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

Re: [alsa-devel] [PATCH] [snd-usb-audio] Add match for JMTek USB sound card
by Takashi Iwai 30 Oct '09

30 Oct '09

At Mon, 12 Oct 2009 10:03:53 +0200, Lubomir Rintel wrote: > > On Mon, 2009-10-12 at 09:18 +0200, Takashi Iwai wrote: > > At Mon, 12 Oct 2009 08:46:26 +0200, > > Clemens Ladisch wrote: > > > > > > Lubomir Rintel wrote: > > > > Entry for JMTek LLC., SSS USB Headphone Set in the quirk table. > > > > > > Please add an explaination why this entry is needed. At first glance, > > > this entry seems to describe a class-compliant device that should not > > > need a quirk. > > > > Ah right. It'd really helpful if Lubomir can give more details... > > A 2.6.31.1-based kernel on my Fedora 12 workstation at hone seemed to > require that, only the input driver attached to the device when plugged > in, the snd-usb-audio didn't seem to load and did not care about the > device when loaded manually. I did not have an idea why, since "alias: > usb:v*p*d*dc*dsc*dp*ic01isc01ip*" really seemed to match my device. > Nevertheless, after finding out that adding an entry to the quirk table > solves my problem I concluded that my understanding (or lack of thereof) > was wrong and that alias is really not meant to match my device. > > Now I plugged the adapter into my work lappy with 2.6.30.8-based Fedora > 11 and the audio interfaces on my adapter got instantly recognized and > claimed by snd-usb-audio (and the input interface by the input > subsystem), without modifying anything. > > Any clues what could have gone wrong then? Hm, I don't see any affecting changes between 2.6.30 and 31. Just as a test, could you copy sound/usb/*.[ch] from 2.6.30 tree to 2.6.31 or 32-rc and check whether the same problem appears? Takashi

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[alsa-devel] [PATCH] ASoC: sh: FSI: Add capture support
by Kuninori Morimoto 30 Oct '09

30 Oct '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi.c | 93 ++++++++++++++++++++++++++++++++++++++++++++++++---- 1 files changed, 86 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9742a28..e1a3d1a 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -447,6 +447,75 @@ static int fsi_data_push(struct fsi_priv *fsi) return 0; } +static int fsi_data_pop(struct fsi_priv *fsi) +{ + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream = NULL; + int free; + int fifo_fill; + int width; + u8 *start; + int i; + + if (!fsi || + !fsi->substream || + !fsi->substream->runtime) + return -EINVAL; + + runtime = fsi->substream->runtime; + + /* FSI FIFO has limit. + * So, this driver can not send periods data at a time + */ + if (fsi->byte_offset >= + fsi->period_len * (fsi->periods + 1)) { + + substream = fsi->substream; + fsi->periods = (fsi->periods + 1) % runtime->periods; + + if (0 == fsi->periods) + fsi->byte_offset = 0; + } + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + /* get free space for alsa */ + free = (fsi->buffer_len - fsi->byte_offset) / width; + + /* get recv size */ + fifo_fill = fsi_get_fifo_residue(fsi, 0); + + if (free < fifo_fill) + fifo_fill = free; + + start = runtime->dma_area; + start += fsi->byte_offset; + + switch (width) { + case 2: + for (i = 0; i < fifo_fill; i++) + *((u16 *)start + i) = + (u16)(fsi_reg_read(fsi, DIDT) >> 8); + break; + case 4: + for (i = 0; i < fifo_fill; i++) + *((u32 *)start + i) = fsi_reg_read(fsi, DIDT); + break; + default: + return -EINVAL; + } + + fsi->byte_offset += fifo_fill * width; + + fsi_irq_enable(fsi, 0); + + if (substream) + snd_pcm_period_elapsed(substream); + + return 0; +} + static irqreturn_t fsi_interrupt(int irq, void *data) { u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; @@ -460,6 +529,10 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_data_push(&master->fsia); if (int_st & INT_B_OUT) fsi_data_push(&master->fsib); + if (int_st & INT_A_IN) + fsi_data_pop(&master->fsia); + if (int_st & INT_B_IN) + fsi_data_pop(&master->fsib); fsi_master_write(INT_ST, 0x0000000); @@ -612,16 +685,12 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret = 0; - /* capture not supported */ - if (!is_play) - return -ENODEV; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: fsi_stream_push(fsi, substream, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); - ret = fsi_data_push(fsi); + ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); break; case SNDRV_PCM_TRIGGER_STOP: fsi_irq_disable(fsi, is_play); @@ -757,7 +826,12 @@ struct snd_soc_dai fsi_soc_dai[] = { .channels_min = 1, .channels_max = 8, }, - /* capture not supported */ + .capture = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, .ops = &fsi_dai_ops, }, { @@ -769,7 +843,12 @@ struct snd_soc_dai fsi_soc_dai[] = { .channels_min = 1, .channels_max = 8, }, - /* capture not supported */ + .capture = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, .ops = &fsi_dai_ops, }, }; -- 1.6.0.4

