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

[alsa-devel] 2.6.32-gitX: oops in ac97 codec registration
by Manuel Lauss 18 Dec '09

18 Dec '09

Hi Mark, current 2.6.32-git spits out the following oops when I load my machine driver with the generic AC97 codec: No device for DAI au1xpsc_ac97 AC97 SoC Audio Codec 0.6 asoc: AC97 HiFi <-> au1xpsc_ac97 mapping ok CPU 0 Unable to handle kernel paging request at virtual address 000001f0, epc == 80366320, ra == 803662c0 Oops[#1]: Cpu 0 $ 0 : 00000000 10003c00 00000000 00000000 $ 4 : 00000000 00004b39 00000001 00004b39 $ 8 : 00000000 00000002 8041a7a0 8d140af0 $12 : 00000004 ad55ad55 00000003 00000001 $16 : 00000000 c1179a94 8d1f2520 c007b420 $20 : 8d49cfe0 8d1f2520 804fc03c 804fc130 $24 : 00000000 0000ad55 $28 : 8d590000 8d591af0 00000000 803662c0 Hi : 00000000 Lo : 00000004 epc : 80366320 snd_soc_new_pcms+0x300/0x358 Not tainted ra : 803662c0 snd_soc_new_pcms+0x2a0/0x358 Status: 10003c03 KERNEL EXL IE Cause : 0080800c BadVA : 000001f0 PrId : 04030202 (Au1250) Modules linked in: snd_soc_exm32cexx(+) snd_soc_cs4251x snd_soc_ac97 snd_ac97_codec snd_soc_au1xpsc_ac97 snd_soc_au1xpsc_i2s Process modprobe (pid: 673, threadinfo=8d590000, task=8d430938, tls=2aacf2d0) Stack : 8d591b30 c007b3c4 c0074afc 00010000 00000001 8d591b08 8d17d1c0 37394341 46694820 43412069 48203739 2d694669 8d170030 804fc16c 8d17d2c8 0000001c 10003c01 8d17d2c8 804fc16c 8d17d2c8 10003c01 8d17d2d0 00000024 80500000 c0074b58 00000001 c11799f4 00000000 8d1f2528 00000001 fffffff4 8d1f2520 c11798f0 00000001 8d17d0c0 00000001 804fc130 00000000 804f4980 c007b288 ... Call Trace: [<80366320>] snd_soc_new_pcms+0x300/0x358 [<c007b288>] ac97_soc_probe+0x108/0x1d0 [snd_soc_ac97] [<80365108>] snd_soc_instantiate_cards+0x22c/0x694 [<803655ec>] soc_probe+0x7c/0xc4 [<802c73b8>] driver_probe_device+0xbc/0x194 [<802c6748>] bus_for_each_drv+0x60/0xb0 [<802c7638>] device_attach+0x60/0x88 [<802c6524>] bus_probe_device+0x30/0x54 [<802c4d74>] device_add+0x3a4/0x53c [<802c8e48>] platform_device_add+0x14c/0x1b8 [<c11794ec>] cexx_audio_probe+0x310/0x374 [snd_soc_exm32cexx] [<802c73b8>] driver_probe_device+0xbc/0x194 [<802c74fc>] __driver_attach+0x6c/0xa4 [<802c6afc>] bus_for_each_dev+0x60/0xb0 [<802c6264>] bus_add_driver+0xc8/0x268 [<802c78ac>] driver_register+0xdc/0x194 [<8010a010>] do_one_initcall+0x68/0x1f4 [<80154268>] sys_init_module+0xec/0x23c [<801022a0>] stack_done+0x20/0x3c Oops is inside snd_ac97_dev_add_pdata() call: seems ac97->dev is NULL. I'm not entirely sure that this isn't somehow the fault of my machine driver; has the procedure to register the asoc device changed since 2.6.32? Thanks! Manuel Lauss

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[alsa-devel] [PATCH 3/3] ASoC: OMAP4: Add McPDM platform driver
by Candelaria Villareal, Jorge 17 Dec '09

