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[alsa-devel] [PATCH] ALSA: HDA: Pinfix quirk for HP Z200 Workstation
by David Henningsson 24 Jun '11

24 Jun '11

BIOS lists the internal speaker as an internal line-out. Change to internal speaker + model=auto for better auto-mute capabilities. Relevant alsa-info is here: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/754964/+attachme… -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 4

[alsa-devel] [ASoc] TDM over one SSC channel on AT91 possible?
by Friedrich Lobenstock 24 Jun '11

24 Jun '11

Hello! The background: I'm just designing an ATMEL AT91RM9200 based board which got two TLV320AIC33 on board. It is either possible to hook up the two TLV320AIC33 to a separate SSC (synchronous serial control) channel each or use one SSC channel and configure it for TDM (time division multiplex) mode. There is already a driver for TLV320AIC14k/TLV320AIC2k in the development tree, so adding support for the TLV320AIC33 should be feasible. What I'm not sure about is the fact if ASOC is prepared for using channels, especially SSC, in TDM mode. Is the core ASOC structure prepared to handle that? Would we be better off instantiate one drive twice each time with different parameters for the SSC channel? Thanks for any feedback. -- MfG / Regards Friedrich Lobenstock _________________________________________________________ SCOTTY Group Austria GmbH a SCOTTY Group plc company, http://www.scottygroup.com/ _________________________________________________________

4 3

[alsa-devel] [PATCH] ALSA: ASoC: pxa-ssp: Correct check for stream presence
by Daniel Mack 23 Jun '11

23 Jun '11

Don't rely on the codec's channels_min information to decide wheter or not allocate a substream's DMA buffer. Rather check if the substream itself was allocated previously. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/pxa/pxa2xx-pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index d73d6f6..da28394 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -97,14 +97,14 @@ static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) -- 1.7.5.4

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[alsa-devel] [patch 1/2] ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entry
by Arnaud Patard 23 Jun '11

23 Jun '11

SND_MXC_SOC_SSI looks to be unused, so kill it. Signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> Index: linux-2.6-submit/sound/soc/imx/Kconfig =================================================================== --- linux-2.6-submit.orig/sound/soc/imx/Kconfig 2011-06-22 09:20:08.000000000 +0200 +++ linux-2.6-submit/sound/soc/imx/Kconfig 2011-06-22 09:32:48.000000000 +0200 @@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC -config SND_MXC_SOC_SSI - tristate - config SND_MXC_SOC_FIQ tristate @@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable support for audio on the i.MX31ADS with the WM1133-EV1 @@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 select SND_SOC_TVL320AIC32X4 - select SND_MXC_SOC_SSI select SND_MXC_SOC_MX2 help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 select SND_SOC_WM9712 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320 || MACH_EUKREA_MBIMXSD35_BASEBOARD \ || MACH_EUKREA_MBIMXSD51_BASEBOARD select SND_SOC_TLV320AIC23 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable I2S based access to the TLV320AIC23B codec attached

4 3

[alsa-devel] snd_pcm_link : Function not implemented
by ashwanik 23 Jun '11

23 Jun '11

Hi All, I have download and compile latency.c application from http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.ht…. This application is working fine on Linux PC. But when I am using this application on my omap4 board(using ubuntu10.10) is not working. I am getting"Streams link error: Function not implemented" error. I debug the application and found this error coming from the *snd_pcm_link* function. I am using ALSA Driver Version 1.0.23. please find pointers. Thanks in Advance Ashwani

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[alsa-devel] [PATCH] ASoC: Add basic WM8918 support
by Mark Brown 23 Jun '11

23 Jun '11

The WM8918 is register compatible with the WM8904 with a subset of the functionality. Add the device ID, a subsequent patch will ensure that only the relevant functionality is exported to userspace. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8904.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9b3bba4..b085575 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2560,6 +2560,7 @@ static __devexit int wm8904_i2c_remove(struct i2c_client *client) static const struct i2c_device_id wm8904_i2c_id[] = { { "wm8904", WM8904 }, { "wm8912", WM8912 }, + { "wm8918", WM8904 }, /* Actually a subset, updates to follow */ { } }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Trigger wm_hubs series update startup off a separate flag
by Mark Brown 23 Jun '11

23 Jun '11

Allowing the two to be used independently. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 1 + sound/soc/codecs/wm8994.c | 1 + sound/soc/codecs/wm_hubs.c | 3 +-- sound/soc/codecs/wm_hubs.h | 1 + 4 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 9e5ff78..03af7ad 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1434,6 +1434,7 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->hubs_data.hp_startup_mode = 1; wm8993->hubs_data.dcs_codes = -2; + wm8993->hubs_data.series_startup = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret != 0) { diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 86e67e7..44153be 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2918,6 +2918,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_codes = -5; wm8994->hubs.hp_startup_mode = 1; wm8994->hubs.dcs_readback_mode = 1; + wm8994->hubs.series_startup = 1; break; default: wm8994->hubs.dcs_readback_mode = 1; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 586265b..8042999 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -107,8 +107,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) return; } - /* Devices not using a DCS code correction have startup mode */ - if (hubs->dcs_codes) { + if (hubs->series_startup) { /* Set for 32 series updates */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, WM8993_DCS_SERIES_NO_01_MASK, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index f8a5e97..aeb5871 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -23,6 +23,7 @@ struct wm_hubs_data { int dcs_codes; int dcs_readback_mode; int hp_startup_mode; + int series_startup; bool class_w; u16 class_w_dcs; -- 1.7.5.4

2 2

[alsa-devel] [patch 2/2] ASoC: imx: add missing module informations
by Arnaud Patard 23 Jun '11

23 Jun '11

- add some modules aliases - add module license to avoid tainted kernel when loading the imx-pcm-audio driver Signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> Index: sound-2.6/sound/soc/imx/imx-pcm-dma-mx2.c =================================================================== --- sound-2.6.orig/sound/soc/imx/imx-pcm-dma-mx2.c 2011-06-22 15:09:06.000000000 +0200 +++ sound-2.6/sound/soc/imx/imx-pcm-dma-mx2.c 2011-06-22 15:36:36.000000000 +0200 @@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void platform_driver_unregister(&imx_pcm_driver); } module_exit(snd_imx_pcm_exit); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:imx-pcm-audio"); Index: sound-2.6/sound/soc/imx/imx-ssi.c =================================================================== --- sound-2.6.orig/sound/soc/imx/imx-ssi.c 2011-06-22 15:09:06.000000000 +0200 +++ sound-2.6/sound/soc/imx/imx-ssi.c 2011-06-22 15:36:36.000000000 +0200 @@ -775,4 +775,4 @@ module_exit(imx_ssi_exit); MODULE_AUTHOR("Sascha Hauer, <s.hauer(a)pengutronix.de>"); MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:imx-ssi");

2 1

[alsa-devel] model needed for Clevo P150HM (Realtek ALC892)
by Julian Sikorski 23 Jun '11

23 Jun '11

Hi, I am coming here from pulseaudio-general mailing list [1]. I recently got myself a Clevo P150HM laptop. It has 2.1 speakers, a built-in microphone and the following connectors: - headphone out - microphone in; can be switched to center/lfe out - spdif out; can be switched to rear speaker out - line in; can be switched to side speaker out Alsa does not seem to reflect that the device is capable of 5.1 and 7.1 sound, and there are two configs showing up in pavucontrol (analog output and analog headphones) which seem to be the same. I have uploaded the output of alsa-info.sh [2]. Please let me know if more information is needed. Regards, Julian [1] http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/10096 [2] http://www.alsa-project.org/db/?f=7c91e874f4272387c2c51e448337567202c0edfe

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[alsa-devel] [PATCH 1/4] ASoC: Add ADAV80x codec driver
by Lars-Peter Clausen 23 Jun '11

23 Jun '11

This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 846 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adav80x.h | 23 ++ 4 files changed, 875 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/adav80x.c create mode 100644 sound/soc/codecs/adav80x.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2998e65..ff43405 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAV80X select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -137,6 +138,9 @@ config SND_SOC_ADAU1701 select SIGMA tristate +config SND_SOC_ADAV80X + tristate + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 51cd3d4..4957431 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o +snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -99,6 +100,7 @@ obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o +obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c new file mode 100644 index 0000000..d2f3d08 --- /dev/null +++ b/sound/soc/codecs/adav80x.c @@ -0,0 +1,846 @@ +/* + * ADAV80X Audio Codec driver supporting ADAV801, ADAV803 + * + * Copyright 2011 Analog Devices Inc. + * Author: Yi Li <yi.li(a)analog.com> + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> + +#include "adav80x.h" + +enum adav80x_clk_src { + ADAV80X_CLK_XIN , + ADAV80X_CLK_PLL1, + ADAV80X_CLK_PLL2, + ADAV80X_CLK_INTERNAL1, + ADAV80X_CLK_INTERNAL2, + ADAV80X_CLK_MCLKI, +}; + +#define ADAV80X_PLAYBACK_CTRL 0x04 +#define ADAV80X_AUX_IN_CTRL 0x05 +#define ADAV80X_REC_CTRL 0x06 +#define ADAV80X_AUX_OUT_CTRL 0x07 +#define ADAV80X_DPATH_CTRL1 0x62 +#define ADAV80X_DPATH_CTRL2 0x63 +#define ADAV80X_DAC_CTRL1 0x64 +#define ADAV80X_DAC_CTRL2 0x65 +#define ADAV80X_DAC_CTRL3 0x66 +#define ADAV80X_DAC_L_VOL 0x68 +#define ADAV80X_DAC_R_VOL 0x69 +#define ADAV80X_PGA_L_VOL 0x6c +#define ADAV80X_PGA_R_VOL 0x6d +#define ADAV80X_ADC_CTRL1 0x6e +#define ADAV80X_ADC_CTRL2 0x6f +#define ADAV80X_ADC_L_VOL 0x70 +#define ADAV80X_ADC_R_VOL 0x71 +#define ADAV80X_PLL_CTRL1 0x74 +#define ADAV80X_PLL_CTRL2 0x75 +#define ADAV80X_ICLK_CTRL1 0x76 +#define ADAV80X_ICLK_CTRL2 0x77 +#define ADAV80X_PLL_CLK_SRC 0x78 +#define ADAV80X_PLL_OUTE 0x7a + +#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll) 0x00 +#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll) (0x40 << (pll)) +#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll) (0x40 << (pll)) + +#define ADAV80X_PLL_CTRL1_PLLDIV 0x10 +#define ADAV80X_PLL_CTRL1_PLL2PD 0x08 +#define ADAV80X_PLL_CTRL1_PLL1PD 0x04 +#define ADAV80X_PLL_CTRL1_XTLPD 0x02 + +#define ADAV80X_PLL_CTRL2_FIELD(pll, x) ((x) << ((pll) * 4)) + +#define ADAV80X_PLL_CTRL2_FS_48(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x00) +#define ADAV80X_PLL_CTRL2_FS_32(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x08) +#define ADAV80X_PLL_CTRL2_FS_44(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c) + +#define ADAV80X_PLL_CTRL2_SEL(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x02) +#define ADAV80X_PLL_CTRL2_DOUB(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x01) +#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f) + +#define ADAV80X_ADC_CTRL1_MODULATOR_MASK 0x80 +#define ADAV80X_ADC_CTRL1_MODULATOR_128FS 0x00 +#define ADAV80X_ADC_CTRL1_MODULATOR_64FS 0x80 + +#define ADAV80X_DAC_CTRL1_PD 0x80 + +#define ADAV80X_DAC_CTRL2_DIV1 0x00 +#define ADAV80X_DAC_CTRL2_DIV1_5 0x10 +#define ADAV80X_DAC_CTRL2_DIV2 0x20 +#define ADAV80X_DAC_CTRL2_DIV3 0x30 +#define ADAV80X_DAC_CTRL2_DIV_MASK 0x30 + +#define ADAV80X_DAC_CTRL2_INTERPOL_256FS 0x00 +#define ADAV80X_DAC_CTRL2_INTERPOL_128FS 0x40 +#define ADAV80X_DAC_CTRL2_INTERPOL_64FS 0x80 +#define ADAV80X_DAC_CTRL2_INTERPOL_MASK 0xc0 + +#define ADAV80X_DAC_CTRL2_DEEMPH_NONE 0x00 +#define ADAV80X_DAC_CTRL2_DEEMPH_44 0x01 +#define ADAV80X_DAC_CTRL2_DEEMPH_32 0x02 +#define ADAV80X_DAC_CTRL2_DEEMPH_48 0x03 +#define ADAV80X_DAC_CTRL2_DEEMPH_MASK 0x01 + +#define ADAV80X_CAPTURE_MODE_MASTER 0x20 +#define ADAV80X_CAPTURE_WORD_LEN24 0x00 +#define ADAV80X_CAPTURE_WORD_LEN20 0x04 +#define ADAV80X_CAPTRUE_WORD_LEN18 0x08 +#define ADAV80X_CAPTURE_WORD_LEN16 0x0c +#define ADAV80X_CAPTURE_WORD_LEN_MASK 0x0c + +#define ADAV80X_CAPTURE_MODE_LEFT_J 0x00 +#define ADAV80X_CAPTURE_MODE_I2S 0x01 +#define ADAV80X_CAPTURE_MODE_RIGHT_J 0x03 +#define ADAV80X_CAPTURE_MODE_MASK 0x03 + +#define ADAV80X_PLAYBACK_MODE_MASTER 0x10 +#define ADAV80X_PLAYBACK_MODE_LEFT_J 0x00 +#define ADAV80X_PLAYBACK_MODE_I2S 0x01 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24 0x04 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20 0x05 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18 0x06 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16 0x07 +#define ADAV80X_PLAYBACK_MODE_MASK 0x07 + +static u8 adav80x_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, + 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, + 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, + 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, + 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, + 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, + 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +}; + +struct adav80x { + enum snd_soc_control_type control_type; + + enum adav80x_clk_src clk_src; + + unsigned int dai_fmt[2]; + unsigned int rate; + bool deemph; +}; + +static const char *adav80x_mux_text[] = { + "ADC", + "Playback", + "Aux Playback", +}; + +static const unsigned int adav80x_mux_values[] = { + 0, 2, 3, +}; + +#define ADAV80X_MUX_ENUM(reg, shift) \ + SOC_VALUE_ENUM_SINGLE(reg, shift, 7, ARRAY_SIZE(adav80x_mux_text), \ + adav80x_mux_text, adav80x_mux_values) + +static const struct soc_enum adav80x_mux_enum[] = { + ADAV80X_MUX_ENUM(ADAV80X_DPATH_CTRL1, 0), + ADAV80X_MUX_ENUM(ADAV80X_DPATH_CTRL1, 3), + ADAV80X_MUX_ENUM(ADAV80X_DPATH_CTRL2, 3), +}; + +static const struct snd_kcontrol_new adav80x_mux_ctrl[] = { + SOC_DAPM_VALUE_ENUM("Route", adav80x_mux_enum[0]), + SOC_DAPM_VALUE_ENUM("Route", adav80x_mux_enum[1]), + SOC_DAPM_VALUE_ENUM("Route", adav80x_mux_enum[2]), +}; + +#define ADAV80X_MUX(name, num) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, &adav80x_mux_ctrl[num]) + +static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1), + SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1), + + SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0), + + ADAV80X_MUX("Aux Capture Select", 0), + ADAV80X_MUX("Capture Select", 1), + ADAV80X_MUX("DAC Select", 2), + + SND_SOC_DAPM_INPUT("VINR"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), + SND_SOC_DAPM_OUTPUT("VOUTL"), +}; + +static const struct snd_soc_dapm_route adav80x_dapm_routes[] = { + { "DAC Select", "ADC", "ADC" }, + { "DAC Select", "Playback", "AIFIN" }, + { "DAC Select", "Aux Playback", "AIFAUXIN" }, + { "DAC", NULL, "DAC Select" }, + + { "Capture Select", "ADC", "ADC" }, + { "Capture Select", "Playback", "AIFIN" }, + { "Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFOUT", NULL, "Capture Select" }, + + { "Aux Capture Select", "ADC", "ADC" }, + { "Aux Capture Select", "Playback", "AIFIN" }, + { "Aux Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFAUXOUT", NULL, "Aux Capture Select" }, + + { "VOUTR", NULL, "DAC" }, + { "VOUTL", NULL, "DAC" }, + + { "Left PGA", NULL, "VINL" }, + { "Right PGA", NULL, "VINR" }, + { "ADC", NULL, "Left PGA" }, + { "ADC", NULL, "Right PGA" }, +}; + +static int adav80x_set_deemph(struct snd_soc_codec *codec) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->deemph) { + switch (adav80x->rate) { + case 0: + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + break; + case 32000: + val = ADAV80X_DAC_CTRL2_DEEMPH_32; + break; + case 44100: + val = ADAV80X_DAC_CTRL2_DEEMPH_44; + break; + case 48000: + default: + val = ADAV80X_DAC_CTRL2_DEEMPH_48; + break; + } + + } else { + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + } + + return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); +} + +static int adav80x_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + adav80x->deemph = deemph; + + return adav80x_set_deemph(codec); +} + +static int adav80x_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = adav80x->deemph; + return 0; +}; + +static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0); + +static const struct snd_kcontrol_new adav80x_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL, + ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL, + ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL, + ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv), + + SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0), + SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1), + + SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0), + + SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0, + adav80x_get_deemph, adav80x_put_deemph), +}; + +static unsigned int adav80x_port_ctrl_regs[2][2] = { + { ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, }, + { ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL }, +}; + +static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int capture = 0x00; + unsigned int playback = 0x00; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + capture |= ADAV80X_CAPTURE_MODE_MASTER; + playback |= ADAV80X_PLAYBACK_MODE_MASTER; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + capture |= ADAV80X_CAPTURE_MODE_I2S; + playback |= ADAV80X_PLAYBACK_MODE_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + capture |= ADAV80X_CAPTURE_MODE_LEFT_J; + playback |= ADAV80X_PLAYBACK_MODE_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + capture |= ADAV80X_CAPTURE_MODE_RIGHT_J; + playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, + capture); + snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + + adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + return 0; +} + +static int adav80x_set_adc_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_ADC_CTRL1_MODULATOR_128FS; + else + val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; + + snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); + + return 0; +} + +static int adav80x_set_dac_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS; + else + val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; + + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, + val); + + return 0; +} + +static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + unsigned int val; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_CAPTURE_WORD_LEN16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_CAPTRUE_WORD_LEN18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_CAPTURE_WORD_LEN20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_CAPTURE_WORD_LEN24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_WORD_LEN_MASK, val); + + return 0; +} + +static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + ADAV80X_PLAYBACK_MODE_MASK, val); + + return 0; +} + +static int adav80x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + adav80x_set_playback_pcm_format(codec, dai, + params_format(params)); + adav80x_set_dac_clock(codec, rate); + } else { + adav80x_set_capture_pcm_format(codec, dai, + params_format(params)); + adav80x_set_adc_clock(codec, rate); + } + adav80x->rate = rate; + adav80x_set_deemph(codec); + + return 0; +} + +static int adav80x_configure_pll(struct snd_soc_codec *codec, unsigned int pll, + int source, unsigned int freq_out) +{ + unsigned int pll_ctrl2 = 0; + unsigned int pll_src; + + if (freq_out > 12288000) { + pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll); + freq_out /= 2; + } + + /* freq_out = sample_rate * 256 */ + switch (freq_out) { + case 8192000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll); + break; + case 11289600: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll); + break; + case 12288000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll); + break; + default: + return -EINVAL; + } + + if (source == ADAV80X_CLK_XIN) + pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll); + else + pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll); + + snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + ADAV80X_PLL_CLK_SRC_PLL_MASK(pll), pll_src); + + snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + ADAV80X_PLL_CTRL2_PLL_MASK(pll), pll_ctrl2); + + return 0; +} + +static int adav80x_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + enum adav80x_clk_src new_src; + unsigned int pll_ctrl1 = ADAV80X_PLL_CTRL1_PLL2PD; + + switch (source) { + case ADAV80X_PLL_SRC_XTAL: + case ADAV80X_PLL_SRC_XIN: + new_src = ADAV80X_CLK_XIN; + break; + case ADAV80X_PLL_SRC_MCLKI: + new_src = ADAV80X_CLK_MCLKI; + break; + default: + return -EINVAL; + } + + if (source != ADAV80X_PLL_SRC_XTAL) + pll_ctrl1 |= ADAV80X_PLL_CTRL1_XTLPD; + + if (freq_out) { + switch (freq_in) { + case 27000000: + break; + case 54000000: + pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; + break; + default: + return -EINVAL; + } + + adav80x_configure_pll(codec, 0, new_src, freq_out); + + snd_soc_write(codec, ADAV80X_PLL_CTRL1, pll_ctrl1); + + if (adav80x->clk_src == ADAV80X_CLK_PLL1) + return 0; + + adav80x->clk_src = ADAV80X_CLK_PLL1; + + /* DAC, ADC, ICLK clock source - PLL1 */ + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x4a); + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x10); + + } else { + if (adav80x->clk_src == new_src) + return 0; + + adav80x->clk_src = new_src; + + if (new_src == ADAV80X_CLK_XIN) { + /* DAC, ADC, ICLK clock source - XIN */ + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x00); + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x00); + } else { + /* DAC, ADC, ICLK clock source - MCLKI */ + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, 0x25); + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, 0x01); + } + + pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLL1PD; + snd_soc_write(codec, ADAV80X_PLL_CTRL1, pll_ctrl1); + } + + return 0; +} + +static int adav80x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask = ADAV80X_DAC_CTRL1_PD; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +/* Enforce the same sample rate on all audio interfaces */ +static int adav80x_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active || !adav80x->rate) + return 0; + + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate); +} + +static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active) + adav80x->rate = 0; +} + +static const struct snd_soc_dai_ops adav80x_dai_ops = { + .set_fmt = adav80x_set_dai_fmt, + .hw_params = adav80x_hw_params, + .set_pll = adav80x_set_dai_pll, + .startup = adav80x_dai_startup, + .shutdown = adav80x_dai_shutdown, +}; + +#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000) + +#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + +#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver adav80x_dais[] = { + { + .name = "adav80x-hifi", + .id = 0, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, + { + .name = "adav80x-aux", + .id = 1, + .playback = { + .stream_name = "Aux Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "Aux Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, +}; + +static int adav80x_probe(struct snd_soc_codec *codec) +{ + int ret; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + if (ret) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Power down SYSCLK output, power down S/PDIF receiver */ + snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x27); + /* Disable DAC zero flag */ + snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + + return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +} + +static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int adav80x_resume(struct snd_soc_codec *codec) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +} + +static int adav80x_remove(struct snd_soc_codec *codec) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static struct snd_soc_codec_driver adav80x_codec_driver = { + .probe = adav80x_probe, + .remove = adav80x_remove, + .suspend = adav80x_suspend, + .resume = adav80x_resume, + .set_bias_level = adav80x_set_bias_level, + + .reg_word_size = sizeof(u8), + .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), + .reg_cache_default = adav80x_default_regs, + + .controls = adav80x_controls, + .num_controls = ARRAY_SIZE(adav80x_controls), + .dapm_widgets = adav80x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets), + .dapm_routes = adav80x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), +}; + +static int __devinit adav80x_bus_probe(struct device *dev, + enum snd_soc_control_type control_type) +{ + struct adav80x *adav80x; + int ret; + + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); + if (!adav80x) + return -ENOMEM; + + dev_set_drvdata(dev, adav80x); + adav80x->control_type = control_type; + + ret = snd_soc_register_codec(dev, &adav80x_codec_driver, + adav80x_dais, ARRAY_SIZE(adav80x_dais)); + if (ret) + kfree(adav80x); + + return ret; +} + +static int __devexit adav80x_bus_remove(struct device *dev) +{ + snd_soc_unregister_codec(dev); + kfree(dev_get_drvdata(dev)); + return 0; +} + +#if defined(CONFIG_SPI_MASTER) +static int __devinit adav80x_spi_probe(struct spi_device *spi) +{ + return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); +} + +static int __devexit adav80x_spi_remove(struct spi_device *spi) +{ + return adav80x_bus_remove(&spi->dev); +} + +static struct spi_driver adav80x_spi_driver = { + .driver = { + .name = "adav801", + .owner = THIS_MODULE, + }, + .probe = adav80x_spi_probe, + .remove = __devexit_p(adav80x_spi_remove), +}; +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id adav80x_id[] = { + { "adav803", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adav80x_id); + +static int __devinit adav80x_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return adav80x_bus_probe(&client->dev, SND_SOC_I2C); +} + +static int __devexit adav80x_i2c_remove(struct i2c_client *client) +{ + return adav80x_bus_remove(&client->dev); +} + +static struct i2c_driver adav80x_i2c_driver = { + .driver = { + .name = "adav803", + .owner = THIS_MODULE, + }, + .probe = adav80x_i2c_probe, + .remove = __devexit_p(adav80x_i2c_remove), + .id_table = adav80x_id, +}; +#endif + +static int __init adav80x_init(void) +{ + int ret = 0; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&adav80x_i2c_driver); + if (ret) + return ret; +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&adav80x_spi_driver); +#endif + + return ret; +} +module_init(adav80x_init); + +static void __exit adav80x_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&adav80x_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&adav80x_spi_driver); +#endif +} +module_exit(adav80x_exit); + +MODULE_DESCRIPTION("ASoC ADAV80x driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_AUTHOR("Yi Li <yi.li(a)analog.com>>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h new file mode 100644 index 0000000..fab23cd --- /dev/null +++ b/sound/soc/codecs/adav80x.h @@ -0,0 +1,23 @@ +/* + * header file for ADAV80X parts + * + * Copyright 2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _ADAV80X_H +#define _ADAV80X_H + +enum adav80x_pll_src { + ADAV80X_PLL_SRC_XIN, + ADAV80X_PLL_SRC_XTAL, + ADAV80X_PLL_SRC_MCLKI, +}; + +enum adav80x_pll { + ADAV80X_PLL1 = 0, + ADAV80X_PLL2 = 1, +}; + +#endif -- 1.7.2.5

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[alsa-devel] How to implement capture_switch control?
by Adam Jiang 23 Jun '11

23 Jun '11

Hello list, I have a device with wm8903 codec. To support mute/unmute capture signal, I was trying to enable/disable PGA on this codec from user space application like snd_ctl_elem_value_set_boolean("Left Input PGA Switch", 0, 0) That means the mute/unmute function became platform dependent. I got the idea to support mute/unmute function with enabling/disabling capture switch. However, the function snd_mixer_selem_has_capture_switch() alwasy return false. I guess something is missing in my driver. Could you please give some bright how could I use capture switch to mute/unmute a channel. How could I implement capture_switch control properly? Thank you! /Adam

2 4

[alsa-devel] [PATCH] ALSA - HDA - Add period time constraint to hda-intel
by Raymond Yau 23 Jun '11

23 Jun '11

As there is a 10 seconds schedule_timeout in alsa-kernel The following error occur when aplay/arecord using a period time longer than 10 seconds for for those hda codec which support rate lower than 32000 when prealloc = 4096 ALSA hda_intel.c:1680: azx_pcm_prepare: bufsize=0x1aeb00, format=0x4111 ... ALSA pcm_native.c:1531: playback drain error (DMA or IRQ trouble?) aplay -v -Dhw:1 --period-time=10001000 22050.wav Playing WAVE '22050.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Stereo Hardware PCM card 1 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 22050 exact rate : 22050 (22050/1) msbits : 16 buffer_size : 441024 period_size : 220512 period_time : 10000544 tstamp_mode : NONE period_step : 1 avail_min : 220512 period_event : 0 start_threshold : 441024 stop_threshold : 441024 silence_threshold: 0 silence_size : 0 boundary : 1806434304 appl_ptr : 0 hw_ptr : 0 ALSA hda_intel.c:1680: azx_pcm_prepare: bufsize=0x9c500, format=0x511 ALSA hda_codec.c:1295: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x511 ALSA pcm_lib.c:1789: capture write error (DMA or IRQ trouble?) arecord -Dhw:1 -v --period-time=10005000 -r 8000 -c2 -f S16_LE /dev/null Recording WAVE '/dev/null' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo Hardware PCM card 1 'HDA Intel' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 160064 period_size : 80032 period_time : 10004000 tstamp_mode : NONE period_step : 1 avail_min : 80032 period_event : 0 start_threshold : 1 stop_threshold : 160064 silence_threshold: 0 silence_size : 0 boundary : 1311244288 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:1773: read error: Input/output error

2 4

[alsa-devel] ASoC: Sharing Codec on SSC TDM
by Fox, Jason P. 22 Jun '11

22 Jun '11

I have an application where I need an AD193X codec to be shared between 8 audio streams using one SSC channel on an AT91SAM9263. The AD1938 codec as configured in my driver has 8 channels. I need to be able to send audio to the codec from 8 different tasks simultaneously. Each task is to use only one channel. We have the 8 TDM channels working in the driver. The first task can access the codec and play audio just fine. The second task cannot open the device and gets an EBUSY return from snd_pcm_hw_open() because the card is now owned by the first task. However, the first task is only using one of the 8 channels. I want the second task to use the second channel, and so on. I can create an asound.conf that gives access to any one of the 8 channels, but only to one task at a time. I've tried using dshare and dmix, but that swamps the AT91 and there are lots of audio dropouts. There is no need to mix audio between any of the streams, i.e. they are all to be discrete audio outputs. What I am looking for is a way to structure the ASOC interface to allow use of all 8 channels simultaneously. If anyone can point me in a direction, recommend a solution, or tell me where else to look it would be VERY much appreciated. Thank you for any help, Jason Fox GAI-Tronics ******************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This footnote also confirms that this email message has been swept for the presence of computer viruses. www.Hubbell.com - Hubbell Incorporated**

1 0

[alsa-devel] [PATCH] ASoC: core - Fix dai usage in pcm_new() platform handlers
by Daniel Mack 22 Jun '11

22 Jun '11

Commit 552d1ef ("ASoC: core - Optimise and refactor pcm_new() to pass only rtd") refactored the ASoC internal core to only pass a pointer to a runtime as argument to individual pcm_new() functions. However, the new code that was added to the handlers to get access to the dai now uses rtd->cpu_dai, while before this patch, rtd->codec_dai was used. This patch fixes all the pcm_new() handlers that were affected. Tested on a PXA platform. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Liam Girdwood <lrg(a)ti.com> --- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 2 +- sound/soc/imx/imx-pcm-fiq.c | 2 +- sound/soc/imx/imx-ssi.c | 2 +- sound/soc/jz4740/jz4740-pcm.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mid-x86/sst_platform.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/s6000/s6000-pcm.c | 2 +- sound/soc/samsung/dma.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- 19 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index f81d4c3..d1140c1 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -367,7 +367,7 @@ static u64 atmel_pcm_dmamask = 0xffffffff; static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 9e59f68..fdaed48 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -421,7 +421,7 @@ static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 96d0d90..9bdf1ea 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -251,7 +251,7 @@ static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index c95cc03..2d7e176 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -286,7 +286,7 @@ static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a49e667..35b60f9 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -836,7 +836,7 @@ static u64 davinci_pcm_dmamask = 0xffffffff; static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret; diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 732208c..932da5f 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -297,7 +297,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); int ret; diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 19ad0c1..729b0f8 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -302,7 +302,7 @@ static u64 psc_dma_dmamask = 0xffffffff; static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); size_t size = psc_dma_hardware.buffer_bytes_max; diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 309c59e..dc8296a 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -241,7 +241,7 @@ static int ssi_irq = 0; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 158a91c..5b0972b 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -391,7 +391,7 @@ static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c index a7c9578..5a34dee 100644 --- a/sound/soc/jz4740/jz4740-pcm.c +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -302,7 +302,7 @@ static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cd33de1..75255e4 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -315,7 +315,7 @@ static int kirkwood_dma_preallocate_dma_buffer(struct snd_pcm *pcm, static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret; diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 3e78260..92de19c 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -404,7 +404,7 @@ static void sst_pcm_free(struct snd_pcm *pcm) int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int retval = 0; diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index d589ef1..1964801 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -318,7 +318,7 @@ static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; if (!card->dev->dma_mask) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b2f5751..0787a54 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -369,7 +369,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index d73d6f6..5438a67 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -88,7 +88,7 @@ static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 80c85fd..925192d 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -446,7 +446,7 @@ static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) { struct snd_card *card = runtime->card->snd_card; - struct snd_soc_dai *dai = runtime->cpu_dai; + struct snd_soc_dai *dai = runtime->codec_dai; struct snd_pcm *pcm = runtime->pcm; struct s6000_pcm_dma_params *params; int res; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 9465588..a2475b2 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -428,7 +428,7 @@ static u64 dma_mask = DMA_BIT_MASK(32); static int dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index ff86e5e..efd00a3 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -325,7 +325,7 @@ static u64 tegra_dma_mask = DMA_BIT_MASK(32); static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; int ret = 0; diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 34aa972..5be6191 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -290,7 +290,7 @@ static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = rtd->codec_dai; struct snd_pcm *pcm = rtd->pcm; struct platform_device *pdev = to_platform_device(dai->platform->dev); struct txx9aclc_soc_device *dev; -- 1.7.5.4

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[alsa-devel] ASoC: Patches for an STA32X and WM8782
by Daniel Mack 22 Jun '11

22 Jun '11

Hi, I'm sending two patches to support the ST STA32x codec and the WM8782 ADC. The code was originally written by Johannes Stezenbach for 2.6.36. As the ASoC internals changed quite a bit due to the multi-component reconstruction, I took care for the code transition to make these drivers compatible to more recent versions. If there's anything left to care for in these drivers, please let me know. Daniel [PATCH 1/2] ALSA: ASoC: add STA32X codec driver [PATCH 2/2] ALSA: ASoC: add WM8782 ADC Codec Driver

6 37

[alsa-devel] Sharing PXA SSP ports
by Daniel Mack 22 Jun '11

22 Jun '11

The current API for requesting an SSP port on PXA platforms assumes that the caller of pxa_ssp_request() is the only user of this port, so both input and output streams must be implemented in the same driver. In a new setup I'm working on, this is not sufficient as we connect one codec to the input stream and another one to the output stream of the same CPU DAI "pxa-ssp-dai.0". I'm currently using a hack that disables the use count check in pxa_ssp_request(), but I certainly need a cleaner solution. What would be a good approach? Would we need a multiplexer for this hardware bus, or can we extend the DAI driver in any clever way to achieve that? Or is there a solution already I just don't know about? Thanks, Daniel

2 3

[alsa-devel] Query about SNDRV_PCM_FORMAT_GSM
by Patrick Lai 22 Jun '11

22 Jun '11

Hi, I see SNDRV_PCM_FORMAT_GSM is defined in asound.h. However, I do not find any reference to this macro except pcm_misc.c. I am curious how bps info(i.e 4.75 kbit/s) is passed down to the ALSA driver. Any pointers? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] [PATCH] ALSA: isight: add now required waking of DMA context
by Stefan Richter 22 Jun '11

22 Jun '11

Since commit 13882a82ee16, upper-layer firewire drivers are required to flush after enqueueing a batch of isochronous buffers. Signed-off-by: Stefan Richter <stefanr(a)s5r6.in-berlin.de> --- Actually, snd-isight still worked for me without it but I guess I was just lucky that the controllers (FW643-e, JMB381, XIO2213A) never read up to the current tail of the DMA program. Dismiss this if you already have something like this in the pipeline. sound/firewire/isight.c | 1 + 1 file changed, 1 insertion(+) Index: b/sound/firewire/isight.c =================================================================== --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -213,6 +213,7 @@ static void isight_packet(struct fw_iso_ if (++index >= QUEUE_LENGTH) index = 0; isight->packet_index = index; + fw_iso_context_queue_flush(isight->context); } static int isight_connect(struct isight *isight) -- Stefan Richter -=====-==-== -=-= ===== http://arcgraph.de/sr/

2 3

[PATCH] ALSA: HDA: Pinfix quirk for HP Z200 Workstation
by David Henningsson 22 Jun '11

22 Jun '11

BIOS lists the internal speaker as an internal line-out. Change to internal speaker + model=auto for better auto-mute capabilities. BugLink: http://bugs.launchpad.net/bugs/754964 Reported-by: Marc Legris <marc.legris(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4adf9e2..fe56b2b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12600,6 +12600,7 @@ static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { */ enum { PINFIX_FSC_H270, + PINFIX_HP_Z200, }; static const struct alc_fixup alc262_fixups[] = { @@ -12612,9 +12613,17 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, + [PINFIX_HP_Z200] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x16, 0x99130120 }, /* internal speaker */ + { } + } + }, }; static const struct snd_pci_quirk alc262_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270), {} }; @@ -12731,6 +12740,8 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = { ALC262_HP_BPC), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", + ALC262_AUTO), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), -- 1.7.4.1 --------------050602020501040407030505--

1 0

Re: [alsa-devel] [PATCH] ASoC: Add support for Cirrus Logic CS42L52 Lowpower Stereo Codec
by Mark Brown 22 Jun '11

