio support for DA830 EVM
by Chaithrika U S 07 Aug '09

07 Aug '09

Define resources for McASP1 used on DA830 EVM, add platform device defintion, initialization function. Additionally, this patch also adds version and FIFO related members to platform data structure. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to DaVinci GIT tree arch/arm/mach-davinci/board-da830-evm.c | 22 +++++++++++++ arch/arm/mach-davinci/da830.c | 48 ++++++++++++++++++++++++++-- arch/arm/mach-davinci/include/mach/asp.h | 15 +++++++++ arch/arm/mach-davinci/include/mach/da8xx.h | 2 + 4 files changed, 84 insertions(+), 3 deletions(-) diff --git a/arch/arm/mach-davinci/board-da830-evm.c b/arch/arm/mach-davinci/board-da830-evm.c index 9025669..4643795 100644 --- a/arch/arm/mach-davinci/board-da830-evm.c +++ b/arch/arm/mach-davinci/board-da830-evm.c @@ -23,6 +23,7 @@ #include <mach/irqs.h> #include <mach/cp_intc.h> #include <mach/da8xx.h> +#include <mach/asp.h> #define DA830_EVM_PHY_MASK 0x0 #define DA830_EVM_MDIO_FREQUENCY 2200000 /* PHY bus frequency */ @@ -51,6 +52,25 @@ static struct davinci_uart_config da830_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +static u8 da830_iis_serializer_direction[] = { + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, TX_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da830_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da830_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da830_iis_serializer_direction, + .eventq_no = EVENTQ_0, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + static __init void da830_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -93,6 +113,8 @@ static __init void da830_evm_init(void) davinci_serial_init(&da830_evm_uart_config); i2c_register_board_info(1, da830_evm_i2c_devices, ARRAY_SIZE(da830_evm_i2c_devices)); + + da830_init_mcasp1(&da830_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da830.c b/arch/arm/mach-davinci/da830.c index 3a9b634..a160a7f 100644 --- a/arch/arm/mach-davinci/da830.c +++ b/arch/arm/mach-davinci/da830.c @@ -23,6 +23,7 @@ #include <mach/common.h> #include <mach/time.h> #include <mach/da8xx.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -411,9 +412,9 @@ static struct davinci_clk da830_clks[] = { CLK("eqep.0", NULL, &eqep0_clk), CLK("eqep.1", NULL, &eqep1_clk), CLK("da830_lcdc", NULL, &lcdc_clk), - CLK("soc-audio.0", NULL, &mcasp0_clk), - CLK("soc-audio.1", NULL, &mcasp1_clk), - CLK("soc-audio.2", NULL, &mcasp2_clk), + CLK("davinci-mcasp.0", NULL, &mcasp0_clk), + CLK("davinci-mcasp.1", NULL, &mcasp1_clk), + CLK("davinci-mcasp.2", NULL, &mcasp2_clk), CLK("musb_hdrc", NULL, &usb20_clk), CLK(NULL, "aemif", &aemif_clk), CLK(NULL, "aintc", &aintc_clk), @@ -1198,6 +1199,47 @@ static struct davinci_soc_info davinci_soc_info_da830 = { .emac_pdata = &da8xx_emac_pdata, }; +static struct resource da830_mcasp1_resources[] = { + { + .name = "mcasp1", + .start = DAVINCI_DA830_MCASP1_REG_BASE, + .end = DAVINCI_DA830_MCASP1_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA830_DMA_MCASP1_AXEVT, + .end = DAVINCI_DA830_DMA_MCASP1_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA830_DMA_MCASP1_AREVT, + .end = DAVINCI_DA830_DMA_MCASP1_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da830_mcasp1_device = { + .name = "davinci-mcasp", + .id = 1, + .num_resources = ARRAY_SIZE(da830_mcasp1_resources), + .resource = da830_mcasp1_resources, +}; + +void __init da830_init_mcasp1(struct snd_platform_data *pdata) +{ + int ret; + + ret = da8xx_pinmux_setup(da830_mcasp1_pins); + if (ret) + pr_warning("da830_evm_init: mcasp1 mux setup failed: %d\n", + ret); + + da830_mcasp1_device.dev.platform_data = pdata; + platform_device_register(&da830_mcasp1_device); +} + void __init da830_init(void) { davinci_common_init(&davinci_soc_info_da830); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index cdf1f44..4df3839 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da830 register banks */ +#define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 + /* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 @@ -26,6 +29,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da830 */ +#define DAVINCI_DA830_DMA_MCASP1_AREVT 2 +#define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 + /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT @@ -43,6 +50,14 @@ struct snd_platform_data { u8 op_mode; u8 num_serializer; u8 *serial_dir; + u8 version; + u8 txnumevt; + u8 rxnumevt; +}; + +enum { + MCASP_VERSION_1 = 0, /* DM646x */ + MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; #define INACTIVE_MODE 0 diff --git a/arch/arm/mach-davinci/include/mach/da8xx.h b/arch/arm/mach-davinci/include/mach/da8xx.h index a8cb570..c2aabf1 100644 --- a/arch/arm/mach-davinci/include/mach/da8xx.h +++ b/arch/arm/mach-davinci/include/mach/da8xx.h @@ -15,6 +15,7 @@ #include <mach/edma.h> #include <mach/i2c.h> #include <mach/emac.h> +#include <mach/asp.h> /* * The cp_intc interrupt controller for the da8xx isn't in the same @@ -65,6 +66,7 @@ int da8xx_register_edma(void); int da8xx_register_i2c(int instance, struct davinci_i2c_platform_data *pdata); int da8xx_register_watchdog(void); int da8xx_register_emac(void); +void __init da830_init_mcasp1(struct snd_platform_data *pdata); extern struct platform_device da8xx_serial_device; extern struct emac_platform_data da8xx_emac_pdata; -- 1.5.6