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[alsa-devel] [PATCH 4/4] Audio: Modifying the defconfig for OMAP3517 / AM3517 EVM
by Anuj Aggarwal 29 Oct '09

29 Oct '09

Enabling Audio for OMAP3517 / AM3517 EVM in the defconfig. Also enabling McBSP and I2C drivers as both are required for audio on this EVM. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- arch/arm/configs/am3517_evm_defconfig | 78 +++++++++++++++++++++++++++++++- 1 files changed, 75 insertions(+), 3 deletions(-) diff --git a/arch/arm/configs/am3517_evm_defconfig b/arch/arm/configs/am3517_evm_defconfig index 787f295..316cbaa 100644 --- a/arch/arm/configs/am3517_evm_defconfig +++ b/arch/arm/configs/am3517_evm_defconfig @@ -202,7 +202,7 @@ CONFIG_ARCH_OMAP3=y # CONFIG_OMAP_DEBUG_CLOCKDOMAIN is not set CONFIG_OMAP_RESET_CLOCKS=y # CONFIG_OMAP_MUX is not set -# CONFIG_OMAP_MCBSP is not set +CONFIG_OMAP_MCBSP=y # CONFIG_OMAP_MBOX_FWK is not set # CONFIG_OMAP_MPU_TIMER is not set CONFIG_OMAP_32K_TIMER=y @@ -590,7 +590,46 @@ CONFIG_HW_RANDOM=y # CONFIG_R3964 is not set # CONFIG_RAW_DRIVER is not set # CONFIG_TCG_TPM is not set -# CONFIG_I2C is not set +CONFIG_I2C=y +CONFIG_I2C_BOARDINFO=y +CONFIG_I2C_COMPAT=y +# CONFIG_I2C_CHARDEV is not set +CONFIG_I2C_HELPER_AUTO=y + +# +# I2C Hardware Bus support +# + +# +# I2C system bus drivers (mostly embedded / system-on-chip) +# +# CONFIG_I2C_DESIGNWARE is not set +# CONFIG_I2C_GPIO is not set +# CONFIG_I2C_OCORES is not set +CONFIG_I2C_OMAP=y +# CONFIG_I2C_SIMTEC is not set + +# +# External I2C/SMBus adapter drivers +# +# CONFIG_I2C_PARPORT_LIGHT is not set +# CONFIG_I2C_TAOS_EVM is not set + +# +# Other I2C/SMBus bus drivers +# +# CONFIG_I2C_PCA_PLATFORM is not set +# CONFIG_I2C_STUB is not set + +# +# Miscellaneous I2C Chip support +# +# CONFIG_DS1682 is not set +# CONFIG_SENSORS_TSL2550 is not set +# CONFIG_I2C_DEBUG_CORE is not set +# CONFIG_I2C_DEBUG_ALGO is not set +# CONFIG_I2C_DEBUG_BUS is not set +# CONFIG_I2C_DEBUG_CHIP is not set # CONFIG_SPI is not set # @@ -666,7 +705,40 @@ CONFIG_SSB_POSSIBLE=y # # CONFIG_VGA_CONSOLE is not set CONFIG_DUMMY_CONSOLE=y -# CONFIG_SOUND is not set +CONFIG_SOUND=y +# CONFIG_SOUND_OSS_CORE is not set +CONFIG_SND=y +CONFIG_SND_TIMER=y +CONFIG_SND_PCM=y +CONFIG_SND_JACK=y +# CONFIG_SND_SEQUENCER is not set +# CONFIG_SND_MIXER_OSS is not set +# CONFIG_SND_PCM_OSS is not set +# CONFIG_SND_HRTIMER is not set +# CONFIG_SND_DYNAMIC_MINORS is not set +CONFIG_SND_SUPPORT_OLD_API=y +CONFIG_SND_VERBOSE_PROCFS=y +# CONFIG_SND_VERBOSE_PRINTK is not set +# CONFIG_SND_DEBUG is not set +# CONFIG_SND_RAWMIDI_SEQ is not set +# CONFIG_SND_OPL3_LIB_SEQ is not set +# CONFIG_SND_OPL4_LIB_SEQ is not set +# CONFIG_SND_SBAWE_SEQ is not set +# CONFIG_SND_EMU10K1_SEQ is not set +CONFIG_SND_DRIVERS=y +# CONFIG_SND_DUMMY is not set +# CONFIG_SND_MTPAV is not set +# CONFIG_SND_SERIAL_U16550 is not set +# CONFIG_SND_MPU401 is not set +CONFIG_SND_ARM=y +CONFIG_SND_SOC=y +CONFIG_SND_OMAP_SOC=y +CONFIG_SND_OMAP_SOC_MCBSP=y +CONFIG_SND_OMAP_SOC_AM3517EVM=y +CONFIG_SND_SOC_I2C_AND_SPI=y +# CONFIG_SND_SOC_ALL_CODECS is not set +CONFIG_SND_SOC_TLV320AIC23=y +# CONFIG_SOUND_PRIME is not set # CONFIG_HID_SUPPORT is not set # CONFIG_USB_SUPPORT is not set # CONFIG_MMC is not set -- 1.6.2.4