22 Jun '11

On Thu, Feb 25, 2010 at 06:07:35PM -0600, Brian wrote: This is a quick first pass review only. It'd be best if you could include a Signed-off-by line in submissions so that they can be merged if they're OK, even if you expect problems you may be surprised! Overall this looks fairly good - there's a lot of comments below but the vast majority of them are stylistic issues or needing to update to current APIs rather than major structural problems with the code. The main thing I don't really follow is the register access code, there's quite a bit of stuff going on in there. > + * published by the Free Software Foundation. > + * Revision history > + * Nov 2007 Initial version. > + * Oct 2008 Updated to 2.6.26 > + * Feb 2010 Updated to latest asoc git tree Please remove the changelog, once things are merged git takes care of that. > + */ > +#define DEBUG Not by default in mainline :) > +#ifdef DEBUG > +#define SOCDBG(fmt, arg...) printk(KERN_ERR "%s: %s() " fmt, SOC_CS42L52_NAME, __FUNCTION__, ##arg) > +#else > +#define SOCDBG(fmt, arg...) > +#endif > +#define SOCINF(fmt, args...) printk(KERN_INFO "%s: " fmt, SOC_CS42L52_NAME, ##args) > +#define SOCERR(fmt, args...) printk(KERN_ERR "%s: " fmt, SOC_CS42L52_NAME, ##args) Please convert the driver to use the standard dev_ and pr_ macros for this (where possible the dev_ ones are better). > +static int soc_cs42l52_set_bias_level(struct snd_soc_codec *codec, > + enum snd_soc_bias_level level); > +static int soc_cs42l52_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai); > +static int soc_cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, > + int clk_id, u_int freq, int dir); > +static int soc_cs42l52_digital_mute(struct snd_soc_dai *dai, int mute); > +static int soc_cs42l52_set_fmt(struct snd_soc_dai *codec_dai, > + u_int fmt); > + > +static unsigned int soc_cs42l52_read(struct snd_soc_codec *codec, > + u_int reg); Please use 'unsigned int' rather than 'u_int' for consistency with the rest of ASoC. It's a bit surprising that you need the forward declarations at all but they do no harm. > +/** > + * snd_soc_get_volsw - single mixer get callback > + * @kcontrol: mixer control > + * @uinfo: control element information > + * > + * Callback to get the value of a single mixer control. > + * > + * Returns 0 for success. > + */ > +int snd_soc_cs42l5x_get_volsw(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); > + int reg = kcontrol->private_value & 0xff; The indentation looks wrong - should be tabs for indenting the code blocks. scripts/checkpatch.pl is a pretty good check for this sort of stuff, I'll not mention other issues. > + int shift = (kcontrol->private_value >> 8) & 0x0f; > + int rshift = (kcontrol->private_value >> 12) & 0x0f; > + int max = (kcontrol->private_value >> 16) & 0xff; > + int mask = (1 << fls(max)) - 1; > + int min = (kcontrol->private_value >> 24) & 0xff; > + > + ucontrol->value.integer.value[0] = > + ((snd_soc_read(codec, reg) >> shift) - min) & mask; > + if (shift != rshift) > + ucontrol->value.integer.value[1] = > + ((snd_soc_read(codec, reg) >> rshift) - min) & mask; This and most of the other custom control stuff looks an awful lot like it should be done by adding a SND_SOC_DOUBLE_SIGNED() or something along the lines of the existing _S8() but I didn't read entirely closely. I was speaking to someone on IRC only the other day who needed some of those types for a different device. These should also be updated to use struct soc_mixer_control for the private value rather than shifting everything into the private value, it allows wider ranges and is less error prone. > +static const u8 soc_cs42l52_reg_default[] = { > + 0x00, 0xE0, 0x01, 0x07, 0x05, /*4*/ > + 0xa0, 0x00, 0x00, 0x81, /*8*/ Very nitpicky but you're using a mix of upper and lower case for the letters in the hex digits. > +static inline int soc_cs42l52_read_reg_cache(struct snd_soc_codec *codec, > + u_int reg) > +{ > + u8 *cache = codec->reg_cache; > + > + return reg > SOC_CS42L52_REG_NUM ? -EINVAL : cache[reg]; > +} You should be able to replace all your register I/O code with use of soc-cache.c with 8 bit data, 8 bit register. Except... > + if(info->flags & SOC_CS42L52_ALL_IN_ONE) > + { > + for(i = 0; i < codec->num_dai; i++) > + { > + if(info->machine_handler) > + info->machine_handler(i); ...I'm not sure what's going on here but it could really use an explanation? I'm not sure what ALL_IN_ONE is supposed to do, or what the machine_handler() function is for. > + if(info->flags & SOC_CS42L52_CHIP_SWICTH) > + { > + num = info->flags & SOC_CS42L52_CHIP_MASK; > + if(info->machine_handler) > + info->machine_handler(num); > + } This too. There's a lot of unusual code using this machine handler function. I suspect this can all use ASoC standard stuff. > +static inline int soc_cs42l52_get_revison(struct snd_soc_codec *codec) > +{ > + u8 data; > + u8 addr; > + int num, ret = 0; > + struct soc_codec_cs42l52 *info = (struct soc_codec_cs42l52*)codec->private_data; No need to cast away from void (quite a few examples of this in the code). > + return ret < 0 ? ret : data; I'm not a big fan of the ternery operator... > +static unsigned int soc_cs42l52_read(struct snd_soc_codec *codec, > + u_int reg) > +{ > + u8 data; > + u8 addr; > + int i, ret = 0; > + struct soc_codec_cs42l52 *info = (struct soc_codec_cs42l52*)codec->private_data; > +#ifndef CONFIG_CS42L52_DEBUG > + if(reg == CODEC_CS42L52_SPK_STATUS) > + { > +#endif This really shouldn't be ifdefed. If you need to handle registers that report status from the device with the soc-cache stuff use the volatile_register() callback in the CODEC, any registers that need a CODEC read will call through to the hardware rather than using the cache. > +static const char *cs42l52_adc_mux[] = {"AIN1", "AIN2", "AIN3", "AIN4", "PGA"}; > +static const char *cs42l52_mic_mux[] = {"MIC1", "MIC2"}; > +static const char *cs42l52_stereo_mux[] = {"Mono", "Stereo"}; > +static const char *cs42l52_off[] = {"On", "Off"}; > +static const char *cs42l52_hpmux[] = {"Off", "On"}; > + > +static const struct soc_enum soc_cs42l52_enum[] = { > +SOC_ENUM_DOUBLE(CODEC_CS42L52_ANALOG_HPF_CTL, 4, 6, 2, cs42l52_hpf_freeze), /*0*/ > +SOC_ENUM_SINGLE(CODEC_CS42L52_ADC_HPF_FREQ, 0, 4, cs42l52_hpf_corner_freq), Don't use an array of enums, define individual variables for them. Some drivers use the arrays for historical reasons - they're quite painful to work with if you ever want to change anything in the middle and we've found a few off by one errors in the referencing into the arrays before :/. > +static const struct snd_kcontrol_new soc_cs42l52_controls[] = { > + > +SOC_ENUM("Mic VA Capture Switch", soc_cs42l52_enum[18]), /*0*/ enums shouldn't be called Switch - Switch has a specific meaning to ALSA (see Documentation/sound/alsa/ControlNames.txt, there's quite a few other controls could do with having the components of the name reordered and various other things following that). Normally just omit the ' Switch' from the name. > +SOC_DOUBLE("Analog SR Capture Switch", CODEC_CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1), > +SOC_DOUBLE("Analog ZC Capture Switch", CODEC_CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1), This'd normally be 'Capture ZC Switch' if it's for zero cross. > +SOC_ENUM("HPF corner freq Capture Switch", soc_cs42l52_enum[1]), /*5*/ > + > +SOC_SINGLE("Ganged Ctl Capture Switch", CODEC_CS42L52_ADC_MISC_CTL, 7, 1, 1), /* should be enabled init */ What does the comment mean here? > +SOC_SINGLE("HP Analog Gain Playback Volume", CODEC_CS42L52_PB_CTL1, 5, 7, 0), Just "Headphone Volume"? > +SOC_SINGLE("Playback B=A Volume Playback Switch", CODEC_CS42L52_PB_CTL1, 4, 1, 0), /*10*/ /*should be enabled init*/ If this is the mute for the headphone volume it should have the same name but with Switch instead of Volume so applications like alsamixer can display the two together. > +static int soc_cs42l52_add_controls(struct snd_soc_codec *codec) > +{ > + int i,ret = 0; > + > + for(i = 0; i < ARRAY_SIZE(soc_cs42l52_controls); i++) > + { > + ret = snd_ctl_add(codec->card, > + snd_soc_cnew(&soc_cs42l52_controls[i], codec, NULL)); > + if(ret < 0) > + { > + SOCDBG("add cs42l52 controls failed\n"); > + break; Replace this with snd_soc_add_controls(). > + /* Sum switches */ > + SND_SOC_DAPM_PGA("AIN1A Switch", CODEC_CS42L52_ADC_PGA_A, 0, 0, NULL, 0), Without having checked the datasheet I *suspect* most of these switches should be modelled as inputs to a mixer (possibly one with no power control of its own)? The "Switch" in the name certainly looks very out of place in the name of a DAPM control. > + /* Output path */ > + SND_SOC_DAPM_MUX("Passthrough Left Playback Switch", SND_SOC_NOPM, 0, 0, &cs42l52_hpa_mux), Normally "Bypass" instead of "Passthrough". > + /* Mic Bias */ > + {"Mic Bias Capture Switch", "On", "PGA MICA"}, > + {"Mic Bias Capture Switch", "On", "PGA MICB"}, > + {"Mic-Bias", NULL, "Mic Bias Capture Switch"}, > + {"Mic-Bias", NULL, "Mic Bias Capture Switch"}, > + {"ADC Mux Left Capture Switch", "PGA", "Mic-Bias"}, > + {"ADC Mux Right Capture Switch", "PGA", "Mic-Bias"}, > + {"Passthrough Left Playback Switch", "On", "Mic-Bias"}, > + {"Passthrough Right Playback Switch", "On", "Mic-Bias"}, This looks odd - normally the microphone bias would be connected to the microphone input externally and would not be hooked up as an input wihtin the CODEC? > + /* terminator */ > + {NULL, NULL, NULL}, Not needed any more (I'm vaugely surprised this doesn't generate an error). > +#define SOC_CS42L52_RATES ( SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ > + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ > + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ > + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ > + SNDRV_PCM_RATE_96000 ) /*refer to cs42l52 datasheet page35*/ This is equivalent to SNDRV_PCM_RATE_8000_96000 > +/* #define CONFIG_MANUAL_CLK */ This might use some comments (or should possibly be deleted)? > + soc_cs42l52_set_bias_level(soc_codec, SND_SOC_BIAS_OFF); > + soc_codec->bias_level = SND_SOC_BIAS_STANDBY; > + schedule_delayed_work(&soc_codec->delayed_work, msecs_to_jiffies(1000)); > + soc_cs42l52_required_setup(soc_codec); A comment about what the delayed work is up to wouldn't hurt? > + /*set hp default volume*/ > + soc_cs42l52_write(soc_codec, CODEC_CS42L52_HPA_VOL, DEFAULT_HP_VOL); > + soc_cs42l52_write(soc_codec, CODEC_CS42L52_HPB_VOL, DEFAULT_HP_VOL); For pretty much all the configuration you're doing here just let userspace set things up, there's no telling what's an appropriate setting on any given system and presumably the chip designers made a reasonable effort to ensure that the default state of the chip is at least not harmful. > +#ifdef CONFIG_MANUAL_CLK > + soc_cs42l52_write(soc_codec, CODEC_CS42L52_CLK_CTL, > + (soc_cs42l52_read(soc_codec, CODEC_CS42L52_CLK_CTL) > + & ~CLK_CTL_AUTODECT_ENABLE)); Pass this configuration in as platform data, this will allow multiple boards to build from the same kernel. > + ret = snd_soc_init_card(soc_dev); > + if(ret) > + { > + SOCERR("add snd card failed\n"); > + goto card_err; > + } Please develop against git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 (or whatever the latest kernel version branch is at any given time.) You should also be calling snd_soc_register_codec() and snd_soc_register_dai() from this function. > +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) No need for the ifdefs if you only support I2C, the ifdefs are used by drivers that have multiple control interfaces (typically I2C plus SPI). > + } > + > + > + return ret; > +} > + > +static int cs42l52_i2c_remove(struct i2c_client *client) > +{ > + struct snd_soc_codec *soc_codec; > + if(client) > + { > + soc_codec = i2c_get_clientdata(client); > + if(soc_codec) > + kfree(soc_codec->reg_cache); > + } > + return 0; > +} This should be unregistering from ASoC as well, or refusing to remove if not implemented. > + case SND_SOC_DAIFMT_DSP_B: > + SOCINF("unsupported format\n"); > + ret = -EINVAL; > + goto done; > + default: > + SOCINF("invaild format\n"); > + ret = -EINVAL; > + goto done; Could just squash these together. > + case SND_SOC_DAIFMT_IB_IF: > + SOCDBG("codec dai fmt inversed sclk\n"); > + iface |= IFACE_CTL1_INV_SCLK; > + break; > + case SND_SOC_DAIFMT_IB_NF: > + iface |= IFACE_CTL1_INV_SCLK; > + break; It looks like the hardware only supports the inversion of one of the clocks so only two of the clock polarity combinations should be supported? > + info->format = iface; I'd really expect to see interaction with the hardware here? > + /* 96k */ > + {12288000, 96000, CLK_CTL_S_DS_MODE, CLK_CTL_NOT_32K, CLK_CTL_NOT_27M, CLK_CTL_RATIO_128, 0}, > + {18432000, 96000, CLK_CTL_S_DS_MODE, CLK_CTL_NOT_32K, CLK_CTL_NOT_27M, CLK_CTL_RATIO_128, 0},/*29*/ > + {12000000, 96000, CLK_CTL_S_DS_MODE, CLK_CTL_NOT_32K, CLK_CTL_NOT_27M, CLK_CTL_RATIO_125, 0}, > + {24000000, 96000, CLK_CTL_S_DS_MODE, CLK_CTL_NOT_32K, CLK_CTL_NOT_27M, CLK_CTL_RATIO_125, 1}, If you're feeling adventurous you should be able to use this data to set up constraints for the DAI in the startup() callback so only sample rates supported from the system clock can be used by userspace. > +static int soc_cs42l52_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + int ret = 0; > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_device *soc_dev = rtd->socdev; > + struct snd_soc_codec *soc_codec = soc_dev->card->codec; > + struct soc_codec_cs42l52 *info = (struct soc_codec_cs42l52*)soc_codec->private_data; > + > + u32 clk = 0; > + int index = soc_cs42l52_get_clk(info->sysclk, params_rate(params)); > +static int soc_cs42l52_suspend(struct platform_device *pdev, pm_message_t state) > +{ > + struct snd_soc_device *soc_dev = (struct snd_soc_device*)platform_get_drvdata(pdev); > + struct snd_soc_codec *soc_codec = soc_dev->card->codec; > + > + soc_cs42l52_write(soc_codec, CODEC_CS42L52_PWCTL1, PWCTL1_PDN_CODEC); > + soc_cs42l52_set_bias_level(soc_codec, SND_SOC_BIAS_OFF); I'd expect that set_bias_level() would do what the explict write does? > +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) > + if(soc_codec_data->i2c_addr) > + { > + ret = i2c_add_driver(&cs42l52_i2c_drv); > + > + if(ret) > + { > + SOCERR("register i2c driver failed\n"); > + goto out; > + > + } > + SOCINF("Cirrus Logic ALSA SoC codec cs42l52 driver verison 1.2 2/2010\n"); > + return ret; > + } This should be changed to use standard device model registration - see http://git.kernel.org/?p=linux/kernel/git/broonie/sound-2.6.git;a=commit;h=… for an example of doing the conversion. > +struct soc_codec_cs42l52_data{ > + u_short i2c_addr; > +}; With the device model stuff this won't be needed any more. > +/* > + *CS42L52 internal registers > + */ > +#define CODEC_CS42L52_CHIP 0x01 > +#define CHIP_ID 0xE0 > +#define CHIP_ID_MASK 0xF8 Many of the register names and other definitions for register fields and values need to be namespace to avoid collisions.

4 5

[alsa-devel] [PATCH] ASoC: codecs: Max98095: Fix logging of hardware revision.
by Taylor Hutt 22 Jun '11

22 Jun '11

The base hardware revision of the Maxim 98095 part is 0x40; the code which outputs the revision of the hardware has been updated to properly use uppercase alphabetic values for the revision numbers. Also, the use of a constant for the length 'max98095_dai' has been replaced with ARRAY_SIZE(). Signed-off-by: Taylor Hutt <thutt(a)chromium.org> --- sound/soc/codecs/max98095.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index e1d282d..9a793b3 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2261,11 +2261,11 @@ static int max98095_probe(struct snd_soc_codec *codec) ret = snd_soc_read(codec, M98095_0FF_REV_ID); if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", + dev_err(codec->dev, "Failure reading hardware revision: %d\n", ret); goto err_access; } - dev_info(codec->dev, "revision %c\n", ret + 'A'); + dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A'); snd_soc_write(codec, M98095_097_PWR_SYS, M98095_PWRSV); @@ -2342,8 +2342,8 @@ static int max98095_i2c_probe(struct i2c_client *i2c, max98095->control_data = i2c; max98095->pdata = i2c->dev.platform_data; - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_max98095, &max98095_dai[0], 3); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095, + max98095_dai, ARRAY_SIZE(max98095_dai)); if (ret < 0) kfree(max98095); return ret; -- 1.7.3.1

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[PATCH 2/2] ALSA: HDA: Add model=auto quirk for Acer Aspire 3830TG
by David Henningsson 21 Jun '11

21 Jun '11

Since we're not using the new auto parser as a fallback yet, add it manually as a quirk. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6c9b742..91c8c0c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3075,6 +3075,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { }; static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO), SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), -- 1.7.4.1 --------------000100060101070007060501--

1 0

[PATCH 1/2] ALSA: HDA: Add a new Conexant codec ID (506c)
by David Henningsson 21 Jun '11

21 Jun '11

Conexant ID 506c was found on Acer Aspire 3830TG. As users report no playback, sending to stable should be safe. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/783582 Reported-by: andROOM Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6e86427..6c9b742 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4390,6 +4390,8 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5066 }, { .id = 0x14f15069, .name = "CX20585", .patch = patch_cxt5066 }, + { .id = 0x14f1506c, .name = "CX20588", + .patch = patch_cxt5066 }, { .id = 0x14f1506e, .name = "CX20590", .patch = patch_cxt5066 }, { .id = 0x14f15097, .name = "CX20631", @@ -4418,6 +4420,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_ALIAS("snd-hda-codec-id:14f15067"); MODULE_ALIAS("snd-hda-codec-id:14f15068"); MODULE_ALIAS("snd-hda-codec-id:14f15069"); +MODULE_ALIAS("snd-hda-codec-id:14f1506c"); MODULE_ALIAS("snd-hda-codec-id:14f1506e"); MODULE_ALIAS("snd-hda-codec-id:14f15097"); MODULE_ALIAS("snd-hda-codec-id:14f15098"); -- 1.7.4.1 --------------000100060101070007060501 Content-Type: text/x-patch; name="0002-ALSA-HDA-Add-model-auto-quirk-for-Acer-Aspire-3830TG.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0002-ALSA-HDA-Add-model-auto-quirk-for-Acer-Aspire-3830TG.pa"; filename*1="tch"

1 0

Re: [alsa-devel] ALSA Fireworks driver
by Clemens Ladisch 21 Jun '11

21 Jun '11

Doug McLain wrote: > > > [nostar@iggy firewire]$ cat /proc/asound/cards > > > 3 [AudioFire12 ]: Fireworks - AudioFire12 > > > Echo Digital Audio AudioFire12 v5.0, GUID 0014860fc44a84a6 at fw1.0, S400 > > [nostar@iggy export]$ aplay -D plughw:AudioFire12 wg.wav > aplay: main:660: audio open error: No such file or directory > > [nostar@iggy export]$ aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: NVidia [HDA NVidia], device 0: ALC1200 Analog [ALC1200 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: NVidia [HDA NVidia], device 1: ALC1200 Digital [ALC1200 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 There is not PCM device for the AF12. In theory, it shouldn't be possible for the driver to attach to the device without having created the PCM device. Are you sure that you have the latest version of the driver? Where exactly did you get it from? Please check that the fireworks.c file has 2275 lines, like this one: http://git.alsa-project.org/?p=alsa-kprivate.git;a=blob;hb=fireworks;f=soun… Regards, Clemens

2 4

Re: [alsa-devel] Jack event API - decision needed
by Mark Brown 21 Jun '11

21 Jun '11

Sorry about the top posting, but as I wasn't involved in any of the discussions and am on a mobile device right now and your mail isn't directly legible it would be enormously helpful if you could summarize the issues you're trying to address, your proposed solution and the problems Kay had. David Henningsson <david.henningsson(a)canonical.com> wrote: >Hi, > >I'm still new in the community in that sense that I'm not sure how >decisions are made. But I could use the outcome of such a decision. > >Background: I'm trying to pull together the missing pieces of jack >detection on both kernel / plumbing / application layers, so that when >you plug something in (headset, microphone etc), userspace is notified >and can take appropriate actions (e g routing decisions). > >As part of that I wrote a udev patch a few days ago, which nobody >commented on in alsa-devel [1], but was somewhat disliked by at least >Kay Sievers who maintains udev [2], who preferred we would rewrite our >input layer to do something else within ALSA. > >So before I proceed further I'd like to know if > >1) We're continuing the path with /dev/input devices > >2a) We'll rewrite these devices to be read-only ALSA mixer controls > >2b) We'll rewrite these devices to be something within ALSA, but not >exactly mixer controls. > >For options 2a) and 2b) I guess the existing /dev/input thing should be >deprecated and/or removed. So part of decision should maybe be based on >information about how widespread the usage of these devices are >currently...? > >-- >David Henningsson, Canonical Ltd. >http://launchpad.net/~diwic > >[1] >http://mailman.alsa-project.org/pipermail/alsa-devel/2011-June/040916.html > >[2] e g http://www.spinics.net/lists/hotplug/msg04949.html

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[alsa-devel] @Daniel Mack: Send you a FTU
by Aurélien Leblond 21 Jun '11

21 Jun '11

Hi Daniel, I saw on the mailing that you a Felix have been doing a great deal of work to provide better support for the M-Audio FastTrack Ultra. I tested yesterday the new patch that provides mixer and the possibility to change the internal routing of the soundcard... I can finally use all the input/output to use external FX, that's very handy! Anyway, on the M-Audio forum, Felix has been mentionning several times that there were few bugs that needed to be fixed with this soundcard, but that you needed to get your hand on one device to fix them (and may be even improve things around its support in general). As M-Audio generously denied providing you with one (haaaa, these big corporations), I'm ok getting a brand new one sent to you. That would be part of my donation to the open-source community! So the question really is...How would we organise that? Regards, Aurélien

4 11

[alsa-devel] [PATCH] Remove ALC268 model override for CPR2000
by Tony Vroon 21 Jun '11

21 Jun '11

The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER. This keeps headphone automute and microphone input from operating on at least one laptop from Opti Systems. Without the override, the BIOS parser does a fine job setting the card up and everything works. Tested-By: Peter Schneider <e.at.chi.kaen(a)googlemail.com> Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 61a774b..04e6023 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13872,7 +13872,6 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), {} };

2 1

[alsa-devel] [PATCH 1/1][resubmit] Add patch file and Makefile entry for CA0132 HDA codec
by Ian Minett 21 Jun '11

21 Jun '11

Please find attached the CA0132 codec patch resubmission, which builds in changes based on the comments received last week. The changelog added below is based on a git commit message, please let us know if it needs to be in a different format. Thanks, Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> Add patch file and Makefile entry for CA0132 HDA codec Add snd_hda_power_up/down calls to control callbacks. Add chipio_send helper function for neater chip access code. Remove dead code function. Switch debug outs from snd_printd to snd_printdd. Shorten enum constants to shorten line lengths and remove redundant terms. Move extension enable/disable commands to codec init/free callbacks. --- sound/pci/hda/Kconfig | 13 + sound/pci/hda/Makefile | 4 + sound/pci/hda/patch_ca0132.c | 1096 ++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1113 insertions(+), 0 deletions(-) create mode 100644 sound/pci/hda/patch_ca0132.c diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 0ea5cc6..85217bd 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -171,6 +171,19 @@ config SND_HDA_CODEC_CA0110 snd-hda-codec-ca0110. This module is automatically loaded at probing. +config SND_HDA_CODEC_CA0132 + bool "Build Creative CA0132 codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Creative CA0132 codec support in + snd-hda-intel driver. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-ca0132. + This module is automatically loaded at probing. + config SND_HDA_CODEC_CMEDIA bool "Build C-Media HD-audio codec support" default y diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 17ef365..87365d5 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -13,6 +13,7 @@ snd-hda-codec-idt-objs := patch_sigmatel.o snd-hda-codec-si3054-objs := patch_si3054.o snd-hda-codec-cirrus-objs := patch_cirrus.o snd-hda-codec-ca0110-objs := patch_ca0110.o +snd-hda-codec-ca0132-objs := patch_ca0132.o snd-hda-codec-conexant-objs := patch_conexant.o snd-hda-codec-via-objs := patch_via.o snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o @@ -42,6 +43,9 @@ endif ifdef CONFIG_SND_HDA_CODEC_CA0110 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o endif +ifdef CONFIG_SND_HDA_CODEC_CA0132 +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o +endif ifdef CONFIG_SND_HDA_CODEC_CONEXANT obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o endif diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c new file mode 100644 index 0000000..55f58e7 --- /dev/null +++ b/sound/pci/hda/patch_ca0132.c @@ -0,0 +1,1096 @@ +/* + * HD audio interface patch for Creative CA0132 chip + * + * Copyright (c) 2011, Creative Technology Ltd. + * + * Based on patch_ca0110.c + * Copyright (c) 2008 Takashi Iwai <tiwai(a)suse.de> + * + * This driver 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. + * + * This driver 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + +#define WIDGET_CHIP_CTRL 0x15 +#define WIDGET_DSP_CTRL 0x16 + +#define WUH_MEM_CONNID 10 +#define DSP_MEM_CONNID 16 + +enum hda_cmd_vendor_io { + /* for DspIO node */ + VENDOR_DSPIO_SCP_WRITE_DATA_LOW = 0x000, + VENDOR_DSPIO_SCP_WRITE_DATA_HIGH = 0x100, + + VENDOR_DSPIO_STATUS = 0xF01, + VENDOR_DSPIO_SCP_POST_READ_DATA = 0x702, + VENDOR_DSPIO_SCP_READ_DATA = 0xF02, + VENDOR_DSPIO_DSP_INIT = 0x703, + VENDOR_DSPIO_SCP_POST_COUNT_QUERY = 0x704, + VENDOR_DSPIO_SCP_READ_COUNT = 0xF04, + + /* for ChipIO node */ + VENDOR_CHIPIO_ADDRESS_LOW = 0x000, + VENDOR_CHIPIO_ADDRESS_HIGH = 0x100, + VENDOR_CHIPIO_STREAM_FORMAT = 0x200, + VENDOR_CHIPIO_DATA_LOW = 0x300, + VENDOR_CHIPIO_DATA_HIGH = 0x400, + + VENDOR_CHIPIO_GET_PARAMETER = 0xF00, + VENDOR_CHIPIO_STATUS = 0xF01, + VENDOR_CHIPIO_HIC_POST_READ = 0x702, + VENDOR_CHIPIO_HIC_READ_DATA = 0xF03, + + VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE = 0x70A, + + VENDOR_CHIPIO_PLL_PMU_WRITE = 0x70C, + VENDOR_CHIPIO_PLL_PMU_READ = 0xF0C, + VENDOR_CHIPIO_8051_ADDRESS_LOW = 0x70D, + VENDOR_CHIPIO_8051_ADDRESS_HIGH = 0x70E, + VENDOR_CHIPIO_FLAG_SET = 0x70F, + VENDOR_CHIPIO_FLAGS_GET = 0xF0F, + VENDOR_CHIPIO_PARAMETER_SET = 0x710, + VENDOR_CHIPIO_PARAMETER_GET = 0xF10, + + VENDOR_CHIPIO_PORT_ALLOC_CONFIG_SET = 0x711, + VENDOR_CHIPIO_PORT_ALLOC_SET = 0x712, + VENDOR_CHIPIO_PORT_ALLOC_GET = 0xF12, + VENDOR_CHIPIO_PORT_FREE_SET = 0x713, + + VENDOR_CHIPIO_PARAMETER_EX_ID_GET = 0xF17, + VENDOR_CHIPIO_PARAMETER_EX_ID_SET = 0x717, + VENDOR_CHIPIO_PARAMETER_EX_VALUE_GET = 0xF18, + VENDOR_CHIPIO_PARAMETER_EX_VALUE_SET = 0x718 +}; + +/* + * Control flag IDs + */ +enum control_flag_id { + /* Connection manager stream setup is bypassed/enabled */ + CONTROL_FLAG_C_MGR = 0, + /* DSP DMA is bypassed/enabled */ + CONTROL_FLAG_DMA = 1, + /* 8051 'idle' mode is disabled/enabled */ + CONTROL_FLAG_IDLE_ENABLE = 2, + /* Tracker for the SPDIF-in path is bypassed/enabled */ + CONTROL_FLAG_TRACKER = 3, + /* DigitalOut to Spdif2Out connection is disabled/enabled */ + CONTROL_FLAG_SPDIF2OUT = 4, + /* Digital Microphone is disabled/enabled */ + CONTROL_FLAG_DMIC = 5, + /* ADC_B rate is 48 kHz/96 kHz */ + CONTROL_FLAG_ADC_B_96KHZ = 6, + /* ADC_C rate is 48 kHz/96 kHz */ + CONTROL_FLAG_ADC_C_96KHZ = 7, + /* DAC rate is 48 kHz/96 kHz (affects all DACs) */ + CONTROL_FLAG_DAC_96KHZ = 8, + /* DSP rate is 48 kHz/96 kHz */ + CONTROL_FLAG_DSP_96KHZ = 9, + /* SRC clock is 98 MHz/196 MHz (196 MHz forces rate to 96 KHz) */ + CONTROL_FLAG_SRC_CLOCK_196MHZ = 10, + /* SRC rate is 48 kHz/96 kHz (48 kHz disabled when clock is 196 MHz) */ + CONTROL_FLAG_SRC_RATE_96KHZ = 11, + /* Decode Loop (DSP->SRC->DSP) is disabled/enabled */ + CONTROL_FLAG_DECODE_LOOP = 12, + /* De-emphasis filter on DAC-1 disabled/enabled */ + CONTROL_FLAG_DAC1_DEEMPHASIS = 13, + /* De-emphasis filter on DAC-2 disabled/enabled */ + CONTROL_FLAG_DAC2_DEEMPHASIS = 14, + /* De-emphasis filter on DAC-3 disabled/enabled */ + CONTROL_FLAG_DAC3_DEEMPHASIS = 15, + /* High-pass filter on ADC_B disabled/enabled */ + CONTROL_FLAG_ADC_B_HIGH_PASS = 16, + /* High-pass filter on ADC_C disabled/enabled */ + CONTROL_FLAG_ADC_C_HIGH_PASS = 17, + /* Common mode on Port_A disabled/enabled */ + CONTROL_FLAG_PORT_A_COMMON_MODE = 18, + /* Common mode on Port_D disabled/enabled */ + CONTROL_FLAG_PORT_D_COMMON_MODE = 19, + /* Impedance for ramp generator on Port_A 16 Ohm/10K Ohm */ + CONTROL_FLAG_PORT_A_10KOHM_LOAD = 20, + /* Impedance for ramp generator on Port_D, 16 Ohm/10K Ohm */ + CONTROL_FLAG_PORT_D_10K0HM_LOAD = 21, + /* ASI rate is 48kHz/96kHz */ + CONTROL_FLAG_ASI_96KHZ = 22, + /* DAC power settings able to control attached ports no/yes */ + CONTROL_FLAG_DACS_CONTROL_PORTS = 23, + /* Clock Stop OK reporting is disabled/enabled */ + CONTROL_FLAG_CONTROL_STOP_OK_ENABLE = 24, + /* Number of control flags */ + CONTROL_FLAGS_MAX = (CONTROL_FLAG_CONTROL_STOP_OK_ENABLE+1) +}; + +/* + * Control parameter IDs + */ +enum control_parameter_id { + /* 0: force HDA, 1: allow DSP if HDA Spdif1Out stream is idle */ + CONTROL_PARAM_SPDIF1_SOURCE = 2, + + /* Stream Control */ + + /* Select stream with the given ID */ + CONTROL_PARAM_STREAM_ID = 24, + /* Source connection point for the selected stream */ + CONTROL_PARAM_STREAM_SOURCE_CONN_POINT = 25, + /* Destination connection point for the selected stream */ + CONTROL_PARAM_STREAM_DEST_CONN_POINT = 26, + /* Number of audio channels in the selected stream */ + CONTROL_PARAM_STREAMS_CHANNELS = 27, + /*Enable control for the selected stream */ + CONTROL_PARAM_STREAM_CONTROL = 28, + + /* Connection Point Control */ + + /* Select connection point with the given ID */ + CONTROL_PARAM_CONN_POINT_ID = 29, + /* Connection point sample rate */ + CONTROL_PARAM_CONN_POINT_SAMPLE_RATE = 30, + + /* Node Control */ + + /* Select HDA node with the given ID */ + CONTROL_PARAM_NODE_ID = 31 +}; + +/* + * Dsp Io Status codes + */ +enum hda_vendor_status_dspio { + /* Success */ + VENDOR_STATUS_DSPIO_OK = 0x00, + /* Busy, unable to accept new command, the host must retry */ + VENDOR_STATUS_DSPIO_BUSY = 0x01, + /* SCP command queue is full */ + VENDOR_STATUS_DSPIO_SCP_COMMAND_QUEUE_FULL = 0x02, + /* SCP response queue is empty */ + VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY = 0x03 +}; + +/* + * Chip Io Status codes + */ +enum hda_vendor_status_chipio { + /* Success */ + VENDOR_STATUS_CHIPIO_OK = 0x00, + /* Busy, unable to accept new command, the host must retry */ + VENDOR_STATUS_CHIPIO_BUSY = 0x01 +}; + +/* + * CA0132 sample rate + */ +enum ca0132_sample_rate { + SR_6_000 = 0x00, + SR_8_000 = 0x01, + SR_9_600 = 0x02, + SR_11_025 = 0x03, + SR_16_000 = 0x04, + SR_22_050 = 0x05, + SR_24_000 = 0x06, + SR_32_000 = 0x07, + SR_44_100 = 0x08, + SR_48_000 = 0x09, + SR_88_200 = 0x0A, + SR_96_000 = 0x0B, + SR_144_000 = 0x0C, + SR_176_400 = 0x0D, + SR_192_000 = 0x0E, + SR_384_000 = 0x0F, + + SR_COUNT = 0x10, + + SR_RATE_UNKNOWN = 0x1F +}; + +/* + * Scp Helper function + */ +enum get_set { + IS_SET = 0, + IS_GET = 1, +}; + +/* + * Duplicated from ca0110 codec + */ + +static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) +{ + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + } + if (dac) + snd_hda_codec_write(codec, dac, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); +} + +static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) +{ + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_VREF80); + if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + } + if (adc) + snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); +} + +static char *dirstr[2] = { "Playback", "Capture" }; + +static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, + int chan, int dir) +{ + char namestr[44]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, + int chan, int dir) +{ + char namestr[44]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +#define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0) +#define add_out_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 0) +#define add_in_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 1) +#define add_in_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 1) +#define add_mono_switch(codec, nid, pfx, chan) \ + _add_switch(codec, nid, pfx, chan, 0) +#define add_mono_volume(codec, nid, pfx, chan) \ + _add_volume(codec, nid, pfx, chan, 0) +#define add_in_mono_switch(codec, nid, pfx, chan) \ + _add_switch(codec, nid, pfx, chan, 1) +#define add_in_mono_volume(codec, nid, pfx, chan) \ + _add_volume(codec, nid, pfx, chan, 1) + + +/* + * CA0132 specific + */ + +struct ca0132_spec { + struct auto_pin_cfg autocfg; + struct hda_multi_out multiout; + hda_nid_t out_pins[AUTO_CFG_MAX_OUTS]; + hda_nid_t dacs[AUTO_CFG_MAX_OUTS]; + hda_nid_t hp_dac; + hda_nid_t input_pins[AUTO_PIN_LAST]; + hda_nid_t adcs[AUTO_PIN_LAST]; + hda_nid_t dig_out; + hda_nid_t dig_in; + unsigned int num_inputs; + long curr_hp_switch; + long curr_hp_volume[2]; + long curr_speaker_switch; + struct mutex chipio_mutex; + const char *input_labels[AUTO_PIN_LAST]; + struct hda_pcm pcm_rec[2]; /* PCM information */ +}; + +/* Chip access helper function */ +static int chipio_send(struct hda_codec *codec, + unsigned int reg, + unsigned int data) +{ + unsigned int res; + int retry = 50; + + /* send bits of data specified by reg */ + do { + res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0, + reg, data); + if (res == VENDOR_STATUS_CHIPIO_OK) + return 0; + } while (--retry); + return -EIO; +} + +/* + * Write chip address through the vendor widget -- NOT protected by the Mutex! + */ +static int chipio_write_address(struct hda_codec *codec, + unsigned int chip_addx) +{ + int res; + + /* send low 16 bits of the address */ + res = chipio_send(codec, VENDOR_CHIPIO_ADDRESS_LOW, + chip_addx & 0xffff); + + if (res != -EIO) { + /* send high 16 bits of the address */ + res = chipio_send(codec, VENDOR_CHIPIO_ADDRESS_HIGH, + chip_addx >> 16); + } + + return res; +} + +/* + * Write data through the vendor widget -- NOT protected by the Mutex! + */ + +static int chipio_write_data(struct hda_codec *codec, unsigned int data) +{ + int res; + + /* send low 16 bits of the data */ + res = chipio_send(codec, VENDOR_CHIPIO_DATA_LOW, data & 0xffff); + + if (res != -EIO) { + /* send high 16 bits of the data */ + res = chipio_send(codec, VENDOR_CHIPIO_DATA_HIGH, + data >> 16); + } + + return res; +} + +/* + * Read data through the vendor widget -- NOT protected by the Mutex! + */ +static int chipio_read_data(struct hda_codec *codec, unsigned int *data) +{ + int res; + + /* post read */ + res = chipio_send(codec, VENDOR_CHIPIO_HIC_POST_READ, 0); + + if (res != -EIO) { + /* read status */ + res = chipio_send(codec, VENDOR_CHIPIO_STATUS, 0); + } + + if (res != -EIO) { + /* read data */ + *data = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_HIC_READ_DATA, + 0); + } + + return res; +} + +/* + * Write given value to the given address through the chip I/O widget. + * protected by the Mutex + */ +static int chipio_write(struct hda_codec *codec, + unsigned int chip_addx, const unsigned int data) +{ + struct ca0132_spec *spec = codec->spec; + int err; + + mutex_lock(&spec->chipio_mutex); + + /* write the address, and if successful proceed to write data */ + err = chipio_write_address(codec, chip_addx); + if (err < 0) + goto exit; + + err = chipio_write_data(codec, data); + if (err < 0) + goto exit; + +exit: + mutex_unlock(&spec->chipio_mutex); + return err; +} + +/* + * Read the given address through the chip I/O widget + * protected by the Mutex + */ +static int chipio_read(struct hda_codec *codec, + unsigned int chip_addx, unsigned int *data) +{ + struct ca0132_spec *spec = codec->spec; + int err; + + mutex_lock(&spec->chipio_mutex); + + /* write the address, and if successful proceed to write data */ + err = chipio_write_address(codec, chip_addx); + if (err < 0) + goto exit; + + err = chipio_read_data(codec, data); + if (err < 0) + goto exit; + +exit: + mutex_unlock(&spec->chipio_mutex); + return err; +} + +/* + * PCM stuffs + */ +static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, + u32 stream_tag, + int channel_id, int format) +{ + unsigned int oldval, newval; + + if (!nid) + return; + + snd_printdd("ca0132_setup_stream: " + "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n", + nid, stream_tag, channel_id, format); + + /* update the format-id if changed */ + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_STREAM_FORMAT, + 0); + if (oldval != format) { + msleep(20); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, + format); + } + + oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + newval = (stream_tag << 4) | channel_id; + if (oldval != newval) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + newval); + } +} + +static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) +{ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); +} + +/* + * PCM callbacks + */ +static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); + + return 0; +} + +static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_cleanup_stream(codec, spec->dacs[0]); + + return 0; +} + +/* + * Digital out + */ +static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format); + + return 0; +} + +static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_cleanup_stream(codec, spec->dig_out); + + return 0; +} + +/* + * Analog capture + */ +static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_setup_stream(codec, spec->adcs[substream->number], + stream_tag, 0, format); + + return 0; +} + +static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_cleanup_stream(codec, spec->adcs[substream->number]); + + return 0; +} + +/* + * Digital capture + */ +static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format); + + return 0; +} + +static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_cleanup_stream(codec, spec->dig_in); + + return 0; +} + +/* + */ +static struct hda_pcm_stream ca0132_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0132_playback_pcm_prepare, + .cleanup = ca0132_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0132_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0132_capture_pcm_prepare, + .cleanup = ca0132_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0132_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0132_dig_playback_pcm_prepare, + .cleanup = ca0132_dig_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0132_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0132_dig_capture_pcm_prepare, + .cleanup = ca0132_dig_capture_pcm_cleanup + }, +}; + +static int ca0132_build_pcms(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + + codec->pcm_info = info; + codec->num_pcms = 0; + + info->name = "CA0132 Analog"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = + spec->multiout.max_channels; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; + codec->num_pcms++; + + if (!spec->dig_out && !spec->dig_in) + return 0; + + info++; + info->name = "CA0132 Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->dig_out) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + ca0132_pcm_digital_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out; + } + if (spec->dig_in) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + ca0132_pcm_digital_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in; + } + codec->num_pcms++; + + return 0; +} + +#define REG_CODEC_MUTE 0x18b014 +#define REG_CODEC_HP_VOL_L 0x18b070 +#define REG_CODEC_HP_VOL_R 0x18b074 + +static int ca0132_hp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + + *valp = spec->curr_hp_switch; + return 0; +} + +static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + unsigned int data; + int err; + + /* any change? */ + if (spec->curr_hp_switch == *valp) + return 0; + + snd_hda_power_up(codec); + + err = chipio_read(codec, REG_CODEC_MUTE, &data); + if (err < 0) + return err; + + /* *valp 0 is mute, 1 is unmute */ + data = (data & 0x7f) | (*valp ? 0 : 0x80); + chipio_write(codec, REG_CODEC_MUTE, data); + if (err < 0) + return err; + + spec->curr_hp_switch = *valp; + + snd_hda_power_down(codec); + return 1; +} + +static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + + *valp = spec->curr_speaker_switch; + return 0; +} + +static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + unsigned int data; + int err; + + /* any change? */ + if (spec->curr_speaker_switch == *valp) + return 0; + + snd_hda_power_up(codec); + + err = chipio_read(codec, REG_CODEC_MUTE, &data); + if (err < 0) + return err; + + /* *valp 0 is mute, 1 is unmute */ + data = (data & 0xef) | (*valp ? 0 : 0x10); + chipio_write(codec, REG_CODEC_MUTE, data); + if (err < 0) + return err; + + spec->curr_speaker_switch = *valp; + + snd_hda_power_down(codec); + return 1; +} + +static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + + *valp++ = spec->curr_hp_volume[0]; + *valp = spec->curr_hp_volume[1]; + return 0; +} + +static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + long left_vol, right_vol; + unsigned int data; + int val; + int err; + + left_vol = *valp++; + right_vol = *valp; + + /* any change? */ + if ((spec->curr_hp_volume[0] == left_vol) && + (spec->curr_hp_volume[1] == right_vol)) + return 0; + + snd_hda_power_up(codec); + + err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data); + if (err < 0) + return err; + + val = 31 - left_vol; + data = (data & 0xe0) | val; + chipio_write(codec, REG_CODEC_HP_VOL_L, data); + if (err < 0) + return err; + + val = 31 - right_vol; + data = (data & 0xe0) | val; + chipio_write(codec, REG_CODEC_HP_VOL_R, data); + if (err < 0) + return err; + + spec->curr_hp_volume[0] = left_vol; + spec->curr_hp_volume[1] = right_vol; + + snd_hda_power_down(codec); + return 1; +} + +static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Headphone Playback Switch", + nid, 1, 0, HDA_OUTPUT); + knew.get = ca0132_hp_switch_get; + knew.put = ca0132_hp_switch_put; + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +static int add_hp_volume(struct hda_codec *codec, hda_nid_t nid) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO("Headphone Playback Volume", + nid, 3, 0, HDA_OUTPUT); + knew.get = ca0132_hp_volume_get; + knew.put = ca0132_hp_volume_put; + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +static int add_speaker_switch(struct hda_codec *codec, hda_nid_t nid) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Speaker Playback Switch", + nid, 1, 0, HDA_OUTPUT); + knew.get = ca0132_speaker_switch_get; + knew.put = ca0132_speaker_switch_put; + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +static void ca0132_fix_hp_caps(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int caps; + + /* set mute-capable, 1db step, 32 steps, ofs 6 */ + caps = 0x80031f06; + snd_hda_override_amp_caps(codec, cfg->hp_pins[0], HDA_OUTPUT, caps); +} + +static int ca0132_build_controls(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, err; + + if (spec->multiout.num_dacs) { + err = add_speaker_switch(codec, spec->out_pins[0]); + if (err < 0) + return err; + } + + if (cfg->hp_outs) { + ca0132_fix_hp_caps(codec); + err = add_hp_switch(codec, cfg->hp_pins[0]); + if (err < 0) + return err; + err = add_hp_volume(codec, cfg->hp_pins[0]); + if (err < 0) + return err; + } + + for (i = 0; i < spec->num_inputs; i++) { + const char *label = spec->input_labels[i]; + + err = add_in_switch(codec, spec->adcs[i], label); + if (err < 0) + return err; + err = add_in_volume(codec, spec->adcs[i], label); + if (err < 0) + return err; + if (cfg->inputs[i].type == AUTO_PIN_MIC) { + /* add Mic-Boost */ + err = add_in_mono_volume(codec, spec->input_pins[i], + "Mic Boost", 1); + if (err < 0) + return err; + } + } + + if (spec->dig_out) { + err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out); + if (err < 0) + return err; + err = add_out_volume(codec, spec->dig_out, "IEC958"); + if (err < 0) + return err; + } + + if (spec->dig_in) { + err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); + if (err < 0) + return err; + err = add_in_volume(codec, spec->dig_in, "IEC958"); + } + return 0; +} + + +static void ca0132_set_ct_ext(struct hda_codec *codec, int enable) +{ + /* Set Creative extension */ + snd_printdd("SET CREATIVE EXTENSION\n"); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, + enable); + msleep(20); +} + + +static void ca0132_config(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + + /* line-outs */ + cfg->line_outs = 1; + cfg->line_out_pins[0] = 0x0b; /* front */ + cfg->line_out_type = AUTO_PIN_LINE_OUT; + + spec->dacs[0] = 0x02; + spec->out_pins[0] = 0x0b; + spec->multiout.dac_nids = spec->dacs; + spec->multiout.num_dacs = 1; + spec->multiout.max_channels = 2; + + /* headphone */ + cfg->hp_outs = 1; + cfg->hp_pins[0] = 0x0f; + + spec->hp_dac = 0; + spec->multiout.hp_nid = 0; + + /* inputs */ + cfg->num_inputs = 2; /* Mic-in and line-in */ + cfg->inputs[0].pin = 0x12; + cfg->inputs[0].type = AUTO_PIN_MIC; + cfg->inputs[1].pin = 0x11; + cfg->inputs[1].type = AUTO_PIN_LINE_IN; + + /* Mic-in */ + spec->input_pins[0] = 0x12; + spec->input_labels[0] = "Mic-In"; + spec->adcs[0] = 0x07; + + /* Line-In */ + spec->input_pins[1] = 0x11; + spec->input_labels[1] = "Line-In"; + spec->adcs[1] = 0x08; + spec->num_inputs = 2; +} + +static void ca0132_init_chip(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_init(&spec->chipio_mutex); +} + +static void ca0132_exit_chip(struct hda_codec *codec) +{ + /* put any chip cleanup stuffs here. */ +} + +static int ca0132_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < spec->multiout.num_dacs; i++) { + init_output(codec, spec->out_pins[i], + spec->multiout.dac_nids[i]); + } + init_output(codec, cfg->hp_pins[0], spec->hp_dac); + init_output(codec, cfg->dig_out_pins[0], spec->dig_out); + + for (i = 0; i < spec->num_inputs; i++) + init_input(codec, spec->input_pins[i], spec->adcs[i]); + + init_input(codec, cfg->dig_in_pin, spec->dig_in); + + ca0132_set_ct_ext(codec, 1); + + return 0; +} + + +static void ca0132_free(struct hda_codec *codec) +{ + ca0132_set_ct_ext(codec, 0); + ca0132_exit_chip(codec); + kfree(codec->spec); +} + +static struct hda_codec_ops ca0132_patch_ops = { + .build_controls = ca0132_build_controls, + .build_pcms = ca0132_build_pcms, + .init = ca0132_init, + .free = ca0132_free, +}; + + + +static int patch_ca0132(struct hda_codec *codec) +{ + struct ca0132_spec *spec; + + snd_printdd("patch_ca0132\n"); + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + ca0132_init_chip(codec); + + ca0132_config(codec); + + codec->patch_ops = ca0132_patch_ops; + + return 0; +} + +/* + * patch entries + */ +static struct hda_codec_preset snd_hda_preset_ca0132[] = { + { .id = 0x11020011, .name = "CA0132", .patch = patch_ca0132 }, + {} /* terminator */ +}; + +MODULE_ALIAS("snd-hda-codec-id:11020011"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Creative CA0132, CA0132 HD-audio codec"); + +static struct hda_codec_preset_list ca0132_list = { + .preset = snd_hda_preset_ca0132, + .owner = THIS_MODULE, +}; + +static int __init patch_ca0132_init(void) +{ + return snd_hda_add_codec_preset(&ca0132_list); +} + +static void __exit patch_ca0132_exit(void) +{ + snd_hda_delete_codec_preset(&ca0132_list); +} + +module_init(patch_ca0132_init) +module_exit(patch_ca0132_exit) -- 1.7.0.4

