August 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] new board driver to connect bfin-5xx with ad1836 codec
by Barry Song 13 Aug '09

13 Aug '09

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 8 ++ sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 139 +++++++++++++++++++++++++++++++++++++ 3 files changed, 149 insertions(+), 0 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1836.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 8a4de4d..ac927ff 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -88,6 +88,14 @@ config SND_BF5XX_SOC_AC97 select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_AD1980 tristate "SoC AD1980/1 Audio support for BF5xx" depends on SND_BF5XX_AC97 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index f4d7607..87e3042 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -16,11 +16,13 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support +snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad1938-objs := bf5xx-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c new file mode 100644 index 0000000..c0ef965 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -0,0 +1,139 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad1836.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 4 2009 + * Description: Board driver for ad1836 sound chip + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad1836.h" +#include "bf5xx-sport.h" + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" + +static struct snd_soc_card bf5xx_ad1836; + +static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret = 0; + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI slots, 8 channels, mask is defined as slot seq + * 0x73625140 means audio channel 0 uses slot 0 + * channel 1 uses slot 4 + * channel 2 uses slot 1 + * channel 3 uses slot 5 + * ... + */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x73625140, 8, 32); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops bf5xx_ad1836_ops = { + .startup = bf5xx_ad1836_startup, + .hw_params = bf5xx_ad1836_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .cpu_dai = &bf5xx_tdm_dai, + .codec_dai = &ad1836_dai, + .ops = &bf5xx_ad1836_ops, +}; + +static struct snd_soc_card bf5xx_ad1836 = { + .name = "bf5xx_ad1836", + .platform = &bf5xx_tdm_soc_platform, + .dai_link = &bf5xx_ad1836_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad1836_snd_devdata = { + .card = &bf5xx_ad1836, + .codec_dev = &soc_codec_dev_ad1836, +}; + +static struct platform_device *bfxx_ad1836_snd_device; + +static int __init bf5xx_ad1836_init(void) +{ + int ret; + + bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad1836_snd_device) + return -ENOMEM; + + platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836_snd_devdata); + bf5xx_ad1836_snd_devdata.dev = &bfxx_ad1836_snd_device->dev; + ret = platform_device_add(bfxx_ad1836_snd_device); + + if (ret) + platform_device_put(bfxx_ad1836_snd_device); + + return ret; +} + +static void __exit bf5xx_ad1836_exit(void) +{ + platform_device_unregister(bfxx_ad1836_snd_device); +} + +module_init(bf5xx_ad1836_init); +module_exit(bf5xx_ad1836_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ALSA SoC AD1836 board driver"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

1 0

[alsa-devel] ASoC update for 2.6.32
by Mark Brown 13 Aug '09

13 Aug '09

The following changes since commit e0026beac0f1a6253488478b3f022ff243b012b9: Mark Brown (1): ASoC: Update WM9081 for tdm_slot() API change 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 Mark Brown (1): ASoC: Update AD1938 for new TDM slot API sound/soc/codecs/ad1938.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

2 1

[alsa-devel] Intel card master and PCM channels doesn't work
by Mantas Pilipavičius 13 Aug '09

13 Aug '09

http://www.alsa-project.org/db/?f=afa01dd48ed59d895d18407af62dbd1fa138686c Everything works great, I can play the sound, but PCM and Master channels don't work, so I can't increase or decrease volume. Only thing I can do is mute/unmute PCM channel

2 1

[alsa-devel] [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM
by Chaithrika U S 12 Aug '09

12 Aug '09

Define resources for McASP1 used on DA830/OMAP-L137 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. In this version of the patch, the init fucntion, resource and platform device definitions have neen moved to the devices-da8xx.c file. arch/arm/mach-davinci/board-da830-evm.c | 27 +++++++++++++++++++++ arch/arm/mach-davinci/da830.c | 7 +++-- arch/arm/mach-davinci/devices-da8xx.c | 36 ++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 15 +++++++++++ arch/arm/mach-davinci/include/mach/da8xx.h | 3 ++ 5 files changed, 85 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..a45340d 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 +113,13 @@ 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)); + + ret = da8xx_pinmux_setup(da830_mcasp1_pins); + if (ret) + pr_warning("da830_evm_init: mcasp1 mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(1, &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..19b2748 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), diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index 11c0971..fe0baaf 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -281,7 +281,43 @@ static struct platform_device da8xx_emac_device = { .resource = da8xx_emac_resources, }; +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, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); } + +void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) +{ + if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + da830_mcasp1_device.dev.platform_data = pdata; + platform_device_register(&da830_mcasp1_device); + } +} diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index cdf1f44..f3c97ac 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 McASP1 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 McASP1 */ +#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..30c5c40 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 da8xx_init_mcasp(int id, struct snd_platform_data *pdata); extern struct platform_device da8xx_serial_device; extern struct emac_platform_data da8xx_emac_pdata; @@ -100,6 +102,7 @@ extern const short da850_uart2_pins[]; extern const short da850_i2c0_pins[]; extern const short da850_i2c1_pins[]; extern const short da850_cpgmac_pins[]; +extern const short da850_mcasp_pins[]; int da8xx_pinmux_setup(const short pins[]); -- 1.5.6