17 Dec '09

McPDM platform driver is configured to use sDMA in order to transfer to/from memory. Support for interfacing with ABE will be added later. McPDM dai currently supports up to 4 downlink channels and 2 uplink channels simultaneously, as well as 88.2 and 96 KHz, and a sample size of 32 bits. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/omap-mcpdm.c | 248 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++++ 3 files changed, 278 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index bf8b388..d6382fa 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,7 +1,7 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o -snd-soc-omap-mcpdm-objs := mcpdm.o +snd-soc-omap-mcpdm-objs := mcpdm.o omap-mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c new file mode 100644 index 0000000..9dc0ff5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.c @@ -0,0 +1,248 @@ +/* + * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port + * + * Copyright (C) 2009 Texas Instruments + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * Contact: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Margarita Olaya <magi.olaya(a)ti.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/control.h> +#include <mach/dma.h> +#include <mach/mcbsp.h> +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" + +struct omap_mcpdm_data { + struct omap_mcpdm_link *links; + int active; +}; + +static struct omap_mcpdm_link omap_mcpdm_links[] = { + /* downlink */ + { + .irq_mask = DN_IRQ_EMTPY | DN_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, + /* uplink */ + { + .irq_mask = UP_IRQ_EMPTY | UP_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, +}; + +static struct omap_mcpdm_data mcpdm_data = { + .links = omap_mcpdm_links, + .active = 0, +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { + { + .name = "Audio downlink", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_ELEMENT, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + }, + { + .name = "Audio uplink", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_ELEMENT, + .port_addr = OMAP44XX_MCPDM_BASE + MCPDM_UP_DATA, + }, +}; + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err = 0; + + if (!cpu_dai->active) + err = omap_mcpdm_request(); + + return err; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (!cpu_dai->active) + omap_mcpdm_free(); +} + +static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + int stream = substream->stream; + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!mcpdm_priv->active++) + omap_mcpdm_start(stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!--mcpdm_priv->active) + omap_mcpdm_stop(stream); + break; + default: + err = -EINVAL; + } + + return err; +} + +static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int channels, err, link_mask = 0; + + cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + + channels = params_channels(params); + switch (channels) { + case 4: + if (stream) + /* up to 2 channels for uplink */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream) + /* up to 2 channels for uplink */ + return -EINVAL; + link_mask |= 1 << 2; + case 2: + link_mask |= 1 << 1; + case 1: + link_mask |= 1 << 0; + break; + default: + /* unsupported number of channels */ + return -EINVAL; + } + + if (stream) { + mcpdm_links[stream].channels = link_mask << 0; + err = omap_mcpdm_set_uplink(&mcpdm_links[stream]); + } else { + mcpdm_links[stream].channels = link_mask << 3; + err = omap_mcpdm_set_downlink(&mcpdm_links[stream]); + } + + return err; +} + +static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int err; + + if (substream->stream) + err = omap_mcpdm_clr_uplink(&mcpdm_links[substream->stream]); + else + err = omap_mcpdm_clr_downlink(&mcpdm_links[substream->stream]); + + return err; +} + +static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { + .startup = omap_mcpdm_dai_startup, + .shutdown = omap_mcpdm_dai_shutdown, + .trigger = omap_mcpdm_dai_trigger, + .hw_params = omap_mcpdm_dai_hw_params, + .hw_free = omap_mcpdm_dai_hw_free, +}; + +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai omap_mcpdm_dai = { + .name = "omap-mcpdm", + .id = -1, + .playback = { + .channels_min = 1, + .channels_max = 4, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .ops = &omap_mcpdm_dai_ops, + .private_data = &mcpdm_data, +}; +EXPORT_SYMBOL_GPL(omap_mcpdm_dai); + +static int __init snd_omap_mcpdm_init(void) +{ + return snd_soc_register_dai(&omap_mcpdm_dai); +} +module_init(snd_omap_mcpdm_init); + +static void __exit snd_omap_mcpdm_exit(void) +{ + snd_soc_unregister_dai(&omap_mcpdm_dai); +} +module_exit(snd_omap_mcpdm_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("OMAP PDM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 0000000..73b80d5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,29 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 Texas Instruments + * + * Contact: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +extern struct snd_soc_dai omap_mcpdm_dai; + +#endif /* End of __OMAP_MCPDM_H__ */ -- 1.6.0.4

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[alsa-devel] [PATCH 1/3] OMAP4: IRQ: Add McPDM IRQ definition.
by Candelaria Villareal, Jorge 17 Dec '09

17 Dec '09

Support for McPDM IRQs in OMAP4 Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/include/plat/irqs.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/arm/plat-omap/include/plat/irqs.h b/arch/arm/plat-omap/include/plat/irqs.h index 97d6c50..7fb2464 100644 --- a/arch/arm/plat-omap/include/plat/irqs.h +++ b/arch/arm/plat-omap/include/plat/irqs.h @@ -432,7 +432,7 @@ #define INT_44XX_PARTHASH_IRQ (79 + IRQ_GIC_START) #define INT_44XX_MMC3_IRQ (94 + IRQ_GIC_START) #define INT_44XX_MMC4_IRQ (96 + IRQ_GIC_START) - +#define INT_44XX_MCPDM_IRQ (112 + IRQ_GIC_START) /* Max. 128 level 2 IRQs (OMAP1610), 192 GPIOs (OMAP730/850) and * 16 MPUIO lines */ -- 1.6.0.4