2 3

Re: [alsa-devel] 答复: [PATCH 1/3] ALSA: VIA HDA: Modify initial verbs list for VT1718S.
by Takashi Iwai 21 Jun '11

21 Jun '11

At Tue, 21 Jun 2011 09:36:14 +0800, <LydiaWang(a)viatech.com.cn> wrote: > > > -----邮件原件----- > > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > > 发送时间: 2011年6月20日 22:32 > > 收件人: Lydia Wang > > 抄送: alsa-devel(a)alsa-project.org; Harald Welte > > 主题: Re: [PATCH 1/3] ALSA: VIA HDA: Modify initial verbs list for VT1718S. > > > > At Mon, 20 Jun 2011 14:14:37 +0800, > > lydiawang wrote: > > > > > > From: Lydia Wang <lydiawang(a)viatech.com.cn> > > > Subject: ALSA: VIA HDA: Modify initial verbs list for VT1718S. > > > > > > Remove some invalid initial verbs and correct some wrong initial verbs > > > for VT1718S codec. > > > > > > Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> > > > > Are these three patches are needed for fixing the bugs or just clean-ups? > > If these are real fixes for 3.0-rc kernel, I'll merge them now. > > > > If not, they can be irrelevant now, as I've been working on major > > rework of VIA codec driver. See the later post. > > > > > > thanks, > > > > Takashi > > These patches are to fix the problem reported by alex.baldacchino.alsasub. > You can refer the attached mail. Fair enough. I'll take them for 3.0-rc, then. thanks, Takashi

1 0

[alsa-devel] [PATCH] [resend] tea575x: remove useless input ioctls
by Ondrej Zary 20 Jun '11

20 Jun '11

Remove empty and useless g_input and s_input ioctls. This fixes one fail of v4l2-compliance test. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-06-11 15:29:18.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-06-11 15:29:51.000000000 +0200 @@ -269,19 +269,6 @@ static int tea575x_s_ctrl(struct v4l2_ct return -EINVAL; } -static int vidioc_g_input(struct file *filp, void *priv, unsigned int *i) -{ - *i = 0; - return 0; -} - -static int vidioc_s_input(struct file *filp, void *priv, unsigned int i) -{ - if (i != 0) - return -EINVAL; - return 0; -} - static const struct v4l2_file_operations tea575x_fops = { .owner = THIS_MODULE, .unlocked_ioctl = video_ioctl2, @@ -293,8 +280,6 @@ static const struct v4l2_ioctl_ops tea57 .vidioc_s_tuner = vidioc_s_tuner, .vidioc_g_audio = vidioc_g_audio, .vidioc_s_audio = vidioc_s_audio, - .vidioc_g_input = vidioc_g_input, - .vidioc_s_input = vidioc_s_input, .vidioc_g_frequency = vidioc_g_frequency, .vidioc_s_frequency = vidioc_s_frequency, }; -- Ondrej Zary

1 0

[alsa-devel] [PATCH] [resend] tea575x: allow multiple opens
by Ondrej Zary 20 Jun '11

20 Jun '11

Change locking to allow tea575x-radio device to be opened multiple times. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/include/sound/tea575x-tuner.h 2011-06-11 15:21:50.000000000 +0200 +++ linux-2.6.39-rc2/include/sound/tea575x-tuner.h 2011-06-11 14:50:55.000000000 +0200 @@ -49,7 +49,7 @@ struct snd_tea575x { bool tuned; /* tuned to a station */ unsigned int val; /* hw value */ unsigned long freq; /* frequency */ - unsigned long in_use; /* set if the device is in use */ + struct mutex mutex; struct snd_tea575x_ops *ops; void *private_data; u8 card[32]; --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-06-11 15:21:50.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-06-11 14:57:28.000000000 +0200 @@ -282,26 +282,9 @@ static int vidioc_s_input(struct file *f return 0; } -static int snd_tea575x_exclusive_open(struct file *file) -{ - struct snd_tea575x *tea = video_drvdata(file); - - return test_and_set_bit(0, &tea->in_use) ? -EBUSY : 0; -} - -static int snd_tea575x_exclusive_release(struct file *file) -{ - struct snd_tea575x *tea = video_drvdata(file); - - clear_bit(0, &tea->in_use); - return 0; -} - static const struct v4l2_file_operations tea575x_fops = { .owner = THIS_MODULE, - .open = snd_tea575x_exclusive_open, - .release = snd_tea575x_exclusive_release, - .ioctl = video_ioctl2, + .unlocked_ioctl = video_ioctl2, }; static const struct v4l2_ioctl_ops tea575x_ioctl_ops = { @@ -340,13 +323,14 @@ int snd_tea575x_init(struct snd_tea575x if (snd_tea575x_read(tea) != 0x55AA) return -ENODEV; - tea->in_use = 0; tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; tea->freq = 90500 * 16; /* 90.5Mhz default */ snd_tea575x_set_freq(tea); tea->vd = tea575x_radio; video_set_drvdata(&tea->vd, tea); + mutex_init(&tea->mutex); + tea->vd.lock = &tea->mutex; v4l2_ctrl_handler_init(&tea->ctrl_handler, 1); tea->vd.ctrl_handler = &tea->ctrl_handler; -- Ondrej Zary

1 0

[alsa-devel] [patch] ASoC: adau1701: signedness bug in adau1701_write()
by Dan Carpenter 20 Jun '11

20 Jun '11

"ret" is supposed to be signed here. The current code will only return -EIO on error, instead of a more appropriate error code such as -EAGAIN etc. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 6c01bb6..2758d5f 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -140,9 +140,10 @@ static unsigned int adau1701_register_size(struct snd_soc_codec *codec, static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - unsigned int i, ret; + unsigned int i; unsigned int size; uint8_t buf[4]; + int ret; size = adau1701_register_size(codec, reg); if (size == 0)

4 3

[alsa-devel] [PATCH] ALSA: HDA: Remove quirk for an HP device
by David Henningsson 20 Jun '11

20 Jun '11

The reporter, who is running kernel 2.6.38, reports that he needs to set model=auto for the headphone output to work correctly. Relevant codec info is in http://bugs.launchpad.net/bugs/761022 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH 1/3] ALSA: VIA HDA: Modify initial verbs list for VT1718S.
by lydiawang 20 Jun '11

20 Jun '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Modify initial verbs list for VT1718S. Remove some invalid initial verbs and correct some wrong initial verbs for VT1718S codec. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2011-06-20 13:25:07.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2011-06-20 13:31:19.000000000 +0800 @@ -4286,9 +4286,6 @@ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, - - /* Setup default input of Front HP to MW9 */ - {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, /* PW9 PW10 Output enable */ {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, @@ -4297,10 +4294,10 @@ /* Enable Boost Volume backdoor */ {0x1, 0xf88, 0x8}, /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -4310,8 +4307,6 @@ /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */ {0x34, AC_VERB_SET_CONNECT_SEL, 0x2}, {0x35, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Unmute MW4's index 0 */ - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { } };

2 1

[alsa-devel] Via VT2020: issues with kernel 2.6.38.{2, 3} (alsa 1.0.23) - working with 2.6.33.2 (alsa 1.0.21)
by alex dot baldacchino dot alsasub at gmail dot com 20 Jun '11

20 Jun '11

Hello, First of all let me apologize if I'm not fulfilling any requirements for this list, and for any errors in my English. I've sent a mail with same subject and content to this list as a non-subscriber on April, 26, but perhaps that's gone lost, or perhaps is still under moderation or didn't pass it, so I'm resending this from a different address as a subscriber. So, please, ignore my previous mail if it came out from moderation. As per subject, I'm experiencing problems with my motherboard built-in audio device (an Asrock with ATI SB8xx), embedding a Via VT2020 codec (I haven't tested the ATI HDMI part in same conditions). The card is recognized 'about' correctly (later in this mail something about dmesg and other issues), and stuff in /proc/asound seems correct, also I have no apparent problem with /dev special files, but I get no sound after having installed kernel 2.6.38.2 and updated to version 2.6.38.3 (both compiled from source); no mixer seems to work (later specific errors), but everything still works with older kernel 2.6.33.2 I've kept in my system (built with the same build environment). Since I'm using the very same userspace environment in both cases, I deem it to be a driver(s) related issue more likely; though, I've updated my alsa-lib and -utils to latest versions (were such when I downloaded sources - alsa-lib 1.0.24.1 & alsa-utils 1.0.24.2) and still get the very same results. But let me go into more details. Here is an excerpt from dmesg called with kernel 2.6.33.2 (sound works, but I'm just using headphone jack, so I can't tell about anything related with later issue with smart5.1 and hda I've seen in list archives): [...] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 HDA Intel 0000:01:05.1: PCI INT B -> GSI 19 (level, low) -> IRQ 19 HDA Intel 0000:01:05.1: irq 28 for MSI/MSI-X HDA Intel 0000:01:05.1: setting latency timer to 64 [...] ohci1394 0000:04:00.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16 ohci1394 0000:04:00.0: setting latency timer to 64 [...] ohci1394: fw-host0: OHCI-1394 1.1 (PCI): IRQ=[16] MMIO=[fe9ff800-fe9fffff] Max Packet=[2048] IR/IT contexts=[4/8] ieee1394: Host added: ID:BUS[0-00:1023] GUID[008f130043910900] [...] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x000f0001 [...] Notice last line is an harmless warning (sound works) and is there because msi seems to be enabled by default; by adding "options snd-hda-intel enable_msi=0" to /etc/modprobe.d/alsa-base.conf both the warning and any reference to MSI/MSI-X disappear; same effect with pci=nomsi on command line (this time affecting other drivers, such as r8169). VT2020 is part of device 0000:00:14.2 sharing IRQ 16 with ohci1394; device 0000:01:05.1 is ATI HDMI, while device 0000:05.0 should be embedded radeon vga (bridged together according with lspci -t with 0000:00:01.0 on top); for each bridge I can find with lspci -t corresponding msi_bus file in sys filesystem (in /sys/bus/pci/devices/*) is always 1, regardless of any boot command (this with kernel 2.6.33.2). Now an excerpt from dmesg in kernel 2.6.38.3: [...] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 hda-codec: no NID for mapping control Independent HP:0:0 HDA Intel 0000:01:05.1: PCI INT B -> GSI 19 (level, low) -> IRQ 19 HDA Intel 0000:01:05.1: setting latency timer to 64 [...] r8169 0000:03:00.0: irq 43 for MSI/MSI-X [...] firewire_ohci 0000:04:00.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16 firewire_ohci 0000:04:00.0: setting latency timer to 64 [...] firewire_ohci: Added fw-ohci device 0000:04:00.0, OHCI v1.10, 4 IR + 8 IT contexts, quirks 0x11 firewire_core: created device fw0: GUID 008f130043910900, S400 [...] Now firewire_ohci replaces ohci1394, again sharing IRQ 16 with hda-intel for vt2020. Line about r8169 is just for reference, since it gets a different irq for msi now. Similar behaviour for MSI: enabled/disabled with pci=nomsi (seems to be enabled by default without pci=msi) and msi_bus == 1 for bridges regardless of boot options, *but for bridge on top of ati hdmi device*: now msi seems to be disabled regardless of both 'enable_msi' option and pci=* boot command and corresponding bridge (device at bus 0000:00:01.0) holds a value of 0 for msi_bus... Notice there are no more warnings about response timeouts, as it happened in older kernel after disabling msi (both globally and for alsa through alsa config options). Notice also line "hda-codec: no NID for mapping control Independent HP:0:0". That's because function 'snd_hda_get_connections' (patch_via.c) called by 'via_hp_build' returns 3 ( test for 'nums <= 1' fails and the function doesn't return), but function 'side_mute_channel', called to init knew->subdevice when 'registering' via_hp_mixer[1], returns 0 (default value) and so condition 'if(nid > 0)', in function 'snd_hda_add_nid' (hda_codec.c), is not met and the error is triggered (by the way, param nid is of type hda_nid_t which in turn is an unsigned integer (u16), so, should that be 'if(nid != 0)' instead? just to avoid the risk of giving the impression that negative values are possible but mistaken (hda_nid_t is defined elsewhere). Also: param @nid, in function comment/documentation is defined as optional, but it doesn't seem to be). Incidentally, in /proc/asound/card0/codec#0 node 0x35 looks much like node 0x34 (with both older and newer kernel; posting excerpt for kernel 2.6.38.3): Node 0x34 [Audio Selector] wcaps 0x300501: Stereo Control: name="Independent HP", index=0, device=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 3 0x08 0x0b 0x0c* Node 0x35 [Audio Selector] wcaps 0x300501: Stereo Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 3 0x08 0x0b* 0x0c (notice connections and wcaps similarity, if relevant at all) Since I'm quite crazy, I've tried to modify 'side_mute_channel' to have it returning '0x35' for codec type 'VT1718S', but to no avail; I also tried to bypass 'snd_hda_add_nid', both avoiding 'via_hp_build' to clone via_hp_mixer[<something>] and/or modifying 'via_build_controls' and 'snd_hda_add_new_ctls' (hda_control.c) so that "Independent HP" ctls were handled by the latter (achieving this was just a matter of a pair of strcmp's to choose skipping the rest of an iteration or not), as I deem it was done in alsa 1.0.21 (embedded in kernel 2.6.33.2), but it didn't work (the only result I got this ways was to avoid that error message about nids in system logs). Additional infos from userspace applications: mplayer works fine with kernel 2.6.33.2 (again, only using one exit, I haven't tested all jacks combinations); called from cmdline (under X - but that's the same shutting X down) with default options ('mplayer <path/to/file>') it gives the following output (relevant excerpt, as I deem it, after codec choice, can provide more if needed): socket(): Address family not supported by protocol AO: [pulse] Init failed: Connection refused Failed to initialize audio driver 'pulse' AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) Starting playback... (notice I'm not using alsa-tools, such as pulse-audio; with the above invocation sound/video works) running mplayer with option '-ao sdl' (using sdl audio, same file) I get the following output: [AO SDL] Samplerate: 44100Hz Channels: Stereo Format s16le AO: [sdl] 44100Hz 2ch s16le (2 bytes per sample) Starting playback... Again, sound works. In both cases, I can notice a change in /proc/asound/card0/codec#0 as follows: In nodes 0x08, 0x09, 0x0a, 0x0e (all [Audio Output] nodes, first 3 with 'wcaps 0x41d: Stereo Amp-Out', last with 'wcaps 0x611: Stereo Digital'), the following line: Converter: stream=0, channel=0 changes into: Converter: stream=5, channel=0 With kernel 2.6.38.x I get the following: running mplayer with no option, I get no sound and following output: socket(): Address family not supported by protocol AO: [pulse] Init failed: Connection refused Failed to initialize audio driver 'pulse' [AO_ALSA] alsa-lib: confmisc.c:768:(parse_card) cannot find card '0' [AO_ALSA] alsa-lib: conf.c:4184:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device [AO_ALSA] alsa-lib: confmisc.c:392:(snd_func_concat) error evaluating strings [AO_ALSA] alsa-lib: conf.c:4184:(_snd_config_evaluate) function snd_func_concat returned error: No such device [AO_ALSA] alsa-lib: confmisc.c:1251:(snd_func_refer) error evaluating name [AO_ALSA] alsa-lib: conf.c:4184:(_snd_config_evaluate) function snd_func_refer returned error: No such device [AO_ALSA] alsa-lib: conf.c:4663:(snd_config_expand) Evaluate error: No such device [AO_ALSA] alsa-lib: pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM default [AO_ALSA] Playback open error: No such device Failed to initialize audio driver 'alsa' [AO SDL] Samplerate: 44100Hz Channels: Stereo Format s16le [AO SDL] using aalib audio driver. [AO SDL] Unable to open audio: No available audio device Failed to initialize audio driver 'sdl:aalib' Could not open/initialize audio device -> no sound. Audio: no sound and nothing changes in codec#0 file; with option -ao sdl, instead: [AO SDL] Samplerate: 44100Hz Channels: Stereo Format s16le ALSA lib confmisc.c:768:(parse_card) cannot find card '0' ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:4663:(snd_config_expand) Evaluate error: No such device ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM default AO: [sdl] 44100Hz 2ch s16le (2 bytes per sample) And sound works! Same changes as per (working) playback with kernel 33.2 (alsa 1.0.21) are noticeable in /proc/asound/card0/codec#0 in this case (that is, when getting sound with kernel 2.6.38.x); in addition, node 0x0c, now changes as well (again, '[Audio Output] wcaps 0x41d: Stereo Amp-Out.' and 'Converter: stream=5, channel=0') Moreover: with kernel 38.2, 38.3 alsamixer fails telling about problems finding device: cannot open mixer: No such device And amixer says: amixer: Mixer attach default error: No such device But there's no problem with /dev stuff (and same devices works pretty fine with kernel 2.6.33.2) - aplay -l doesn't work as well: aplay: device_list:240: no soundcards found... With kernel 2.6.33.2, aplay -l says: **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT2020 Analog [VT2020 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 1: VT2020 Digital [VT2020 Digital] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 A 'third party' mixer embedded in my distribution aborts by hitting a failed assertion in function 'snd_hctl_load' (hcontrol.c) failing assertion being 'assert(hctl->count == 0)' - I've added a few simple lines above the assertion to print out some info on stderr, and got the following results: In kernel 2.6.33.2: hctl->count: 0 hctl->ctl: 134748144 (0x80817F0) hctl->ctl->name: hw:0 which seems reasonable - according to /proc/iomem address 0x80817F0 is in range 00100000-cf83ffff (system ram) and higher than kernel stuff subranges ( 01000000-012c3be7: Kernel code - 012c3be8-0141dadf : Kernel data - 01481000-014b1e03 : Kernel bss); In kernel 2.6.38.2 & .38.3: hctl->count: 1156 hctl->ctl: 5398 (0x1516) Segmentation fault suggesting memory corruption (that's always the same address) - according to /proc/iomem address 0x1516 is in first range (0x0000-0xffff) which is reserved. I thought (and hoped) that could have been somehow related to cached/noncached pages issue solved by commit c27b92295ab4c6b90b1cee94c4c9c1b4732e1c2e, but applying diff patch for kernel 2.6.38.3 didn't change anything. After that, I've read changelog for kermel 2.6.38.4 but such didn't seem to me to present related material, so I didn't updgrade. I hope this post can help solving this issue and improving alsa infrastracture (I cannot easily go further). Best regards.

2 19

[alsa-devel] Issues with HDSPM
by Andre Schramm 20 Jun '11

20 Jun '11

Hello, the HDSPM driver seems to be broken in the current ALSA release. We're using CentOS 5 with 2.6.33.9-rt31.64.el5rt and ALSA 1.0.24 compiled from source, the card is a PCIe Hammerfall DSP MADI (rev d2). As already mentioned in [1], the card operates in master mode, but the system clock mode can't be set to anything higher than 48kHz. # amixer cget numid=3 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=0 # amixer cset numid=2 5 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=5 # amixer cget numid=6 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=48000 When switching to a PCI version of the card, the kernel completely crashes on boot. And with [2], there is another issue which unfortunately hasn't been merged yet and still exists in the current release. Regards, Andre [1] http://sourceforge.net/mailarchive/message.php?msg_id=27520432 [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006723.html

3 9

[alsa-devel] SPDIF In on ALC889 to speakers?
by Ady Deac 20 Jun '11

20 Jun '11

Hi guys, I have an GA-890FXA-UD5 mobo with an intel-hda sound card. I've purchased an additional SPDIF In adapter (bracket) to be able to play digital content on my 5.1 speakers. I have the speakers connected to the jacks (ie. not to the SPDIF Out). I've checked the Windows drivers, but these also only support bypassing SPDIF In to SPDIF Out. Is there any way to have this done by the card itself? Maybe some developers tweaks? Some chipset routing maybe? :D I know that the CM106 chipset needs some special initialization (eg: wr 00 20 05 30 01) to actually bypass the SPDIF In to the speakers, but it works! Anything that can be done in this case? Else, is there an easy way of bypassing the SPDIF Input to the speakers but also to allow the use of the card for playback? Thank you! -- Deac Mihai-Adrian W: www.mikesoftware.com P: +40-745-256.364, +40-723-85.44.44 F: +40-264-566.033 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40) # cat /proc/asound/card0/codec* Codec: Realtek ALC889 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0889 Subsystem Id: 0x1458a102 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=2 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=4 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=6 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x13 0x13] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x570]: 32000 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x1f 0x1f] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003c: IN OUT HP Detect Pin Default 0x01014410: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003c: IN OUT HP Detect Pin Default 0x01011412: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x01016411: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x01012414: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19c50: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19c60: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x6, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181345f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN Detect Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4005e601: [N/A] Line Out at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x014b6130: [Jack] SPDIF Out at Ext Rear Conn = Comb, Color = Orange DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000020: IN Pin Default 0x01cb7170: [Jack] SPDIF In at Ext Rear Conn = Comb, Color = Yellow DefAssociation = 0x7, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=28 Processing Coefficient: 0x00 Coefficient Index: 0x0f Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 12 0x18* 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x25 0x0b

1 0

[PATCH] ALSA: HDA: Remove quirk for an HP device
by David Henningsson 20 Jun '11

20 Jun '11

The reporter, who is running kernel 2.6.38, reports that he needs to set model=auto for the headphone output to work correctly. BugLink: http://bugs.launchpad.net/bugs/761022 Cc: stable(a)kernel.org (v2.6.38+) Reported-by: Jo Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b97af9..c174865 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4884,7 +4884,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST), SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V), SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), -- 1.7.4.1 --------------080401020001090605070601--

1 0

[alsa-devel] [PATCH 3/3] ALSA: VIA HDA: Create a master amplifier control for VT1718S.
by lydiawang 20 Jun '11

20 Jun '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Create a master amplifier control for VT1718S. Create a master volume and mute control of playback for VT1718S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2011-06-20 13:49:00.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2011-06-20 14:01:15.000000000 +0800 @@ -4465,6 +4465,19 @@ if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT) { + /* add control to mixer index 0 */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x21, 3, 5, + HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x21, 3, 5, + HDA_INPUT)); + if (err < 0) + return err; /* Front */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(

1 0

[alsa-devel] [PATCH 2/3] ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S.
by lydiawang 20 Jun '11

20 Jun '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S. When switch HP independent mode, mute/unmute connctions of mixer which is connected to headphone for VT1718S. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2011-06-20 13:41:04.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2011-06-20 13:48:07.000000000 +0800 @@ -745,12 +745,23 @@ struct via_spec *spec = codec->spec; hda_nid_t nid = kcontrol->private_value; unsigned int pinsel = ucontrol->value.enumerated.item[0]; + unsigned int parm0, parm1; /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - if (spec->codec_type == VT1718S) + if (spec->codec_type == VT1718S) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + /* Set correct mute switch for MW3 */ + parm0 = spec->hp_independent_mode ? + AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0); + parm1 = spec->hp_independent_mode ? + AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm0); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm1); + } else snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel);

1 0

[alsa-devel] [PATCH] ASoC: Fix DAPM sequence run for per-widget I/O methods
by Mark Brown 20 Jun '11

20 Jun '11

Previously we were using the DAPM context rather than a widget as the argument for update_bits() so we didn't need to care that our list walk of widgets left us one beyond the end of the list. Now we're using them for the register update we need to make sure we're pointing at an actual widget not the list_head. Fix originally suggested by Liam on IM. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 605c225..ceb2ba4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -918,6 +918,12 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, } if (reg >= 0) { + /* Any widget will do, they should all be updating the + * same register. + */ + w = list_first_entry(pending, struct snd_soc_dapm_widget, + power_list); + pop_dbg(dapm->dev, card->pop_time, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); -- 1.7.5.4

2 1

[alsa-devel] [PATCH 00/11] em28xx improvements
by Mauro Carvalho Chehab 19 Jun '11

19 Jun '11

Douglas borrowed me a couple devices some time ago. I've made some patches for the first one, but never had time to look at the second device, until this weekend, when I re-discovered it: An em2861-based device: a Kworld 305. On em2861, the audio interface is provided by a separate device, breaking the audio detection rules. With this device in hands, and using xawtv with audio support, I decided to make it work. This patch series contains: - a few em28xx driver random fixes; - support for compiling em28xx driver without IR support; - Mixer control for em28xx-based devices with AC97 mixer; - Add support for the USB vendor audio class at interface 1. Most em28xx hardware comes with an AC97 mixer (in general, emp202, also manufactured by Empiatech). In general, different types of input (TV, Svideo, Composite) uses different audio inputs at the mixer. Some devices also provide two different outputs (one for the audio out plug, and another for DMA transfers). Each vendor has its own way to connect the device. The em28xx cards contain the logic to mute/unmute the entries associated with each input, so the direct access to the mixer is interesting only if someone wants to do things like using more than one entry at the same time, or controlling the volume for whatever reason. With this setup, pulseaudio does the wrong thing: it just assumes that the master volume controls the device. On all tree devices I used for tests, this volume doesn't control the PCM output. I think that, on one of them, the Master is used to control the Audio Out jack output (seldomly used, on an usb device). Anyway, this is something that needs to be solved at pulseaudio, not inside the kernel (or, at least, I dunno any way for a driver to tell pulseaudio to not try to mask the audio controls). Mauro Carvalho Chehab (11): [media] em28xx: Don't initialize a var if won't be using it [media] em28xx: Fix a wrong enum at the ac97 control tables [media] em28xx: Allow to compile it without RC/input support [media] em28xx-alsa: add mixer support for AC97 volume controls [media] em28xx-audio: add support for mute controls [media] em28xx-audio: volumes are inverted [media] em28xx-audio: add debug info for the volume control [media] em28xx-audio: Properly report failures to start stream [media] em28xx-audio: Some Alsa API fixes [media] em28xx: Add support for devices with a separate audio interface [media] em28xx: Mark Kworld 305 as validated drivers/media/video/em28xx/Kconfig | 10 +- drivers/media/video/em28xx/Makefile | 6 +- drivers/media/video/em28xx/em28xx-audio.c | 253 ++++++++++++++++++++++++++--- drivers/media/video/em28xx/em28xx-cards.c | 104 +++++++++---- drivers/media/video/em28xx/em28xx-core.c | 37 +++-- drivers/media/video/em28xx/em28xx-i2c.c | 2 - drivers/media/video/em28xx/em28xx.h | 20 +++ 7 files changed, 355 insertions(+), 77 deletions(-)

1 0

[alsa-devel] [PATCH 09/11] [media] em28xx-audio: Some Alsa API fixes
by Mauro Carvalho Chehab 19 Jun '11

19 Jun '11

Mark the alsa stream with SNDRV_PCM_INFO_BATCH, as the timing to get audio streams can vary. Also, add SNDRV_PCM_TRIGGER for pause/release. while here, fix the stop indicator, to be sure that audio will be properly released at the stop events. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)redhat.com> diff --git a/drivers/media/video/em28xx/em28xx-audio.c b/drivers/media/video/em28xx/em28xx-audio.c index 5381f6d..47f21a3 100644 --- a/drivers/media/video/em28xx/em28xx-audio.c +++ b/drivers/media/video/em28xx/em28xx-audio.c @@ -249,6 +249,7 @@ static struct snd_pcm_hardware snd_em28xx_hw_capture = { .info = SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE, @@ -401,11 +402,15 @@ static int snd_em28xx_capture_trigger(struct snd_pcm_substream *substream, int retval = 0; switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: atomic_set(&dev->stream_started, 1); break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ case SNDRV_PCM_TRIGGER_STOP: - atomic_set(&dev->stream_started, 1); + atomic_set(&dev->stream_started, 0); break; default: retval = -EINVAL; -- 1.7.1

1 0

[alsa-devel] [PATCH 06/11] [media] em28xx-audio: volumes are inverted
by Mauro Carvalho Chehab 19 Jun '11

19 Jun '11

While here, fix volume mask to 5 bits Signed-off-by: Mauro Carvalho Chehab <mchehab(a)redhat.com> diff --git a/drivers/media/video/em28xx/em28xx-audio.c b/drivers/media/video/em28xx/em28xx-audio.c index c7b96b4..302553a 100644 --- a/drivers/media/video/em28xx/em28xx-audio.c +++ b/drivers/media/video/em28xx/em28xx-audio.c @@ -452,8 +452,8 @@ static int em28xx_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *value) { struct em28xx *dev = snd_kcontrol_chip(kcontrol); - u16 val = (value->value.integer.value[0] & 0x1f) | - (value->value.integer.value[1] & 0x1f) << 8; + u16 val = (0x1f - (value->value.integer.value[0] & 0x1f)) | + (0x1f - (value->value.integer.value[1] & 0x1f)) << 8; int rc; mutex_lock(&dev->lock); @@ -482,8 +482,8 @@ static int em28xx_vol_get(struct snd_kcontrol *kcontrol, if (val < 0) return val; - value->value.integer.value[0] = val & 0x1f; - value->value.integer.value[1] = (val << 8) & 0x1f; + value->value.integer.value[0] = 0x1f - (val & 0x1f); + value->value.integer.value[1] = 0x1f - ((val << 8) & 0x1f); return 0; } @@ -501,9 +501,9 @@ static int em28xx_vol_put_mute(struct snd_kcontrol *kcontrol, goto err; if (val) + rc &= 0x1f1f; + else rc |= 0x8000; - else - rc &= 0x7f7f; rc = em28xx_write_ac97(dev, kcontrol->private_value, rc); @@ -525,9 +525,9 @@ static int em28xx_vol_get_mute(struct snd_kcontrol *kcontrol, return val; if (val & 0x8000) - value->value.integer.value[0] = 1; - else value->value.integer.value[0] = 0; + else + value->value.integer.value[0] = 1; return 0; } -- 1.7.1

1 0

[alsa-devel] [PATCH 05/11] [media] em28xx-audio: add support for mute controls
by Mauro Carvalho Chehab 19 Jun '11

19 Jun '11

Signed-off-by: Mauro Carvalho Chehab <mchehab(a)redhat.com> diff --git a/drivers/media/video/em28xx/em28xx-audio.c b/drivers/media/video/em28xx/em28xx-audio.c index a75c779..c7b96b4 100644 --- a/drivers/media/video/em28xx/em28xx-audio.c +++ b/drivers/media/video/em28xx/em28xx-audio.c @@ -448,8 +448,6 @@ static int em28xx_vol_info(struct snd_kcontrol *kcontrol, return 0; } -/* FIXME: should also add mute controls for each */ - static int em28xx_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *value) { @@ -463,7 +461,7 @@ static int em28xx_vol_put(struct snd_kcontrol *kcontrol, if (rc < 0) goto err; - val |= rc & 0x8080; /* Preserve the mute flags */ + val |= rc & 0x8000; /* Preserve the mute flag */ rc = em28xx_write_ac97(dev, kcontrol->private_value, val); @@ -490,25 +488,87 @@ static int em28xx_vol_get(struct snd_kcontrol *kcontrol, return 0; } +static int em28xx_vol_put_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct em28xx *dev = snd_kcontrol_chip(kcontrol); + u16 val = value->value.integer.value[0]; + int rc; + + mutex_lock(&dev->lock); + rc = em28xx_read_ac97(dev, kcontrol->private_value); + if (rc < 0) + goto err; + + if (val) + rc |= 0x8000; + else + rc &= 0x7f7f; + + rc = em28xx_write_ac97(dev, kcontrol->private_value, rc); + +err: + mutex_unlock(&dev->lock); + return rc; +} + +static int em28xx_vol_get_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct em28xx *dev = snd_kcontrol_chip(kcontrol); + int val; + + mutex_lock(&dev->lock); + val = em28xx_read_ac97(dev, kcontrol->private_value); + mutex_unlock(&dev->lock); + if (val < 0) + return val; + + if (val & 0x8000) + value->value.integer.value[0] = 1; + else + value->value.integer.value[0] = 0; + + return 0; +} + static const DECLARE_TLV_DB_SCALE(em28xx_db_scale, -3450, 150, 0); static int em28xx_cvol_new(struct snd_card *card, struct em28xx *dev, char *name, int id) { int err; + char ctl_name[44]; struct snd_kcontrol *kctl; - struct snd_kcontrol_new tmp = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = name, - .info = em28xx_vol_info, - .get = em28xx_vol_get, - .put = em28xx_vol_put, - .private_value = id, - .tlv.p = em28xx_db_scale, - }; + struct snd_kcontrol_new tmp; + memset (&tmp, 0, sizeof(tmp)); + tmp.iface = SNDRV_CTL_ELEM_IFACE_MIXER, + tmp.private_value = id, + tmp.name = ctl_name, + + /* Add Mute Control */ + sprintf(ctl_name, "%s Switch", name); + tmp.get = em28xx_vol_get_mute; + tmp.put = em28xx_vol_put_mute; + tmp.info = snd_ctl_boolean_mono_info; kctl = snd_ctl_new1(&tmp, dev); + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + + memset (&tmp, 0, sizeof(tmp)); + tmp.iface = SNDRV_CTL_ELEM_IFACE_MIXER, + tmp.private_value = id, + tmp.name = ctl_name, + /* Add Volume Control */ + sprintf(ctl_name, "%s Volume", name); + tmp.get = em28xx_vol_get; + tmp.put = em28xx_vol_put; + tmp.info = em28xx_vol_info; + tmp.tlv.p = em28xx_db_scale, + kctl = snd_ctl_new1(&tmp, dev); err = snd_ctl_add(card, kctl); if (err < 0) return err; @@ -516,7 +576,6 @@ static int em28xx_cvol_new(struct snd_card *card, struct em28xx *dev, return 0; } - /* * register/unregister code and data */ -- 1.7.1

1 0

[alsa-devel] [PATCH 04/11] [media] em28xx-alsa: add mixer support for AC97 volume controls
by Mauro Carvalho Chehab 19 Jun '11

19 Jun '11

Export ac97 volume controls via mixer. Pulseaudio will probably handle it very badly, as it has no idea about how volumes are wired, and how are they associated with each TV input. Those wirings are card model dependent, and we don't have the wiring mappings for each supported device. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)redhat.com> diff --git a/drivers/media/video/em28xx/em28xx-audio.c b/drivers/media/video/em28xx/em28xx-audio.c index a24e177..a75c779 100644 --- a/drivers/media/video/em28xx/em28xx-audio.c +++ b/drivers/media/video/em28xx/em28xx-audio.c @@ -41,6 +41,7 @@ #include <sound/info.h> #include <sound/initval.h> #include <sound/control.h> +#include <sound/tlv.h> #include <media/v4l2-common.h> #include "em28xx.h" @@ -433,6 +434,92 @@ static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, return vmalloc_to_page(pageptr); } +/* + * AC97 volume control support + */ +static int em28xx_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0; + info->value.integer.max = 0x1f; + + return 0; +} + +/* FIXME: should also add mute controls for each */ + +static int em28xx_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct em28xx *dev = snd_kcontrol_chip(kcontrol); + u16 val = (value->value.integer.value[0] & 0x1f) | + (value->value.integer.value[1] & 0x1f) << 8; + int rc; + + mutex_lock(&dev->lock); + rc = em28xx_read_ac97(dev, kcontrol->private_value); + if (rc < 0) + goto err; + + val |= rc & 0x8080; /* Preserve the mute flags */ + + rc = em28xx_write_ac97(dev, kcontrol->private_value, val); + +err: + mutex_unlock(&dev->lock); + return rc; +} + +static int em28xx_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct em28xx *dev = snd_kcontrol_chip(kcontrol); + int val; + + mutex_lock(&dev->lock); + val = em28xx_read_ac97(dev, kcontrol->private_value); + mutex_unlock(&dev->lock); + if (val < 0) + return val; + + value->value.integer.value[0] = val & 0x1f; + value->value.integer.value[1] = (val << 8) & 0x1f; + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(em28xx_db_scale, -3450, 150, 0); + +static int em28xx_cvol_new(struct snd_card *card, struct em28xx *dev, + char *name, int id) +{ + int err; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new tmp = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = name, + .info = em28xx_vol_info, + .get = em28xx_vol_get, + .put = em28xx_vol_put, + .private_value = id, + .tlv.p = em28xx_db_scale, + }; + + kctl = snd_ctl_new1(&tmp, dev); + + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + + return 0; +} + + +/* + * register/unregister code and data + */ static struct snd_pcm_ops snd_em28xx_pcm_capture = { .open = snd_em28xx_capture_open, .close = snd_em28xx_pcm_close, @@ -489,6 +576,22 @@ static int em28xx_audio_init(struct em28xx *dev) INIT_WORK(&dev->wq_trigger, audio_trigger); + if (dev->audio_mode.ac97 != EM28XX_NO_AC97) { + em28xx_cvol_new(card, dev, "Video", AC97_VIDEO_VOL); + em28xx_cvol_new(card, dev, "Line In", AC97_LINEIN_VOL); + em28xx_cvol_new(card, dev, "Phone", AC97_PHONE_VOL); + em28xx_cvol_new(card, dev, "Microphone", AC97_PHONE_VOL); + em28xx_cvol_new(card, dev, "CD", AC97_CD_VOL); + em28xx_cvol_new(card, dev, "AUX", AC97_AUX_VOL); + em28xx_cvol_new(card, dev, "PCM", AC97_PCM_OUT_VOL); + + em28xx_cvol_new(card, dev, "Master", AC97_MASTER_VOL); + em28xx_cvol_new(card, dev, "Line", AC97_LINE_LEVEL_VOL); + em28xx_cvol_new(card, dev, "Mono", AC97_MASTER_MONO_VOL); + em28xx_cvol_new(card, dev, "LFE", AC97_LFE_MASTER_VOL); + em28xx_cvol_new(card, dev, "Surround", AC97_SURR_MASTER_VOL); + } + err = snd_card_register(card); if (err < 0) { snd_card_free(card); diff --git a/drivers/media/video/em28xx/em28xx-core.c b/drivers/media/video/em28xx/em28xx-core.c index 7bf3a86..752d4ed 100644 --- a/drivers/media/video/em28xx/em28xx-core.c +++ b/drivers/media/video/em28xx/em28xx-core.c @@ -286,6 +286,7 @@ int em28xx_read_ac97(struct em28xx *dev, u8 reg) return ret; return le16_to_cpu(val); } +EXPORT_SYMBOL_GPL(em28xx_read_ac97); /* * em28xx_write_ac97() @@ -313,6 +314,7 @@ int em28xx_write_ac97(struct em28xx *dev, u8 reg, u16 val) return 0; } +EXPORT_SYMBOL_GPL(em28xx_write_ac97); struct em28xx_vol_itable { enum em28xx_amux mux; -- 1.7.1

1 0

[alsa-devel] [PATCH 08/11] [media] em28xx-audio: Properly report failures to start stream
by Mauro Carvalho Chehab 19 Jun '11

19 Jun '11

If the audio stream fails for any reason, it should: 1) Report an error via dmesg; 2) Mark internally that the stream didn't started. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)redhat.com> diff --git a/drivers/media/video/em28xx/em28xx-audio.c b/drivers/media/video/em28xx/em28xx-audio.c index 56739a4..5381f6d 100644 --- a/drivers/media/video/em28xx/em28xx-audio.c +++ b/drivers/media/video/em28xx/em28xx-audio.c @@ -213,9 +213,12 @@ static int em28xx_init_audio_isoc(struct em28xx *dev) for (i = 0; i < EM28XX_AUDIO_BUFS; i++) { errCode = usb_submit_urb(dev->adev.urb[i], GFP_ATOMIC); if (errCode) { + em28xx_errdev("submit of audio urb failed\n"); em28xx_deinit_isoc_audio(dev); + atomic_set(&dev->stream_started, 0); return errCode; } + } return 0; -- 1.7.1