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[alsa-devel] [PATCH 3/4] Audio: Modifying board-evm file for audio codec
by Anuj Aggarwal 29 Oct '09

29 Oct '09

Adding I2C bus registration code in board-evm file for OMAP3517 / AM3517 EVM for AIC23 audio codec. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- arch/arm/mach-omap2/board-am3517evm.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-omap2/board-am3517evm.c b/arch/arm/mach-omap2/board-am3517evm.c index 23cf949..5f5f3ee 100644 --- a/arch/arm/mach-omap2/board-am3517evm.c +++ b/arch/arm/mach-omap2/board-am3517evm.c @@ -30,6 +30,25 @@ #include <plat/common.h> /* + * I2C + */ +static struct i2c_board_info __initdata am3517evm_i2c_boardinfo[] = { + { + I2C_BOARD_INFO("tlv320aic23", 0x1A), + }, +}; + +static int __init am3517_evm_i2c_init(void) +{ + omap_register_i2c_bus(1, 400, NULL, 0); + omap_register_i2c_bus(2, 400, am3517evm_i2c_boardinfo, + ARRAY_SIZE(am3517evm_i2c_boardinfo)); + omap_register_i2c_bus(3, 400, NULL, 0); + + return 0; +} + +/* * Board initialization */ static struct omap_board_config_kernel am3517_evm_config[] __initdata = { @@ -50,6 +69,8 @@ static void __init am3517_evm_init_irq(void) static void __init am3517_evm_init(void) { + am3517_evm_i2c_init(); + platform_add_devices(am3517_evm_devices, ARRAY_SIZE(am3517_evm_devices)); -- 1.6.2.4

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[alsa-devel] [PATCH 2/4] Audio: Modifying Kconfig/Makefile for AM3517 EVM
by Anuj Aggarwal 29 Oct '09

29 Oct '09

Modifying the Kconfig and Makefile in sound/soc/omap folder to add support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Kconfig | 9 +++++++++ sound/soc/omap/Makefile | 2 ++ 2 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 2dee983..6344456 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -55,6 +55,15 @@ config SND_OMAP_SOC_OMAP3EVM help Say Y if you want to add support for SoC audio on the omap3evm board. +config SND_OMAP_SOC_AM3517EVM + tristate "SoC Audio support for OMAP3517 / AM3517 EVM" + depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517 + EVM. + config SND_OMAP_SOC_SDP3430 tristate "SoC Audio support for Texas Instruments SDP3430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d6947..0c78ae4 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -12,6 +12,7 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-omap3evm-objs := omap3evm.o +snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o @@ -23,6 +24,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o +obj-$(CONFIG_MACH_OMAP3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- 1.6.2.4

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[alsa-devel] [PATCH 1/4] Audio: Adding OMAP3517 / AM3517 EVM support in ASOC
by Anuj Aggarwal 29 Oct '09