2 1

[alsa-devel] [PATCH] alsa: hdsp - allow proc reporting with disconnected io box
by Tim Blechmann 12 Aug '09

12 Aug '09

the hdsp driver refuses to report any information via the proc interface, if the io box is not connected. with this patch, the content of the control and status registers is printed before the iobox check. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 37 ++++++++++++++++++++----------------- 1 files changed, 20 insertions(+), 17 deletions(-)

2 3

[alsa-devel] [PATCH] alsa-info: Use sysfs if available instead of dmidecode
by Ozan Çağlayan 12 Aug '09

12 Aug '09

Using sysfs for acquiring DMI data requires no root privileges. Use it if available instead of dmidecode. Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- utils/alsa-info.sh | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index 0ab1ddf..8f401c0 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -417,7 +417,12 @@ ARTSINST=$(which artsd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) JACKINST=$(which jackd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) DMIDECODE=$(which dmidecode 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) -if [ -x $DMIDECODE ]; then +#Check for DMI data +if [ -d /sys/class/dmi/id ]; then + # No root privileges are required when using sysfs method + DMI_SYSTEM_MANUFACTURER=$(cat /sys/class/dmi/id/sys_vendor 2>/dev/null) + DMI_SYSTEM_PRODUCT_NAME=$(cat /sys/class/dmi/id/product_name 2>/dev/null) +elif [ -x $DMIDECODE ]; then DMI_SYSTEM_MANUFACTURER=$($DMIDECODE -s system-manufacturer 2>/dev/null) DMI_SYSTEM_PRODUCT_NAME=$($DMIDECODE -s system-product-name 2>/dev/null) fi -- 1.6.3.4

2 1

[alsa-devel] [PATCH] add output/input widgets in ad1938 to make dac/adc dynamic PM work
by Barry Song 12 Aug '09

12 Aug '09

According to the function dapm_dac_check_power() in sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any output widget as sink. And according to dapm_adc_check_power(), adc power can't be on/off stand-alone without any input widget as source. So we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC to hope their power can be managed dynamically. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad1938.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index aafda74..9b8cc5c 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -99,11 +99,23 @@ static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_OUTPUT("DAC4OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), }; static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "DAC4OUT", "DAC4 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, }; /* -- 1.5.6.3

2 1

[alsa-devel] [PATCHv2 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 12 Aug '09

12 Aug '09

Hello guys, Here is version 2 of this series. There were lots of little changes and also one significant change. Here is a changelog: - I tried to avoid additional ifdefs in mcbsp code - The WAKEUPEN_ALL macro was removed - I added a REVISIT prefix for the additional delay on mcbsp start - New added constants are now following mcbsp.h style - While reading DMA operation mode of mcbsp, now it prints also some help text to explain the meaning of each value (even though you still write numbers). Besides, I re-wrote the code for ASoC part, now the threshold and dma op mode usage are un-glued from mcbsp code. I tried to do not leave omap-pcm.c dirty, but, there was still one needed callback for setting the mcbsp threshold, as that needs to be done before DMA is started. In this series I've left THRESHOLD mode as default for mcbsp 2 only. But as I saw in previous comments, this is still an open issue :-) Comments, as usual are welcome. Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 375 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 67 +++++- sound/soc/omap/omap-pcm.c | 33 +++- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 517 insertions(+), 14 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 23

[alsa-devel] [PATCH] ALSA: hda - fix noise issue when recording from digital mic with alc268
by Herton Ronaldo Krzesinski 12 Aug '09