1 0

[alsa-devel] [PATCH 07/11] [media] em28xx-audio: add debug info for the volume control
by Mauro Carvalho Chehab 19 Jun '11

19 Jun '11

Signed-off-by: Mauro Carvalho Chehab <mchehab(a)redhat.com> diff --git a/drivers/media/video/em28xx/em28xx-audio.c b/drivers/media/video/em28xx/em28xx-audio.c index 302553a..56739a4 100644 --- a/drivers/media/video/em28xx/em28xx-audio.c +++ b/drivers/media/video/em28xx/em28xx-audio.c @@ -464,6 +464,13 @@ static int em28xx_vol_put(struct snd_kcontrol *kcontrol, val |= rc & 0x8000; /* Preserve the mute flag */ rc = em28xx_write_ac97(dev, kcontrol->private_value, val); + if (rc < 0) + goto err; + + dprintk("%sleft vol %d, right vol %d (0x%04x) to ac97 volume control 0x%04x\n", + (val & 0x8000) ? "muted " : "", + 0x1f - ((val >> 8) & 0x1f), 0x1f - (val & 0x1f), + val, (int)kcontrol->private_value); err: mutex_unlock(&dev->lock); @@ -482,6 +489,11 @@ static int em28xx_vol_get(struct snd_kcontrol *kcontrol, if (val < 0) return val; + dprintk("%sleft vol %d, right vol %d (0x%04x) from ac97 volume control 0x%04x\n", + (val & 0x8000) ? "muted " : "", + 0x1f - ((val >> 8) & 0x1f), 0x1f - (val & 0x1f), + val, (int)kcontrol->private_value); + value->value.integer.value[0] = 0x1f - (val & 0x1f); value->value.integer.value[1] = 0x1f - ((val << 8) & 0x1f); @@ -506,6 +518,13 @@ static int em28xx_vol_put_mute(struct snd_kcontrol *kcontrol, rc |= 0x8000; rc = em28xx_write_ac97(dev, kcontrol->private_value, rc); + if (rc < 0) + goto err; + + dprintk("%sleft vol %d, right vol %d (0x%04x) to ac97 volume control 0x%04x\n", + (val & 0x8000) ? "muted " : "", + 0x1f - ((val >> 8) & 0x1f), 0x1f - (val & 0x1f), + val, (int)kcontrol->private_value); err: mutex_unlock(&dev->lock); @@ -529,6 +548,11 @@ static int em28xx_vol_get_mute(struct snd_kcontrol *kcontrol, else value->value.integer.value[0] = 1; + dprintk("%sleft vol %d, right vol %d (0x%04x) from ac97 volume control 0x%04x\n", + (val & 0x8000) ? "muted " : "", + 0x1f - ((val >> 8) & 0x1f), 0x1f - (val & 0x1f), + val, (int)kcontrol->private_value); + return 0; } @@ -556,6 +580,8 @@ static int em28xx_cvol_new(struct snd_card *card, struct em28xx *dev, err = snd_ctl_add(card, kctl); if (err < 0) return err; + dprintk("Added control %s for ac97 volume control 0x%04x\n", + ctl_name, id); memset (&tmp, 0, sizeof(tmp)); tmp.iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -572,6 +598,8 @@ static int em28xx_cvol_new(struct snd_card *card, struct em28xx *dev, err = snd_ctl_add(card, kctl); if (err < 0) return err; + dprintk("Added control %s for ac97 volume control 0x%04x\n", + ctl_name, id); return 0; } -- 1.7.1

1 0

[alsa-devel] [PATCH 10/11] [media] em28xx: Add support for devices with a separate audio interface
by Mauro Carvalho Chehab 19 Jun '11

19 Jun '11

Some devices use a separate interface for the vendor audio class. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)redhat.com> diff --git a/drivers/media/video/em28xx/em28xx-audio.c b/drivers/media/video/em28xx/em28xx-audio.c index 47f21a3..cff0768 100644 --- a/drivers/media/video/em28xx/em28xx-audio.c +++ b/drivers/media/video/em28xx/em28xx-audio.c @@ -3,9 +3,9 @@ * * Copyright (C) 2006 Markus Rechberger <mrechberger(a)gmail.com> * - * Copyright (C) 2007 Mauro Carvalho Chehab <mchehab(a)infradead.org> + * Copyright (C) 2007-2011 Mauro Carvalho Chehab <mchehab(a)redhat.com> * - Port to work with the in-kernel driver - * - Several cleanups + * - Cleanups, fixes, alsa-controls, etc. * * This driver is based on my previous au600 usb pstn audio driver * and inherits all the copyrights @@ -281,24 +281,28 @@ static int snd_em28xx_capture_open(struct snd_pcm_substream *substream) return -ENODEV; } - /* Sets volume, mute, etc */ - - dev->mute = 0; - mutex_lock(&dev->lock); - ret = em28xx_audio_analog_set(dev); - if (ret < 0) - goto err; - runtime->hw = snd_em28xx_hw_capture; - if (dev->alt == 0 && dev->adev.users == 0) { - dev->alt = 7; - dprintk("changing alternate number to 7\n"); - usb_set_interface(dev->udev, 0, 7); + if ((dev->alt == 0 || dev->audio_ifnum) && dev->adev.users == 0) { + if (dev->audio_ifnum) + dev->alt = 1; + else + dev->alt = 7; + + dprintk("changing alternate number on interface %d to %d\n", + dev->audio_ifnum, dev->alt); + usb_set_interface(dev->udev, dev->audio_ifnum, dev->alt); + + /* Sets volume, mute, etc */ + dev->mute = 0; + mutex_lock(&dev->lock); + ret = em28xx_audio_analog_set(dev); + if (ret < 0) + goto err; + + dev->adev.users++; + mutex_unlock(&dev->lock); } - dev->adev.users++; - mutex_unlock(&dev->lock); - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); dev->adev.capture_pcm_substream = substream; runtime->private_data = dev; @@ -635,17 +639,17 @@ static int em28xx_audio_init(struct em28xx *dev) static int devnr; int err; - if (dev->has_alsa_audio != 1) { + if (!dev->has_alsa_audio || dev->audio_ifnum < 0) { /* This device does not support the extension (in this case the device is expecting the snd-usb-audio module or doesn't have analog audio support at all) */ return 0; } - printk(KERN_INFO "em28xx-audio.c: probing for em28x1 " - "non standard usbaudio\n"); + printk(KERN_INFO "em28xx-audio.c: probing for em28xx Audio Vendor Class\n"); printk(KERN_INFO "em28xx-audio.c: Copyright (C) 2006 Markus " "Rechberger\n"); + printk(KERN_INFO "em28xx-audio.c: Copyright (C) 2007-2011 Mauro Carvalho Chehab\n"); err = snd_card_create(index[devnr], "Em28xx Audio", THIS_MODULE, 0, &card); @@ -737,7 +741,7 @@ static void __exit em28xx_alsa_unregister(void) MODULE_LICENSE("GPL"); MODULE_AUTHOR("Markus Rechberger <mrechberger(a)gmail.com>"); -MODULE_AUTHOR("Mauro Carvalho Chehab <mchehab(a)infradead.org>"); +MODULE_AUTHOR("Mauro Carvalho Chehab <mchehab(a)redhat.com>"); MODULE_DESCRIPTION("Em28xx Audio driver"); module_init(em28xx_alsa_register); diff --git a/drivers/media/video/em28xx/em28xx-cards.c b/drivers/media/video/em28xx/em28xx-cards.c index bbd67d7..c445bea 100644 --- a/drivers/media/video/em28xx/em28xx-cards.c +++ b/drivers/media/video/em28xx/em28xx-cards.c @@ -2846,6 +2846,16 @@ static int em28xx_init_dev(struct em28xx **devhandle, struct usb_device *udev, } } + if (dev->is_audio_only) { + errCode = em28xx_audio_setup(dev); + if (errCode) + return -ENODEV; + em28xx_add_into_devlist(dev); + em28xx_init_extension(dev); + + return 0; + } + /* Prepopulate cached GPO register content */ retval = em28xx_read_reg(dev, dev->reg_gpo_num); if (retval >= 0) @@ -2946,6 +2956,9 @@ fail_reg_devices: return retval; } +/* high bandwidth multiplier, as encoded in highspeed endpoint descriptors */ +#define hb_mult(wMaxPacketSize) (1 + (((wMaxPacketSize) >> 11) & 0x03)) + /* * em28xx_usb_probe() * checks for supported devices @@ -2955,15 +2968,15 @@ static int em28xx_usb_probe(struct usb_interface *interface, { const struct usb_endpoint_descriptor *endpoint; struct usb_device *udev; - struct usb_interface *uif; struct em28xx *dev = NULL; int retval; - int i, nr, ifnum, isoc_pipe; + bool is_audio_only = false, has_audio = false; + int i, nr, isoc_pipe; + const int ifnum = interface->altsetting[0].desc.bInterfaceNumber; char *speed; char descr[255] = ""; udev = usb_get_dev(interface_to_usbdev(interface)); - ifnum = interface->altsetting[0].desc.bInterfaceNumber; /* Check to see next free device and mark as used */ nr = find_first_zero_bit(&em28xx_devused, EM28XX_MAXBOARDS); @@ -2983,6 +2996,19 @@ static int em28xx_usb_probe(struct usb_interface *interface, goto err; } + /* Get endpoints */ + for (i = 0; i < interface->num_altsetting; i++) { + int ep; + + for (ep = 0; ep < interface->altsetting[i].desc.bNumEndpoints; ep++) { + struct usb_host_endpoint *e; + e = &interface->altsetting[i].endpoint[ep]; + + if (e->desc.bEndpointAddress == 0x83) + has_audio = true; + } + } + endpoint = &interface->cur_altsetting->endpoint[0].desc; /* check if the device has the iso in endpoint at the correct place */ @@ -3002,19 +3028,22 @@ static int em28xx_usb_probe(struct usb_interface *interface, check_interface = 0; if (!check_interface) { - em28xx_err(DRIVER_NAME " video device (%04x:%04x): " - "interface %i, class %i found.\n", - le16_to_cpu(udev->descriptor.idVendor), - le16_to_cpu(udev->descriptor.idProduct), - ifnum, - interface->altsetting[0].desc.bInterfaceClass); + if (has_audio) { + is_audio_only = true; + } else { + em28xx_err(DRIVER_NAME " video device (%04x:%04x): " + "interface %i, class %i found.\n", + le16_to_cpu(udev->descriptor.idVendor), + le16_to_cpu(udev->descriptor.idProduct), + ifnum, + interface->altsetting[0].desc.bInterfaceClass); + em28xx_err(DRIVER_NAME " This is an anciliary " + "interface not used by the driver\n"); - em28xx_err(DRIVER_NAME " This is an anciliary " - "interface not used by the driver\n"); - - em28xx_devused &= ~(1<<nr); - retval = -ENODEV; - goto err; + em28xx_devused &= ~(1<<nr); + retval = -ENODEV; + goto err; + } } } @@ -3044,8 +3073,8 @@ static int em28xx_usb_probe(struct usb_interface *interface, if (*descr) strlcat(descr, " ", sizeof(descr)); - printk(DRIVER_NAME ": New device %s@ %s Mbps " - "(%04x:%04x, interface %d, class %d)\n", + printk(KERN_INFO DRIVER_NAME + ": New device %s@ %s Mbps (%04x:%04x, interface %d, class %d)\n", descr, speed, le16_to_cpu(udev->descriptor.idVendor), @@ -3053,6 +3082,11 @@ static int em28xx_usb_probe(struct usb_interface *interface, ifnum, interface->altsetting->desc.bInterfaceNumber); + if (has_audio) + printk(KERN_INFO DRIVER_NAME + ": Audio Vendor Class interface %i found\n", + ifnum); + /* * Make sure we have 480 Mbps of bandwidth, otherwise things like * video stream wouldn't likely work, since 12 Mbps is generally @@ -3088,10 +3122,13 @@ static int em28xx_usb_probe(struct usb_interface *interface, dev->devno = nr; dev->model = id->driver_info; dev->alt = -1; + dev->is_audio_only = is_audio_only; + dev->has_alsa_audio = has_audio; + dev->audio_ifnum = ifnum; /* Checks if audio is provided by some interface */ for (i = 0; i < udev->config->desc.bNumInterfaces; i++) { - uif = udev->config->interface[i]; + struct usb_interface *uif = udev->config->interface[i]; if (uif->altsetting[0].desc.bInterfaceClass == USB_CLASS_AUDIO) { dev->has_audio_class = 1; break; @@ -3099,9 +3136,7 @@ static int em28xx_usb_probe(struct usb_interface *interface, } /* compute alternate max packet sizes */ - uif = udev->actconfig->interface[0]; - - dev->num_alt = uif->num_altsetting; + dev->num_alt = interface->num_altsetting; dev->alt_max_pkt_size = kmalloc(32 * dev->num_alt, GFP_KERNEL); if (dev->alt_max_pkt_size == NULL) { @@ -3113,14 +3148,21 @@ static int em28xx_usb_probe(struct usb_interface *interface, } for (i = 0; i < dev->num_alt ; i++) { - u16 tmp = le16_to_cpu(uif->altsetting[i].endpoint[isoc_pipe].desc.wMaxPacketSize); - dev->alt_max_pkt_size[i] = - (tmp & 0x07ff) * (((tmp & 0x1800) >> 11) + 1); + u16 tmp = le16_to_cpu(interface->altsetting[i].endpoint[isoc_pipe].desc.wMaxPacketSize); + unsigned int size = tmp & 0x7ff; + + if (udev->speed == USB_SPEED_HIGH) + size = size * hb_mult(tmp); + + dev->alt_max_pkt_size[i] = size; } if ((card[nr] >= 0) && (card[nr] < em28xx_bcount)) dev->model = card[nr]; + /* save our data pointer in this interface device */ + usb_set_intfdata(interface, dev); + /* allocate device struct */ mutex_init(&dev->lock); mutex_lock(&dev->lock); @@ -3132,9 +3174,6 @@ static int em28xx_usb_probe(struct usb_interface *interface, goto err; } - /* save our data pointer in this interface device */ - usb_set_intfdata(interface, dev); - request_modules(dev); /* Should be the last thing to do, to avoid newer udev's to @@ -3163,6 +3202,13 @@ static void em28xx_usb_disconnect(struct usb_interface *interface) if (!dev) return; + if (dev->is_audio_only) { + mutex_lock(&dev->lock); + em28xx_close_extension(dev); + mutex_unlock(&dev->lock); + return; + } + em28xx_info("disconnecting %s\n", dev->vdev->name); flush_request_modules(dev); diff --git a/drivers/media/video/em28xx/em28xx-core.c b/drivers/media/video/em28xx/em28xx-core.c index 752d4ed..16c9b73 100644 --- a/drivers/media/video/em28xx/em28xx-core.c +++ b/drivers/media/video/em28xx/em28xx-core.c @@ -499,17 +499,13 @@ int em28xx_audio_setup(struct em28xx *dev) if (dev->chip_id == CHIP_ID_EM2870 || dev->chip_id == CHIP_ID_EM2874 || dev->chip_id == CHIP_ID_EM28174) { /* Digital only device - don't load any alsa module */ - dev->audio_mode.has_audio = 0; - dev->has_audio_class = 0; - dev->has_alsa_audio = 0; + dev->audio_mode.has_audio = false; + dev->has_audio_class = false; + dev->has_alsa_audio = false; return 0; } - /* If device doesn't support Usb Audio Class, use vendor class */ - if (!dev->has_audio_class) - dev->has_alsa_audio = 1; - - dev->audio_mode.has_audio = 1; + dev->audio_mode.has_audio = true; /* See how this device is configured */ cfg = em28xx_read_reg(dev, EM28XX_R00_CHIPCFG); @@ -519,8 +515,8 @@ int em28xx_audio_setup(struct em28xx *dev) cfg = EM28XX_CHIPCFG_AC97; /* Be conservative */ } else if ((cfg & EM28XX_CHIPCFG_AUDIOMASK) == 0x00) { /* The device doesn't have vendor audio at all */ - dev->has_alsa_audio = 0; - dev->audio_mode.has_audio = 0; + dev->has_alsa_audio = false; + dev->audio_mode.has_audio = false; return 0; } else if ((cfg & EM28XX_CHIPCFG_AUDIOMASK) == EM28XX_CHIPCFG_I2S_3_SAMPRATES) { @@ -549,8 +545,8 @@ int em28xx_audio_setup(struct em28xx *dev) */ em28xx_warn("AC97 chip type couldn't be determined\n"); dev->audio_mode.ac97 = EM28XX_NO_AC97; - dev->has_alsa_audio = 0; - dev->audio_mode.has_audio = 0; + dev->has_alsa_audio = false; + dev->audio_mode.has_audio = false; goto init_audio; } diff --git a/drivers/media/video/em28xx/em28xx.h b/drivers/media/video/em28xx/em28xx.h index f9b77b4..e03849f 100644 --- a/drivers/media/video/em28xx/em28xx.h +++ b/drivers/media/video/em28xx/em28xx.h @@ -487,6 +487,8 @@ struct em28xx { int devno; /* marks the number of this device */ enum em28xx_chip_id chip_id; + int audio_ifnum; + struct v4l2_device v4l2_dev; struct em28xx_board board; @@ -503,6 +505,7 @@ struct em28xx { unsigned int has_audio_class:1; unsigned int has_alsa_audio:1; + unsigned int is_audio_only:1; /* Controls audio streaming */ struct work_struct wq_trigger; /* Trigger to start/stop audio for alsa module */ -- 1.7.1

1 0

[alsa-devel] [PATCH] ASoC: soc-dapm: Fix breakage with incorrect codec pointer
by Dimitris Papastamos 19 Jun '11

19 Jun '11

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 605c225..a877dc6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -922,7 +922,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - soc_widget_update_bits(w, reg, mask, value); + snd_soc_update_bits(dapm->codec, reg, mask, value); } list_for_each_entry(w, pending, power_list) { -- 1.7.5.2

3 5

[alsa-devel] Loud pop from speakers when plugging out headphones/shutdown (ALC665 Intel HDA)
by breakdown＠paranoici.org 19 Jun '11

19 Jun '11

Hi all, I have a rather annoying problem with the audio system on my Dell XPS L502X laptop (ALC665 chipset). Basically, every time I plug out headphones or perform a shutdown, a loud "pop" sound (usually referred to as "static noise", but way louder) can be heard from the left speaker. I would exclude this is an hardware problem, since with Windows 7 the problem is absent. The "pop" sound is louder when headphones are first plugged out after a certain amount of time listening to music, if I plug in/out immediately after, no poppish sound can be heard. Steps I've tried to do: - checked the "power_save=0" module option (i have a custom 2.6.39 kernel and I disabled that option during the configuration process, I've also tried to enable it in the kernel and then added that option, with no results) - changed the "model" parameter when loading the snd-hda-intel driver (I've tried nearly all of the "dell" parameters, with no success) - muted all audio channels - disabled the pulseaudio system to only use ALSA - changed headphones, same problem The only workaround I've discovered is to manually remove the snd-hda-intel module (or, in alternative, passing the "dell" module parameter, which on this card actually disable both left and right speakers). Detailed info as follows (attached and uploaded): lspci -vvnn output (only relevant lines): http://pastebin.com/88un2GUv alsa-info.sh output: http://www.alsa-project.org/db/?f=88ceaf1f9de818d205f8b93f77f8ac57f19fd27d /proc/asound/card*/codec#*: http://pastebin.com/JgGHAQ23 Thank you in advance. Cheers, Angelo

1 0

[alsa-devel] [Fwd: [kde] Low maximum volume with pulseaudio on kde]
by Sérgio Basto 19 Jun '11

19 Jun '11

-------- Forwarded Message -------- From: Sérgio Basto <sergio(a)serjux.com> Reply-to: kde(a)mail.kde.org To: KDE ML <kde(a)mail.kde.org> Subject: [kde] Low maximum volume with pulseaudio on kde Date: Wed, 15 Jun 2011 23:04:26 +0100 Hi, try again, 1st - edit /etc/pulse/client.conf and change autospawn = no 2nd - killall pulseaudio 3rd - run on user env: pulseaudio -vvv lets test with kmix I see on pulseaudio logs at maximum : D: alsa-sink.c: Requested volume: 0: 100% 1: 100% D: alsa-sink.c: Got hardware volume: 0: 100% 1: 100% D: alsa-sink.c: Calculated software volume: 0: 100% 1: 100% (accurate-enough=yes) with gnome-control-center -> sound I see on pulseaudio logs at maximum : D: protocol-native.c: Client gnome-control-center changes volume of sink alsa_output.pci-0000_00_1b.0.analog-stereo. D: alsa-sink.c: Requested volume: 0: 153% 1: 153% D: alsa-sink.c: Got hardware volume: 0: 100% 1: 100% D: alsa-sink.c: Calculated software volume: 0: 153% 1: 153% (accurate-enough=no) so how I put kmix also control "software volume" since system use pulse-audio ? I read this articles http://www.webupd8.org/2009/08/increase-maximum-sound-level-in-ubuntu.html http://alsa.opensrc.org/How_to_use_softvol_to_control_the_master_volume but can't figure out. thanks, -- Sérgio M. B. ___________________________________________________ This message is from the kde mailing list. Account management: https://mail.kde.org/mailman/listinfo/kde. Archives: http://lists.kde.org/. More info: http://www.kde.org/faq.html. -- Sérgio M. B.

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Re: [alsa-devel] [Fwd: [kde] Low maximum volume with pulseaudio on kde]
by Sérgio Basto 18 Jun '11

18 Jun '11

On Sat, 2011-06-18 at 13:12 -0400, Daniel Chen wrote: > > On Jun 18, 2011 1:05 PM, "Sérgio Basto" <sergio(a)serjux.com> wrote: > > > > On Sat, 2011-06-18 at 10:19 -0400, Daniel Chen wrote: > > > > > > Does the GNOME volume control in Sound Preferences show that the > > > slider is to the right of 100%? > > > > yes go to 150% > > > > -- > > Sérgio M. B. > > > > You should probably get in touch with Colin Guthrie, who made > modifications to Kmix to support PulseAudio. This isn't an alsa-devel > issue per se. so how I put kmix also control "software volume" since system use pulseaudio ? http://alsa.opensrc.org/How_to_use_softvol_to_control_the_master_volume and if I disable pulseaudio , how I configure softvols with kmix or AlsaMixer ? > Cheers, > -Dan > -- Sérgio M. B.

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Re: [alsa-devel] [PATCH v4 11/18] input: Add initial support for TWL6040 vibrator
by Péter Ujfalusi 18 Jun '11

18 Jun '11

On Wednesday 15 June 2011 10:23:01 Tejun Heo wrote: > On Wed, Jun 15, 2011 at 10:18:58AM +0200, Tejun Heo wrote: > > No human being can feel 120usec difference and I can't see how using > > HIGHPRI is justified here (which is what the code is doing > > _accidentally_ by using singlethread_workqueue). > > Ooh, one more thing, and even if you insist on using HIGHPRI (please > don't), you don't need to create workqueue for each device. You can > just create one for the whole driver in init and destroy it from exit. > What matters is the HIGHPRI attribute of the workqueue. The number of > workqueues is completely irrelevant. Fair enough. I'll move to create_workqueue. If we later find issues with this (in a 'live' system), we can figure out a way to fix it. Thank you for your time on this. I'll make the changes accordingly. Regards, Péter

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[alsa-devel] [PATCH] ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency
by Mark Brown 18 Jun '11

18 Jun '11

The Sigma code is in drivers/firmware which is only included on a very small subset of architectures and so ends up breaking the build on others. There's a pending patch to make the directory build as standard but it's not merged yet. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- I already applied this due to the build breakage. sound/soc/codecs/Kconfig | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f745e55..7a2e426 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,7 +17,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 - select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C -- 1.7.5.4

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[alsa-devel] [PATCH 1/2] ASoC: Fix ADAU1701 build in SND_SOC_ALL_CODECS
by Mark Brown 18 Jun '11

18 Jun '11

Since select doesn't obey dependencies we need ALL_CODECS to select SIGMA directly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bb90642..33ceb06 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" + select SIGMA if I2C # Analog devices DSP select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS -- 1.7.5.4

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[alsa-devel] [PATCH] ALSA: HDA: Remove redundant LPIB quirks for ATI chipset
by David Henningsson 17 Jun '11

17 Jun '11

Now that we have changed the position_fix default for ATI and AMD to be LPIB (see commit 50e3bbf989), we can remove the quirks that were added for ATI chipsets. I went through the commits, most of them pointing to launchpad, where the PCI ID is easily available. I'm also attaching my notes in case it is helpful for anyone. Seeing that note list, we should probably also set "VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10)" to use LPIB as well. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] ALSA Setta Mixer Values
by Alfs Kurmis 17 Jun '11

17 Jun '11

Hi Experts ! How with C program i do List/Check available mixer channels for ALSA snd card X, how to determine is it input or output , how to set [and read] value(s) for any channel(s) , [if necessary] [ different for Left-Right if necessary ] Or simplest case - How to just setta Master and PCM to given values ? Which include(s)--Structures--Functions must be used ? Is somewhere any C copy-paste example or template available ? Tnx for examples in advance @ all. Alfs Kurmis ----

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[alsa-devel] [PATCH] ASoC: Blackfin: allow SPI for SSM2602 parts
by Mike Frysinger 17 Jun '11

17 Jun '11

This board has hardware switches for selecting SPI or I2C, so don't require I2C for this driver. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/Kconfig | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 4d361bd..871308f 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -10,10 +10,9 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) select SND_BF5XX_SOC_I2S select SND_SOC_SSM2602 - select I2C help Say Y if you want to add support for SoC audio on BF527-EZKIT. -- 1.7.5.3

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[PATCH] ALSA: HDA: Remove redundant LPIB quirks for ATI chipset
by David Henningsson 17 Jun '11

17 Jun '11

Now that we have changed the position_fix default for ATI and AMD to be LPIB (see commit 50e3bbf989), we can remove the quirks that were added for ATI chipsets. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 8 -------- 1 files changed, 0 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fdfd748..10819c8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2361,28 +2361,20 @@ static int azx_dev_free(struct snd_device *device) * white/black-listing for position_fix */ static struct snd_pci_quirk position_fix_list[] __devinitdata = { - SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1028, 0x0470, "Dell Inspiron 1120", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB), - SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} }; -- 1.7.4.1 --------------070903030003090708030002 Content-Type: text/plain; name="lpibcommits2.txt" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="lpibcommits2.txt" commit 7a68be94e22e7643038726ebc14360752a91800b BugLink: https://launchpad.net/bugs/583983 Audio device [0403]: ATI Technologies Inc IXP SB4x0 High Definition Audio Controller [1002:437b] (rev 01) Subsystem: Acer Incorporated [ALI] Device [1025:009f] commit dd5a089edfa51a74692604b4b427953d8e16bc35 BugLink: https://launchpad.net/bugs/685161 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] Subsystem: Acer Incorporated [ALI] Device [1025:026f] commit d2e1c97331d37a477ef7e9a272b4de4e53b893be Unknown hw commit 9919c7619c52d01e89103bca405cc3d4a2b1ac31 BugLink: https://launchpad.net/bugs/530346 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] Subsystem: Dell Device [1028:01f6] commit e03fa055bc126e536c7f65862e08a9b143138ea9 Hardware info not present in buglink, but I know for sure it's ATI chipset. commit 2f703e7a2ea5f6d5ea14a7b2cd0d31be07839ac6 Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) 0403: 8086:293e (rev 03) Subsystem: 103c:306d commit dd37f8e8659bc617c3f2a84e007a4824ccdac458 BugLink: https://launchpad.net/bugs/465942 VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASUSTeK Computer Inc. Device [1043:81b3] commit 9f75c1b12c5ef392ddcea575b13560842c28b1b3 BugLink: https://launchpad.net/bugs/587546 VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASUSTeK Computer Inc. Device [1043:81e7] commit b540afc2b3d6e4cd1d1f137ef6d9e9c78d67fecd BugLink: http://bugs.launchpad.net/bugs/718402 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] (rev 40) Subsystem: ASUSTeK Computer Inc. Device [1043:8410] commit e96d3127760a2fc509bca6bf7e61e8bc61497aeb BugLink: https://launchpad.net/bugs/586347 Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05) 0403: 8086:3b56 (rev 05) Subsystem: 104d:9069 commit 4e0938dba7fccf37a4aecba4d937da7f312b5d55 BugLink: https://launchpad.net/bugs/549560 ATI Technologies Inc IXP SB4x0 High Definition Audio Controller [1002:437b] (rev 01) Subsystem: Toshiba America Info Systems Device [1179:ff10] commit 61bb42c37dfa9016dcacc86bcd41362ab2457d4a BugLink: https://launchpad.net/bugs/551949 Intel Corporation N10/ICH 7 Family High Definition Audio Controller [8086:27d8] (rev 01) Subsystem: Holco Enterprise Co, Ltd/Shuttle Computer Device [1297:3166] commit 9ec8ddad59fadd8021adfea4cb716a49b0e232e9 BugLink: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=575669 Unknown hw commit 45d4ebf1a6255f2234a041685789cbecac3453f1 unknown hw commit 0321b69569eadbc13242922925a4316754c5f744 BugLink: https://launchpad.net/bugs/523953 ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] Subsystem: Biostar Microtech Int'l Corp Device [1565:820f] commit 8815cd030fdd73932a791d1f06194c8db807cde7 unknown hw commit b90c076424da8166797bdc34187660fd0124f530 BugLink: https://launchpad.net/bugs/580749 VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASRock Incorporation Device [1849:0888] commit 0e0280dc2b0c7395a880d25544b47f3e3e3f79db BugLink: https://launchpad.net/bugs/459083 Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 02) Subsystem: Intel Corporation Device [8086:2503] commit 572c0e3c73341755f3e7dfaaef6b26df12bd709c BugLink: https://bugs.launchpad.net/bugs/538895 ATI Technologies Inc IXP SB4x0 High Definition Audio Controller [1002:437b] (rev 01) Subsystem: Intel Corporation Device [8086:d601] --------------070903030003090708030002--

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[alsa-devel] ASoC: "No soundcards found" but 'snd_soc_instantiate_card' succeeds afterwards
by Lambrecht Jürgen 17 Jun '11

17 Jun '11

Hello, I have a imx27 ARM9 with MC13783 codec (all from Freescale). alsa init does not find a sound card, but finally there is a sound card. cat /proc/asound/cards: 0 [iMXmc13783audio]: - iMX-mc13783-audio iMX-mc13783-audio Maybe the inits are done in a wrong order, but how to fix? Here some parts of the prints: [start] bus: 'platform': add driver mc13783-codec bus: 'platform': driver_probe_device: matched device mc13783-codec with driver mc13783-codec bus: 'platform': really_probe: probing driver mc13783-codec with device mc13783-codec mc13783-codec mc13783-codec: codec register mc13783-codec mc13783-codec mc13783-codec: dai register mc13783-codec #1 Registered DAI 'mc13783-hifi' Registered codec 'mc13783-codec' driver: 'mc13783-codec': driver_bound: bound to device 'mc13783-codec' bus: 'platform': really_probe: bound device mc13783-codec to driver mc13783-codec bus: 'platform': add driver imx-ssi bus: 'platform': driver_probe_device: matched device imx-ssi.0 with driver imx-ssi bus: 'platform': really_probe: probing driver imx-ssi with device imx-ssi.0 imx-ssi imx-ssi.0: dai register imx-ssi.0 Registered DAI 'imx-ssi.0' Registering platform device 'imx-fiq-pcm-audio.0'. Parent at platform device: 'imx-fiq-pcm-audio.0': device_add bus: 'platform': add device imx-fiq-pcm-audio.0 Registering platform device 'imx-pcm-audio.0'. Parent at platform device: 'imx-pcm-audio.0': device_add bus: 'platform': add device imx-pcm-audio.0 driver: 'imx-ssi.0': driver_bound: bound to device 'imx-ssi' bus: 'platform': really_probe: bound device imx-ssi.0 to driver imx-ssi bus: 'platform': add driver imx-pcm-audio bus: 'platform': driver_probe_device: matched device imx-pcm-audio.0 with driver imx-pcm-audio bus: 'platform': really_probe: probing driver imx-pcm-audio with device imx-pcm-audio.0 imx-pcm-audio imx-pcm-audio.0: platform register imx-pcm-audio.0 asoc: snd_soc_instantiate_cards Registered platform 'imx-pcm-audio.0' driver: 'imx-pcm-audio.0': driver_bound: bound to device 'imx-pcm-audio' bus: 'platform': really_probe: bound device imx-pcm-audio.0 to driver imx-pcm-audio ALSA device list: No soundcards found. [end] -> then a sound card is registered by calling 'late_initcall(imx_mc13783_audio_init);' [start] Registering platform device 'soc-audio'. Parent at platform device: 'soc-audio': device_add bus: 'platform': add device soc-audio bus: 'platform': driver_probe_device: matched device soc-audio with driver soc-audio bus: 'platform': really_probe: probing driver soc-audio with device soc-audio asoc: snd_soc_instantiate_card iMX-mc13783-audio soc-audio soc-audio: binding MC13783 Playback at idx 0 soc-audio soc-audio: binding MC13783 Capture at idx 1 asoc: created sound card for card iMX-mc13783-audio asoc: sound card iMX-mc13783-audio probed soc-audio soc-audio: probe iMX-mc13783-audio dai link 0 device: 'MC13783 Playback': device_add soc-audio soc-audio: registered pcm #0 Playback mc13783-hifi-0 asoc: mc13783-hifi <-> imx-ssi.0 mapping ok asoc: sound card iMX-mc13783-audio instantiated soc-audio soc-audio: probe iMX-mc13783-audio dai link 1 device: 'MC13783 Capture': device_add soc-audio soc-audio: registered pcm #1 Capture mc13783-hifi-1 asoc: mc13783-hifi <-> imx-ssi.0 mapping ok asoc: sound card iMX-mc13783-audio instantiated device: 'card0': device_add device: 'pcmC0D1p': device_add device: 'pcmC0D1c': device_add device: 'adsp': device_add device: 'pcmC0D0p': device_add device: 'pcmC0D0c': device_add device: 'dsp': device_add device: 'audio': device_add device: 'controlC0': device_add device: 'mixer': device_add asoc: sound card iMX-mc13783-audio registered [end] Regards, Jürgen P.S.: to enable pr_debug and dev_dbg I had to add them myself in soc-core.c (see my previous mail) -- Jürgen Lambrecht R&D Associate Tel: +32 (0)51 303045 Fax: +32 (0)51 310670 http://www.televic-rail.com Televic Rail NV - Leo Bekaertlaan 1 - 8870 Izegem - Belgium Company number 0825.539.581 - RPR Kortrijk

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[alsa-devel] [PATCH] sound: codec: tlv320aic3x: remove mono controls
by Jan Weitzel 17 Jun '11

17 Jun '11

if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite registers for class-d amplifier. classd amplifier controls are only used for AIC3X_MODEL_3007. Removing all mono snd_kcontrol_new, snd_soc_dapm_widget, snd_soc_dapm_route and aic3x_init stuff from common code and call only for not AIC3X_MODEL_3007 codecs. Testet only with AIC3X_MODEL_3007 Signed-off-by: Jan Weitzel <j.weitzel(a)phytec.de> --- sound/soc/codecs/tlv320aic3x.c | 106 ++++++++++++++++++++++++--------------- 1 files changed, 65 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c3d96fc..7146832 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -324,16 +324,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", - LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", - PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", - DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, - 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), @@ -357,7 +347,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output pin mute controls */ SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, 0x01, 0), - SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, @@ -377,6 +366,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; +static const struct snd_kcontrol_new aic3x_mono_controls[] = { + SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", + LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume", + PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", + DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), +}; + /* * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps */ @@ -526,9 +528,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0), - /* Mono Output */ - SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), - /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, @@ -590,9 +589,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, &aic3x_right_line_mixer_controls[0], ARRAY_SIZE(aic3x_right_line_mixer_controls)), - SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_mono_mixer_controls[0], - ARRAY_SIZE(aic3x_mono_mixer_controls)), SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_hp_mixer_controls[0], ARRAY_SIZE(aic3x_left_hp_mixer_controls)), @@ -608,7 +604,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LLOUT"), SND_SOC_DAPM_OUTPUT("RLOUT"), - SND_SOC_DAPM_OUTPUT("MONO_LOUT"), SND_SOC_DAPM_OUTPUT("HPLOUT"), SND_SOC_DAPM_OUTPUT("HPROUT"), SND_SOC_DAPM_OUTPUT("HPLCOM"), @@ -630,6 +625,18 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Detection"), }; +static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = { + /* Mono Output */ + SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_mono_mixer_controls[0], + ARRAY_SIZE(aic3x_mono_mixer_controls)), + + SND_SOC_DAPM_OUTPUT("MONO_LOUT"), +}; + static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = { /* Class-D outputs */ SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0), @@ -714,17 +721,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line Out", NULL, "Right DAC Mux"}, {"RLOUT", NULL, "Right Line Out"}, - /* Mono Output */ - {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, - {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, - {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, - {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, - {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, - - {"Mono Out", NULL, "Mono Mixer"}, - {"MONO_LOUT", NULL, "Mono Out"}, - /* Left HP Output */ {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, @@ -780,6 +776,19 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPRCOM", NULL, "Right HP Com"}, }; +static const struct snd_soc_dapm_route intercon_mono[] = { + /* Mono Output */ + {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, + {"Mono Mixer", "DACL1 Switch", "Left DAC Mux"}, + {"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + {"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, + {"Mono Mixer", "DACR1 Switch", "Right DAC Mux"}, + + {"Mono Out", NULL, "Mono Mixer"}, + {"MONO_LOUT", NULL, "Mono Out"}, +}; + static const struct snd_soc_dapm_route intercon_3007[] = { /* Class-D outputs */ {"Left Class-D Out", NULL, "Left Line Out"}, @@ -804,6 +813,11 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) ARRAY_SIZE(aic3007_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_3007, ARRAY_SIZE(intercon_3007)); + } else { + snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets, + ARRAY_SIZE(aic3x_dapm_mono_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_mono, + ARRAY_SIZE(intercon_mono)); } return 0; @@ -1290,17 +1304,12 @@ static int aic3x_init(struct snd_soc_codec *codec) /* DAC to Line Out default volume and route to Output mixer */ snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - /* DAC to Mono Line Out default volume and route to Output mixer */ - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ reg = snd_soc_read(codec, LLOPM_CTRL); snd_soc_write(codec, LLOPM_CTRL, reg | UNMUTE); reg = snd_soc_read(codec, RLOPM_CTRL); snd_soc_write(codec, RLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, MONOLOPM_CTRL); - snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); reg = snd_soc_read(codec, HPLOUT_CTRL); snd_soc_write(codec, HPLOUT_CTRL, reg | UNMUTE); reg = snd_soc_read(codec, HPROUT_CTRL); @@ -1325,9 +1334,6 @@ static int aic3x_init(struct snd_soc_codec *codec) /* PGA to Line Out default volume, disconnect from Output Mixer */ snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); @@ -1337,14 +1343,30 @@ static int aic3x_init(struct snd_soc_codec *codec) /* Line2 Line Out default volume, disconnect from Output Mixer */ snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to Mono Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); if (aic3x->model == AIC3X_MODEL_3007) { aic3x_init_3007(codec); snd_soc_write(codec, CLASSD_CTRL, 0); + + return 0; } + /* Not on AIC3X_MODEL_3007 init */ + + /* DAC to Mono Line Out default volume and route to Output mixer */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + + /* unmute outputs */ + reg = snd_soc_read(codec, MONOLOPM_CTRL); + snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); + + /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); + + /* Line2 to Mono Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); return 0; } @@ -1423,7 +1445,9 @@ static int aic3x_probe(struct snd_soc_codec *codec) ARRAY_SIZE(aic3x_snd_controls)); if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); - + else + snd_soc_add_controls(codec, aic3x_mono_controls, + ARRAY_SIZE(aic3x_mono_controls)); aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); -- 1.7.0.4

3 4

[alsa-devel] [PATCH v5 00/18] MFD/ASoC/Input: TWL4030/TWL60X0 changes
by Peter Ujfalusi 17 Jun '11