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[alsa-devel] suspicious code in aplay.c
by Dmitri Vorobiev 07 Aug '09

07 Aug '09

Hello, The following code in alsa-utils/aplay/aplay.c looks suspicious to me: 1215 if (!quiet_mode) 1216 fprintf(stderr, _("Suspended. Trying resume. ")); fflush(stderr); Is it really necessary to flush the stderr stream even if there was no call to fprintf()? Perhaps these lines have to be like that: if (!quiet_mode) { fprintf(stderr, _("Suspended. Trying resume. ")); fflush(stderr); } Thanks, Dmitri P.S. I'm not subscribed to the mailing list, so please Cc: me on reply.

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 07 Aug '09

07 Aug '09

The following changes since commit 6d7f68a1eab3d5b3c0a6a5cf9434c77cd3a8c6ac: Janusz Krzysztofik (1): ASoC: add support for Amstrad E3 (Delta) machine are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 There's a merge up of a for-2.6.31 change in here which I've not pushed yet since the change is relatively invasive for so late in the release cycle (the OMAP simultaneous audio change). I didn't redo the branch in case it does get pushed; worst case it's just an extra merge commit which isn't the end of the world. Daniel Mack (1): ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device Janusz Krzysztofik (3): ASoC: CX20442: add some debugging ASoC: CX20442: simplify codec controller usage TTY/ASoC: Rename N_AMSDELTA line discipline to N_V253 Jarkko Nikula (1): ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/stop John Bonesio (1): ASoC: MPC5200: Support for buffer wrap around Mark Brown (12): ASoC: Begin to factor out register cache I/O functions ASoC: Remove use of hw_read from TLV320AIC3x driver ASoC: Factor out WM8580 register cache code Merge branch 'tlv320aic3x' into reg-cache ASoC: Add I/O control bus information to factored out cache setup ASoC: Factor out I2C 8 bit address 16 bit data I/O ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs ASoC: Factor out 7 bit register 9 bit data SPI write ASoC: Existing S3C24xx AC97 drivers should depend on S3C24xx ASoC: Add WM8776 CODEC driver Merge branch 'for-2.6.31' into for-2.6.32 Merge branch 'reg-cache' into for-2.6.32 Troy Kisky (2): ASoC: DaVinci: i2s: don't bounce through rtd to get dai ASoC: DaVinci: pcm, constrain buffer size to multiple of period arch/arm/plat-omap/include/mach/mcbsp.h | 4 +- arch/arm/plat-omap/mcbsp.c | 50 ++- include/linux/tty.h | 2 +- include/sound/soc.h | 11 +- sound/soc/Makefile | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4270.c | 27 +- sound/soc/codecs/cx20442.c | 3 +- sound/soc/codecs/tlv320aic3x.c | 11 +- sound/soc/codecs/wm8510.c | 174 +++----- sound/soc/codecs/wm8523.c | 98 +--- sound/soc/codecs/wm8580.c | 154 ++----- sound/soc/codecs/wm8728.c | 111 +---- sound/soc/codecs/wm8731.c | 141 ++----- sound/soc/codecs/wm8750.c | 154 ++----- sound/soc/codecs/wm8776.c | 781 +++++++++++++++++++++++++++++++ sound/soc/codecs/wm8776.h | 51 ++ sound/soc/codecs/wm8900.c | 319 +++++-------- sound/soc/codecs/wm8903.c | 244 ++++------- sound/soc/codecs/wm8940.c | 143 ++---- sound/soc/codecs/wm8960.c | 200 +++----- sound/soc/codecs/wm8961.c | 231 ++++------ sound/soc/codecs/wm8971.c | 127 ++---- sound/soc/codecs/wm8988.c | 138 ++---- sound/soc/codecs/wm8990.c | 194 +++----- sound/soc/codecs/wm9081.c | 229 ++++------ sound/soc/davinci/davinci-i2s.c | 14 +- sound/soc/davinci/davinci-pcm.c | 5 + sound/soc/fsl/mpc5200_dma.c | 17 + sound/soc/omap/ams-delta.c | 6 +- sound/soc/omap/omap-mcbsp.c | 10 +- sound/soc/s3c24xx/Kconfig | 4 +- sound/soc/soc-cache.c | 218 +++++++++ 34 files changed, 2048 insertions(+), 1831 deletions(-) create mode 100644 sound/soc/codecs/wm8776.c create mode 100644 sound/soc/codecs/wm8776.h create mode 100644 sound/soc/soc-cache.c

2 1

[alsa-devel] [PATCH 0/2] Give the N_AMSDELTA line discipline more generic name
by Janusz Krzysztofik 07 Aug '09

07 Aug '09

This series of patches renames the N_AMSDELTA line discipline, used by the Amstrad Delta ASoC machine driver and the CX20442 ASoC codec driver for controlling the CX20442 voice codec over the CX81801 voice modem, giving it a more generic name: N_V253. The name has been chosen after the voice modem standard codename V.253. I think that even if the code of the line discipline is currently implemented inside those two specific drivers, the idea, if not the code, could be reused with other similar hardware configurations, if necessary. As the line discipline is conceived to be registered from a machine driver, there should be no conflicts, even if used by several different drivers that happen to be be built-in side-by-side. [PATCH 1/2] TTY: Rename N_AMSDELTA line discipline to N_V253 [PATCH 2/2] ASoC: AMS_DELTA: Use updated line discipline name Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