29 Oct '09

Adding support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/am3517evm.c | 202 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 202 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/am3517evm.c diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c new file mode 100644 index 0000000..135901b --- /dev/null +++ b/sound/soc/omap/am3517evm.c @@ -0,0 +1,202 @@ +/* + * am3517evm.c -- ALSA SoC support for OMAP3517 / AM3517 EVM + * + * Author: Anuj Aggarwal <anuj.aggarwal(a)ti.com> + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2009 Texas Instruments Incorporated + * + * 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 version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#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 <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" + +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static int am3517evm_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 *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_CLKR_SRC_CLKX, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_CLKR_SRC_CLKX\n"); + return ret; + } + + snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops am3517evm_ops = { + .hw_params = am3517evm_hw_params, +}; + +/* am3517evm machine dapm widgets */ +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LOUT"}, + {"Line Out", NULL, "ROUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic In"}, +}; + +static int am3517evm_aic23_init(struct snd_soc_codec *codec) +{ + /* Add am3517-evm specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic In"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link am3517evm_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = am3517evm_aic23_init, + .ops = &am3517evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_am3517evm = { + .name = "am3517evm", + .platform = &omap_soc_platform, + .dai_link = &am3517evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device am3517evm_snd_devdata = { + .card = &snd_soc_am3517evm, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *am3517evm_snd_device; + +static int __init am3517evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3517evm()) { + pr_err("Not OMAP3517 / AM3517 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3517 / AM3517 EVM SoC init\n"); + + am3517evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!am3517evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(am3517evm_snd_device, &am3517evm_snd_devdata); + am3517evm_snd_devdata.dev = &am3517evm_snd_device->dev; + *(unsigned int *)am3517evm_dai.cpu_dai->private_data = 0; /* McBSP1 */ + + ret = platform_device_add(am3517evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(am3517evm_snd_device); + + return ret; +} + +static void __exit am3517evm_soc_exit(void) +{ + platform_device_unregister(am3517evm_snd_device); +} + +module_init(am3517evm_soc_init); +module_exit(am3517evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM"); +MODULE_LICENSE("GPL v2"); -- 1.6.2.4

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[alsa-devel] What am i doing wrong?
by Neil Jones 29 Oct '09

29 Oct '09

Im developing ASoC Driver for an SoC (I2S based). When testing playback with aplay, I get 1 buffers worth of data played back (10 periods) then playback stops due to a XRUN when my dma list wraps back to the start because the available data >= stop_threshold ?? Here are my logs, interrupts are occurring at the correct time and my buffer pointer increments by period size each time. Any help would be appreciated Aplay log: # /usr/bin/aplay -vvv /var/sample96KHz_edit.wav Playing WAVE '/var/sample96KHz_edit.wav' : Signed 16 bit Little Endian, Rate 96000 Hz, Stereo Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0*0.5 + 1*0.5 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 96000 exact rate : 96000 (96000/1) msbits : 16 buffer_size : 48000 period_size : 4096 period_time : 42666 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 48000 stop_threshold : 48000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 Slave: Hardware PCM card 0 'Metamorph-Base' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 96000 exact rate : 96000 (96000/1) msbits : 16 buffer_size : 48000 period_size : 4096 period_time : 42666 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 48000 stop_threshold : 48000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 Max peak (8192 samples): 0x00002594 ###### 29% Max peak (8192 samples): 0x00000df8 ### 10% Max peak (8192 samples): 0x00007fff #################### 99% Max peak (8192 samples): 0x00006074 ################ 75% Max peak (8192 samples): 0x00003bff ########## 46% Max peak (8192 samples): 0x00003e7b ########## 48% Max peak (8192 samples): 0x00001abf ##### 20% Max peak (8192 samples): 0x000009e0 ## 7% Max peak (8192 samples): 0x000067df ################# 81% Max peak (8192 samples): 0x00001b45 ##### 21% Max peak (8192 samples): 0x00000b91 ## 9% Max peak (5888 samples): 0x000044a5 ########### 53% underrun!!! (at least 83.448 ms long) Status: state : XRUN trigger_time: 4784.5629989 tstamp : 4784.88220967 delay : 0 avail : 48000 avail_max : 48000 Kernel Log: PCM DMA IRQ: status 100000, timestamp 4483380 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x1000/0x1000/0xbb80, hwptr=0x0, hw_base=0x0, hw_intr=0x0 PCM DMA IRQ: status 120000, timestamp 4483430 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x2000/0x1000/0xbb80, hwptr=0x1000, hw_base=0x0, hw_intr=0x1000 PCM DMA IRQ: status 120000, timestamp 4483480 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x3000/0x1000/0xbb80, hwptr=0x2000, hw_base=0x0, hw_intr=0x2000 PCM DMA IRQ: status 120000, timestamp 4483530 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x4000/0x1000/0xbb80, hwptr=0x3000, hw_base=0x0, hw_intr=0x3000 PCM DMA IRQ: status 120000, timestamp 4483580 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x5000/0x1000/0xbb80, hwptr=0x4000, hw_base=0x0, hw_intr=0x4000 PCM DMA IRQ: status 120000, timestamp 4483630 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x6000/0x1000/0xbb80, hwptr=0x5000, hw_base=0x0, hw_intr=0x5000 PCM DMA IRQ: status 120000, timestamp 4483680 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x7000/0x1000/0xbb80, hwptr=0x6000, hw_base=0x0, hw_intr=0x6000 PCM DMA IRQ: status 120000, timestamp 4483730 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x8000/0x1000/0xbb80, hwptr=0x7000, hw_base=0x0, hw_intr=0x7000 PCM DMA IRQ: status 120000, timestamp 4483790 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x9000/0x1000/0xbb80, hwptr=0x8000, hw_base=0x0, hw_intr=0x8000 PCM DMA IRQ: status 120000, timestamp 4483840 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0xa000/0x1000/0xbb80, hwptr=0x9000, hw_base=0x0, hw_intr=0x9000 PCM DMA IRQ: status 120000, timestamp 4483890 ms, ALSA sound/core/pcm_lib.c:250: period_update: pcmC0D0p:0: pos=0x0/0x1000/0xbb80, hwptr=0xa000, hw_base=0x0, hw_intr=0xa000 ALSA sound/core/pcm_lib.c:160: XRUN: pcmC0D0p:0 - This message is subject to Imagination Technologies' e-mail terms: http://www.imgtec.com/e-mail.htm Imagination Technologies Ltd is a limited company registered in England No: 1306335 Registered Office: Imagination House, Home Park Estate, Kings Langley, Hertfordshire, WD4 8LZ. Email to and from the company may be monitored for compliance and other administrative purposes. -