17 Jun '11

Hello, Changes since v4: - Comments from Dmitry Torokhov addressed for the twl6030-vibra driver. Intro message from the v4 series: The series will do five major things, and they are in one series, because they pretty much depending on each other, so it is easier to handle them together. 1. Move the common pmic related configurations out from the board files. New twl-common.c/h file has been created for this purpose. OMAP3 board files received two patches (pdata, and regualtor changes), OMAP4 board files has been changed with one patch. 2. Rename the twl4030-codec MFD driver to twl4030-audio Since the ASoC multicomponent introduction (2.6.37) the twl4030-codec MFD dirver had been using twl4030-audio to register (twl4030-codec is the ASoC codec driver). The documentation refers to this part of twl as audio block, so it is better to rename the driver as well to avoid confusion. 3. Introduction of MFD driver for twl6040. The TWL6040 audio IC has codec and vibra functionality. Convert the existing ASoC codec driver to use the new MFD driver's interface. 4. Vibrator driver for TWL6040 (Input/ForceFeedback driver) The driver itself, and support for the vibrators in SDP4430 board. 5. Updates for twl6040 ASoC codec driver HS ramp configuration, new ABE dais, updated sample rate constraints, SDP4430 configuration. Regards, Peter --- Axel Castaneda Gonzalez (1): ASoC: twl6040: Configure ramp step based on platform Liam Girdwood (3): ASoC: twl6040: add all ABE DAIs ASoC: twl6040: Support other sample rates in constraints. ASoC: twl6040: set default constraints. Misael Lopez Cruz (5): mfd: twl6040: Add initial support ASoC: twl6040: Convert into TWL6040 MFD child input: Add initial support for TWL6040 vibrator ASoC: twl6040: Remove pll and headset mode dependency OMAP4: SDP4430: Add twl6040 codec platform support Peter Ujfalusi (9): OMAP: New twl-common for common TWL configuration OMAP4: Move common twl6030 configuration to twl-common OMAP3: Move common twl configuration to twl-common OMAP3: Move common regulator configuration to twl-common MFD: twl4030-codec: Rename internals from codec to audio MFD: twl4030-codec -> twl4030-audio: Rename the driver MFD: twl4030-audio: Rename platform data MFD: twl6040: Change platform data for soc codec driver OMAP4: SDP4430: Add twl6040 vibrator platform support arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/board-3430sdp.c | 89 +--- arch/arm/mach-omap2/board-4430sdp.c | 187 ++---- arch/arm/mach-omap2/board-cm-t35.c | 48 +-- arch/arm/mach-omap2/board-devkit8000.c | 37 +- arch/arm/mach-omap2/board-igep0020.c | 44 +-- arch/arm/mach-omap2/board-ldp.c | 15 +- arch/arm/mach-omap2/board-omap3beagle.c | 60 +-- arch/arm/mach-omap2/board-omap3evm.c | 67 +-- arch/arm/mach-omap2/board-omap3pandora.c | 59 +-- arch/arm/mach-omap2/board-omap3stalker.c | 70 +-- arch/arm/mach-omap2/board-omap3touchbook.c | 57 +-- arch/arm/mach-omap2/board-omap4panda.c | 149 +----- arch/arm/mach-omap2/board-overo.c | 59 +-- arch/arm/mach-omap2/board-rm680.c | 8 +- arch/arm/mach-omap2/board-rx51-peripherals.c | 44 +-- arch/arm/mach-omap2/board-zoom-peripherals.c | 88 +--- arch/arm/mach-omap2/common-board-devices.c | 21 - arch/arm/mach-omap2/common-board-devices.h | 26 +- arch/arm/mach-omap2/twl-common.c | 304 ++++++++++ arch/arm/mach-omap2/twl-common.h | 59 ++ arch/arm/plat-omap/include/plat/irqs.h | 12 +- drivers/input/misc/Kconfig | 13 +- drivers/input/misc/Makefile | 1 + drivers/input/misc/twl4030-vibra.c | 12 +- drivers/input/misc/twl6040-vibra.c | 425 +++++++++++++ drivers/mfd/Kconfig | 8 +- drivers/mfd/Makefile | 3 +- drivers/mfd/twl-core.c | 13 +- drivers/mfd/twl4030-audio.c | 277 +++++++++ drivers/mfd/twl4030-codec.c | 277 --------- drivers/mfd/twl6040-core.c | 601 +++++++++++++++++++ drivers/mfd/twl6040-irq.c | 205 +++++++ include/linux/i2c/twl.h | 25 +- .../linux/mfd/{twl4030-codec.h => twl4030-audio.h} | 16 +- include/linux/mfd/twl6040.h | 261 ++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl4030.c | 22 +- sound/soc/codecs/twl6040.c | 630 ++++++++------------ sound/soc/codecs/twl6040.h | 118 ---- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/sdp4430.c | 2 + sound/soc/omap/zoom2.c | 2 +- 43 files changed, 2648 insertions(+), 1773 deletions(-) create mode 100644 arch/arm/mach-omap2/twl-common.c create mode 100644 arch/arm/mach-omap2/twl-common.h create mode 100644 drivers/input/misc/twl6040-vibra.c create mode 100644 drivers/mfd/twl4030-audio.c delete mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 drivers/mfd/twl6040-core.c create mode 100644 drivers/mfd/twl6040-irq.c rename include/linux/mfd/{twl4030-codec.h => twl4030-audio.h} (96%) create mode 100644 include/linux/mfd/twl6040.h -- 1.7.5.3

2 25

[alsa-devel] Can I get audio via HDMI with radeon mobility HD 5400?
by Tom Horsley 17 Jun '11

17 Jun '11

If I install the binary ATI catalyst driver, I can get audio out the HDMI connection, and alsa-info says it is using the snd-hda-intel driver. If I use the open source radeon driver, I cannot get audio out the HDMI connection, yet alsa-info says it is using the same snd-hda-intel driver. Does anyone know how to persuade snd-hda-intel to work with the open source radeon driver? (Because the video quality is about 1000% better with the open source driver). With binary catalyst driver, I see this from alsa-info: http://home.comcast.net/~tomhorsley/hardware/zino/catalyst-alsa-info.txt With open source radeon driver, I see this: http://home.comcast.net/~tomhorsley/hardware/zino/radeon-alsa-info.txt

1 0

[alsa-devel] unsigned int 'return -1' in /sound/soc/codecs/tlv320aic23.c
by Lambrecht Jürgen 17 Jun '11

17 Jun '11

Hello, I see this error in /sound/soc/codecs/tlv320aic23.c: static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec ^^^^^^^^ *codec, unsigned int reg) { u16 *cache = codec->reg_cache; if (reg >= ARRAY_SIZE(tlv320aic23_reg)) return -1; ^^ return cache[reg]; } Maybe better return 0? Best read the codec datasheet to find a safe or non-existing value? And also a "printk(KERN_ERR " then? Actually, better just remove the check, because if you want to write a codec register, you must know what you are doing? I am busy with the codec driver for mc13783. Currently, the mc13783 (from Sascha Hauer) code does not check 'reg'. (I want to supply a patch finally, but as newbie: from which GIT?, because mine is a merge of 2 branches of the Pengutronix GIT) Regards, Jürgen -- Jürgen Lambrecht R&D Associate Tel: +32 (0)51 303045 Fax: +32 (0)51 310670 http://www.televic-rail.com Televic Rail NV - Leo Bekaertlaan 1 - 8870 Izegem - Belgium Company number 0825.539.581 - RPR Kortrijk

1 0

[alsa-devel] [PATCH v4 00/18] MFD/ASoC/Input: TWL4030/TWL60X0 changes
by Peter Ujfalusi 17 Jun '11

17 Jun '11

Hello, Changes since v3: - patches generated with git format-patch -M to handle renames better - Included depending updates for twl6040 soc codec driver: - HS ramp configuration, new ABE dais, updated sample rate constraints - Configure the twl6040 codec on SDP4430 board Updated intro message from the v3 series: The series will do five major things, and they are in one series, because they pretty much depending on each other, so it is easier to handle them together. 1. Move the common pmic related configurations out from the board files. New twl-common.c/h file has been created for this purpose. OMAP3 board files received two patches (pdata, and regualtor changes), OMAP4 board files has been changed with one patch. 2. Rename the twl4030-codec MFD driver to twl4030-audio Since the ASoC multicomponent introduction (2.6.37) the twl4030-codec MFD dirver had been using twl4030-audio to register (twl4030-codec is the ASoC codec driver). The documentation refers to this part of twl as audio block, so it is better to rename the driver as well to avoid confusion. 3. Introduction of MFD driver for twl6040. The TWL6040 audio IC has codec and vibra functionality. Convert the existing ASoC codec driver to use the new MFD driver's interface. 4. Vibrator driver for TWL6040 (Input/ForceFeedback driver) The driver itself, and support for the vibrators in SDP4430 board. 5. Updates for twl6040 ASoC codec driver HS ramp configuration, new ABE dais, updated sample rate constraints, SDP4430 configuration. Regards, Peter --- Axel Castaneda Gonzalez (1): ASoC: twl6040: Configure ramp step based on platform Liam Girdwood (3): ASoC: twl6040: add all ABE DAIs ASoC: twl6040: Support other sample rates in constraints. ASoC: twl6040: set default constraints. Misael Lopez Cruz (5): mfd: twl6040: Add initial support ASoC: twl6040: Convert into TWL6040 MFD child input: Add initial support for TWL6040 vibrator ASoC: twl6040: Remove pll and headset mode dependency OMAP4: SDP4430: Add twl6040 codec platform support Peter Ujfalusi (9): OMAP: New twl-common for common TWL configuration OMAP4: Move common twl6030 configuration to twl-common OMAP3: Move common twl configuration to twl-common OMAP3: Move common regulator configuration to twl-common MFD: twl4030-codec: Rename internals from codec to audio MFD: twl4030-codec -> twl4030-audio: Rename the driver MFD: twl4030-audio: Rename platform data MFD: twl6040: Change platform data for soc codec driver OMAP4: SDP4430: Add twl6040 vibrator platform support arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/board-3430sdp.c | 89 +--- arch/arm/mach-omap2/board-4430sdp.c | 187 ++---- arch/arm/mach-omap2/board-cm-t35.c | 48 +-- arch/arm/mach-omap2/board-devkit8000.c | 37 +- arch/arm/mach-omap2/board-igep0020.c | 44 +-- arch/arm/mach-omap2/board-ldp.c | 15 +- arch/arm/mach-omap2/board-omap3beagle.c | 60 +-- arch/arm/mach-omap2/board-omap3evm.c | 67 +-- arch/arm/mach-omap2/board-omap3pandora.c | 59 +-- arch/arm/mach-omap2/board-omap3stalker.c | 70 +-- arch/arm/mach-omap2/board-omap3touchbook.c | 57 +-- arch/arm/mach-omap2/board-omap4panda.c | 149 +----- arch/arm/mach-omap2/board-overo.c | 59 +-- arch/arm/mach-omap2/board-rm680.c | 8 +- arch/arm/mach-omap2/board-rx51-peripherals.c | 44 +-- arch/arm/mach-omap2/board-zoom-peripherals.c | 88 +--- arch/arm/mach-omap2/common-board-devices.c | 21 - arch/arm/mach-omap2/common-board-devices.h | 26 +- arch/arm/mach-omap2/twl-common.c | 304 ++++++++++ arch/arm/mach-omap2/twl-common.h | 59 ++ arch/arm/plat-omap/include/plat/irqs.h | 12 +- drivers/input/misc/Kconfig | 13 +- drivers/input/misc/Makefile | 1 + drivers/input/misc/twl4030-vibra.c | 12 +- drivers/input/misc/twl6040-vibra.c | 428 +++++++++++++ drivers/mfd/Kconfig | 8 +- drivers/mfd/Makefile | 3 +- drivers/mfd/twl-core.c | 13 +- drivers/mfd/twl4030-audio.c | 277 +++++++++ drivers/mfd/twl4030-codec.c | 277 --------- drivers/mfd/twl6040-core.c | 601 +++++++++++++++++++ drivers/mfd/twl6040-irq.c | 205 +++++++ include/linux/i2c/twl.h | 25 +- .../linux/mfd/{twl4030-codec.h => twl4030-audio.h} | 16 +- include/linux/mfd/twl6040.h | 261 ++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl4030.c | 22 +- sound/soc/codecs/twl6040.c | 630 ++++++++------------ sound/soc/codecs/twl6040.h | 118 ---- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/sdp4430.c | 2 + sound/soc/omap/zoom2.c | 2 +- 43 files changed, 2651 insertions(+), 1773 deletions(-) create mode 100644 arch/arm/mach-omap2/twl-common.c create mode 100644 arch/arm/mach-omap2/twl-common.h create mode 100644 drivers/input/misc/twl6040-vibra.c create mode 100644 drivers/mfd/twl4030-audio.c delete mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 drivers/mfd/twl6040-core.c create mode 100644 drivers/mfd/twl6040-irq.c rename include/linux/mfd/{twl4030-codec.h => twl4030-audio.h} (96%) create mode 100644 include/linux/mfd/twl6040.h -- 1.7.5.3

4 32

[alsa-devel] WM8731 set sample rate
by MIroslav Dušek 17 Jun '11

17 Jun '11

Hi all, I am using processor S3C2416 as master and WM8731 as slave and I am trying to set right sample rate of ADC DAC data between them. I set I2S clock to EPLL output (96MHz) , and I mainly need calculate 8KHz sample rate. In user manual of S3C2416 is poor explanation of clocking I2S bus. What is BCLKMaster and where can I set 1/M ratio and (Figure 23-2 in UM rev 1.3 S3C2416) ?? What exactly means values in registers (BFS, RFS) ??Is Re-scaler set by IIS_PSR really 1 / N rescaller in Figure 23-2 in UM rev 1.3 S3C2416. Thanks for any help. Mirek

1 0

[alsa-devel] [PATCH V2 0/2] Add WM8994 PCM Machine driver for Exynos4210
by Sangbeom Kim 17 Jun '11

17 Jun '11

Hi, This is 2nd patchset for WM8994 pcm machine driver that supports PCM audio on SMDKV310, SMDKC210 and test is done. This is minor change version. AS jassi comment, I added new pcm structure for quirk handling. And added Liam's acked. Based on these patches WM8994 pcm machine driver supports followings: o Playback supports 8kHz sampling rates. o Capture supports 8kHz sampling rates. This patchset contains followings o To Kukjin Kim and Ben Dooks, [PATCH 1/2] ARM: EXYNOS4: Add PCM audio support for WM8994 o To Jassi Brar, Mark Brown and Liam Girdwood, [PATCH 2/2] ASoC: SAMSUNG: Add WM8580 PCM Machine driver Best Regards, SB Kim (Sangbeom Kim)

5 10

[alsa-devel] [PATCH 1/3 v2] ASoC: Add ADAU1701 codec driver
by Lars-Peter Clausen 17 Jun '11

17 Jun '11

This patch adds support for the Analog Devices ADAU1701 SigmaDSP. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: Move the error message for unsupported register addresses to adau1701_register_size. --- sound/soc/codecs/Kconfig | 7 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adau1701.c | 548 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adau1701.h | 17 ++ 4 files changed, 573 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/adau1701.c create mode 100644 sound/soc/codecs/adau1701.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..f745e55 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -130,7 +131,11 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate - + +config SND_SOC_ADAU1701 + select SIGMA + tristate + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..30a4c63 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-adau1701-objs := adau1701.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -95,6 +96,7 @@ obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c new file mode 100644 index 0000000..6c01bb6 --- /dev/null +++ b/sound/soc/codecs/adau1701.c @@ -0,0 +1,548 @@ +/* + * Driver for ADAU1701 SigmaDSP processor + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * based on an inital version by Cliff Cai <cliff.cai(a)analog.com> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/sigma.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau1701.h" + +#define ADAU1701_DSPCTRL 0x1c +#define ADAU1701_SEROCTL 0x1e +#define ADAU1701_SERICTL 0x1f + +#define ADAU1701_AUXNPOW 0x22 + +#define ADAU1701_OSCIPOW 0x26 +#define ADAU1701_DACSET 0x27 + +#define ADAU1701_NUM_REGS 0x28 + +#define ADAU1701_DSPCTRL_CR (1 << 2) +#define ADAU1701_DSPCTRL_DAM (1 << 3) +#define ADAU1701_DSPCTRL_ADM (1 << 4) +#define ADAU1701_DSPCTRL_SR_48 0x00 +#define ADAU1701_DSPCTRL_SR_96 0x01 +#define ADAU1701_DSPCTRL_SR_192 0x02 +#define ADAU1701_DSPCTRL_SR_MASK 0x03 + +#define ADAU1701_SEROCTL_INV_LRCLK 0x2000 +#define ADAU1701_SEROCTL_INV_BCLK 0x1000 +#define ADAU1701_SEROCTL_MASTER 0x0800 + +#define ADAU1701_SEROCTL_OBF16 0x0000 +#define ADAU1701_SEROCTL_OBF8 0x0200 +#define ADAU1701_SEROCTL_OBF4 0x0400 +#define ADAU1701_SEROCTL_OBF2 0x0600 +#define ADAU1701_SEROCTL_OBF_MASK 0x0600 + +#define ADAU1701_SEROCTL_OLF1024 0x0000 +#define ADAU1701_SEROCTL_OLF512 0x0080 +#define ADAU1701_SEROCTL_OLF256 0x0100 +#define ADAU1701_SEROCTL_OLF_MASK 0x0180 + +#define ADAU1701_SEROCTL_MSB_DEALY1 0x0000 +#define ADAU1701_SEROCTL_MSB_DEALY0 0x0004 +#define ADAU1701_SEROCTL_MSB_DEALY8 0x0008 +#define ADAU1701_SEROCTL_MSB_DEALY12 0x000c +#define ADAU1701_SEROCTL_MSB_DEALY16 0x0010 +#define ADAU1701_SEROCTL_MSB_DEALY_MASK 0x001c + +#define ADAU1701_SEROCTL_WORD_LEN_24 0x0000 +#define ADAU1701_SEROCTL_WORD_LEN_20 0x0001 +#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010 +#define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003 + +#define ADAU1701_AUXNPOW_VBPD 0x40 +#define ADAU1701_AUXNPOW_VRPD 0x20 + +#define ADAU1701_SERICTL_I2S 0 +#define ADAU1701_SERICTL_LEFTJ 1 +#define ADAU1701_SERICTL_TDM 2 +#define ADAU1701_SERICTL_RIGHTJ_24 3 +#define ADAU1701_SERICTL_RIGHTJ_20 4 +#define ADAU1701_SERICTL_RIGHTJ_18 5 +#define ADAU1701_SERICTL_RIGHTJ_16 6 +#define ADAU1701_SERICTL_MODE_MASK 7 +#define ADAU1701_SERICTL_INV_BCLK BIT(3) +#define ADAU1701_SERICTL_INV_LRCLK BIT(4) + +#define ADAU1701_OSCIPOW_OPD 0x04 +#define ADAU1701_DACSET_DACINIT 1 + +#define ADAU1701_FIRMWARE "adau1701.bin" + +struct adau1701 { + unsigned int dai_fmt; +}; + +static const struct snd_kcontrol_new adau1701_controls[] = { + SOC_SINGLE("Master Capture Switch", ADAU1701_DSPCTRL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau1701_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC0", "Playback", ADAU1701_AUXNPOW, 3, 1), + SND_SOC_DAPM_DAC("DAC1", "Playback", ADAU1701_AUXNPOW, 2, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", ADAU1701_AUXNPOW, 1, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", ADAU1701_AUXNPOW, 0, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", ADAU1701_AUXNPOW, 7, 1), + + SND_SOC_DAPM_OUTPUT("OUT0"), + SND_SOC_DAPM_OUTPUT("OUT1"), + SND_SOC_DAPM_OUTPUT("OUT2"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_INPUT("IN0"), + SND_SOC_DAPM_INPUT("IN1"), +}; + +static const struct snd_soc_dapm_route adau1701_dapm_routes[] = { + { "OUT0", NULL, "DAC0" }, + { "OUT1", NULL, "DAC1" }, + { "OUT2", NULL, "DAC2" }, + { "OUT3", NULL, "DAC3" }, + + { "ADC", NULL, "IN0" }, + { "ADC", NULL, "IN1" }, +}; + +static unsigned int adau1701_register_size(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case ADAU1701_DSPCTRL: + case ADAU1701_SEROCTL: + case ADAU1701_AUXNPOW: + case ADAU1701_OSCIPOW: + case ADAU1701_DACSET: + return 2; + case ADAU1701_SERICTL: + return 1; + } + + dev_err(codec->dev, "Unsupported register address: %d\n", reg); + return 0; +} + +static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + unsigned int i, ret; + unsigned int size; + uint8_t buf[4]; + + size = adau1701_register_size(codec, reg); + if (size == 0) + return -EINVAL; + + snd_soc_cache_write(codec, reg, value); + + buf[0] = 0x08; + buf[1] = reg; + + for (i = size + 1; i >= 2; --i) { + buf[i] = value; + value >>= 8; + } + + ret = i2c_master_send(to_i2c_client(codec->dev), buf, size + 2); + if (ret == size + 2) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int value; + unsigned int ret; + + ret = snd_soc_cache_read(codec, reg, &value); + if (ret) + return ret; + + return value; +} + +static int adau1701_load_firmware(struct snd_soc_codec *codec) +{ + return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE); +} + +static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK; + unsigned int val; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU1701_SEROCTL_WORD_LEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAU1701_SEROCTL_WORD_LEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU1701_SEROCTL_WORD_LEN_24; + break; + default: + return -EINVAL; + } + + if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) { + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val |= ADAU1701_SEROCTL_MSB_DEALY16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= ADAU1701_SEROCTL_MSB_DEALY12; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= ADAU1701_SEROCTL_MSB_DEALY8; + break; + } + mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; + } + + snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + + return 0; +} + +static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU1701_SERICTL_RIGHTJ_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAU1701_SERICTL_RIGHTJ_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU1701_SERICTL_RIGHTJ_24; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_SERICTL, + ADAU1701_SERICTL_MODE_MASK, val); + + return 0; +} + +static int adau1701_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_codec *codec = rtd->codec; + snd_pcm_format_t format; + unsigned int val; + + switch (params_rate(params)) { + case 192000: + val = ADAU1701_DSPCTRL_SR_192; + break; + case 96000: + val = ADAU1701_DSPCTRL_SR_96; + break; + case 48000: + val = ADAU1701_DSPCTRL_SR_48; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + ADAU1701_DSPCTRL_SR_MASK, val); + + format = params_format(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return adau1701_set_playback_pcm_format(codec, format); + else + return adau1701_set_capture_pcm_format(codec, format); +} + +static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int serictl = 0x00, seroctl = 0x00; + bool invert_lrclk; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* master, 64-bits per sample, 1 frame per sample */ + seroctl |= ADAU1701_SEROCTL_MASTER | ADAU1701_SEROCTL_OBF16 + | ADAU1701_SEROCTL_OLF1024; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_lrclk = true; + break; + case SND_SOC_DAIFMT_IB_NF: + invert_lrclk = false; + serictl |= ADAU1701_SERICTL_INV_BCLK; + seroctl |= ADAU1701_SEROCTL_INV_BCLK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert_lrclk = true; + serictl |= ADAU1701_SERICTL_INV_BCLK; + seroctl |= ADAU1701_SEROCTL_INV_BCLK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + serictl |= ADAU1701_SERICTL_LEFTJ; + seroctl |= ADAU1701_SEROCTL_MSB_DEALY0; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serictl |= ADAU1701_SERICTL_RIGHTJ_24; + seroctl |= ADAU1701_SEROCTL_MSB_DEALY8; + invert_lrclk = !invert_lrclk; + break; + default: + return -EINVAL; + } + + if (invert_lrclk) { + seroctl |= ADAU1701_SEROCTL_INV_LRCLK; + serictl |= ADAU1701_SERICTL_INV_LRCLK; + } + + adau1701->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + snd_soc_write(codec, ADAU1701_SERICTL, serictl); + snd_soc_update_bits(codec, ADAU1701_SEROCTL, + ~ADAU1701_SEROCTL_WORD_LEN_MASK, seroctl); + + return 0; +} + +static int adau1701_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* Enable VREF and VREF buffer */ + snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + break; + case SND_SOC_BIAS_OFF: + /* Disable VREF and VREF buffer */ + snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int mask = ADAU1701_DSPCTRL_DAM; + unsigned int val; + + if (mute) + val = 0; + else + val = mask; + + snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + + return 0; +} + +static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir) +{ + unsigned int val; + + switch (clk_id) { + case ADAU1701_CLK_SRC_OSC: + val = 0x0; + break; + case ADAU1701_CLK_SRC_MCLK: + val = ADAU1701_OSCIPOW_OPD; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + + return 0; +} + +#define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) + +#define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops adau1701_dai_ops = { + .set_fmt = adau1701_set_dai_fmt, + .hw_params = adau1701_hw_params, + .digital_mute = adau1701_digital_mute, +}; + +static struct snd_soc_dai_driver adau1701_dai = { + .name = "adau1701", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = ADAU1701_RATES, + .formats = ADAU1701_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 8, + .rates = ADAU1701_RATES, + .formats = ADAU1701_FORMATS, + }, + .ops = &adau1701_dai_ops, + .symmetric_rates = 1, +}; + +static int adau1701_probe(struct snd_soc_codec *codec) +{ + int ret; + + codec->dapm.idle_bias_off = 1; + + ret = adau1701_load_firmware(codec); + if (ret) + dev_warn(codec->dev, "Failed to load firmware\n"); + + snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); + snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); + + return 0; +} + +static struct snd_soc_codec_driver adau1701_codec_drv = { + .probe = adau1701_probe, + .set_bias_level = adau1701_set_bias_level, + + .reg_cache_size = ADAU1701_NUM_REGS, + .reg_word_size = sizeof(u16), + + .controls = adau1701_controls, + .num_controls = ARRAY_SIZE(adau1701_controls), + .dapm_widgets = adau1701_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1701_dapm_widgets), + .dapm_routes = adau1701_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1701_dapm_routes), + + .write = adau1701_write, + .read = adau1701_read, + + .set_sysclk = adau1701_set_sysclk, +}; + +static __devinit int adau1701_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct adau1701 *adau1701; + int ret; + + adau1701 = kzalloc(sizeof(*adau1701), GFP_KERNEL); + if (!adau1701) + return -ENOMEM; + + i2c_set_clientdata(client, adau1701); + ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, + &adau1701_dai, 1); + if (ret < 0) + kfree(adau1701); + + return ret; +} + +static __devexit int adau1701_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id adau1701_i2c_id[] = { + { "adau1701", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); + +static struct i2c_driver adau1701_i2c_driver = { + .driver = { + .name = "adau1701", + .owner = THIS_MODULE, + }, + .probe = adau1701_i2c_probe, + .remove = __devexit_p(adau1701_i2c_remove), + .id_table = adau1701_i2c_id, +}; + +static int __init adau1701_init(void) +{ + return i2c_add_driver(&adau1701_i2c_driver); +} +module_init(adau1701_init); + +static void __exit adau1701_exit(void) +{ + i2c_del_driver(&adau1701_i2c_driver); +} +module_exit(adau1701_exit); + +MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver"); +MODULE_AUTHOR("Cliff Cai <cliff.cai(a)analog.com>"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1701.h b/sound/soc/codecs/adau1701.h new file mode 100644 index 0000000..8d0949a --- /dev/null +++ b/sound/soc/codecs/adau1701.h @@ -0,0 +1,17 @@ +/* + * header file for ADAU1701 SigmaDSP processor + * + * Copyright 2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _ADAU1701_H +#define _ADAU1701_H + +enum adau1701_clk_src { + ADAU1701_CLK_SRC_OSC, + ADAU1701_CLK_SRC_MCLK, +}; + +#endif -- 1.7.2.5

3 5

[alsa-devel] ALSA Fireworks driver
by Doug McLain 17 Jun '11

17 Jun '11

Hi Clemens, I came across your fireworks driver the other day, and would like to get involved. Has there been any discussion on this anywhere or have you been completely on your own? I did a search of alsa-devel and didn't find anything, suprisingly. It seems like native alsa support of what seems to me to be the first FW device would be the talk of the town. :) I built the driver as a module on F15 and loaded it on my AF12, and found it in /proc/asound/cards, and tried to start jack on it and it failed to start for duplex, capture, or playback, just FYI. Doug

2 2

[alsa-devel] [PATCH] ASoC: Clean up some coding style nits in the bf5xx-i2s-pcm driver
by Mark Brown 17 Jun '11

17 Jun '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index a611483..819d3bd 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -169,7 +169,7 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); - ret = snd_pcm_hw_constraint_integer(runtime, \ + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) goto out; @@ -307,8 +307,8 @@ static int __devexit bfin_i2s_soc_platform_remove(struct platform_device *pdev) static struct platform_driver bfin_i2s_pcm_driver = { .driver = { - .name = "bfin-i2s-pcm-audio", - .owner = THIS_MODULE, + .name = "bfin-i2s-pcm-audio", + .owner = THIS_MODULE, }, .probe = bfin_i2s_soc_platform_probe, -- 1.7.5.4

3 2

[alsa-devel] Codec VIA VT2020/VT1718S and questions about commit 4ab2d53a99b6dcee86837d2a9739bfb9f468db45
by alex dot baldacchino dot alsasub at gmail dot com 17 Jun '11

17 Jun '11

Hello, I've been looking into the code of alsa drivers (mainly hda and patch_via.c), with help by Raymond Yau (many thanks to him) after a problem making my system work with a sound driver newer than version 1.0.21 (at last, I fixed it by recompiling alsa-lib to use device files in /dev as created by udev with correct minor numbers - original subject of that discussion on alsa-devel mailing list is "Via VT2020: issues with kernel 2.6.38.{2, 3} (alsa 1.0.23) - working with 2.6.33.2 (alsa 1.0.21)"). My system is a no-brand desktop, with an asrock motherbord (model 890GX Extreme3) embedding an audio codec VIA VT2020 (beside ATI HDMI). That should support 7.1 speaker systems, plus eventually a headset connected to a front audio panel in multistream mode (since VT1718S seems to support as much as 10 playback channles, as per channels_max field in static const struct hda_pcm_stream vt1718S_pcm_analog_playback, file patch_via.c). Though, my motherboard holds no grey (side) jack (would be nid 0x27), and it would seem that blue (line-in) jack (0x2a) should be retasked as side output (it would use AOW 0x0c, actually used for Independent HP mode, but I've tested, with help by Raymond Yau, and can confirm 0x0b can be used instead by front panel line-out - it's in the connection list for nid 0x34, labeled "Independent HP", which connects to 0x1b and thus to 0x28, "Headphone Playback Switch"). We were trying to fix it (most of the code, about all the code about that, coming from him, I'm just testing/reviewing it and looking into some other, perhaps more general and maybe minor, issues). But now we'd have a concern about a piece of code within struct hda_verb vt1718S_volume_init_verbs[], as modified by you in commit 4ab2d53a99b6dcee86837d2a9739bfb9f468db45, specifically the following part: @@ -4264,7 +4265,7 @@ static struct hda_verb vt1718S_volume_init_verbs[] = { {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, /* Setup default input of Front HP to MW9 */ {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, My system (codec vt2020) has only 5 connections to nid 0x21: Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x1f 0x1f] [0x1f 0x1f] [0x97 0x97] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 and that seems to be the same for other vt1718s codec implementations (by asrock): https://launchpadlibrarian.net/70606587/Card0.Codecs.codec.0.txt (motherboard ASRock H55M Pro, more info at https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/772638 ), therefore your code seems to access a non-existing connection. Since such looks like not very safe/correct, Raymond Yau had suggested to change it into: - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, which seems to work (or at least to cause no problem) on my system. However, after finding your commit, we wonder if that could be necessary for the machine (e.g. because of internal functioning, I'm _not_ asking you to disclose, of course), so I'm writing you to ask for some clarification, as far as possible. Moreover, I have some doubts about the rest of that patch: @@ -4255,7 +4255,8 @@ static struct hda_verb vt1718S_volume_init_verbs[] = { {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Enable MW0 adjust Gain 5 */ + {0x1, 0xfb2, 0x10}, /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback * mixer widget */ In my system, MW0 (first audio mixer widget) seems to be nid 0x18 (0x1 being AFG Function Id) and I haven't been able to find infos about verb 0xfb2; commenting it out seems to cause no problem, however that may look like an implementation-specific verb to be sent to a special connection (as a part of VIA internals I'll never ask you to disclose), so I'll be grateful to you if you can just tell me if that's specifically needed and must be there. Furthermore: /* Setup default input of Front HP to MW9 */ {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, (this part not created by that commit of yours) My system seems to have only 6 Audio Mixer Widgets (0x18, 0x19, 0x1a, 0x1b, 0x1c and 0x21 - unless any Vendor Defined Widget is taken into account), as it seems to be for vt1718S codec as described in links above; in both cases 0x28 has one only connection: vt2020 (mine): Node 0x28 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=21, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x1b vt1718s (linked above): Node 0x28 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=05, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x1b Thus, it would seem (to me, at least) more natural (and correct) to initialize it by selecting connection 0 (the only existing one - or none at all, since the only one should be selected by the hardware by default), unless there's some specific reason for that (e.g. other codec implementations with more than one connection to 0x28, or internal need for that). Could there be the need to create different initialization verb arrays for different flavours of VT1718S? Many thanks in advance for your answers. Best regards.

2 1

Re: [alsa-devel] Driver for TerraTec DMX 6Fire USB
by Takashi Iwai 16 Jun '11

16 Jun '11

At Tue, 18 Jan 2011 14:41:47 +0100, torsten.schenk wrote: > > Hello, > > (to all those who already got this mail: sorry, I sent the attachment accidentally in html format so it was ignored by alsa-devel ML distributor) > > i recently got a TerraTec DMX 6Fire USB. Since it's not supported by alsa, I decided to look into communications between the windows driver and the card. Here is the result. > > What should be revised: > - I was unsure about naming conventions, so please check the name of card/midi/pcm/module. > - Also the device needs firmware to be able to run, so I included the firmware from the windows driver in the attachment. I read about that Linus Torvalds doesn't like .ihex-parsing in drivers, but I cannot convert the files easily into a binary format because the offsets are not contiguous. Of course there could be alternate ways to be thought about, which would imply that original files from TT are not supported since they are in ihex. > - I didn't manage to create a configuration in /usr/share/alsa/cards. > > What is working: Everything except SPDIF > - Hardware Master volume > - PCM 44-192kHz@24 bits, 6 channels out, 4 channels in (analog) > - MIDI in/out > - firmware loading for coldplugged devices > - phono/line switching > > If anything else comes to your attention, just let me know. Sorry for the late reply. It took some time until I have free slot. First off, there are lots of coding-style issues to be fixed. Try to run scripts/checkpatch.pl once with your patch. Not all warnings have to be fixed (e.g. 80-chars) but would be nice if fixed. Here are some review comments through a very quirk glance: > +static int __devinit probe(struct usb_interface *intf, const struct usb_device_id *usb_id) Adding a short prefix (e.g. usb6fire_probe()) is better in general. Otherwise the stack trace would be difficult to read. Of course, too long names are also annoying, so it's a matter of taste. ... > + chip = (struct sfire_chip*)card->private_data; You can avoid cast here. > + chips[regidx] = chip; > + chip->dev = device; > + chip->regidx = regidx; > + chip->intf_count = 1; > + chip->card = card; > + chip->shutdown = FALSE; > + chip->midi = NULL; > + chip->comm = NULL; > + chip->control = NULL; > + chip->pcm = NULL; NULLs are automatically filled, so these can be skipped. > + ret = snd_usb_6fire_comm_init(chip); > + if(ret < 0) > + { > + snd_card_free(card); > + return ret; > + } > + > + ret = snd_usb_6fire_midi_init(chip); > + if(ret < 0) > + { > + snd_usb_6fire_comm_destroy(chip); > + snd_card_free(card); > + return ret; > + } > + > + ret = snd_usb_6fire_pcm_init(chip); > + if(ret < 0) > + { > + snd_usb_6fire_midi_destroy(chip); > + snd_usb_6fire_comm_destroy(chip); > + snd_card_free(card); > + return ret; > + } > + > + ret = snd_usb_6fire_control_init(chip); > + if(ret < 0) > + { > + snd_usb_6fire_pcm_destroy(chip); > + snd_usb_6fire_midi_destroy(chip); > + snd_usb_6fire_comm_destroy(chip); > + snd_card_free(card); > + return ret; > + } > + > + ret = snd_card_register(card); > + if(ret < 0) > + { > + snd_usb_6fire_pcm_destroy(chip); > + snd_usb_6fire_midi_destroy(chip); > + snd_usb_6fire_comm_destroy(chip); > + snd_usb_6fire_control_destroy(chip); > + snd_card_free(card); A general way for error-handling is either using goto or using a common destructor. In either way, you can call the *_free or *_destroy with the non-NULL check of each instance. It's a cleaner in general. > +static void disconnect(struct usb_interface *intf) > +{ > + struct sfire_chip *chip; > + struct snd_card *card; > + > + chip = usb_get_intfdata(intf); > + if(chip) //if !chip, fw upload has been performed so that we don't have a chip. > + { > + card = chip->card; > + chip->intf_count--; > + if (chip->intf_count == 0) > + { > + chip->shutdown = TRUE; > + snd_card_disconnect(card); > + snd_usb_6fire_comm_destroy(chip); > + snd_usb_6fire_control_destroy(chip); > + snd_usb_6fire_pcm_destroy(chip); > + snd_usb_6fire_midi_destroy(chip); > + snd_card_free(card); > + usb_set_intfdata(intf, NULL); The same destructor as in probe can be used here. > +static struct usb_device_id device_table[] = > +{ > + { > + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, > + .idVendor = 0x0ccd, > + .idProduct = 0x0080 > + }, > + { } > +}; Usually MODULE_DEVICE_TABLE() follows just after the definition. > diff -Nur a/sound/usb/6fire/chip.h b/sound/usb/6fire/chip.h > --- a/sound/usb/6fire/chip.h 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/chip.h 2011-01-18 13:58:06.000000000 +0100 > @@ -0,0 +1,30 @@ > +/* > + * Linux driver for TerraTec DMX 6Fire USB > + * > + * Author: Torsten Schenk > + * Created: Jan 01, 2011 > + * Copyright: (C) Torsten Schenk > + * > + * 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. > + */ > +#pragma once Don't use prgram. Also, add #ifdef xxx for protecting the double inclusion. > +#include "common.h" > + > +struct sfire_chip > +{ > + struct usb_device *dev; > + struct snd_card *card; > + int intf_count; //number of registered interfaces > + int regidx; //index in module parameter arrays > + BOOL shutdown; Don't use own bool type. Use the generic "bool" type, and use true/false instead of TRUE/FALSE (unless the size really matters). > diff -Nur a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c > --- a/sound/usb/6fire/comm.c 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/comm.c 2011-01-18 13:58:21.000000000 +0100 ... > +static int write_02(struct comm_runtime *rt, __u8 reg, __u16 value) Use u8 and u16. > diff -Nur a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h > --- a/sound/usb/6fire/comm.h 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/comm.h 2011-01-18 13:58:29.000000000 +0100 > @@ -0,0 +1,42 @@ > +/* > + * Linux driver for TerraTec DMX 6Fire USB > + * > + * Author: Torsten Schenk > + * Created: Jan 01, 2011 > + * Copyright: (C) Torsten Schenk > + * > + * 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. > + */ > + > +#pragma once Ditto. No pragma and add ifdef. > diff -Nur a/sound/usb/6fire/common.h b/sound/usb/6fire/common.h > --- a/sound/usb/6fire/common.h 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/common.h 2011-01-18 13:58:38.000000000 +0100 > +#pragma once > + > +#include <linux/slab.h> > +#include <linux/usb.h> > +#include <sound/core.h> > + > +#define BOOL char > +#define TRUE 1 > +#define FALSE 0 Don't use these. > +#define MIN(X, Y) ((X) < (Y) ? (X) : (Y)) > +#define MAX(X, Y) ((X) > (Y) ? (X) : (Y)) Use standard min() / max() macros. > +#define ERR KERN_ERR "6fire: " > +#define WARNING KERN_WARNING "6fire: " Hmm... These are also better to be avoided. At most, define "6fire: " as PREFIX or PFX or whatever. But KERN_ERR should be put plainly. > diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c > --- a/sound/usb/6fire/control.c 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/control.c 2011-01-18 13:58:57.000000000 +0100 > diff -Nur a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c > --- a/sound/usb/6fire/firmware.c 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/firmware.c 2011-01-18 13:57:15.000000000 +0100 ... > +static inline int fpga_write(struct usb_device *device, char *data, int len) Avoid inline unless you really need it. Also add MODULE_FIRMWARE() definitions in this file. > diff -Nur a/sound/usb/6fire/Makefile b/sound/usb/6fire/Makefile > --- a/sound/usb/6fire/Makefile 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/Makefile 2011-01-16 23:14:22.000000000 +0100 > @@ -0,0 +1,13 @@ > +obj-m += snd-usb-6fire.o > +snd-usb-6fire-objs += chip.o comm.o midi.o control.o firmware.o pcm.o > + > +all: > + make CONFIG_DEBUG_SECTION_MISMATCH=y -C /lib/modules/$(shell uname -r)/build M=$(PWD) modules > + > +clean: > + make -C /lib/modules/$(shell uname -r)/build M=$(PWD) clean > + > +install: > + mkdir -p /lib/modules/$(shell uname -r)/extras > + cp snd-usb-6fire.ko /lib/modules/$(shell uname -r)/extras Remove all, clean, install things. > diff -Nur a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c > --- a/sound/usb/6fire/pcm.c 1970-01-01 01:00:00.000000000 +0100 > +++ b/sound/usb/6fire/pcm.c 2011-01-18 13:59:57.000000000 +0100 > diff -Nur a/sound/usb/Kconfig b/sound/usb/Kconfig > --- a/sound/usb/Kconfig 2011-01-16 23:20:41.000000000 +0100 > +++ b/sound/usb/Kconfig 2011-01-16 23:14:22.000000000 +0100 > @@ -32,6 +32,17 @@ > To compile this driver as a module, choose M here: the module > will be called snd-ua101. > +config SND_USB_6FIRE > + tristate "TerraTec DMX 6Fire USB" > + depends on X86 > + select SND_HWDEP Do you really need hwdep? Rather select FW_LOADER, if any. > + select SND_PCM Missing SND_RAWMIDI dependency. > @@ -65,7 +76,6 @@ > * Native Instruments Guitar Rig Session I/O > * Native Instruments Guitar Rig mobile > * Native Instruments Traktor Kontrol X1 > - * Native Instruments Traktor Kontrol S4 > To compile this driver as a module, choose M here: the module > will be called snd-usb-caiaq. > @@ -83,7 +93,6 @@ > * Native Instruments Kore Controller > * Native Instruments Kore Controller 2 > * Native Instruments Audio Kontrol 1 > - * Native Instruments Traktor Kontrol S4 Don't bother others ;) thanks, Takashi

2 20

[alsa-devel] [PATCH] sound: fix lola section mismatch warnings
by Randy Dunlap 16 Jun '11

16 Jun '11

From: Randy Dunlap <randy.dunlap(a)oracle.com> Fix section mismatch warnings in lola sound driver: WARNING: sound/pci/lola/snd-lola.o(.text+0xda4): Section mismatch in reference from the function lola_parse_tree() to the function .devinit.text:lola_init_pcm() The function lola_parse_tree() references the function __devinit lola_init_pcm(). This is often because lola_parse_tree lacks a __devinit annotation or the annotation of lola_init_pcm is wrong. WARNING: sound/pci/lola/snd-lola.o(.text+0xdf5): Section mismatch in reference from the function lola_parse_tree() to the function .devinit.text:lola_init_pins() The function lola_parse_tree() references the function __devinit lola_init_pins(). This is often because lola_parse_tree lacks a __devinit annotation or the annotation of lola_init_pins is wrong. WARNING: sound/pci/lola/snd-lola.o(.text+0xe52): Section mismatch in reference from the function lola_parse_tree() to the function .devinit.text:lola_init_clock_widget() The function lola_parse_tree() references the function __devinit lola_init_clock_widget(). This is often because lola_parse_tree lacks a __devinit annotation or the annotation of lola_init_clock_widget is wrong. Signed-off-by: Randy Dunlap <randy.dunlap(a)oracle.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/lola/lola.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-30-rc3.orig/sound/pci/lola/lola.c +++ lnx-30-rc3/sound/pci/lola/lola.c @@ -445,7 +445,7 @@ static void lola_reset_setups(struct lol lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */ } -static int lola_parse_tree(struct lola *chip) +static int __devinit lola_parse_tree(struct lola *chip) { unsigned int val; int nid, err;