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Re: [alsa-devel] [alsa-cvslog] alsa-driver: ALSA driver repository - compatibility code, new drivers and standalone building branch, master now at v1.0.22-1-g79c453f
by Takashi Iwai 17 Dec '09

17 Dec '09

At Thu, 17 Dec 2009 17:41:37 +0100 (CET), noreply-git(a)alsa-project.org wrote: > > Also, missing isa/dt019x.c is added. The driver was removed in the recent version. Takashi

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[alsa-devel] PLL computation in TLV320AIC3x SoC driver
by Peter Meerwald 17 Dec '09

17 Dec '09

Hello, I'm trying to use the SoC TLV320AIC3x codec driver with sysclk 16384000 and ran into some problems with setting PLL; below is a patch against linux-2.6-asoc note that the original code uses variables pll_r and pll_p instead of the loop variable r and p to compute tmp, this seems broken further, the original code does not respect the constraints on j (>= 4, <= 55 for d==0) according to the codec's datasheet, and similarly for d!=0 I've tested the code with a number of reasonable sysclk values and got sane PLL values; please apply if acceptable thanks, regards, p. diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 3395cf9..e84e473 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -766,9 +766,10 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; struct aic3x_priv *aic3x = codec->private_data; int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; - u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; - u16 pll_d = 1; + u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; + u16 d, pll_d = 1; u8 reg; + int clk; /* select data word length */ data = @@ -835,47 +836,62 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, return 0; /* Use PLL - * find an apropriate setup for j, d, r and p by iterating over - * p and r - j and d are calculated for each fraction. - * Up to 128 values are probed, the closest one wins the game. + * find an appropriate setup for j, d, r and p by iterating over + * p, r and j first, then trying to compute the fraction d. + * Up to 6528 values are probed, the closest one wins the game. * The sysclk is divided by 1000 to prevent integer overflows. */ codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000); - for (r = 1; r <= 16; r++) - for (p = 1; p <= 8; p++) { - int clk, tmp = (codec_clk * pll_r * 10) / pll_p; - u8 j = tmp / 10000; - u16 d = tmp % 10000; - - if (j > 63) - continue; - - if (d != 0 && aic3x->sysclk < 10000000) - continue; - - /* This is actually 1000 * ((j + (d/10000)) * r) / p - * The term had to be converted to get rid of the - * division by 10000 */ - clk = ((10000 * j * r) + (d * r)) / (10 * p); - - /* check whether this values get closer than the best - * ones we had before */ - if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { - pll_j = j; pll_d = d; pll_r = r; pll_p = p; - last_clk = clk; - } - - /* Early exit for exact matches */ - if (clk == codec_clk) - break; - } + for (r = 1; r <= 16; r++) + for (p = 1; p <= 8; p++) { + for (j = 4; j <= 55; j++) { + /* This is actually 1000 * ((j + (d/10000)) * r) / p + * The term had to be converted to get rid of the + * division by 10000; d = 0 here */ + int clk = (1000 * j * r) / p; + + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = 0; pll_r = r; pll_p = p; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + goto found; + } + } + + /* try with d != 0 */ + for (p = 1; p <= 8; p++) { + j = codec_clk * p / 1000; + + if (j < 4 || j > 11) continue; + + /* do not use codec_clk here since we'd loose precision */ + d = ((2048 * fsref * 10) / (aic3x->sysclk / 1000)) % 10000; + clk = (10000 * j + d) / (10 * p); + + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = d; pll_r = 1; pll_p = 1; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + goto found; + } if (last_clk == 0) { printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); return -EINVAL; } +found: data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); -- Peter Meerwald Kaigasse 3 / 8 A-5020 Salzburg / AUSTRIA +43-664-2444418 (mobile)

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[alsa-devel] [PATCH] sound/oss/pss: Fix test of unsigned in pss_reset_dsp() and pss_download_boot()
by Roel Kluin 17 Dec '09