2 1

[alsa-devel] Audio support for Simtec Hermes
by Ben Dooks 07 Aug '09

07 Aug '09

Add support for the TLV320AIC33 codec on the Simtec Hermes board, using the core work done for the TLV320AIC23 codecs. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> --- sound/soc/s3c24xx/Kconfig | 7 + sound/soc/s3c24xx/Makefile | 3 sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 151 ++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+) Index: b/sound/soc/s3c24xx/Kconfig =================================================================== --- a/sound/soc/s3c24xx/Kconfig 2009-08-04 12:27:06.000000000 +0100 +++ b/sound/soc/s3c24xx/Kconfig 2009-08-04 12:29:22.000000000 +0100 @@ -127,3 +127,10 @@ config SND_S3C24XX_SOC_SIMTEC_TLV320AIC2 select SND_S3C24XX_SOC_I2S select SND_SOC_TLV320AIC23 select SND_S3C24XX_SOC_SIMTEC + +config SND_S3C24XX_SOC_SIMTEC_HERMES + tristate "SoC I2S Audio support for Simtec Hermes board" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC3X + select SND_S3C24XX_SOC_SIMTEC Index: b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c 2009-08-04 12:42:00.000000000 +0100 @@ -0,0 +1,151 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic3x.h" + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Out", NULL), + SND_SOC_DAPM_LINE("GSM In", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("ZV", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ + + { "Headphone Jack", NULL, "HPLOUT" }, + { "Headphone Jack", NULL, "HPLCOM" }, + { "Headphone Jack", NULL, "HPROUT" }, + { "Headphone Jack", NULL, "HPRCOM" }, + + /* ZV connected to Line1 */ + + { "LINE1L", NULL, "ZV" }, + { "LINE1R", NULL, "ZV" }, + + /* Line In connected to Line2 */ + + { "LINE2L", NULL, "Line In" }, + { "LINE2R", NULL, "Line In" }, + + /* Microphone connected to MIC3R and MIC_BIAS */ + + { "MIC3L", NULL, "Mic" }, + + /* GSM connected to MONO_LOUT and MIC3L (in) */ + + { "GSM Out", NULL, "MONO_LOUT" }, + { "MIC3L", NULL, "GSM In" }, + + /* Speaker is connected to LINEOUT{LN,LP,RN,RP} */ +}; + +/** + * simtec_hermes_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_hermes_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct aic3x_setup_data codec_setup = { + .i2c_bus = 0, + .i2c_already = 1, +}; + +static struct snd_soc_dai_link simtec_dai_aic33 = { + .name = "tlv320aic33", + .stream_name = "TLV320AIC33", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &aic3x_dai, + .init = simtec_hermes_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic33 = { + .name = "Simtec-Hermes", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic33, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic33 = { + .card = &snd_soc_machine_simtec_aic33, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &codec_setup, +}; + +static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) +{ + dev_info(&pd->dev, "probing....\n"); + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); +} + +static struct platform_driver simtec_audio_hermes_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-hermes-snd", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_hermes_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); + +static int __init simtec_hermes_modinit(void) +{ + return platform_driver_register(&simtec_audio_hermes_platdrv); +} + +static void __exit simtec_hermes_modexit(void) +{ + platform_driver_unregister(&simtec_audio_hermes_platdrv); +} + +module_init(simtec_hermes_modinit); +module_exit(simtec_hermes_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); Index: b/sound/soc/s3c24xx/Makefile =================================================================== --- a/sound/soc/s3c24xx/Makefile 2009-08-04 12:27:06.000000000 +0100 +++ b/sound/soc/s3c24xx/Makefile 2009-08-04 12:29:22.000000000 +0100 @@ -24,6 +24,7 @@ snd-soc-ln2440sbc-alc650-objs := ln2440s snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o @@ -36,4 +37,6 @@ obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_A obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o + -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

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[alsa-devel] PROC_COEF/PROC_INDEX question
by Wu Fengguang 07 Aug '09

07 Aug '09

Hello, I'm not sure of the usage for GET_PROC_COEF/GET_COEF_INDEX, and found that their return value is unstable (changing all the time) for codec: Codec: Realtek ALC888 Address: 2 Function Id: 0x1 Vendor Id: 0x10ec0888 Subsystem Id: 0x80860023 Revision Id: 0x100202 For example: # hda-verb /dev/snd/hwC0D2 0x20 GET_PROC_COEF 0 nid = 0x20, verb = 0xc00, param = 0x0 value = 0x80 # hda-verb /dev/snd/hwC0D2 0x20 GET_COEF_INDEX 0 nid = 0x20, verb = 0xd00, param = 0x0 value = 0xe # hda-verb /dev/snd/hwC0D2 0x20 GET_PROC_COEF 0 nid = 0x20, verb = 0xc00, param = 0x0 value = 0x9c00 # hda-verb /dev/snd/hwC0D2 0x20 GET_COEF_INDEX 0 nid = 0x20, verb = 0xd00, param = 0x0 value = 0xf I wonder if this is the normal case? Thanks, Fengguang

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