2 1

Re: [alsa-devel] Via VT2020: issues with kernel 2.6.38.{2, 3} (alsa 1.0.23) - working with 2.6.33.2 (alsa 1.0.21)
by alex dot baldacchino dot alsasub at gmail dot com 16 Jun '11

16 Jun '11

(Sorry for previous message being too long) 2011/6/14 Raymond Yau <superquad.vortex2(a)gmail.com>: - Nascondi testo citato - > 2011/6/8 alex dot baldacchino dot alsasub at gmail dot com > <alex.baldacchino.alsasub(a)gmail.com>: >> >> >> http://pastebin.ca/2076750 >> >> >> By the way, should something like the above be done for capture >> streams as well? I've noticed (with front panel disabled) that >> controls "Front Mic Playback Volume" and "Front Mic Playback Switch" >> are correctly missing from audio-mixer widget 0x21, but node 0x29 is >> still 'attached' to control "Front Mic Boost Capture Volume", which is >> accessible through a mixer (though I didn't try to play with it; it is >> listed also by amixer output in above linked document). Is this the >> wanted/expected behaviour? > > It seem that 0x2b and 0x29 does not have any mic boost capture volume, > that look like "Playback Switch" when they are retasked as output for > 3 jacks motherboard > > Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > Control: name="Front Mic Boost Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000233c: IN OUT HP Detect > Vref caps: HIZ 50 100 > Pin Default 0x02a19037: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x7 > Pin-ctls: 0x21: IN VREF_50 > Unsolicited: tag=20, enabled=1 > Power states: D0 D1 D2 D3 > Power: setting=D3, actual=D3 > Connection: 1 > 0x1c > > Node 0x2b [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > Control: name="Mic Boost Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00002334: IN OUT Detect > Vref caps: HIZ 50 100 > Pin Default 0x01a19036: [Jack] Mic at Ext Rear > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x6 > Pin-ctls: 0x21: IN VREF_50 > Unsolicited: tag=20, enabled=1 > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > Connection: 2 > 0x0a* 0x0c > That pastebin was alsa-info.sh output with front panel disabled in bios, to show via_build_pcms() can be fixed to avoid creating a sencond playback pcm substream (thus, a second subdevice) when there's no front audio panel in the system (or is disabled), even if snd_hda_find_mixer_ctl() cannot find control "Independent HP" if called from within via_build_pcms: APLAY **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT2020 Analog [VT2020 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 But I might have pasted a wrong file (sorry for that), correct output for that (sort of) patch would be: Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Front Mic Boost Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x42a190f7: [N/A] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x7 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=20, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D3, actual=D3 Connection: 1 0x1c Front audio panel not connected (not seen by the BIOS) -> front mic not available -> power setting D3 (because of set_pin_power_state() checking for presence, I guess) Node 0x2b [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Mic Boost Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00002334: IN OUT Detect Vref caps: HIZ 50 100 Pin Default 0x01a19036: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x6 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=20, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D3, actual=D3 Connection: 2 0x0a* 0x0c Rear jack present (on the motherboard), but nothing connected to it -> power state D3 > /* capture mixer elements */ > static const struct snd_kcontrol_new vt1718S_capture_mixer[] = { > HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), > HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), > HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), > - HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), > - HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, > - HDA_INPUT), > > This would eliminate both boost volume controls at once, but in normal conditions (e.g. front audio panel enabled in bios, headset connected to it) Front Mic Boost works (e.g. I can record sound louder and noisier). I just wondered if creating "Front Mic Boost Capture Volume" could have been done dynamically, instead, to match the driver capability (as is) to avoid creating Volume and Switch controls (on mixer 0x21) for Front Mic: with front audio panel missing (disabled in bios): Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x1f 0x1f] [0x80 0x80] [0x80 0x80] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 otherwise: Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x1f 0x1f] [0x1f 0x1f] [0x80 0x80] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 A possible solution: /* capture mixer elements */ static const struct snd_kcontrol_new vt1718S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, - HDA_INPUT), ... + static const struct snd_kcontrol_new vt1718S_front_mic_boost = + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT); + static int vt1718S_add_front_mic_boost( struct via_spec *spec ){ + snd_kcontrol_new *knew; + + if( spec->front_panel_out == HDA_HW_WITH_FRONT_PANEL){ + knew = via_clone_control( spec, &vt1718S_front_mic_boost ); + if( knew == NULL ) + return -ENOMEM; + return 1; + } + + return 0; + } in struct via_spec: ... struct hda_pcm pcm_rec[ VIA_NUM_REC]; + #define HDA_HW_WITHOUT_FRONT_PANEL 0 /* initial value*/ + #define HDA_HW_WITH_FRONT_PANEL 1 + unsigned int front_panel_out; /* used to determine the correct number of playback pcm substreams */ in via_new_spec(): ... codec->spec = spec; spec->codec = codec; spec->codec_type = get_codec_type(codec); + spec->front_panel_out = HDA_HW_WITHOUT_FRONT_PANEL; in via_hp_build(): ... knew = via_clone_control(spec, &via_hp_mixer[0]); if (knew == NULL){ return -ENOMEM; } knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; knew->private_value = nid; + spec->front_panel_out = HDA_HW_WITH_FRONT_PANEL; /* now we know there's a valid secondary playback pcm substream */ in via_build_pcms(): ... info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + /* is this a multistream-capable environment? let's compute real number of playback pcm substreams */ + info->stream[SNDRV_PCM_STREAM_PLAYBACK].substreams = 1 + spec->front_panel_out; in patch_vt1718S(): ... if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1718S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); get_mux_nids(codec); override_mic_boost(codec, 0x2b, 0, 3, 40); override_mic_boost(codec, 0x29, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; spec->num_mixers++; + vt1718S_add_front_mic_boost( spec ); } It worked for me, AFAICT: With front audio panel _enabled_ in bios: Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=3, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x1f 0x1f] [0x1f 0x1f] [0x80 0x80] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Front Mic Boost Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x02a19037: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x7 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=20, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x1c aplay -l output **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT2020 Analog [VT2020 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 [...] and front mic boost control listed in amixer: Simple mixer control 'Front Mic Boost',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 3 Front Left: Capture 0 [0%] [0.00dB] Front Right: Capture 0 [0%] [0.00dB] With front audio panel _disabled_ in bios: Node 0x21 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x1f 0x1f] [0x80 0x80] [0x80 0x80] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 5 0x2c 0x2b 0x2a 0x29 0x28 Node 0x29 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000233c: IN OUT HP Detect Vref caps: HIZ 50 100 Pin Default 0x42a190f7: [N/A] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x7 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=20, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D3, actual=D3 Connection: 1 0x1c aplay -l output: **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT2020 Analog [VT2020 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 [...] and control "Front Mic Boost" missing in amixer output. Attaching alsa-info.sh outputs. Perhaps other codecs could benefit of that, like VT1708S, VT1716S, VT2002P, VT1812, all creating a "Front Mic Boost Capture Volume" control statically (each one needing its own modification to vtxxxxx_capture_mixer, patch_vtxxxxx(), and vt1718S_add_front_mic_boost() function renamed, moved upward in the file before patch_vt1708S() and modified with a further snd_kcontrol_new argument to be used by all others, e.g. called as vt_add_front_mic_boost( spec, &vt1718S_front_mic_boost ): + static int vt_add_front_mic_boost( struct via_spec *spec, const struct snd_kcontrol_new *fmb_ctl ){ + snd_kcontrol_new *knew; + + if( spec->front_panel_out == HDA_HW_WITH_FRONT_PANEL){ + knew = via_clone_control( spec, fmb_ctl ); + if( knew == NULL ) + return -ENOMEM; + return 1; + } + + return 0; + } ). > > The problem is in hda_verb vt1718S_volume_init_verbs > > /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, > {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, > > node 0x21 of your vt2020 has only 5 connections > > Refer to commit 4ab2d53a99b6dcee86837d2a9739bfb9f468db45 > > you will need to ask the author of this patch since the patch > explicitly change this non exisiting connection of node 0x21 on your > vt2020 > > - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, > + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, > > Will do. Perhaps there could be the need to create a separate vt2020_volume_init_verbs, unless that operation is due to internal implementation of the codec (reacting properly to the non-existing connection write query). Thank you for everything. - Nascondi testo citato - > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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Re: [alsa-devel] linux-next: Tree for May 23 (sound/soc/codecs)
by Randy Dunlap 16 Jun '11

16 Jun '11

On Mon, 23 May 2011 15:45:18 +1000 Stephen Rothwell wrote: > Hi all, > > Changes since 20110520: when CONFIG_GPIOLIB is not enabled and CONFIG_GENERIC_GPIO is not enabled: sound/soc/codecs/ak4641.c:524: error: 'GPIOF_OUT_INIT_LOW' undeclared (first use in this function) sound/soc/codecs/wm8915.c:2857: error: 'GPIOF_OUT_INIT_LOW' undeclared (first use in this function) --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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[alsa-devel] [PATCH] alsa-utils: amixer: Fix SND_CTL_TLVT_DB_RANGE handling
by Lars-Peter Clausen 16 Jun '11

16 Jun '11

Current support for handling SND_CTL_TLVT_DB_RANGE in amixer is completely broken. This patch fixes it. Reported-by: Daniel Mack <zonque(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- amixer/amixer.c | 14 +++++++------- 1 files changed, 7 insertions(+), 7 deletions(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index a177288..9d2855d 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -517,20 +517,20 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ #ifdef SND_CTL_TLVT_DB_RANGE case SND_CTL_TLVT_DB_RANGE: printf("dBrange-\n"); - if ((size / (6 * sizeof(unsigned int))) != 0) { + if ((size % (6 * sizeof(unsigned int))) != 0) { while (size > 0) { printf("0x%08x,", tlv[idx++]); size -= sizeof(unsigned int); } break; } - idx = 0; - while (idx < size) { + while (size > 0) { print_spaces(spaces + 2); - printf("rangemin=%i,", tlv[0]); - printf(",rangemax=%i\n", tlv[1]); - decode_tlv(spaces + 4, tlv + 2, 6 * sizeof(unsigned int)); - idx += 6 * sizeof(unsigned int); + printf("rangemin=%i,", tlv[idx++]); + printf(",rangemax=%i\n", tlv[idx++]); + decode_tlv(spaces + 4, tlv + idx, 4 * sizeof(unsigned int)); + idx += 4; + size -= 6 * sizeof(unsigned int); } break; #endif -- 1.7.2.5

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[alsa-devel] [PATCH 0/5] AD1836 updates part two
by Mike Frysinger 16 Jun '11

16 Jun '11

This replaces the patchset I sent out yesterday. It mostly syncs the missing work we had locally with what Lars-Peter did. I guess we can omit the "ad1836" -> "ad183x" rename. Barry Song (1): ASoC: AD1836: rename suspend/resume funcs Mike Frysinger (4): ASoC: AD1836: drop unnecessary spi register check ASoC: AD1836: fix intermixed tab/space indentation ASoC: AD1836: fix codec name ASoC: AD1836: clean up comment headers sound/soc/codecs/ad1836.c | 78 +++++++++++++++++--------------------------- sound/soc/codecs/ad1836.h | 21 +++--------- 2 files changed, 36 insertions(+), 63 deletions(-) -- 1.7.5.3

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Add WM8994 PCM Machine driver
by Sangbeom Kim 16 Jun '11

16 Jun '11

This patch add WM8994 PCM machine driver to support PCM audio on SMDKV310, SMDKC210 boards. Playback and Capture supports 8kHz sampling rates. and It is tested on SMDKV310, SMDKC210. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/Kconfig | 8 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/smdk_wm8994pcm.c | 176 ++++++++++++++++++++++++++++++++++++ 3 files changed, 185 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/smdk_wm8994pcm.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index b8c7a2e..db2bc49 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -170,3 +170,11 @@ config SND_SOC_SMDK_WM8580_PCM select SND_SAMSUNG_PCM help Say Y if you want to add support for SoC audio on the SMDK. + +config SND_SOC_SMDK_WM8994_PCM + tristate "SoC PCM Audio support for WM8994 on SMDK" + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310) + select SND_SOC_WM8994 + select SND_SAMSUNG_PCM + help + Say Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 6c598fc..4a3a428 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -35,6 +35,7 @@ snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o +snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -53,3 +54,4 @@ obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o +obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c new file mode 100644 index 0000000..756c228 --- /dev/null +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -0,0 +1,176 @@ +/* + * sound/soc/samsung/smdk_wm8994pcm.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.samsung.com + * + * 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 <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "../codecs/wm8994.h" +#include "dma.h" +#include "pcm.h" + +/* + * Board Settings: + * o '1' means 'ON' + * o '0' means 'OFF' + * o 'X' means 'Don't care' + * + * SMDKC210, SMDKV310: CFG3- 1001, CFG5-1000, CFG7-111111 + */ + +/* + * Configure audio route as :- + * $ amixer sset 'DAC1' on,on + * $ amixer sset 'Right Headphone Mux' 'DAC' + * $ amixer sset 'Left Headphone Mux' 'DAC' + * $ amixer sset 'DAC1R Mixer AIF1.1' on + * $ amixer sset 'DAC1L Mixer AIF1.1' on + * $ amixer sset 'IN2L' on + * $ amixer sset 'IN2L PGA IN2LN' on + * $ amixer sset 'MIXINL IN2L' on + * $ amixer sset 'AIF1ADC1L Mixer ADC/DMIC' on + * $ amixer sset 'IN2R' on + * $ amixer sset 'IN2R PGA IN2RN' on + * $ amixer sset 'MIXINR IN2R' on + * $ amixer sset 'AIF1ADC1R Mixer ADC/DMIC' on + */ + +/* SMDK has a 16.9344MHZ crystal attached to WM8994 */ +#define SMDK_WM8994_FREQ 16934400 + +static int smdk_wm8994_pcm_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->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned long mclk_freq; + int rfs, ret; + + switch(params_rate(params)) { + case 8000: + rfs = 512; + break; + default: + printk(KERN_ERR "%s:%d Sampling Rate %u not supported!\n", + __func__, __LINE__, params_rate(params)); + return -EINVAL; + } + + mclk_freq = params_rate(params) * rfs; + + /* Set the codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* Set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + SMDK_WM8994_FREQ, mclk_freq); + if (ret < 0) + return ret; + + /* Set PCM source clock on CPU */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set SCLK_DIV for making bclk */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops smdk_wm8994_pcm_ops = { + .hw_params = smdk_wm8994_pcm_hw_params, +}; + +static struct snd_soc_dai_link smdk_dai[] = { + { + .name = "WM8994 PAIF PCM", + .stream_name = "Primary PCM", + .cpu_dai_name = "samsung-pcm.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm8994-codec", + .ops = &smdk_wm8994_pcm_ops, + }, +}; + +static struct snd_soc_card smdk_pcm = { + .name = "SMDK-PCM", + .dai_link = smdk_dai, + .num_links = 1, +}; + +static int __devinit snd_smdk_probe(struct platform_device *pdev) +{ + int ret = 0; + + smdk_pcm.dev = &pdev->dev; + ret = snd_soc_register_card(&smdk_pcm); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); + return ret; + } + + return 0; +} + +static int __devexit snd_smdk_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&smdk_pcm); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_smdk_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "samsung-smdk-pcm", + }, + .probe = snd_smdk_probe, + .remove = __devexit_p(snd_smdk_remove), +}; + +static int __init smdk_audio_init(void) +{ + return platform_driver_register(&snd_smdk_driver); +} + +module_init(smdk_audio_init); + +static void __exit smdk_audio_exit(void) +{ + platform_driver_unregister(&snd_smdk_driver); +} + +module_exit(smdk_audio_exit); + +MODULE_AUTHOR("Sangbeom Kim, <sbkim73(a)samsung.com>"); +MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM"); +MODULE_LICENSE("GPL"); -- 1.7.1

2 1

Re: [alsa-devel] Creative X-Fi Titanium HD ctxfi bugs 5378 5218 4868
by Takashi Iwai 16 Jun '11

16 Jun '11

At Sat, 11 Jun 2011 18:25:36 +0800, Harry Butterworth wrote: > > Hi, > > I have redone the patch against today's git repositories. > Creative have not yet got back to me about the PLL initialization. > Perhaps we can fix that later. > > I have only put the PCI ID in the pci_ids_compat.h.in file. Is it > going to be necessary to add the PCI device ID somewhere in the kernel > tree to avoid breaking the kernel build? Yes, include/linux/pci_ids.h is the place to change. But I can fix it by myself, too. > > > A few comment about the patch: > > > > - Instead of adding have_dedicated_mic and have_output_switch > > callbacks in every place, I'd put some bit flags indicating the > > capabilities, and set at the initialization appropriately. > > I had a go at this in the patch 0002-ALSA... > > > - A few easy coding-style issues; run scripts/checkpatch.pl and fix > > the errors appropriately. > > checkpatch now reports no errors. > > I've only tested these changes with my Titanium HD card. I don't have > any of the other X-Fi cards so I couldn't do any regression testing. > > If there's anything else I need to do, please let me know. I'm going to check patches in a more detail tomorrow. Thanks! Takashi

2 4

[alsa-devel] [PATCH] udev: Allow ALSA input jacks to be accessed by the current user
by David Henningsson 15 Jun '11

15 Jun '11

I know this is probably not the correct upstream, but I just wanted to run it through alsa-devel first to make sure that you don't think this is all madness. :-) So, one missing piece for userspace (PulseAudio etc) to actually be able to use the jack input devices that ALSA create, is that these devices are accessible by root only. This patch makes the input device nodes accessible by the same users that can access the sound card: the current logged in user, as well as users in the audio group. One thing I was thinking about, was that the udev-acl rule actually grants read-write access to the input device node, where probably only read access is needed. Is this dangerous? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

1 0

[PATCH] udev: Allow ALSA input jacks to be accessed by the current user
by David Henningsson 15 Jun '11

15 Jun '11

One missing piece for userspace (PulseAudio etc) to actually be able to use the jack input devices that ALSA create, is that these devices are accessible by root only. This patch makes the input device nodes accessible by the same users that can access the sound card: the current logged in user, as well as users in the audio group. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- extras/input_id/input_id.c | 23 +++++++++++++++++++++++ extras/udev-acl/70-acl.rules | 1 + rules/rules.d/60-persistent-input.rules | 1 + 3 files changed, 25 insertions(+), 0 deletions(-) diff --git a/extras/input_id/input_id.c b/extras/input_id/input_id.c index ba53df0..814ce8e 100644 --- a/extras/input_id/input_id.c +++ b/extras/input_id/input_id.c @@ -189,6 +189,25 @@ static void test_key (struct udev *udev, puts("ID_INPUT_KEYBOARD=1"); } +static void test_sound (struct udev_device *dev, + const unsigned long* bitmask_sw) +{ + + struct udev_device *snd_dev; + + /* Ensure parent is sound card */ + snd_dev = udev_device_get_parent_with_subsystem_devtype(dev, "sound", NULL); + if (!snd_dev) + return; + + /* Check that there is at least one relevant switch type */ + if (test_bit(SW_HEADPHONE_INSERT, bitmask_sw) || + test_bit(SW_MICROPHONE_INSERT, bitmask_sw) || + test_bit(SW_LINEOUT_INSERT, bitmask_sw) || + test_bit(SW_VIDEOOUT_INSERT, bitmask_sw)) + puts("ID_INPUT_SNDJACK=1"); +} + static void help(void) { printf("Usage: input_id [options] <device path>\n" @@ -212,6 +231,7 @@ int main (int argc, char** argv) unsigned long bitmask_abs[NBITS(ABS_MAX)]; unsigned long bitmask_key[NBITS(KEY_MAX)]; unsigned long bitmask_rel[NBITS(REL_MAX)]; + unsigned long bitmask_sw[NBITS(SW_MAX)]; udev = udev_new(); if (udev == NULL) @@ -272,10 +292,13 @@ int main (int argc, char** argv) get_cap_mask (dev, "capabilities/abs", bitmask_abs, sizeof (bitmask_abs)); get_cap_mask (dev, "capabilities/rel", bitmask_rel, sizeof (bitmask_rel)); get_cap_mask (dev, "capabilities/key", bitmask_key, sizeof (bitmask_key)); + get_cap_mask (dev, "capabilities/sw", bitmask_sw, sizeof (bitmask_sw)); test_pointers(bitmask_ev, bitmask_abs, bitmask_key, bitmask_rel); test_key(udev, bitmask_ev, bitmask_key); + test_sound(dev, bitmask_sw); + return 0; } diff --git a/extras/udev-acl/70-acl.rules b/extras/udev-acl/70-acl.rules index 037349d..a6f54d9 100644 --- a/extras/udev-acl/70-acl.rules +++ b/extras/udev-acl/70-acl.rules @@ -25,6 +25,7 @@ SUBSYSTEM=="scsi_generic", SUBSYSTEMS=="scsi", ATTRS{type}=="4|5", TAG+="udev-ac # sound devices SUBSYSTEM=="sound", TAG+="udev-acl" +SUBSYSTEM=="input", ENV{ID_INPUT_SNDJACK}=="1", TAG+="udev-acl" # ffado is an userspace driver for firewire sound cards SUBSYSTEM=="firewire", ENV{ID_FFADO}=="1", TAG+="udev-acl" diff --git a/rules/rules.d/60-persistent-input.rules b/rules/rules.d/60-persistent-input.rules index cd1de4e..d04c09e 100644 --- a/rules/rules.d/60-persistent-input.rules +++ b/rules/rules.d/60-persistent-input.rules @@ -12,6 +12,7 @@ ENV{ID_INPUT_MOUSE}=="?*", ENV{.INPUT_CLASS}="mouse" ENV{ID_INPUT_TOUCHPAD}=="?*", ENV{.INPUT_CLASS}="mouse" ENV{ID_INPUT_TABLET}=="?*", ENV{.INPUT_CLASS}="mouse" ENV{ID_INPUT_JOYSTICK}=="?*", ENV{.INPUT_CLASS}="joystick" +ENV{ID_INPUT_SNDJACK}=="?*", ENV{.INPUT_CLASS}="sndjack", GROUP="audio" DRIVERS=="pcspkr", ENV{.INPUT_CLASS}="spkr" ATTRS{name}=="*dvb*|*DVB*|* IR *", ENV{.INPUT_CLASS}="ir" -- 1.7.4.1 --------------060705090906040605070702--

1 0

[alsa-devel] Need Help About Submitting Updates to Existing ASoC Codec Driver [Re Post]
by Ashish P. Chavan 15 Jun '11

15 Jun '11

[Sorry for my previous post in which outlook messed up line breaks, Please ignore it. This is repost of the same message.] Hi, We are in the process of upgrading codec driver for Dialog's DA7210 codec (sound/soc/codecs/da7210.c). Existing driver in mainline is just bare minimum and doesn't support many of the chip's features. Our update would be a major functionality update in nature. Our final goal is to get the updated driver merged into mainline kernel. For this we just want to make sure that we follow correct process. As far as I understood from alsa-project.org and other resources, the normal process is to submit the patch against alsa-kenel (git.alsa-project.org:alsa-kernel.git alsa-kernel) and get it reviewed by maintainers there. Once all maintainers are satisfied with the patch, it will be eventually put in to for-linus branch and get in to mainline. Is this the correct approach and path to achieve our goal? Is there any obvious misunderstanding or missing step(s) here? Thanks, -- Ashish ---------------------------------------------------------------------------- || Linux Is User Friendly, It's Just Selective About Who It's Friend' Are || ----------------------------------------------------------------------------

2 1

[alsa-devel] [PATCH 1/8] ASoC: AD1836: drop unnecessary spi register check
by Mike Frysinger 15 Jun '11

15 Jun '11

The only thing the init func does is register a spi driver, so if that fails, we return the value back up to the caller who will display an error message for us. So drop the redundant checking/message. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ad1836.c | 10 +--------- 1 files changed, 1 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 754c496..d368539 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -315,15 +315,7 @@ static struct spi_driver ad1836_spi_driver = { static int __init ad1836_init(void) { - int ret; - - ret = spi_register_driver(&ad1836_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n", - ret); - } - - return ret; + return spi_register_driver(&ad1836_spi_driver); } module_init(ad1836_init); -- 1.7.5.3

3 12

Re: [alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Takashi Iwai 15 Jun '11

15 Jun '11

At Thu, 9 Jun 2011 13:14:01 +0200, Michal Hocko wrote: > > On Thu 09-06-11 11:44:41, Takashi Iwai wrote: > > At Thu, 9 Jun 2011 11:28:26 +0200, > > Michal Hocko wrote: > > > > > > [1 <text/plain; us-ascii (7bit)>] > > > On Thu 09-06-11 11:20:04, Takashi Iwai wrote: > > > > At Thu, 9 Jun 2011 11:06:07 +0200, > > > > Michal Hocko wrote: > > > > > > > > > > On Thu 09-06-11 10:56:16, Michal Hocko wrote: > > > > > > On Thu 09-06-11 08:09:18, Takashi Iwai wrote: > > > > > > > At Wed, 8 Jun 2011 19:48:55 +0200, > > > > > [...] > > > > > > > Thanks, that's helpful. > > > > > > > As a quirk solution, could you try to pass model=auto option to > > > > > > > snd-hda-intel module? > > > > > > > > > > > > Headphones work now but speakers are quite... I have tried to play > > > > > > with Headphone and Speaker controls in alsamixer but no changes. > > > > > > > > > > > > Here is what I get in logs when the module is loaded: > > > > > > [ 5320.360277] HDA Intel 0000:00:1b.0: PCI INT A disabled > > > > > > [ 5326.920247] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > > > > > [ 5326.920262] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > > > > > [ 5326.920279] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 > > > > > > [ 5326.920375] HDA Intel 0000:00:1b.0: irq 42 for MSI/MSI-X > > > > > > [ 5326.920405] HDA Intel 0000:00:1b.0: setting latency timer to 64 > > > > > > [ 5326.981085] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input18 > > > > > > [ 5326.981262] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input19 > > > > > > [ 5326.987934] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input20 > > > > > > > > > > > > [maybe not important but input numbers are increasing for every new load] > > > > > > > > > > > > There is another strange thing. After I stop mplayer and wait for some > > > > > > time (~1min) I can hear a "click" sound and then there is no sound at > > > > > > all... If I start mplayer in the meantime it just works... > > > > > > If I try to remove the module and load it again then things work and stop > > > > > > working after first playback again. > > > > > > > > > > And I have just found out that I do not have to start any playback to > > > > > see this. It is sufficient to load the module and wait a minute and > > > > > there is no sound at all (both from speakers or headphones). > > > > > > > > OK, could you give alsa-info.sh outputs at both working and > > > > non-working states? > > > > > > Sure. Attached. The only change seems to be: > > > Node 0x02 [Audio Output] wcaps 0x11: Stereo > > > Device: name="ALC262 Analog", type="Audio", device=0 > > > - Converter: stream=5, channel=0 > > > + Converter: stream=0, channel=0 > > > > This is OK. It shows that the converter was assigned once to this > > stream when you played back a sound before power-save. > > > > As long as the codec registers show the same value, it's not obviously > > what went wrong. The symptom sounds like a resume issue. If so, > > S3/S4 might show the same problem. > > > > BTW, does my patch fix the problem when you load without model=auto? > > I have just booted to the rc2 with the patch (https://lkml.org/lkml/2011/6/9/25) > and it looks it didn't solve the problem. Playback works only from > speakers when headphones are not plugged in - no sound if plugged in. So, the auto-mute now works but now the headphone doesn't work. Weird... Maybe the same reason the speaker doesn't work with auto-mode. I saw that the only one converter keeps the stream in your alsa-info.sh output although multiple converters (0x02, 0x03, 0x06) should be used simultaneously. > See alsa-info.txt-fresh-boot. Same thing happens if I > modprobe -r snd-hda-intel and then load it again (without any parameters). > > > If yes, does the problem appear with it? I.e. is the silence problem > > specific to model=auto or not? > > If I try to modproble with model=auto I get headphones working but not > output from speakers. Have a look at alsa-info.txt-modprobe_auto. For testing, just copy 2.6.39/sound/pci/*.[ch] into 3.0 tree and rebuild. Does it work as expected? If you have a good-working reference, please take alsa-info.sh output while playing back a stream. Note that the one-minute thing happens only when you unplug AC cable, supposedly. Or, simply write some value (in seconds) to /sys/modules/snd_hda_intel/parameters/powersave and access the device once (either mixer or PCM) to activate the power-saving mode. thanks, Takashi

3 16

[alsa-devel] Need Help About Submitting Updates to Existing ASoC Codec Driver
by Ashish P. Chavan 15 Jun '11

15 Jun '11

Hi, We are in the process of upgrading codec driver for Dialog's DA7210 codec (sound/soc/codecs/da7210.c). Existing driver in mainline is just bare minimum and doesn't support many of the chip's features. Our update would be a major functionality update in nature. Our final goal is to get the updated driver merged into mainline kernel. For this we just want to make sure that we follow correct process. As far as I understood from alsa-project.org and other resources, the normal process is to submit the patch against alsa-kenel (git.alsa-project.org:alsa-kernel.git alsa-kernel) and get it reviewed by maintainers there. Once all maintainers are satisfied with the patch, it will be eventually put in to for-linus branch and get in to mainline. Is this the correct approach and path to achieve our goal? Is there any obvious misunderstanding or missing step(s) here? Thanks, -- Ashish ---------------------------------------------------------------------------- || Linux Is User Friendly, It's Just Selective About Who It's Friend' Are || ----------------------------------------------------------------------------

1 0

[alsa-devel] IEC61937/SPDIF
by ajay khandelwal 14 Jun '11

14 Jun '11

Hi All, I have to develop driver for SPDIF output of my board connected to HDMI. I'm not sure for following (specially IEC61937) 1) How should driver know if it is compressed audio or plain pcm 2) Is their any plugin available for IEC61937, I could see only for IEC958 3) Who should create burst preamble, some plug in or driver. Thanks and Regards, Ajay

3 2

[alsa-devel] [PATCH v3] ASoC: dapm - Refactor widget IO functions in preparation for platform widgets.
by Liam Girdwood 14 Jun '11

14 Jun '11

This time with soc_widget_update_bits reflecting recent soc_update_bits changes. Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP devices have mixer components that can alter power usage and hence require full DAPM support. This provides a generic widget IO operation wrapper in preparation for future patches that implement platform driver DAPM. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 49 +++++++++++++++++++++++++++++++++++++++++++------ 1 files changed, 43 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 34106bc..2e6c351 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,6 +124,43 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) +{ + if (w->codec) + return snd_soc_read(w->codec, reg); + return 0; +} + +static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) +{ + if (w->codec) + return snd_soc_write(w->codec, reg, val); + return 0; +} + +static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, + unsigned short reg, unsigned int mask, unsigned int value) +{ + int change; + unsigned int old, new; + int ret; + + ret = soc_widget_read(w, reg); + if (ret < 0) + return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) { + ret = soc_widget_write(w, reg, new); + if (ret < 0) + return ret; + } + + return change; +} + /** * snd_soc_dapm_set_bias_level - set the bias level for the system * @dapm: DAPM context @@ -181,7 +218,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = snd_soc_read(w->codec, reg); + val = soc_widget_read(w, reg); val = (val >> shift) & mask; if ((invert && !val) || (!invert && val)) @@ -197,7 +234,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; - val = snd_soc_read(w->codec, e->reg); + val = soc_widget_read(w, e->reg); item = (val >> e->shift_l) & (bitmask - 1); p->connect = 0; @@ -227,7 +264,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, w->kcontrol_news[i].private_value; int val, item; - val = snd_soc_read(w->codec, e->reg); + val = soc_widget_read(w, e->reg); val = (val >> e->shift_l) & e->mask; for (item = 0; item < e->max; item++) { if (val == e->values[item]) @@ -669,7 +706,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - snd_soc_update_bits(w->codec, -(w->reg + 1), + soc_widget_update_bits(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -877,7 +914,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - snd_soc_update_bits(dapm->codec, reg, mask, value); + soc_widget_update_bits(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1878,7 +1915,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) /* Read the initial power state from the device */ if (w->reg >= 0) { - val = snd_soc_read(w->codec, w->reg); + val = soc_widget_read(w, w->reg); val &= 1 << w->shift; if (w->invert) val = !val; -- 1.7.4.1

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[alsa-devel] Subject= @Daniel Mack: Send you a FTU
by Aurélien Leblond 14 Jun '11

14 Jun '11

>> I saw on the mailing that you a Felix have been doing a great deal of >> work to provide better support for the M-Audio FastTrack Ultra. >> I tested yesterday the new patch that provides mixer and the >> possibility to change the internal routing of the soundcard... I can >> finally use all the input/output to use external FX, that's very >> handy! >> >> Anyway, on the M-Audio forum, Felix has been mentionning several times >> that there were few bugs that needed to be fixed with this soundcard, >> but that you needed to get your hand on one device to fix them (and >> may be even improve things around its support in general). > >Quite frankly, I'm not fully aware about known issues with these >devices that need attention. Could you summarize which problems are >left to look at? There are still the issue of samples dropping. Tto reproduce it: - Open Audacity and set it up to be using Jack. - Click Generate -> Tone -> Ok. "Normally" you should hear very regular pops...

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Re: [alsa-devel] [PATCH] hdspm.c: make driver name generic to avoid user confusion
by Adrian Knoth 14 Jun '11

14 Jun '11

On Tue, Jun 07, 2011 at 10:44:18AM +0200, Jörn Nettingsmeier wrote: > hi adi, *! Hi! > here's a trivial patch to change the driver name of hdspm.c to avoid > irritating users who don't have the MADI version (just happened on the > LAU list). > static struct pci_driver driver = { > - .name = "RME Hammerfall DSP MADI", > + .name = "RME Hammerfall HDSPe Series", Are braces allowed in driver names? I guess so. "HDSPe" is also confusing, only referring to the PCIe versions. Since we also support the old PCI versions, I suggest to name it "RME Hammerfall HDSP(e) Series" On the other hand, the 'H' in "HDSP" stands for "Hammerfall", so "RME HDSP(e) Series" would be more appropriate. Luckily, we don't collide with the hdsp.c driver name, but still, it's misleading, because there are cards that match "RME HDSP(e) Series" (like the Multiface/Digiface) but are handled by hdsp.c instead. What at mess. ;) Merging hdsp.c into hdspm.c would be the clean approach to this problem, still I don't have time for this in the near future. Maybe in July. Given that all current variants are misleading in one way or another, we could also leave everything as is. What do you think? Cheers -- mail: adi(a)thur.de http://adi.thur.de PGP/GPG: key via keyserver

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[alsa-devel] [PATCH] Sound, 6Fire USB: Fix double-free bug in usb6fire_fw_ezusb_upload()
by Jesper Juhl 14 Jun '11

14 Jun '11

We have a double-free bug in sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload(). We already call release_firmware(fw) on line 258, so when we then do it again after usb6fire_fw_ezusb_write() returns <0, we have a double-free. Easily fixed by just removing the last call to release_firmware(). Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- firmware.c | 1 - 1 file changed, 1 deletion(-) Patch against Linus' tree (head at 40779859de0f73b40390c6401a024d06cf024290). diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index a91719d..1e3ae33 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -270,7 +270,6 @@ static int usb6fire_fw_ezusb_upload( data = 0x00; /* resume ezusb cpu */ ret = usb6fire_fw_ezusb_write(device, 0xa0, 0xe600, &data, 1); if (ret < 0) { - release_firmware(fw); snd_printk(KERN_ERR PREFIX "unable to upload ezusb " "firmware %s: end message.\n", fwname); return ret; -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] Help About Tascam US144
by Leonardo Antonio 14 Jun '11

14 Jun '11

Hello, This is my problem: [ 1432.218389] usb_submit_urb(sk->outurb[0]) returned -90 [ 1432.219384] status=-2 [ 1432.620091] us122l_start error -90 [ 1432.620126] snd-usb-us122l: probe of 2-3:1.1 failed with error -22 [ 1432.620191] usbcore: registered new interface driver snd-usb-us122l Can you help me? Thanks!