17 Dec '09

limit and jiffies are unsigned so the test does not work. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found using coccinelle: http://coccinelle.lip6.fr/ diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 83f5ee2..9325f4d 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -269,7 +269,7 @@ static int pss_reset_dsp(pss_confdata * devc) unsigned long i, limit = jiffies + HZ/10; outw(0x2000, REG(PSS_CONTROL)); - for (i = 0; i < 32768 && (limit-jiffies >= 0); i++) + for (i = 0; i < 32768 && (limit >= jiffies); i++) inw(REG(PSS_CONTROL)); outw(0x0000, REG(PSS_CONTROL)); return 1; @@ -369,11 +369,11 @@ static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size outw(0, REG(PSS_DATA)); limit = jiffies + HZ/10; - for (i = 0; i < 32768 && (limit - jiffies >= 0); i++) + for (i = 0; i < 32768 && (limit >= jiffies); i++) val = inw(REG(PSS_STATUS)); limit = jiffies + HZ/10; - for (i = 0; i < 32768 && (limit-jiffies >= 0); i++) + for (i = 0; i < 32768 && (limit >= jiffies); i++) { val = inw(REG(PSS_STATUS)); if (val & 0x4000)

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[alsa-devel] Two patches for Dell Vostro 1015 (CX20583)
by Einar Rünkaru 17 Dec '09

17 Dec '09

Hi. Attached two patches: 1. Corrected initialization of Conexant CX20583-10Z - forgot proper initialization of internal mic. Int mic was muted until external mic was plugged/unplugged. Also added control for internal mic boost. I renamed 'analog mic boost' to 'ext mic boost'. Alternative was to use names analog-digital, which seemed very confusing. If the renaming breaks something, it can be dropped. 2. Made use of the chips beep device. Now it's possible to change volume of beep. (It was too loud) To make use of it in Ubuntu 9.04 pcspkr must be blacklisted. Einar Rünkaru

2 1

[alsa-devel] [PATCH] ASoC: Fix build of DA7210
by Mark Brown 17 Dec '09

17 Dec '09

From: Mark Brown <broonie(a)finisterre.wolfsonmicro.main> DAC_VOICE_EN was not defined - looks to have been overly enthusiastically deleted from a previous revision of the patch, pull the value from v1. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/da7210.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 14f5f34..fbf3ab4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -81,6 +81,9 @@ #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) +/* DAC_HPF fields */ +#define DA7210_DAC_VOICE_EN (1 << 7) + /* DAC_SEL bit fields */ #define DA7210_DAC_L_SRC_DAI_L (4 << 0) #define DA7210_DAC_L_EN (1 << 3) -- 1.6.5.4

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[alsa-devel] ALSA 1.0.22 release
by Jaroslav Kysela 17 Dec '09

17 Dec '09

Hello all, the ALSA 1.0.22 packages (except alsa-oss and alsa-firmware) are available for download at http://www.alsa-project.org . The changes are listed in this URL: http://www.alsa-project.org/main/index.php/Changes_v1.0.21_v1.0.22 Jaroslav Kysela ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH 2/2] ALSA: pcm - specify amount of data needed in wait_for_avail_min
by jassisinghbrar＠gmail.com 17 Dec '09

17 Dec '09

From: Jassi Brar <jassi.brar(a)samsung.com> For better accuracy, make the waiting time more specific. That helps avoiding users of wait_for_avail_min(i.e, snd_pcm_lib_write1/read1) to wait until atleast avail_min amount is available, whereas all that might be needed is just 1 frame(for example). Since the waiting is not for avail_min, the name has been changed to reflect that. The users of the new function are also updated to pass exact minimum data needed. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/core/pcm_lib.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ee2d7b9..4a92804 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1664,12 +1664,12 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) EXPORT_SYMBOL(snd_pcm_period_elapsed); /* - * Wait until avail_min data becomes available + * Wait until atleast the requested amount(*availp) of data becomes available * Returns a negative error code if any error occurs during operation. * The available space is stored on availp. When err = 0 and avail = 0 * on the capture stream, it indicates the stream is in DRAINING state. */ -static int wait_for_avail_min(struct snd_pcm_substream *substream, +static int wait_for_min(struct snd_pcm_substream *substream, snd_pcm_uframes_t *availp) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1719,7 +1719,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, avail = snd_pcm_playback_avail(runtime); else avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->control->avail_min) + if (avail >= *availp) break; } _endloop: @@ -1794,7 +1794,8 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + avail = size; + err = wait_for_min(substream, &avail); if (err < 0) goto _end_unlock; } @@ -2016,7 +2017,8 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + avail = size; + err = wait_for_min(substream, &avail); if (err < 0) goto _end_unlock; if (!avail) -- 1.6.2.5

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