12 Aug '09

With auto config model of alc268 realtek codec, it allows to select any of possible available digital microphone inputs when only one is available. For example, when only digital mic in nid 0x12 is available, on second input source it will allow you to select unavailable digital mic in nid 0x13. The problem is that selecting unavailable digital mic creates a source of noise when recording (I'm not sure if this happens on all machines with alc268 and only one digital mic input, but testing on a quanta uw1 netbook a lot of noise is introduced in recording from digital mic 0x12/first input source, when you select the unavailable digital mic 0x13 for capture source 0x24 in the second input source in mixer). Then to avoid noise when recording from digital mic with auto model in this case, prevent a digital mic input source to be selected if microphone is not available. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++----------------- 1 files changed, 36 insertions(+), 26 deletions(-) I'm attaching the codec info too here of the machine with the problem. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac3207e..7ff2934 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12232,32 +12232,40 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { struct hda_input_mux *imux = &spec->private_imux[0]; - int i, idx1; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - switch(cfg->input_pins[i]) { - case 0x18: - idx1 = 0; /* Mic 1 */ - break; - case 0x19: - idx1 = 1; /* Mic 2 */ - break; - case 0x1a: - idx1 = 2; /* Line In */ - break; - case 0x1c: - idx1 = 3; /* CD */ - break; - case 0x12: - case 0x13: - idx1 = 6; /* digital mics */ - break; - default: - continue; + int i, idx1, dmic_nid; + + dmic_nid = 0x12; + while (dmic_nid <= 0x13) { + for (i = 0; i < AUTO_PIN_LAST; i++) { + switch (cfg->input_pins[i]) { + case 0x18: + idx1 = 0; /* Mic 1 */ + break; + case 0x19: + idx1 = 1; /* Mic 2 */ + break; + case 0x1a: + idx1 = 2; /* Line In */ + break; + case 0x1c: + idx1 = 3; /* CD */ + break; + case 0x12: + case 0x13: + if (cfg->input_pins[i] != dmic_nid) + continue; + idx1 = 6; /* digital mics */ + break; + default: + continue; + } + imux->items[imux->num_items].label = + auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx1; + imux->num_items++; } - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = idx1; - imux->num_items++; + imux++; + dmic_nid++; } return 0; } @@ -12355,7 +12363,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) add_mixer(spec, alc268_beep_mixer); add_verb(spec, alc268_volume_init_verbs); - spec->num_mux_defs = 1; + spec->num_mux_defs = 2; spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); @@ -12665,6 +12673,8 @@ static int patch_alc268(struct hda_codec *codec) for (i = 0; i < spec->num_adc_nids; i++) snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], 0, AC_VERB_SET_CONNECT_SEL, + i < spec->num_mux_defs ? + spec->input_mux[i].items[0].index : spec->input_mux->items[0].index); } -- 1.6.4

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM
by Chaithrika U S 11 Aug '09

11 Aug '09

There is one instance of McASP on DA850/OMAP-L138 SoC. This is connected to TLV320AIC3106 codec for audio playback and capture. This patch adds audio support on this platform. Some of the structure prefix names which are common for DA830/OMAP-L137 EVM and DA850/OMAP-L138 EVM have been renamed to da8xx from da830. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/Kconfig | 13 +++++++++++-- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +++++++++++++++++++++++------ 3 files changed, 35 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 677a538..4dfd4ad 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -43,12 +43,21 @@ config SND_DAVINCI_SOC_SFFSDR Lyrtech SFFSDR board. config SND_DA830_SOC_EVM - tristate "SoC Audio support for DA830/OMAPL137 EVM" + tristate "SoC Audio support for DA830/OMAP-L137 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM select SND_DAVINCI_SOC_MCASP select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DA830/OMAPL137 EVM + DA830/OMAP-L137 EVM + +config SND_DA850_SOC_EVM + tristate "SoC Audio support for DA850/OMAP-L138 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DA850/OMAP-L138 EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 5e2195f..a6939d7 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -14,4 +14,5 @@ snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index a5a9212..46c1b0c 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -54,7 +54,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; - else if (machine_is_davinci_da830_evm()) + else if (machine_is_davinci_da830_evm() || + machine_is_davinci_da850_evm()) sysclk = 24576000; else @@ -165,7 +166,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_ops, }, }; -static struct snd_soc_dai_link da830_evm_dai = { +static struct snd_soc_dai_link da8xx_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], @@ -191,8 +192,15 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { }; static struct snd_soc_card da830_snd_soc_card = { - .name = "DA830 EVM", - .dai_link = &da830_evm_dai, + .name = "DA830/OMAP-L137 EVM", + .dai_link = &da8xx_evm_dai, + .platform = &davinci_soc_platform, + .num_links = 1, +}; + +static struct snd_soc_card da850_snd_soc_card = { + .name = "DA850/OMAP-L138 EVM", + .dai_link = &da8xx_evm_dai, .platform = &davinci_soc_platform, .num_links = 1, }; @@ -209,7 +217,7 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; -static struct aic3x_setup_data da830_evm_aic3x_setup = { +static struct aic3x_setup_data da8xx_evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x18, }; @@ -232,7 +240,13 @@ static struct snd_soc_device dm6467_evm_snd_devdata = { static struct snd_soc_device da830_evm_snd_devdata = { .card = &da830_snd_soc_card, .codec_dev = &soc_codec_dev_aic3x, - .codec_data = &da830_evm_aic3x_setup, + .codec_data = &da8xx_evm_aic3x_setup, +}; + +static struct snd_soc_device da850_evm_snd_devdata = { + .card = &da850_snd_soc_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &da8xx_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; @@ -255,6 +269,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_da830_evm()) { evm_snd_dev_data = &da830_evm_snd_devdata; index = 1; + } else if (machine_is_davinci_da850_evm()) { + evm_snd_dev_data = &da850_evm_snd_devdata; + index = 0; } else return -EINVAL; -- 1.5.6

1 0