1 0

[alsa-devel] [PATCH] ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O bus
by Mark Brown 13 Jun '11

13 Jun '11

This will be removed in -next so let's drop it from mainline as soon as we can in order to minimise surprises. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This posting is against -next, sorry, but the update for 3.0 is trivially obvious. include/sound/soc.h | 3 +-- sound/soc/soc-io.c | 3 --- 2 files changed, 1 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4334ab2..6424b10 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -258,8 +258,7 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { - SND_SOC_CUSTOM = 1, - SND_SOC_I2C, + SND_SOC_I2C = 1, SND_SOC_SPI, }; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index df9338a..cca490c 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -367,9 +367,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; switch (control) { - case SND_SOC_CUSTOM: - break; - case SND_SOC_I2C: #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) codec->hw_write = (hw_write_t)i2c_master_send; -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Don't use codec->control_data in bulk write
by Mark Brown 13 Jun '11

13 Jun '11

In order to facilitate merging with the register map I/O replace the use of control_data for the bulk writes with direct lookup of the client data from the device. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-io.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index fbea6c9..cca490c 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -269,12 +269,12 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r switch (codec->control_type) { #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) case SND_SOC_I2C: - ret = i2c_master_send(codec->control_data, data, len); + ret = i2c_master_send(to_i2c_client(codec->dev), data, len); break; #endif #if defined(CONFIG_SPI_MASTER) case SND_SOC_SPI: - ret = spi_write(codec->control_data, data, len); + ret = spi_write(to_spi_device(codec->dev), data, len); break; #endif default: -- 1.7.5.4

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[alsa-devel] soc-dsp questions
by pl bossart 13 Jun '11

13 Jun '11

To prepare next week's ALSA-Asoc meeting, I reviewed Liam's dsp-upstream code, and I am a bit unclear on the 'no_host_mode' supported by some DAIs. Apparently these are regular ALSA PCM substreams, except that there are no data exchanges to/from the host. - If these substreams are known, an application can in theory open/close them. How would it know that no data is to be written/read? I would think additional changes are required in alsa-lib? - is this really important that such PCM devices be known in userspace? Or are they declared only so that ALSA controls are enabled for these streams? - along the same lines, how useful is the .pointer routine for such devices? Since no data will be provided by the host, is this really needed? I see it's only meaningful for the low-power playback mode, but I am unsure if the information provided would actually be used by anyone? - what is the meaning of the SND_SOC_DAI_LINK_OPT_HOST definition? It doesn't seem to be updated depending on which back-end is used, so it this really needed? Thanks -Pierre

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[alsa-devel] [PATCH] ASoC: Move register I/O code into a separate file
by Mark Brown 13 Jun '11

13 Jun '11

For clarity and to help ongoing refactoring in this area create a new file to contain the physical I/O functions, separating them out from the cache operations. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/Makefile | 3 +- sound/soc/soc-cache.c | 380 ---------------------------------------------- sound/soc/soc-io.c | 399 +++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 401 insertions(+), 381 deletions(-) create mode 100644 sound/soc/soc-io.c diff --git a/sound/soc/Makefile b/sound/soc/Makefile index adb5719..4f91387 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o soc-pcm.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o +snd-soc-core-objs += soc-pcm.o soc-io.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index b88a61f..d9f8ade 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,386 +20,6 @@ #include <trace/events/asoc.h> -#ifdef CONFIG_SPI_MASTER -static int do_spi_write(void *control, const char *data, int len) -{ - struct spi_device *spi = control; - int ret; - - ret = spi_write(spi, data, len); - if (ret < 0) - return ret; - - return len; -} -#endif - -static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value, const void *data, int len) -{ - int ret; - - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - - ret = codec->hw_write(codec->control_data, data, len); - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; -} - -static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) -{ - int ret; - unsigned int val; - - if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; - - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); - } - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret < 0) - return -1; - return val; -} - -static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 9) | (value & 0x1ff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - reg &= 0xff; - data[0] = reg; - data[1] = value & 0xff; - - return do_hw_write(codec, reg, value, data, 2); -} - -static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 val = cpu_to_be16(value); - - data[0] = reg; - memcpy(&data[1], &val, sizeof(val)); - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int do_i2c_read(struct snd_soc_codec *codec, - void *reg, int reglen, - void *data, int datalen) -{ - struct i2c_msg xfer[2]; - int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = reglen; - xfer[0].buf = reg; - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = datalen; - xfer[1].buf = data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret == 2) - return 0; - else if (ret < 0) - return ret; - else - return -EIO; -} -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, &reg, 1, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_8_8_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u16 data; - int ret; - - ret = do_i2c_read(codec, &reg, 1, &data, 2); - if (ret < 0) - return 0; - return (data >> 8) | ((data & 0xff) << 8); -} -#else -#define snd_soc_8_16_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, &reg, 2, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_16_8_read_i2c NULL -#endif - -static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 rval = cpu_to_be16(reg); - - memcpy(data, &rval, sizeof(rval)); - data[2] = value; - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = cpu_to_be16(r); - u16 data; - int ret; - - ret = do_i2c_read(codec, &reg, 2, &data, 2); - if (ret < 0) - return 0; - return be16_to_cpu(data); -} -#else -#define snd_soc_16_16_read_i2c NULL -#endif - -static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data[2]; - - data[0] = cpu_to_be16(reg); - data[1] = cpu_to_be16(value); - - return do_hw_write(codec, reg, value, data, sizeof(data)); -} - -/* Primitive bulk write support for soc-cache. The data pointed to by - * `data' needs to already be in the form the hardware expects - * including any leading register specific data. Any data written - * through this function will not go through the cache as it only - * handles writing to volatile or out of bounds registers. - */ -static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, - const void *data, size_t len) -{ - int ret; - - /* To ensure that we don't get out of sync with the cache, check - * whether the base register is volatile or if we've directly asked - * to bypass the cache. Out of bounds registers are considered - * volatile. - */ - if (!codec->cache_bypass - && !snd_soc_codec_volatile_register(codec, reg) - && reg < codec->driver->reg_cache_size) - return -EINVAL; - - switch (codec->control_type) { -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - case SND_SOC_I2C: - ret = i2c_master_send(codec->control_data, data, len); - break; -#endif -#if defined(CONFIG_SPI_MASTER) - case SND_SOC_SPI: - ret = spi_write(codec->control_data, data, len); - break; -#endif - default: - BUG(); - } - - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; -} - -static struct { - int addr_bits; - int data_bits; - int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); -} io_types[] = { - { - .addr_bits = 4, .data_bits = 12, - .write = snd_soc_4_12_write, - }, - { - .addr_bits = 7, .data_bits = 9, - .write = snd_soc_7_9_write, - }, - { - .addr_bits = 8, .data_bits = 8, - .write = snd_soc_8_8_write, - .i2c_read = snd_soc_8_8_read_i2c, - }, - { - .addr_bits = 8, .data_bits = 16, - .write = snd_soc_8_16_write, - .i2c_read = snd_soc_8_16_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 8, - .write = snd_soc_16_8_write, - .i2c_read = snd_soc_16_8_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 16, - .write = snd_soc_16_16_write, - .i2c_read = snd_soc_16_16_read_i2c, - }, -}; - -/** - * snd_soc_codec_set_cache_io: Set up standard I/O functions. - * - * @codec: CODEC to configure. - * @addr_bits: Number of bits of register address data. - * @data_bits: Number of bits of data per register. - * @control: Control bus used. - * - * Register formats are frequently shared between many I2C and SPI - * devices. In order to promote code reuse the ASoC core provides - * some standard implementations of CODEC read and write operations - * which can be set up using this function. - * - * The caller is responsible for allocating and initialising the - * actual cache. - * - * Note that at present this code cannot be used by CODECs with - * volatile registers. - */ -int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, - enum snd_soc_control_type control) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(io_types); i++) - if (io_types[i].addr_bits == addr_bits && - io_types[i].data_bits == data_bits) - break; - if (i == ARRAY_SIZE(io_types)) { - printk(KERN_ERR - "No I/O functions for %d bit address %d bit data\n", - addr_bits, data_bits); - return -EINVAL; - } - - codec->write = io_types[i].write; - codec->read = hw_read; - codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; - - switch (control) { - case SND_SOC_CUSTOM: - break; - - case SND_SOC_I2C: -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - codec->hw_write = (hw_write_t)i2c_master_send; -#endif - if (io_types[i].i2c_read) - codec->hw_read = io_types[i].i2c_read; - - codec->control_data = container_of(codec->dev, - struct i2c_client, - dev); - break; - - case SND_SOC_SPI: -#ifdef CONFIG_SPI_MASTER - codec->hw_write = do_spi_write; -#endif - - codec->control_data = container_of(codec->dev, - struct spi_device, - dev); - break; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); - static bool snd_soc_set_cache_val(void *base, unsigned int idx, unsigned int val, unsigned int word_size) { diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c new file mode 100644 index 0000000..855e5cd --- /dev/null +++ b/sound/soc/soc-io.c @@ -0,0 +1,399 @@ +/* + * soc-io.c -- ASoC register I/O helpers + * + * Copyright 2009-2011 Wolfson Microelectronics PLC. + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * 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/i2c.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> + +#include <trace/events/asoc.h> + +#ifdef CONFIG_SPI_MASTER +static int do_spi_write(void *control, const char *data, int len) +{ + struct spi_device *spi = control; + int ret; + + ret = spi_write(spi, data, len); + if (ret < 0) + return ret; + + return len; +} +#endif + +static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value, const void *data, int len) +{ + int ret; + + if (!snd_soc_codec_volatile_register(codec, reg) && + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } + + if (codec->cache_only) { + codec->cache_sync = 1; + return 0; + } + + ret = codec->hw_write(codec->control_data, data, len); + if (ret == len) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) +{ + int ret; + unsigned int val; + + if (reg >= codec->driver->reg_cache_size || + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { + if (codec->cache_only) + return -1; + + BUG_ON(!codec->hw_read); + return codec->hw_read(codec, reg); + } + + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + return val; +} + +static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 data; + + data = cpu_to_be16((reg << 12) | (value & 0xffffff)); + + return do_hw_write(codec, reg, value, &data, 2); +} + +static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 data; + + data = cpu_to_be16((reg << 9) | (value & 0x1ff)); + + return do_hw_write(codec, reg, value, &data, 2); +} + +static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + reg &= 0xff; + data[0] = reg; + data[1] = value & 0xff; + + return do_hw_write(codec, reg, value, data, 2); +} + +static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + u16 val = cpu_to_be16(value); + + data[0] = reg; + memcpy(&data[1], &val, sizeof(val)); + + return do_hw_write(codec, reg, value, data, 3); +} + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int do_i2c_read(struct snd_soc_codec *codec, + void *reg, int reglen, + void *data, int datalen) +{ + struct i2c_msg xfer[2]; + int ret; + struct i2c_client *client = codec->control_data; + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = reglen; + xfer[0].buf = reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = datalen; + xfer[1].buf = data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret == 2) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} +#endif + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + u8 reg = r; + u8 data; + int ret; + + ret = do_i2c_read(codec, &reg, 1, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_8_8_read_i2c NULL +#endif + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + u8 reg = r; + u16 data; + int ret; + + ret = do_i2c_read(codec, &reg, 1, &data, 2); + if (ret < 0) + return 0; + return (data >> 8) | ((data & 0xff) << 8); +} +#else +#define snd_soc_8_16_read_i2c NULL +#endif + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + u16 reg = r; + u8 data; + int ret; + + ret = do_i2c_read(codec, &reg, 2, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_16_8_read_i2c NULL +#endif + +static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + u16 rval = cpu_to_be16(reg); + + memcpy(data, &rval, sizeof(rval)); + data[2] = value; + + return do_hw_write(codec, reg, value, data, 3); +} + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + u16 reg = cpu_to_be16(r); + u16 data; + int ret; + + ret = do_i2c_read(codec, &reg, 2, &data, 2); + if (ret < 0) + return 0; + return be16_to_cpu(data); +} +#else +#define snd_soc_16_16_read_i2c NULL +#endif + +static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 data[2]; + + data[0] = cpu_to_be16(reg); + data[1] = cpu_to_be16(value); + + return do_hw_write(codec, reg, value, data, sizeof(data)); +} + +/* Primitive bulk write support for soc-cache. The data pointed to by + * `data' needs to already be in the form the hardware expects + * including any leading register specific data. Any data written + * through this function will not go through the cache as it only + * handles writing to volatile or out of bounds registers. + */ +static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, + const void *data, size_t len) +{ + int ret; + + /* To ensure that we don't get out of sync with the cache, check + * whether the base register is volatile or if we've directly asked + * to bypass the cache. Out of bounds registers are considered + * volatile. + */ + if (!codec->cache_bypass + && !snd_soc_codec_volatile_register(codec, reg) + && reg < codec->driver->reg_cache_size) + return -EINVAL; + + switch (codec->control_type) { +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) + case SND_SOC_I2C: + ret = i2c_master_send(codec->control_data, data, len); + break; +#endif +#if defined(CONFIG_SPI_MASTER) + case SND_SOC_SPI: + ret = spi_write(codec->control_data, data, len); + break; +#endif + default: + BUG(); + } + + if (ret == len) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static struct { + int addr_bits; + int data_bits; + int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); + unsigned int (*read)(struct snd_soc_codec *, unsigned int); + unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); +} io_types[] = { + { + .addr_bits = 4, .data_bits = 12, + .write = snd_soc_4_12_write, + }, + { + .addr_bits = 7, .data_bits = 9, + .write = snd_soc_7_9_write, + }, + { + .addr_bits = 8, .data_bits = 8, + .write = snd_soc_8_8_write, + .i2c_read = snd_soc_8_8_read_i2c, + }, + { + .addr_bits = 8, .data_bits = 16, + .write = snd_soc_8_16_write, + .i2c_read = snd_soc_8_16_read_i2c, + }, + { + .addr_bits = 16, .data_bits = 8, + .write = snd_soc_16_8_write, + .i2c_read = snd_soc_16_8_read_i2c, + }, + { + .addr_bits = 16, .data_bits = 16, + .write = snd_soc_16_16_write, + .i2c_read = snd_soc_16_16_read_i2c, + }, +}; + +/** + * snd_soc_codec_set_cache_io: Set up standard I/O functions. + * + * @codec: CODEC to configure. + * @addr_bits: Number of bits of register address data. + * @data_bits: Number of bits of data per register. + * @control: Control bus used. + * + * Register formats are frequently shared between many I2C and SPI + * devices. In order to promote code reuse the ASoC core provides + * some standard implementations of CODEC read and write operations + * which can be set up using this function. + * + * The caller is responsible for allocating and initialising the + * actual cache. + * + * Note that at present this code cannot be used by CODECs with + * volatile registers. + */ +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits, + enum snd_soc_control_type control) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(io_types); i++) + if (io_types[i].addr_bits == addr_bits && + io_types[i].data_bits == data_bits) + break; + if (i == ARRAY_SIZE(io_types)) { + printk(KERN_ERR + "No I/O functions for %d bit address %d bit data\n", + addr_bits, data_bits); + return -EINVAL; + } + + codec->write = io_types[i].write; + codec->read = hw_read; + codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; + + switch (control) { + case SND_SOC_CUSTOM: + break; + + case SND_SOC_I2C: +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) + codec->hw_write = (hw_write_t)i2c_master_send; +#endif + if (io_types[i].i2c_read) + codec->hw_read = io_types[i].i2c_read; + + codec->control_data = container_of(codec->dev, + struct i2c_client, + dev); + break; + + case SND_SOC_SPI: +#ifdef CONFIG_SPI_MASTER + codec->hw_write = do_spi_write; +#endif + + codec->control_data = container_of(codec->dev, + struct spi_device, + dev); + break; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); + -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Add weak routes for sidetone style paths
by Mark Brown 13 Jun '11

13 Jun '11

Normally DAPM will power up any connected audio path. This is not ideal for sidetone paths as with sidetone paths the audio path is not wanted in itself, it is only desired if the two paths it provides a sidetone between are both active. If the sidetone path causes a power up then it can be hard to minimise pops as we first power up either the sidetone or the main output path and then power the other, with the second power up potentially introducing a DC offset. Address this by introducing the concept of a weak path. If a path is marked as weak then DAPM will ignore that path when walking the graph, though all the relevant controls are still available to the application layer to allow these paths to be configured. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 + sound/soc/soc-dapm.c | 87 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 89 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 7415878..7c5465b 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -348,6 +348,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm); int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); +int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route, int num); /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fd2d774..746349f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -593,6 +593,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { + if (path->weak) + continue; + if (path->walked) continue; @@ -643,6 +646,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { + if (path->weak) + continue; + if (path->walked) continue; @@ -724,6 +730,9 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { + if (path->weak) + continue; + if (path->connected && !path->connected(path->source, path->sink)) continue; @@ -1806,6 +1815,84 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); +static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route) +{ + struct snd_soc_dapm_widget *source = dapm_find_widget(dapm, + route->source, + true); + struct snd_soc_dapm_widget *sink = dapm_find_widget(dapm, + route->sink, + true); + struct snd_soc_dapm_path *path; + int count = 0; + + if (!source) { + dev_err(dapm->dev, "Unable to find source %s for weak route\n", + route->source); + return -ENODEV; + } + + if (!sink) { + dev_err(dapm->dev, "Unable to find sink %s for weak route\n", + route->sink); + return -ENODEV; + } + + if (route->control || route->connected) + dev_warn(dapm->dev, "Ignoring control for weak route %s->%s\n", + route->source, route->sink); + + list_for_each_entry(path, &source->sinks, list_source) { + if (path->sink == sink) { + path->weak = 1; + count++; + } + } + + if (count == 0) + dev_err(dapm->dev, "No path found for weak route %s->%s\n", + route->source, route->sink); + if (count > 1) + dev_warn(dapm->dev, "%d paths found for weak route %s->%s\n", + count, route->source, route->sink); + + return 0; +} + +/** + * snd_soc_dapm_weak_routes - Mark routes between DAPM widgets as weak + * @dapm: DAPM context + * @route: audio routes + * @num: number of routes + * + * Mark existing routes matching those specified in the passed array + * as being weak, meaning that they are ignored for the purpose of + * power decisions. The main intended use case is for sidetone paths + * which couple audio between other independent paths if they are both + * active in order to make the combination work better at the user + * level but which aren't intended to be "used". + * + * Note that CODEC drivers should not use this as sidetone type paths + * can frequently also be used as bypass paths. + */ +int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route, int num) +{ + int i, err; + int ret = 0; + + for (i = 0; i < num; i++) { + err = snd_soc_dapm_weak_route(dapm, route); + if (err) + ret = err; + route++; + } + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); + /** * snd_soc_dapm_new_widgets - add new dapm widgets * @dapm: DAPM context -- 1.7.5.4

2 1

Re: [alsa-devel] [PATCH] hda: check make_exec_verb() return value
by Takashi Iwai 13 Jun '11

13 Jun '11

At Mon, 13 Jun 2011 07:45:45 -0700, Greg Thelen wrote: > > If given a -1 cmd parameter then make_exec_verb() returns -1 without > setting the res output value. > > Prior to this change snd_hda_codec_read() assumed that make_exec_verb() > unconditionally set res regardless of the cmd value. > > This change explicitly checks the make_exec_verb() return value before > consuming the potentially unset res value. > > Signed-off-by: Greg Thelen <gthelen(a)google.com> Applied now. Thanks. Takashi > --- > sound/pci/hda/hda_codec.c | 3 ++- > 1 files changed, 2 insertions(+), 1 deletions(-) > > diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c > index 45b4a8d..f06dd5f 100644 > --- a/sound/pci/hda/hda_codec.c > +++ b/sound/pci/hda/hda_codec.c > @@ -243,7 +243,8 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, > { > unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm); > unsigned int res; > - codec_exec_verb(codec, cmd, &res); > + if (codec_exec_verb(codec, cmd, &res)) > + return -1; > return res; > } > EXPORT_SYMBOL_HDA(snd_hda_codec_read); > -- > 1.7.3.1 >

1 0

[alsa-devel] [PATCH] ASoC: Factor out redundant read() functions
by Mark Brown 13 Jun '11

13 Jun '11

We've got a whole bunch of functions which just call straight through to do_hw_read(). Simplify this situation by removing them and using hw_read() directly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 52 +++++++----------------------------------------- 1 files changed, 8 insertions(+), 44 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9a88a27..b88a61f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -61,7 +61,7 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, return -EIO; } -static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg) +static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) { int ret; unsigned int val; @@ -82,12 +82,6 @@ static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg) return val; } -static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -98,12 +92,6 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, return do_hw_write(codec, reg, value, &data, 2); } -static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -126,12 +114,6 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, return do_hw_write(codec, reg, value, data, 2); } -static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -144,12 +126,6 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, return do_hw_write(codec, reg, value, data, 3); } -static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int do_i2c_read(struct snd_soc_codec *codec, void *reg, int reglen, @@ -232,12 +208,6 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #define snd_soc_16_8_read_i2c NULL #endif -static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -267,12 +237,6 @@ static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, #define snd_soc_16_16_read_i2c NULL #endif -static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -337,30 +301,30 @@ static struct { } io_types[] = { { .addr_bits = 4, .data_bits = 12, - .write = snd_soc_4_12_write, .read = snd_soc_4_12_read, + .write = snd_soc_4_12_write, }, { .addr_bits = 7, .data_bits = 9, - .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, + .write = snd_soc_7_9_write, }, { .addr_bits = 8, .data_bits = 8, - .write = snd_soc_8_8_write, .read = snd_soc_8_8_read, + .write = snd_soc_8_8_write, .i2c_read = snd_soc_8_8_read_i2c, }, { .addr_bits = 8, .data_bits = 16, - .write = snd_soc_8_16_write, .read = snd_soc_8_16_read, + .write = snd_soc_8_16_write, .i2c_read = snd_soc_8_16_read_i2c, }, { .addr_bits = 16, .data_bits = 8, - .write = snd_soc_16_8_write, .read = snd_soc_16_8_read, + .write = snd_soc_16_8_write, .i2c_read = snd_soc_16_8_read_i2c, }, { .addr_bits = 16, .data_bits = 16, - .write = snd_soc_16_16_write, .read = snd_soc_16_16_read, + .write = snd_soc_16_16_write, .i2c_read = snd_soc_16_16_read_i2c, }, }; @@ -402,7 +366,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, } codec->write = io_types[i].write; - codec->read = io_types[i].read; + codec->read = hw_read; codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; switch (control) { -- 1.7.5.3

2 1

[alsa-devel] [PATCH 0/4] Adding idma driver for samsung SoCs
by Sangbeom Kim 13 Jun '11

13 Jun '11

I2S in samsung audio subsytem has a small internal dma(idma). IDMA can be used in low power audio mode and I2S0 secondary fifo data tranfer. In this patchsets, idma driver is added. This patch is tested on SMDKV310, Nexus-s(S5PC110) with lakkyung jung (lakkyung.jung(a)samsung.com) This patchset contains followings o To Jassi Brar, Mark Brown and Liam Girdwood, [PATCH 1/4] ASoC: SAMSUNG: Modify I2S driver to support idma [PATCH 2/4] ASoC: SAMSUNG: Add I2S0 internal dma driver [PATCH 3/4] ASoC: SAMSUNG: Change platform driver for SMDKs o To Kukjin Kim and Ben Dooks, [PATCH 4/4] ARM: SAMSUNG: Add platform device for idma Best Regards, SB Kim

5 25

[alsa-devel] [PATCH] tea575x: allow multiple opens
by Ondrej Zary 12 Jun '11

12 Jun '11

Change locking to allow tea575x-radio device to be opened multiple times. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/include/sound/tea575x-tuner.h 2011-06-11 15:21:50.000000000 +0200 +++ linux-2.6.39-rc2/include/sound/tea575x-tuner.h 2011-06-11 14:50:55.000000000 +0200 @@ -49,7 +49,7 @@ struct snd_tea575x { bool tuned; /* tuned to a station */ unsigned int val; /* hw value */ unsigned long freq; /* frequency */ - unsigned long in_use; /* set if the device is in use */ + struct mutex mutex; struct snd_tea575x_ops *ops; void *private_data; u8 card[32]; --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-06-11 15:21:50.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-06-11 14:57:28.000000000 +0200 @@ -282,26 +282,9 @@ static int vidioc_s_input(struct file *f return 0; } -static int snd_tea575x_exclusive_open(struct file *file) -{ - struct snd_tea575x *tea = video_drvdata(file); - - return test_and_set_bit(0, &tea->in_use) ? -EBUSY : 0; -} - -static int snd_tea575x_exclusive_release(struct file *file) -{ - struct snd_tea575x *tea = video_drvdata(file); - - clear_bit(0, &tea->in_use); - return 0; -} - static const struct v4l2_file_operations tea575x_fops = { .owner = THIS_MODULE, - .open = snd_tea575x_exclusive_open, - .release = snd_tea575x_exclusive_release, - .ioctl = video_ioctl2, + .unlocked_ioctl = video_ioctl2, }; static const struct v4l2_ioctl_ops tea575x_ioctl_ops = { @@ -340,13 +323,14 @@ int snd_tea575x_init(struct snd_tea575x if (snd_tea575x_read(tea) != 0x55AA) return -ENODEV; - tea->in_use = 0; tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; tea->freq = 90500 * 16; /* 90.5Mhz default */ snd_tea575x_set_freq(tea); tea->vd = tea575x_radio; video_set_drvdata(&tea->vd, tea); + mutex_init(&tea->mutex); + tea->vd.lock = &tea->mutex; v4l2_ctrl_handler_init(&tea->ctrl_handler, 1); tea->vd.ctrl_handler = &tea->ctrl_handler; -- Ondrej Zary

3 4

[alsa-devel] [PATCH] [ALSA] emu10k1: Add details for E-mu 0404 PCIe version
by Florian Zeitz 12 Jun '11

12 Jun '11

This patch adds the necessary details to support the PCIe version of E-MU's 0404 card. I have tested this for some time now and it appears to be working fine. _From comparing the PCBs it seems the PCIe version just added a PCIe chipset and left all other components pretty much in place. For anyone intrigued to take a look at the PCB there are pictures I took at <http://babelmonkeys.de/~florob/E-MU%200404/>. This modification was made on top of Takashi Iwai's fix/misc branch. Regards, Florian Zeitz --- sound/pci/emu10k1/emu10k1_main.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 5e619a8..15f0161 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1440,6 +1440,14 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ + /* EMU0404 PCIe */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40051102, + .driver = "Audigy2", .name = "E-mu 0404 PCIe [MAEM8984]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 PCIe ver_03 */ /* Note that all E-mu cards require kernel 2.6 or newer. */ {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]", -- 1.7.5.4

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[alsa-devel] [PATCH 0/3] hdspm - Series of small corrections
by Adrian Knoth 12 Jun '11

12 Jun '11

Hi! Here's a series of small corrections to the hdspm driver. First patch was tested by Joern Nettingsmeier, the remaining two by IOSONO-sound.com together with me. Cheers Adrian Knoth (3): ALSA: hdspm - Fix locking in snd_hdspm_midi_input_read ALSA: hdspm - Fix jumping external wordclock frequency in AutoSync mode ALSA: hdspm - Add firmware revision ID for RME MADI PCI version sound/pci/rme9652/hdspm.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) -- 1.7.5.4

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[alsa-devel] Microphone array (playstation eye) unsupported rates by driver
by Michael Brentson 12 Jun '11

12 Jun '11

Hello, I am trying to use the microphone array of an Playstation Eye usb device. The problem is that the only valid sample rate seems to be 16kHz, even though the usb device(according to the vendor's site) should support sampling rates up to 48kHz. I would have two questions: 1. I would like to try to modify the driver to make 48kHz as a supported rate. Could this be dangerous for the hardware, or if I will add an unsupported rate in the driver, the device will simply not work? 2. I'm not familiar with the "under the hood" aspects of ALSA, and I've started reading* Writing an ALSA Driver by Takashi Iwai.* From what I can tell, I would have to modify the snd-usb-audio driver, is this a correct assumption?(The data on which this assumption is made is presented below)* arecord --list-devices **** List of CAPTURE Hardware Devices **** ..... card 2: CameraB409241 [USB Camera-B4.09.24.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 * lsusb -s 1:3 Bus 001 Device 003: ID 1415:2000 Nam Tai E&E Products Ltd. or OmniVision Technologies, Inc. Sony Playstation Eye lsusb -t .... /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/6p, 480M |__ Port 4: Dev 3, If 0, Class=vend., Driver=ov534, 480M |__ Port 4: Dev 3, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 4: Dev 3, If 2, Class=audio, Driver=snd-usb-audio, 480M Best Regards, Michael

1 0

[alsa-devel] snd_hda_intel
by Donn Washburn 12 Jun '11

12 Jun '11

Hello Alsa; This MB is a ASUS M4A87TD and a VIA VT1818 High Definition Audio 8 channel CODEC which supports192Khz and 24bit BD Lossless Sound I have a problem with a new motherboard and openSuSE's 11.4 version of alsa (alsa-1.0.24.1-3.1.x86_64). The sound is very good however, all of the mixers may be the problem. The problem is that the pots of a mixer are all coming up incorrect. Like the Line In is showing up as Mic 1 and Master does work at all. I am including the output of hwinfo --sound to show modules are mount 19: PCI 14.2: 0403 Audio device [Created at pci.318] Unique ID: 5Dex.BDUuYMOWyw7 SysFS ID: /devices/pci0000:00/0000:00:14.2 SysFS BusID: 0000:00:14.2 Hardware Class: sound Model: "ATI SBx00 Azalia (Intel HDA)" Vendor: pci 0x1002 "ATI Technologies Inc" Device: pci 0x4383 "SBx00 Azalia (Intel HDA)" SubVendor: pci 0x1002 "ATI Technologies Inc" SubDevice: pci 0x4383 Revision: 0x40 Driver: "HDA Intel" Driver Modules: "snd_hda_intel" Memory Range: 0xfcef4000-0xfcef7fff (rw,non-prefetchable) IRQ: 16 (9521 events) Module Alias: "pci:v00001002d00004383sv00001002sd00004383bc04sc03i00" Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: "modprobe snd_hda_intel" Config Status: cfg=yes, avail=yes, need=no, active=unknown 32: PCI 500.1: 0403 Audio device [Created at pci.318] Unique ID: 5yAR.MFsh1X_ykN2 Parent ID: _Znp.ZJmKoWxd6BF SysFS ID: /devices/pci0000:00/0000:00:02.0/0000:05:00.1 SysFS BusID: 0000:05:00.1 Hardware Class: sound Model: "nVidia Audio device" Vendor: pci 0x10de "nVidia Corporation" Device: pci 0x0bea SubVendor: pci 0x3842 "eVga.com. Corp." SubDevice: pci 0x1441 Revision: 0xa1 Driver: "HDA Intel" Driver Modules: "snd_hda_intel" Memory Range: 0xfe97c000-0xfe97ffff (rw,non-prefetchable) IRQ: 19 (211 events) Module Alias: "pci:v000010DEd00000BEAsv00003842sd00001441bc04sc03i00" Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: "modprobe snd_hda_intel" Config Status: cfg=yes, avail=yes, need=no, active=unknown Attached to: #9 (PCI bridge) Sorry about the length of this sound file I do hope this may offer an idea. Is there a better module to try. Or is this all a KDE Kmix (one pot only) or gmixer (about 5 pots) problem. Please advise! -- 73 de Donn Washburn 307 Savoy Street Email:" n5xwb(a)comcast.net " Sugar Land, TX 77478 LL# 1.281.242.3256 Ham Callsign N5XWB HAMs : " n5xwb(a)arrl.net " VoIP via Skype:n5xwbg BMWMOA #:4146 Ambassador " http://counter.li.org " #279316

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[PATCH] [ALSA] emu10k1: Add details for E-mu 0404 PCIe version
by Florian Zeitz 12 Jun '11

12 Jun '11

This patch adds the necessary details to support the PCIe version of E-MU's 0404 card.

1 0

[alsa-devel] [PATCH] tea575x: remove useless input ioctls
by Ondrej Zary 11 Jun '11

11 Jun '11

Remove empty and useless g_input and s_input ioctls. This fixes one fail of v4l2-compliance test. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-06-11 15:29:18.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-06-11 15:29:51.000000000 +0200 @@ -269,19 +269,6 @@ static int tea575x_s_ctrl(struct v4l2_ct return -EINVAL; } -static int vidioc_g_input(struct file *filp, void *priv, unsigned int *i) -{ - *i = 0; - return 0; -} - -static int vidioc_s_input(struct file *filp, void *priv, unsigned int i) -{ - if (i != 0) - return -EINVAL; - return 0; -} - static const struct v4l2_file_operations tea575x_fops = { .owner = THIS_MODULE, .unlocked_ioctl = video_ioctl2, @@ -293,8 +280,6 @@ static const struct v4l2_ioctl_ops tea57 .vidioc_s_tuner = vidioc_s_tuner, .vidioc_g_audio = vidioc_g_audio, .vidioc_s_audio = vidioc_s_audio, - .vidioc_g_input = vidioc_g_input, - .vidioc_s_input = vidioc_s_input, .vidioc_g_frequency = vidioc_g_frequency, .vidioc_s_frequency = vidioc_s_frequency, }; -- Ondrej Zary

1 0

[alsa-devel] Creative X-Fi Titanium HD ctxfi bugs 5378 5218 4868
by Harry Butterworth 11 Jun '11

11 Jun '11

Hello folks, I bought an X-Fi Titanium HD thinking it would work with the ctxfi driver but it is a new card and has a few differences from the earlier cards. Creative technical support have been quite helpful in answering my questions. This email is a heads-up to let you know that I'm doing some work on the driver. I'm not finished yet. In particular the PLL initialization still does not work reliably, the capture source selection is a bit broken and one of the ADC chips is being driven out-of-spec when the card is run at 192kHz. I have attached the current state of my changes which are enough to get the Titanium HD card working but the patch shouldn't be applied because my hack at the PLL code and a couple of other bits may break existing cards. I have been working against the ALSA 1.0.24 release so this is a patch against 1.0.24. I'm waiting for creative to answer my questions about the PLL initialization and correct setup for 196kHz operation. Once I get the answers I'll try to fix the issues and submit a cleaner patch. Comments on the current version are welcome. Harry

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Re: [alsa-devel] HELP:How to power down TWL5030 Voice/Bluetooth sidetone PGA?
by Péter Ujfalusi 11 Jun '11

11 Jun '11

On Thursday 09 June 2011 16:40:15 onlyfever wrote: > Hello, > I want to power down Voice/Bluetooth sidetone PGA of TWL5030. > I set the VSTPGA Register/BTSTPGA Register to 0x00,but it seems that > the Voice/Bluetooth sidetone PGA still work, > and even set VSTPGA/BTSTPGA to Mute,there still have sidetone come out. > > How can I power down Voice/Bluetooth sidetone PGA? > I don't need sidetone. Are you sure, that the sidetone is added by the VSTPGA/BTSTPGA? Could you check, if you accidentally left either the analog loopbacks enabled (or digital loopbacks)? Check for these: Right1/2 Analog Loopback, Left1/2 Analog Loopback, Voice Analog Loopback, Left/Right/Voice Digital Loopback Could you describe your scenario (uplink, downlink routing, etc)? -- Péter

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[alsa-devel] OMAP4 DMIC only works on 2 channels but 4 or 6 channels
by hong zhang 11 Jun '11

11 Jun '11

Liam, OMAP4 DMIC only works on 2 channels but 4 or 6 channels by means of arecord and ABE diagram. 1. "arecord -f dat -c 4 plughw:0,0 dmic-4.wav" returns Recording WAVE 'dmic-4.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channe[ 929.988220] omap-dmic-dai omap-dmic-dai: is ls 4 [ 929.996002] asoc: interface omap-dmic-abe-dai-0 hw params failed [ 930.002563] SDP4430 Media: omap_abe_dai_hw_params: backend hw_params failed -22 [ 930.011016] asoc: interface MultiMedia1 hw params failed arecord: set_params:1116: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 64 CHANNELS: 4 RATE: 48000 PERIOD_TIME: 42625 PERIOD_SIZE: 2046 PERIOD_BYTES: 16368 PERIODS: 4 BUFFER_TIME: 170500 BUFFER_SIZE: 8184 BUFFER_BYTES: 65472 TICK_TIME: 0 2. "arecord -f dat -c 2 plughw:0,0 dmic-4.wav" collects wrong data, no voice is capture in dmic-4.wav. if following is set. amixer cset numid=58 4 amixer cset numid=59 3 amixer cset numid=60 2 amixer cset numid=61 1 item 1 indicates DMIC refuses to take more than 2 channels using arecord. item 2 indicates DMIC capture no voice data if 2 DMIC pins (4 channels) are enabled and arecord using 2 channels. amixer does not handle any channel number. And ABE does not overwrite number of channels set by arecord. Do you think it is bug? Thanks. ---henry

1 0

[alsa-devel] [PATCH 1/3] ASoC: Add ADAU1701 codec driver
by Lars-Peter Clausen 10 Jun '11

10 Jun '11

This patch adds support for the Analog Devices ADAU1701 SigmaDSP. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/Kconfig | 7 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adau1701.c | 547 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adau1701.h | 17 ++ 4 files changed, 572 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/adau1701.c create mode 100644 sound/soc/codecs/adau1701.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..f745e55 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -130,7 +131,11 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate - + +config SND_SOC_ADAU1701 + select SIGMA + tristate + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..30a4c63 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-adau1701-objs := adau1701.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -95,6 +96,7 @@ obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c new file mode 100644 index 0000000..df2b1c3 --- /dev/null +++ b/sound/soc/codecs/adau1701.c @@ -0,0 +1,547 @@ +/* + * Driver for ADAU1701 SigmaDSP processor + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * based on an inital version by Cliff Cai <cliff.cai(a)analog.com> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/sigma.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau1701.h" + +#define ADAU1701_DSPCTRL 0x1c +#define ADAU1701_SEROCTL 0x1e +#define ADAU1701_SERICTL 0x1f + +#define ADAU1701_AUXNPOW 0x22 + +#define ADAU1701_OSCIPOW 0x26 +#define ADAU1701_DACSET 0x27 + +#define ADAU1701_NUM_REGS 0x28 + +#define ADAU1701_DSPCTRL_CR (1 << 2) +#define ADAU1701_DSPCTRL_DAM (1 << 3) +#define ADAU1701_DSPCTRL_ADM (1 << 4) +#define ADAU1701_DSPCTRL_SR_48 0x00 +#define ADAU1701_DSPCTRL_SR_96 0x01 +#define ADAU1701_DSPCTRL_SR_192 0x02 +#define ADAU1701_DSPCTRL_SR_MASK 0x03 + +#define ADAU1701_SEROCTL_INV_LRCLK 0x2000 +#define ADAU1701_SEROCTL_INV_BCLK 0x1000 +#define ADAU1701_SEROCTL_MASTER 0x0800 + +#define ADAU1701_SEROCTL_OBF16 0x0000 +#define ADAU1701_SEROCTL_OBF8 0x0200 +#define ADAU1701_SEROCTL_OBF4 0x0400 +#define ADAU1701_SEROCTL_OBF2 0x0600 +#define ADAU1701_SEROCTL_OBF_MASK 0x0600 + +#define ADAU1701_SEROCTL_OLF1024 0x0000 +#define ADAU1701_SEROCTL_OLF512 0x0080 +#define ADAU1701_SEROCTL_OLF256 0x0100 +#define ADAU1701_SEROCTL_OLF_MASK 0x0180 + +#define ADAU1701_SEROCTL_MSB_DEALY1 0x0000 +#define ADAU1701_SEROCTL_MSB_DEALY0 0x0004 +#define ADAU1701_SEROCTL_MSB_DEALY8 0x0008 +#define ADAU1701_SEROCTL_MSB_DEALY12 0x000c +#define ADAU1701_SEROCTL_MSB_DEALY16 0x0010 +#define ADAU1701_SEROCTL_MSB_DEALY_MASK 0x001c + +#define ADAU1701_SEROCTL_WORD_LEN_24 0x0000 +#define ADAU1701_SEROCTL_WORD_LEN_20 0x0001 +#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010 +#define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003 + +#define ADAU1701_AUXNPOW_VBPD 0x40 +#define ADAU1701_AUXNPOW_VRPD 0x20 + +#define ADAU1701_SERICTL_I2S 0 +#define ADAU1701_SERICTL_LEFTJ 1 +#define ADAU1701_SERICTL_TDM 2 +#define ADAU1701_SERICTL_RIGHTJ_24 3 +#define ADAU1701_SERICTL_RIGHTJ_20 4 +#define ADAU1701_SERICTL_RIGHTJ_18 5 +#define ADAU1701_SERICTL_RIGHTJ_16 6 +#define ADAU1701_SERICTL_MODE_MASK 7 +#define ADAU1701_SERICTL_INV_BCLK BIT(3) +#define ADAU1701_SERICTL_INV_LRCLK BIT(4) + +#define ADAU1701_OSCIPOW_OPD 0x04 +#define ADAU1701_DACSET_DACINIT 1 + +#define ADAU1701_FIRMWARE "adau1701.bin" + +struct adau1701 { + unsigned int dai_fmt; +}; + +static const struct snd_kcontrol_new adau1701_controls[] = { + SOC_SINGLE("Master Capture Switch", ADAU1701_DSPCTRL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau1701_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC0", "Playback", ADAU1701_AUXNPOW, 3, 1), + SND_SOC_DAPM_DAC("DAC1", "Playback", ADAU1701_AUXNPOW, 2, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", ADAU1701_AUXNPOW, 1, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", ADAU1701_AUXNPOW, 0, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", ADAU1701_AUXNPOW, 7, 1), + + SND_SOC_DAPM_OUTPUT("OUT0"), + SND_SOC_DAPM_OUTPUT("OUT1"), + SND_SOC_DAPM_OUTPUT("OUT2"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_INPUT("IN0"), + SND_SOC_DAPM_INPUT("IN1"), +}; + +static const struct snd_soc_dapm_route adau1701_dapm_routes[] = { + { "OUT0", NULL, "DAC0" }, + { "OUT1", NULL, "DAC1" }, + { "OUT2", NULL, "DAC2" }, + { "OUT3", NULL, "DAC3" }, + + { "ADC", NULL, "IN0" }, + { "ADC", NULL, "IN1" }, +}; + +static unsigned int adau1701_register_size(unsigned int reg) +{ + switch (reg) { + case ADAU1701_DSPCTRL: + case ADAU1701_SEROCTL: + case ADAU1701_AUXNPOW: + case ADAU1701_OSCIPOW: + case ADAU1701_DACSET: + return 2; + case ADAU1701_SERICTL: + return 1; + } + + return 0; +} + +static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + unsigned int i, ret; + unsigned int size; + uint8_t buf[4]; + + size = adau1701_register_size(reg); + if (size == 0) { + dev_err(codec->dev, "Unsupported register address: %d\n", reg); + return -EINVAL; + } + + snd_soc_cache_write(codec, reg, value); + + buf[0] = 0x08; + buf[1] = reg; + + for (i = size + 1; i >= 2; --i) { + buf[i] = value; + value >>= 8; + } + + ret = i2c_master_send(to_i2c_client(codec->dev), buf, size + 2); + if (ret == size + 2) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int value; + unsigned int ret; + + ret = snd_soc_cache_read(codec, reg, &value); + if (ret) + return ret; + + return value; +} + +static int adau1701_load_firmware(struct snd_soc_codec *codec) +{ + return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE); +} + +static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK; + unsigned int val; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU1701_SEROCTL_WORD_LEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAU1701_SEROCTL_WORD_LEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU1701_SEROCTL_WORD_LEN_24; + break; + default: + return -EINVAL; + } + + if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) { + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val |= ADAU1701_SEROCTL_MSB_DEALY16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= ADAU1701_SEROCTL_MSB_DEALY12; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= ADAU1701_SEROCTL_MSB_DEALY8; + break; + } + mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; + } + + snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + + return 0; +} + +static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU1701_SERICTL_RIGHTJ_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAU1701_SERICTL_RIGHTJ_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU1701_SERICTL_RIGHTJ_24; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_SERICTL, + ADAU1701_SERICTL_MODE_MASK, val); + + return 0; +} + +static int adau1701_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_codec *codec = rtd->codec; + snd_pcm_format_t format; + unsigned int val; + + switch (params_rate(params)) { + case 192000: + val = ADAU1701_DSPCTRL_SR_192; + break; + case 96000: + val = ADAU1701_DSPCTRL_SR_96; + break; + case 48000: + val = ADAU1701_DSPCTRL_SR_48; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + ADAU1701_DSPCTRL_SR_MASK, val); + + format = params_format(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return adau1701_set_playback_pcm_format(codec, format); + else + return adau1701_set_capture_pcm_format(codec, format); +} + +static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int serictl = 0x00, seroctl = 0x00; + bool invert_lrclk; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* master, 64-bits per sample, 1 frame per sample */ + seroctl |= ADAU1701_SEROCTL_MASTER | ADAU1701_SEROCTL_OBF16 + | ADAU1701_SEROCTL_OLF1024; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_lrclk = true; + break; + case SND_SOC_DAIFMT_IB_NF: + invert_lrclk = false; + serictl |= ADAU1701_SERICTL_INV_BCLK; + seroctl |= ADAU1701_SEROCTL_INV_BCLK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert_lrclk = true; + serictl |= ADAU1701_SERICTL_INV_BCLK; + seroctl |= ADAU1701_SEROCTL_INV_BCLK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + serictl |= ADAU1701_SERICTL_LEFTJ; + seroctl |= ADAU1701_SEROCTL_MSB_DEALY0; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serictl |= ADAU1701_SERICTL_RIGHTJ_24; + seroctl |= ADAU1701_SEROCTL_MSB_DEALY8; + invert_lrclk = !invert_lrclk; + break; + default: + return -EINVAL; + } + + if (invert_lrclk) { + seroctl |= ADAU1701_SEROCTL_INV_LRCLK; + serictl |= ADAU1701_SERICTL_INV_LRCLK; + } + + adau1701->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + snd_soc_write(codec, ADAU1701_SERICTL, serictl); + snd_soc_update_bits(codec, ADAU1701_SEROCTL, + ~ADAU1701_SEROCTL_WORD_LEN_MASK, seroctl); + + return 0; +} + +static int adau1701_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* Enable VREF and VREF buffer */ + snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + break; + case SND_SOC_BIAS_OFF: + /* Disable VREF and VREF buffer */ + snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int mask = ADAU1701_DSPCTRL_DAM; + unsigned int val; + + if (mute) + val = 0; + else + val = mask; + + snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + + return 0; +} + +static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir) +{ + unsigned int val; + + switch (clk_id) { + case ADAU1701_CLK_SRC_OSC: + val = 0x0; + break; + case ADAU1701_CLK_SRC_MCLK: + val = ADAU1701_OSCIPOW_OPD; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + + return 0; +} + +#define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) + +#define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops adau1701_dai_ops = { + .set_fmt = adau1701_set_dai_fmt, + .hw_params = adau1701_hw_params, + .digital_mute = adau1701_digital_mute, +}; + +static struct snd_soc_dai_driver adau1701_dai = { + .name = "adau1701", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = ADAU1701_RATES, + .formats = ADAU1701_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 8, + .rates = ADAU1701_RATES, + .formats = ADAU1701_FORMATS, + }, + .ops = &adau1701_dai_ops, + .symmetric_rates = 1, +}; + +static int adau1701_probe(struct snd_soc_codec *codec) +{ + int ret; + + codec->dapm.idle_bias_off = 1; + + ret = adau1701_load_firmware(codec); + if (ret) + dev_warn(codec->dev, "Failed to load firmware\n"); + + snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); + snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); + + return 0; +} + +static struct snd_soc_codec_driver adau1701_codec_drv = { + .probe = adau1701_probe, + .set_bias_level = adau1701_set_bias_level, + + .reg_cache_size = ADAU1701_NUM_REGS, + .reg_word_size = sizeof(u16), + + .controls = adau1701_controls, + .num_controls = ARRAY_SIZE(adau1701_controls), + .dapm_widgets = adau1701_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1701_dapm_widgets), + .dapm_routes = adau1701_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1701_dapm_routes), + + .write = adau1701_write, + .read = adau1701_read, + + .set_sysclk = adau1701_set_sysclk, +}; + +static __devinit int adau1701_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct adau1701 *adau1701; + int ret; + + adau1701 = kzalloc(sizeof(*adau1701), GFP_KERNEL); + if (!adau1701) + return -ENOMEM; + + i2c_set_clientdata(client, adau1701); + ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, + &adau1701_dai, 1); + if (ret < 0) + kfree(adau1701); + + return ret; +} + +static __devexit int adau1701_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id adau1701_i2c_id[] = { + { "adau1701", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); + +static struct i2c_driver adau1701_i2c_driver = { + .driver = { + .name = "adau1701", + .owner = THIS_MODULE, + }, + .probe = adau1701_i2c_probe, + .remove = __devexit_p(adau1701_i2c_remove), + .id_tabl = adau1701_i2c_id, +}; + +static int __init adau1701_init(void) +{ + return i2c_add_driver(&adau1701_i2c_driver); +} +module_init(adau1701_init); + +static void __exit adau1701_exit(void) +{ + i2c_del_driver(&adau1701_i2c_driver); +} +module_exit(adau1701_exit); + +MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1701.h b/sound/soc/codecs/adau1701.h new file mode 100644 index 0000000..8d0949a --- /dev/null +++ b/sound/soc/codecs/adau1701.h @@ -0,0 +1,17 @@ +/* + * header file for ADAU1701 SigmaDSP processor + * + * Copyright 2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _ADAU1701_H +#define _ADAU1701_H + +enum adau1701_clk_src { + ADAU1701_CLK_SRC_OSC, + ADAU1701_CLK_SRC_MCLK, +}; + +#endif -- 1.7.2.5