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[alsa-devel] [PATCH] ASoC: TWL4030: Add APLL supply for the capture path
by Peter Ujfalusi 29 Oct '09

29 Oct '09

Capture path also need the APLL enabled, adding DAPM_SUPPLY for the Virtual ADCs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- I have forgot to run git add, so the capture path DAPM_SUPPLY was missing from the previous series. This also applies on top of Mark's tree, for-2.6.33-pending branch at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git and depends on the "ASoC: TWL4030: Change codec_muted to apll_enabled" patch. Sorry about this. sound/soc/codecs/twl4030.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 277e99c..f9121ef 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1449,6 +1449,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, + {"ADC Virtual Left1", NULL, "APLL Enable"}, + {"ADC Virtual Right1", NULL, "APLL Enable"}, + {"ADC Virtual Left2", NULL, "APLL Enable"}, + {"ADC Virtual Right2", NULL, "APLL Enable"}, + /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, {"Left1 Analog Loopback", "Switch", "Analog Left"}, -- 1.6.5.1

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[alsa-devel] [PATCH 0/2] ASoC: TWL4030: Vibra and APLL fixes
by Peter Ujfalusi 29 Oct '09

29 Oct '09

Hello, Audio driven vibrator output fix: To have correct audio routed to the vibra output, we also need to configure the VIBRA_SET register, which in theory has nothing to do with the vibra, when it is driven with audio. and fix for the APLL powering sequence: The APLL in the past was enabled at set_dai_sysclk function, but it has been fixed, which introduced some really strange signals on the outputs. This series applies on top of Mark's tree, for-2.6.33 branch at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git Since the underlying twl4030 patches are not yet in the topic/asoc of Takashi's tree. --- Jari Vanhala (1): ASoC: TWL4030: Vibra motor stop fix when it is driven with audio Peter Ujfalusi (1): ASoC: TWL4030: Change APLL powering sequence sound/soc/codecs/twl4030.c | 38 ++++++++++++++++++++++++++++++++++---- 1 files changed, 34 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH] ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal
by Anuj Aggarwal 29 Oct '09

29 Oct '09

The pop-removal specific values are configured for TWL4030 codec for OMAP3EVM through this patch. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/omap3evm.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 9114c26..8deb59b 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -93,10 +93,17 @@ static struct snd_soc_card snd_soc_omap3evm = { .num_links = 1, }; +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 4, + .sysclk = 26000, +}; + /* Audio subsystem */ static struct snd_soc_device omap3evm_snd_devdata = { .card = &snd_soc_omap3evm, .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, }; static struct platform_device *omap3evm_snd_device; -- 1.6.2.4

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