4 22

[alsa-devel] [PATCHv2] ASoC: dapm - Refactor widget IO functions in preparation for platform widgets.
by Liam Girdwood 10 Jun '11

10 Jun '11

Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP devices have mixer components that can alter power usage and hence require full DAPM support. This provides a generic widget IO operation wrapper in preparation for future patches that implement platform driver DAPM. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 41 +++++++++++++++++++++++++++++++++++------ 1 files changed, 35 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 34106bc..ea6c2a5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,6 +124,35 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) +{ + if (w->codec) + return snd_soc_read(w->codec, reg); + return 0; +} + +static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) +{ + if (w->codec) + return snd_soc_write(w->codec, reg, val); + return 0; +} + +static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, + unsigned short reg, unsigned int mask, unsigned int value) +{ + int change; + unsigned int old, new; + + old = soc_widget_read(w, reg); + new = (old & ~mask) | value; + change = old != new; + if (change) + soc_widget_write(w, reg, new); + + return change; +} + /** * snd_soc_dapm_set_bias_level - set the bias level for the system * @dapm: DAPM context @@ -181,7 +210,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = snd_soc_read(w->codec, reg); + val = soc_widget_read(w, reg); val = (val >> shift) & mask; if ((invert && !val) || (!invert && val)) @@ -197,7 +226,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; - val = snd_soc_read(w->codec, e->reg); + val = soc_widget_read(w, e->reg); item = (val >> e->shift_l) & (bitmask - 1); p->connect = 0; @@ -227,7 +256,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, w->kcontrol_news[i].private_value; int val, item; - val = snd_soc_read(w->codec, e->reg); + val = soc_widget_read(w, e->reg); val = (val >> e->shift_l) & e->mask; for (item = 0; item < e->max; item++) { if (val == e->values[item]) @@ -669,7 +698,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - snd_soc_update_bits(w->codec, -(w->reg + 1), + soc_widget_update_bits(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -877,7 +906,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - snd_soc_update_bits(dapm->codec, reg, mask, value); + soc_widget_update_bits(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1878,7 +1907,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) /* Read the initial power state from the device */ if (w->reg >= 0) { - val = snd_soc_read(w->codec, w->reg); + val = soc_widget_read(w, w->reg); val &= 1 << w->shift; if (w->invert) val = !val; -- 1.7.4.1

1 0

[alsa-devel] [PATCH] ALSA: hda: Fix inaudible internal speakers on CyberpowerPC Gamer Xplorer N57001 laptop
by Daniel Chen 10 Jun '11

10 Jun '11

Hi, Attached is a patch against current sound-2.6 master HEAD that resolves the original reporter's inaudible internal speakers by using the model=asus quirk. Please apply. Cheers, -Dan

2 5

[alsa-devel] Regression in 3.0: oops with module RO/NX in snd_printk when loading snd-hda-intel (gold linker bug?)
by Maarten Lankhorst 10 Jun '11

10 Jun '11

Since 3.0.0rc1 I get an oops in alc_auto_parse_customize_define, when it's calling snd_printk, I have module NX/RO enabled, and linked the kernel with gold. the only command I do is 'modprobe snd debug=8; modprobe snd-hda-intel' after init=/bin/bash ld --version: GNU gold (GNU Binutils for Ubuntu 2.21.0.20110327) 1.11 [ 43.327003] HDA Intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 43.327012] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 43.327063] HDA Intel 0000:00:1b.0: irq 46 for MSI/MSI-X [ 43.327090] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 43.327097] BUG: unable to handle kernel paging request at ffffffffa003c80a [ 43.327253] IP: [<ffffffffa002f791>] __snd_printk+0x71/0x110 [snd] [ 43.327392] PGD 1a25067 PUD 1a29063 PMD 231bce067 PTE 0 [ 43.327523] Oops: 0000 [#1] PREEMPT SMP [ 43.327620] CPU 1 [ 43.327660] Modules linked in: snd_hda_intel(+) snd_hda_codec snd_hwdep snd_pcm_oss snd_pcm snd_page_alloc snd_seq_midi snd_seq_midi_event snd_rawmidi snd_seq snd_seq_device snd_timer snd soundcore [ 43.328104] [ 43.328134] Pid: 149, comm: modprobe Not tainted 3.0.0-rc2-patser+ #34 To Be Filled By O.E.M. To Be Filled By O.E.M./P67 Pro3 [ 43.328393] RIP: 0010:[<ffffffffa002f791>] [<ffffffffa002f791>] __snd_printk+0x71/0x110 [snd] [ 43.328581] RSP: 0018:ffff8802310a9c38 EFLAGS: 00010216 [ 43.328696] RAX: ffff8802310a9c78 RBX: 0000000000000002 RCX: ffffffffa00bea30 [ 43.328851] RDX: 0000000000000a1a RSI: ffffffffa00be00a RDI: ffffffffa00be00a [ 43.329006] RBP: ffff8802310a9cb8 R08: 0000000000004401 R09: 0000000000000040 [ 43.329161] R10: 00000000000000d8 R11: 0000000000000002 R12: ffffffffa00bea30 [ 43.329316] R13: ffff88023110e800 R14: ffff88023110e800 R15: 0000000000004401 [ 43.329472] FS: 00007fc62e05c720(0000) GS:ffff88023f480000(0000) knlGS:0000000000000000 [ 43.329647] CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b [ 43.329772] CR2: ffffffffa003c80a CR3: 00000002310b6000 CR4: 00000000000406e0 [ 43.329927] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 43.330082] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 43.330237] Process modprobe (pid: 149, threadinfo ffff8802310a8000, task ffff8802311216f0) [ 43.330418] Stack: [ 43.330462] ffff8802310a9c58 ffffffff81a05080 ffff880200000020 ffff8802310a9cc8 [ 43.330640] ffff8802310a9c78 ffffffff810bab78 ffff8802310a9c98 0000000005411735 [ 43.330815] ffff88023110e800 40ff880232ad3000 ffff880232ad3000 ffff880232ad3000 [ 43.330991] Call Trace: [ 43.331044] [<ffffffff810bab78>] ? synchronize_irq+0x58/0xf0 [ 43.336434] [<ffffffffa00bd191>] azx_probe+0x48b/0x9fb [snd_hda_intel] [ 43.341867] [<ffffffff81317e8f>] local_pci_probe+0x5f/0xd0 [ 43.347300] [<ffffffff813185f9>] pci_device_probe+0x119/0x120 [ 43.352744] [<ffffffff813b27aa>] ? driver_sysfs_add+0x7a/0xb0 [ 43.358193] [<ffffffff813b2b23>] driver_probe_device+0xa3/0x2a0 [ 43.363649] [<ffffffff813b2dcb>] __driver_attach+0xab/0xb0 [ 43.369038] [<ffffffff813b2d20>] ? driver_probe_device+0x2a0/0x2a0 [ 43.374387] [<ffffffff813b193e>] bus_for_each_dev+0x5e/0x90 [ 43.379725] [<ffffffff813b272e>] driver_attach+0x1e/0x20 [ 43.385029] [<ffffffff813b22d0>] bus_add_driver+0x160/0x2b0 [ 43.390172] [<ffffffffa0021000>] ? 0xffffffffa0020fff [ 43.395151] [<ffffffff813b3394>] driver_register+0x74/0x130 [ 43.400133] [<ffffffffa0021000>] ? 0xffffffffa0020fff [ 43.405010] [<ffffffff81318876>] __pci_register_driver+0x56/0xd0 [ 43.409827] [<ffffffff8102b653>] ? set_memory_nx+0x43/0x50 [ 43.414645] [<ffffffffa002101e>] alsa_card_azx_init+0x1e/0x1000 [snd_hda_intel] [ 43.419550] [<ffffffff810002d5>] do_one_initcall+0x45/0x170 [ 43.424489] [<ffffffff8109505a>] sys_init_module+0xfa/0x250 [ 43.429420] [<ffffffff8157c52b>] system_call_fastpath+0x16/0x1b [ 43.434403] Code: 00 00 00 75 6b 48 8b 5d f0 4c 8b 65 f8 c9 c3 48 8d 45 10 c7 45 90 20 00 00 00 48 89 45 98 48 8d 45 c0 48 89 45 a0 80 3e 2f 74 78 <8b> 05 73 d0 00 00 85 db 89 45 b0 48 c7 c6 b3 23 03 a0 48 c7 c0 [ 43.444891] RIP [<ffffffffa002f791>] __snd_printk+0x71/0x110 [snd] [ 43.450012] RSP <ffff8802310a9c38> [ 43.450013] CR2: ffffffffa003c80a [ 43.450014] ---[ end trace 890ffb2c861edf22 ]--- everything sound loaded modular, all hda codecs too. CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y module RO/NX is set, which I assume is causing the problem. I don't understand the oops, but disassembly of the specific offset is below, from objdump -S: 7de1 is the location of the crash #ifdef CONFIG_SND_VERBOSE_PRINTK /* strip the leading path if the given path is absolute */ static const char *sanity_file_name(const char *path) { if (*path == '/') 7ddc: 80 3e 2f cmpb $0x2f,(%rsi) 7ddf: 74 78 je 7e59 <__snd_printk+0xe9> /* print file and line with a certain printk prefix */ static int print_snd_pfx(unsigned int level, const char *path, int line, const char *format) { const char *file = sanity_file_name(path); char tmp[] = "<0>"; 7de1: 8b 05 00 00 00 00 mov 0x0(%rip),%eax # 7de7 <__snd_printk+0x77> const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT; 7de7: 85 db test %ebx,%ebx /* print file and line with a certain printk prefix */ static int print_snd_pfx(unsigned int level, const char *path, int line, const char *format) { const char *file = sanity_file_name(path); char tmp[] = "<0>"; 7de9: 89 45 b0 mov %eax,-0x50(%rbp) const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT; 7dec: 48 c7 c6 00 00 00 00 mov $0x0,%rsi 7df3: 48 c7 c0 00 00 00 00 mov $0x0,%rax 7dfa: 48 0f 44 f0 cmove %rax,%rsi --- Seems to me like it's segfaulting when loading <0>\0 (4 bytes, so fits in an integer) to %eax. building all modules with the normal binutils works fine, and makes this crash disappear. Is this bug interesting enough that I'll see if I can find the commit which caused this regression? ~Maarten

2 3

[alsa-devel] ASoC pr_debug disabled in soc-core.c
by Lambrecht Jürgen 10 Jun '11

10 Jun '11

Hello, I am rather new to linux, and I don't find why 'pr_debug' is disabled in /sound/soc/soc-core.c but enabled in /drivers/base/dd.c (can see the messages with 'dmesg' or with setting 'loglevel=8'). (printk(KERN_DEBUG ...) does work though) I enabled those configuration options: - DEBUG_DRIVER - SND_DEBUG - SND_VERBOSE_PRINTK - DEBUG_KERNEL (SND_PCM_XRUN_DEBUG is disabled) What am I missing? Regards, Jürgen -- Jürgen Lambrecht R&D Associate Tel: +32 (0)51 303045 Fax: +32 (0)51 310670 http://www.televic-rail.com Televic Rail NV - Leo Bekaertlaan 1 - 8870 Izegem - Belgium Company number 0825.539.581 - RPR Kortrijk

1 0

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
by Jarkko Nikula 10 Jun '11

10 Jun '11

Attempt to change McBSP CLKS source while another stream is active is not safe after commit d135865 ("OMAP: McBSP: implement functional clock switching via clock framework") in 2.6.37. CLKS parent clock switching using clock framework have to idle the McBSP before switching and then activate it again. This short break can cause a DMA transaction error to already running stream which halts and recovers only by closing and restarting the stream. This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a severe timeout error and does BUG_ON(). Fix this by checking is the McBSP active in omap_mcbsp_dai_set_dai_sysclk before attempting to change any clocking configuration. This test should have been here just from the beginning anyway. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: stable(a)kernel.org --- For 2.6.37 and onward. This issue is valid to those machine drivers that use the McBSP as a DAI link master and are configuring the CLKS source (like sound/soc/omap/omap3pandora.c in mainline). --- sound/soc/omap/omap-mcbsp.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 07b7723..fc896c1 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,6 +516,9 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + if (mcbsp_data->active) + return 0; + /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || -- 1.7.4.4

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 10 Jun '11

10 Jun '11

The following changes since commit d21685ec258f803d3badae5eae821383a34815a9: Merge branch 'for-2.6.40' into for-2.6.41 (2011-05-30 10:54:18 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Due to the merge up of 3.0-rc1 I wasn't able to get a sensible changelog or diff out of git easily - pretty much everything has been sitting in -next except for a small Samsung patch.

2 2

[alsa-devel] ASoC updates for 3.0
by Mark Brown 10 Jun '11

10 Jun '11

The following changes since commit 59c5f46fbe01a00eedf54a23789634438bb80603: Linux 3.0-rc2 (2011-06-06 18:06:33 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 This also includes a merge up of 3.0-rc1. The core change is the relatively large but straightforward patch to pass the DAPM context around when creating new mixers. Joachim Eastwood (1): ASoC: atmel_ssc: Don't try to free ssc if request failed Lars-Peter Clausen (3): ASoC: AD1836: Fix setting the PCM format ASoC: Blackfin: bf5xx-ad1836: Fix codec device name ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context Mark Brown (6): Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into for-3.0 ASoC: Only update SYSCLK_ENA when pausing WM8915 SYSCLK ASoC: Add missing break in WM8915 FLL source selection ASoC: Check for NULL register bank in snd_soc_get_cache_val() ASoC: Fix WM8962 headphone volume update for use of advanced caches ASoC: WM8804 does not support sample rates below 32kHz Sangbeom Kim (1): ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register Timur Tabi (1): ASoC: fsl: fix initialization of DMA buffers sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- sound/soc/blackfin/bf5xx-ad1836.c | 4 ++-- sound/soc/codecs/ad1836.c | 14 +++++++------- sound/soc/codecs/ad1836.h | 6 ++++++ sound/soc/codecs/wm8804.c | 9 +++++++-- sound/soc/codecs/wm8915.c | 3 ++- sound/soc/codecs/wm8962.c | 4 ++-- sound/soc/codecs/wm_hubs.c | 8 ++++---- sound/soc/fsl/fsl_dma.c | 9 +++++---- sound/soc/samsung/i2s.c | 4 ++-- sound/soc/soc-cache.c | 3 +++ sound/soc/soc-dapm.c | 22 ++++++++++++---------- 12 files changed, 55 insertions(+), 36 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register
by Sangbeom Kim 10 Jun '11

10 Jun '11

If DMA active status should be checked, I2SCON register should be referenced. In this patch, Fix the incorrect referencing of I2SCON register. Reported-by : Lakkyung Jung <lakkyung.jung(a)samsung.com> Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ffa09b3..992a732 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -191,7 +191,7 @@ static inline bool tx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SMOD); + active = readl(i2s->addr + I2SCON); if (is_secondary(i2s)) active &= CON_TXSDMA_ACTIVE; @@ -223,7 +223,7 @@ static inline bool rx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SMOD) & CON_RXDMA_ACTIVE; + active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE; return active ? true : false; } -- 1.7.1

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Re: [alsa-devel] [PATCH 1/1] ASoC: codecs: max98088: Combined DAI operations for DAI1 and DAI2.
by Jinyoung Park 10 Jun '11

10 Jun '11

Submitting again the patch what applied Peter's comments. >> These added local variables and their initialization repeats. >> It may be better to put them in struct max98088_cdata[]. > I got it. I will change register offset into struct max98088_cdata[]. > >> DAI2 does not have an OSR bit in the _CLOCK register. >> In the combined function, this should be conditionally for DAI1 only. > I will fix it. Thanks, Jin. Combined DAI operations for DAI1 and DAI2. For this, added id into DAI runtime data to identify each DAI. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com> --- sound/soc/codecs/max98088.c | 278 +++++++++++-------------------------------- 1 files changed, 69 insertions(+), 209 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ac65a2d..3c0e4f7 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -36,6 +36,14 @@ struct max98088_cdata { unsigned int rate; unsigned int fmt; int eq_sel; + + unsigned short clkmode_reg; + unsigned short clk_reg; + unsigned short clkcfg_hi_reg; + unsigned short clkcfg_lo_reg; + unsigned short fmt_reg; + unsigned short play_lvl_reg; + unsigned short filters_reg; }; struct max98088_priv { @@ -1253,29 +1261,26 @@ static inline int rate_value(int rate, u8 *value) return -EINVAL; } -static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, +static int max98088_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); - struct max98088_cdata *cdata; + struct max98088_cdata *cdata = &max98088->dai[dai->id - 1]; unsigned long long ni; unsigned int rate; - u8 regval; - - cdata = &max98088->dai[0]; + u8 val; rate = params_rate(params); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, - M98088_DAI_WS, 0); + snd_soc_update_bits(codec, cdata->fmt_reg, M98088_DAI_WS, 0); break; case SNDRV_PCM_FORMAT_S24_LE: - snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, - M98088_DAI_WS, M98088_DAI_WS); + snd_soc_update_bits(codec, cdata->fmt_reg, M98088_DAI_WS, + M98088_DAI_WS); break; default: return -EINVAL; @@ -1283,16 +1288,14 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, M98088_SHDNRUN, 0); - if (rate_value(rate, &regval)) + if (rate_value(rate, &val)) return -EINVAL; - snd_soc_update_bits(codec, M98088_REG_11_DAI1_CLKMODE, - M98088_CLKMODE_MASK, regval); + snd_soc_update_bits(codec, cdata->clkmode_reg, M98088_CLKMODE_MASK, val); cdata->rate = rate; /* Configure NI when operating as master */ - if (snd_soc_read(codec, M98088_REG_14_DAI1_FORMAT) - & M98088_DAI_MAS) { + if (snd_soc_read(codec, cdata->fmt_reg) & M98088_DAI_MAS) { if (max98088->sysclk == 0) { dev_err(codec->dev, "Invalid system clock frequency\n"); return -EINVAL; @@ -1300,86 +1303,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; do_div(ni, (unsigned long long int)max98088->sysclk); - snd_soc_write(codec, M98088_REG_12_DAI1_CLKCFG_HI, - (ni >> 8) & 0x7F); - snd_soc_write(codec, M98088_REG_13_DAI1_CLKCFG_LO, - ni & 0xFF); + snd_soc_write(codec, cdata->clkcfg_hi_reg, (ni >> 8) & 0x7F); + snd_soc_write(codec, cdata->clkcfg_lo_reg, ni & 0xFF); } /* Update sample rate mode */ if (rate < 50000) - snd_soc_update_bits(codec, M98088_REG_18_DAI1_FILTERS, - M98088_DAI_DHF, 0); + snd_soc_update_bits(codec, cdata->filters_reg, M98088_DAI_DHF, 0); else - snd_soc_update_bits(codec, M98088_REG_18_DAI1_FILTERS, - M98088_DAI_DHF, M98088_DAI_DHF); - - snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, M98088_SHDNRUN, - M98088_SHDNRUN); - - return 0; -} - -static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); - struct max98088_cdata *cdata; - unsigned long long ni; - unsigned int rate; - u8 regval; - - cdata = &max98088->dai[1]; - - rate = params_rate(params); - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, - M98088_DAI_WS, 0); - break; - case SNDRV_PCM_FORMAT_S24_LE: - snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, - M98088_DAI_WS, M98088_DAI_WS); - break; - default: - return -EINVAL; - } - - snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, M98088_SHDNRUN, 0); - - if (rate_value(rate, &regval)) - return -EINVAL; - - snd_soc_update_bits(codec, M98088_REG_19_DAI2_CLKMODE, - M98088_CLKMODE_MASK, regval); - cdata->rate = rate; - - /* Configure NI when operating as master */ - if (snd_soc_read(codec, M98088_REG_1C_DAI2_FORMAT) - & M98088_DAI_MAS) { - if (max98088->sysclk == 0) { - dev_err(codec->dev, "Invalid system clock frequency\n"); - return -EINVAL; - } - ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) - * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); - snd_soc_write(codec, M98088_REG_1A_DAI2_CLKCFG_HI, - (ni >> 8) & 0x7F); - snd_soc_write(codec, M98088_REG_1B_DAI2_CLKCFG_LO, - ni & 0xFF); - } - - /* Update sample rate mode */ - if (rate < 50000) - snd_soc_update_bits(codec, M98088_REG_20_DAI2_FILTERS, - M98088_DAI_DHF, 0); - else - snd_soc_update_bits(codec, M98088_REG_20_DAI2_FILTERS, - M98088_DAI_DHF, M98088_DAI_DHF); + snd_soc_update_bits(codec, cdata->filters_reg, M98088_DAI_DHF, + M98088_DAI_DHF); snd_soc_update_bits(codec, M98088_REG_51_PWR_SYS, M98088_SHDNRUN, M98088_SHDNRUN); @@ -1423,16 +1356,13 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, return 0; } -static int max98088_dai1_set_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) +static int max98088_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_codec *codec = dai->codec; struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); - struct max98088_cdata *cdata; - u8 reg15val; - u8 reg14val = 0; - - cdata = &max98088->dai[0]; + struct max98088_cdata *cdata = &max98088->dai[dai->id - 1]; + u8 clk_val; + u8 fmt_val = 0; if (fmt != cdata->fmt) { cdata->fmt = fmt; @@ -1440,14 +1370,12 @@ static int max98088_dai1_set_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: /* Slave mode PLL */ - snd_soc_write(codec, M98088_REG_12_DAI1_CLKCFG_HI, - 0x80); - snd_soc_write(codec, M98088_REG_13_DAI1_CLKCFG_LO, - 0x00); + snd_soc_write(codec, cdata->clkcfg_hi_reg, 0x80); + snd_soc_write(codec, cdata->clkcfg_lo_reg, 0x00); break; case SND_SOC_DAIFMT_CBM_CFM: /* Set to master mode */ - reg14val |= M98088_DAI_MAS; + fmt_val |= M98088_DAI_MAS; break; case SND_SOC_DAIFMT_CBS_CFM: case SND_SOC_DAIFMT_CBM_CFS: @@ -1458,7 +1386,7 @@ static int max98088_dai1_set_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - reg14val |= M98088_DAI_DLY; + fmt_val |= M98088_DAI_DLY; break; case SND_SOC_DAIFMT_LEFT_J: break; @@ -1470,127 +1398,43 @@ static int max98088_dai1_set_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_NB_IF: - reg14val |= M98088_DAI_WCI; + fmt_val |= M98088_DAI_WCI; break; case SND_SOC_DAIFMT_IB_NF: - reg14val |= M98088_DAI_BCI; + fmt_val |= M98088_DAI_BCI; break; case SND_SOC_DAIFMT_IB_IF: - reg14val |= M98088_DAI_BCI|M98088_DAI_WCI; + fmt_val |= M98088_DAI_BCI|M98088_DAI_WCI; break; default: return -EINVAL; } - snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, + snd_soc_update_bits(codec, cdata->fmt_reg, M98088_DAI_MAS | M98088_DAI_DLY | M98088_DAI_BCI | - M98088_DAI_WCI, reg14val); + M98088_DAI_WCI, fmt_val); - reg15val = M98088_DAI_BSEL64; - if (max98088->digmic) - reg15val |= M98088_DAI_OSR64; - snd_soc_write(codec, M98088_REG_15_DAI1_CLOCK, reg15val); + clk_val = M98088_DAI_BSEL64; + if ((dai->id == 1) && max98088->digmic) + clk_val |= M98088_DAI_OSR64; + snd_soc_write(codec, cdata->clk_reg, clk_val); } return 0; } -static int max98088_dai2_set_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) +static int max98088_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_codec *codec = dai->codec; struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); - struct max98088_cdata *cdata; - u8 reg1Cval = 0; - - cdata = &max98088->dai[1]; - - if (fmt != cdata->fmt) { - cdata->fmt = fmt; - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* Slave mode PLL */ - snd_soc_write(codec, M98088_REG_1A_DAI2_CLKCFG_HI, - 0x80); - snd_soc_write(codec, M98088_REG_1B_DAI2_CLKCFG_LO, - 0x00); - break; - case SND_SOC_DAIFMT_CBM_CFM: - /* Set to master mode */ - reg1Cval |= M98088_DAI_MAS; - break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: - default: - dev_err(codec->dev, "Clock mode unsupported"); - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - reg1Cval |= M98088_DAI_DLY; - break; - case SND_SOC_DAIFMT_LEFT_J: - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - reg1Cval |= M98088_DAI_WCI; - break; - case SND_SOC_DAIFMT_IB_NF: - reg1Cval |= M98088_DAI_BCI; - break; - case SND_SOC_DAIFMT_IB_IF: - reg1Cval |= M98088_DAI_BCI|M98088_DAI_WCI; - break; - default: - return -EINVAL; - } - - snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, - M98088_DAI_MAS | M98088_DAI_DLY | M98088_DAI_BCI | - M98088_DAI_WCI, reg1Cval); - - snd_soc_write(codec, M98088_REG_1D_DAI2_CLOCK, - M98088_DAI_BSEL64); - } - - return 0; -} - -static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int mute) -{ - struct snd_soc_codec *codec = codec_dai->codec; - int reg; - - if (mute) - reg = M98088_DAI_MUTE; - else - reg = 0; - - snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY, - M98088_DAI_MUTE_MASK, reg); - return 0; -} - -static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) -{ - struct snd_soc_codec *codec = codec_dai->codec; - int reg; + struct max98088_cdata *cdata = &max98088->dai[dai->id - 1]; + int val = 0; if (mute) - reg = M98088_DAI_MUTE; - else - reg = 0; + val = M98088_DAI_MUTE; - snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY, - M98088_DAI_MUTE_MASK, reg); + snd_soc_update_bits(codec, cdata->play_lvl_reg, M98088_DAI_MUTE_MASK, + val); return 0; } @@ -1653,21 +1497,22 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec, static struct snd_soc_dai_ops max98088_dai1_ops = { .set_sysclk = max98088_dai_set_sysclk, - .set_fmt = max98088_dai1_set_fmt, - .hw_params = max98088_dai1_hw_params, - .digital_mute = max98088_dai1_digital_mute, + .set_fmt = max98088_dai_set_fmt, + .hw_params = max98088_dai_hw_params, + .digital_mute = max98088_dai_digital_mute, }; static struct snd_soc_dai_ops max98088_dai2_ops = { .set_sysclk = max98088_dai_set_sysclk, - .set_fmt = max98088_dai2_set_fmt, - .hw_params = max98088_dai2_hw_params, - .digital_mute = max98088_dai2_digital_mute, + .set_fmt = max98088_dai_set_fmt, + .hw_params = max98088_dai_hw_params, + .digital_mute = max98088_dai_digital_mute, }; static struct snd_soc_dai_driver max98088_dai[] = { { .name = "HiFi", + .id = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, @@ -1686,6 +1531,7 @@ static struct snd_soc_dai_driver max98088_dai[] = { }, { .name = "Aux", + .id = 2, .playback = { .stream_name = "Aux Playback", .channels_min = 1, @@ -1977,11 +1823,25 @@ static int max98088_probe(struct snd_soc_codec *codec) cdata->rate = (unsigned)-1; cdata->fmt = (unsigned)-1; cdata->eq_sel = 0; + cdata->clkmode_reg = M98088_REG_11_DAI1_CLKMODE; + cdata->clk_reg = M98088_REG_15_DAI1_CLOCK; + cdata->clkcfg_hi_reg = M98088_REG_12_DAI1_CLKCFG_HI; + cdata->clkcfg_lo_reg = M98088_REG_13_DAI1_CLKCFG_LO; + cdata->fmt_reg = M98088_REG_14_DAI1_FORMAT; + cdata->play_lvl_reg = M98088_REG_2F_LVL_DAI1_PLAY; + cdata->filters_reg = M98088_REG_18_DAI1_FILTERS; cdata = &max98088->dai[1]; cdata->rate = (unsigned)-1; cdata->fmt = (unsigned)-1; cdata->eq_sel = 0; + cdata->clkmode_reg = M98088_REG_19_DAI2_CLKMODE; + cdata->clk_reg = M98088_REG_1D_DAI2_CLOCK; + cdata->clkcfg_hi_reg = M98088_REG_1A_DAI2_CLKCFG_HI; + cdata->clkcfg_lo_reg = M98088_REG_1B_DAI2_CLKCFG_LO; + cdata->fmt_reg = M98088_REG_1C_DAI2_FORMAT; + cdata->play_lvl_reg = M98088_REG_31_LVL_DAI2_PLAY; + cdata->filters_reg = M98088_REG_20_DAI2_FILTERS; max98088->ina_state = 0; max98088->inb_state = 0; -- 1.7.1

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[alsa-devel] [PATCH] sound/atmel_ssc_dai: add a missing space to an error message
by Uwe Kleine-König 10 Jun '11

10 Jun '11

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 7fbfa05..7d6ab47 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -402,7 +402,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S && bits > 16) { printk(KERN_WARNING - "atmel_ssc_dai: sample size %d" + "atmel_ssc_dai: sample size %d " "is too large for I2S\n", bits); return -EINVAL; } -- 1.7.5.3

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[PATCH] Revert sound/pci/*.[ch] to 2.6.39
by Michal Hocko 10 Jun '11

10 Jun '11

Takashi has suggested to revert these files to 2.6.39 to test auto mute breakage in 3.0.0-rc[12] --- sound/pci/es1968.c | 78 ---------- sound/pci/fm801.c | 378 +++++++++++++++++++++++++++++++++++++----------- sound/pci/intel8x0m.c | 4 +- 3 files changed, 293 insertions(+), 167 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index ab0a615..7c17f45 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -112,10 +112,6 @@ #include <sound/ac97_codec.h> #include <sound/initval.h> -#ifdef CONFIG_SND_ES1968_RADIO -#include <sound/tea575x-tuner.h> -#endif - #define CARD_NAME "ESS Maestro1/2" #define DRIVER_NAME "ES1968" @@ -557,10 +553,6 @@ struct es1968 { spinlock_t ac97_lock; struct tasklet_struct hwvol_tq; #endif - -#ifdef CONFIG_SND_ES1968_RADIO - struct snd_tea575x tea; -#endif }; static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id); @@ -2579,63 +2571,6 @@ static int __devinit snd_es1968_input_register(struct es1968 *chip) } #endif /* CONFIG_SND_ES1968_INPUT */ -#ifdef CONFIG_SND_ES1968_RADIO -#define GPIO_DATA 0x60 -#define IO_MASK 4 /* mask register offset from GPIO_DATA - bits 1=unmask write to given bit */ -#define IO_DIR 8 /* direction register offset from GPIO_DATA - bits 0/1=read/write direction */ -/* mask bits for GPIO lines */ -#define STR_DATA 0x0040 /* GPIO6 */ -#define STR_CLK 0x0080 /* GPIO7 */ -#define STR_WREN 0x0100 /* GPIO8 */ -#define STR_MOST 0x0200 /* GPIO9 */ - -static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) -{ - struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = 0; - - val |= (pins & TEA575X_DATA) ? STR_DATA : 0; - val |= (pins & TEA575X_CLK) ? STR_CLK : 0; - val |= (pins & TEA575X_WREN) ? STR_WREN : 0; - - outw(val, io); -} - -static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) -{ - struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = inw(io); - - return (val & STR_DATA) ? TEA575X_DATA : 0 | - (val & STR_MOST) ? TEA575X_MOST : 0; -} - -static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool output) -{ - struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 odir = inw(io + IO_DIR); - - if (output) { - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); - } else { - outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); - outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); - } -} - -static struct snd_tea575x_ops snd_es1968_tea_ops = { - .set_pins = snd_es1968_tea575x_set_pins, - .get_pins = snd_es1968_tea575x_get_pins, - .set_direction = snd_es1968_tea575x_set_direction, -}; -#endif - static int snd_es1968_free(struct es1968 *chip) { #ifdef CONFIG_SND_ES1968_INPUT @@ -2650,10 +2585,6 @@ static int snd_es1968_free(struct es1968 *chip) outw(0, chip->io_port + ESM_PORT_HOST_IRQ); /* disable IRQ */ } -#ifdef CONFIG_SND_ES1968_RADIO - snd_tea575x_exit(&chip->tea); -#endif - if (chip->irq >= 0) free_irq(chip->irq, chip); snd_es1968_free_gameport(chip); @@ -2792,15 +2723,6 @@ static int __devinit snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); -#ifdef CONFIG_SND_ES1968_RADIO - chip->tea.private_data = chip; - chip->tea.ops = &snd_es1968_tea_ops; - strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); - sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea)) - printk(KERN_INFO "es1968: detected TEA575x radio\n"); -#endif - *chip_ret = chip; return 0; diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index a7ec703..e1baad7 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -38,6 +38,7 @@ #ifdef CONFIG_SND_FM801_TEA575X_BOOL #include <sound/tea575x-tuner.h> +#define TEA575X_RADIO 1 #endif MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); @@ -52,7 +53,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * /* * Enable TEA575x tuner * 1 = MediaForte 256-PCS - * 2 = MediaForte 256-PCP + * 2 = MediaForte 256-PCPR * 3 = MediaForte 64-PCR * 16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card * High 16-bits are video (radio) device number + 1 @@ -66,7 +67,7 @@ MODULE_PARM_DESC(id, "ID string for the FM801 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); -MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only)."); +MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only)."); #define TUNER_ONLY (1<<4) #define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) @@ -195,7 +196,7 @@ struct fm801 { spinlock_t reg_lock; struct snd_info_entry *proc_entry; -#ifdef CONFIG_SND_FM801_TEA575X_BOOL +#ifdef TEA575X_RADIO struct snd_tea575x tea; #endif @@ -714,89 +715,310 @@ static int __devinit snd_fm801_pcm(struct fm801 *chip, int device, struct snd_pc * TEA5757 radio */ -#ifdef CONFIG_SND_FM801_TEA575X_BOOL +#ifdef TEA575X_RADIO -/* GPIO to TEA575x maps */ -struct snd_fm801_tea575x_gpio { - u8 data, clk, wren, most; - char *name; -}; +/* 256PCS GPIO numbers */ +#define TEA_256PCS_DATA 1 +#define TEA_256PCS_WRITE_ENABLE 2 /* inverted */ +#define TEA_256PCS_BUS_CLOCK 3 -static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = { - { .data = 1, .clk = 3, .wren = 2, .most = 0, .name = "SF256-PCS" }, - { .data = 1, .clk = 0, .wren = 2, .most = 3, .name = "SF256-PCP" }, - { .data = 2, .clk = 0, .wren = 1, .most = 3, .name = "SF64-PCR" }, -}; +static void snd_fm801_tea575x_256pcs_write(struct snd_tea575x *tea, unsigned int val) +{ + struct fm801 *chip = tea->private_data; + unsigned short reg; + int i = 25; -static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) + spin_lock_irq(&chip->reg_lock); + reg = inw(FM801_REG(chip, GPIO_CTRL)); + /* use GPIO lines and set write enable bit */ + reg |= FM801_GPIO_GS(TEA_256PCS_DATA) | + FM801_GPIO_GS(TEA_256PCS_WRITE_ENABLE) | + FM801_GPIO_GS(TEA_256PCS_BUS_CLOCK); + /* all of lines are in the write direction */ + /* clear data and clock lines */ + reg &= ~(FM801_GPIO_GD(TEA_256PCS_DATA) | + FM801_GPIO_GD(TEA_256PCS_WRITE_ENABLE) | + FM801_GPIO_GD(TEA_256PCS_BUS_CLOCK) | + FM801_GPIO_GP(TEA_256PCS_DATA) | + FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK) | + FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE)); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + + while (i--) { + if (val & (1 << i)) + reg |= FM801_GPIO_GP(TEA_256PCS_DATA); + else + reg &= ~FM801_GPIO_GP(TEA_256PCS_DATA); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + reg |= FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + reg &= ~FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + } + + /* and reset the write enable bit */ + reg |= FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE) | + FM801_GPIO_GP(TEA_256PCS_DATA); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + spin_unlock_irq(&chip->reg_lock); +} + +static unsigned int snd_fm801_tea575x_256pcs_read(struct snd_tea575x *tea) { struct fm801 *chip = tea->private_data; - unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); - struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; + unsigned short reg; + unsigned int val = 0; + int i; + + spin_lock_irq(&chip->reg_lock); + reg = inw(FM801_REG(chip, GPIO_CTRL)); + /* use GPIO lines, set data direction to input */ + reg |= FM801_GPIO_GS(TEA_256PCS_DATA) | + FM801_GPIO_GS(TEA_256PCS_WRITE_ENABLE) | + FM801_GPIO_GS(TEA_256PCS_BUS_CLOCK) | + FM801_GPIO_GD(TEA_256PCS_DATA) | + FM801_GPIO_GP(TEA_256PCS_DATA) | + FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE); + /* all of lines are in the write direction, except data */ + /* clear data, write enable and clock lines */ + reg &= ~(FM801_GPIO_GD(TEA_256PCS_WRITE_ENABLE) | + FM801_GPIO_GD(TEA_256PCS_BUS_CLOCK) | + FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK)); + + for (i = 0; i < 24; i++) { + reg &= ~FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + reg |= FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + val <<= 1; + if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_256PCS_DATA)) + val |= 1; + } - reg &= ~(FM801_GPIO_GP(gpio.data) | - FM801_GPIO_GP(gpio.clk) | - FM801_GPIO_GP(gpio.wren)); + spin_unlock_irq(&chip->reg_lock); + + return val; +} - reg |= (pins & TEA575X_DATA) ? FM801_GPIO_GP(gpio.data) : 0; - reg |= (pins & TEA575X_CLK) ? FM801_GPIO_GP(gpio.clk) : 0; - /* WRITE_ENABLE is inverted */ - reg |= (pins & TEA575X_WREN) ? 0 : FM801_GPIO_GP(gpio.wren); +/* 256PCPR GPIO numbers */ +#define TEA_256PCPR_BUS_CLOCK 0 +#define TEA_256PCPR_DATA 1 +#define TEA_256PCPR_WRITE_ENABLE 2 /* inverted */ + +static void snd_fm801_tea575x_256pcpr_write(struct snd_tea575x *tea, unsigned int val) +{ + struct fm801 *chip = tea->private_data; + unsigned short reg; + int i = 25; + + spin_lock_irq(&chip->reg_lock); + reg = inw(FM801_REG(chip, GPIO_CTRL)); + /* use GPIO lines and set write enable bit */ + reg |= FM801_GPIO_GS(TEA_256PCPR_DATA) | + FM801_GPIO_GS(TEA_256PCPR_WRITE_ENABLE) | + FM801_GPIO_GS(TEA_256PCPR_BUS_CLOCK); + /* all of lines are in the write direction */ + /* clear data and clock lines */ + reg &= ~(FM801_GPIO_GD(TEA_256PCPR_DATA) | + FM801_GPIO_GD(TEA_256PCPR_WRITE_ENABLE) | + FM801_GPIO_GD(TEA_256PCPR_BUS_CLOCK) | + FM801_GPIO_GP(TEA_256PCPR_DATA) | + FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK) | + FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE)); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + + while (i--) { + if (val & (1 << i)) + reg |= FM801_GPIO_GP(TEA_256PCPR_DATA); + else + reg &= ~FM801_GPIO_GP(TEA_256PCPR_DATA); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + reg |= FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + reg &= ~FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + } + /* and reset the write enable bit */ + reg |= FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE) | + FM801_GPIO_GP(TEA_256PCPR_DATA); outw(reg, FM801_REG(chip, GPIO_CTRL)); + spin_unlock_irq(&chip->reg_lock); } -static u8 snd_fm801_tea575x_get_pins(struct snd_tea575x *tea) +static unsigned int snd_fm801_tea575x_256pcpr_read(struct snd_tea575x *tea) { struct fm801 *chip = tea->private_data; - unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); - struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; + unsigned short reg; + unsigned int val = 0; + int i; + + spin_lock_irq(&chip->reg_lock); + reg = inw(FM801_REG(chip, GPIO_CTRL)); + /* use GPIO lines, set data direction to input */ + reg |= FM801_GPIO_GS(TEA_256PCPR_DATA) | + FM801_GPIO_GS(TEA_256PCPR_WRITE_ENABLE) | + FM801_GPIO_GS(TEA_256PCPR_BUS_CLOCK) | + FM801_GPIO_GD(TEA_256PCPR_DATA) | + FM801_GPIO_GP(TEA_256PCPR_DATA) | + FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE); + /* all of lines are in the write direction, except data */ + /* clear data, write enable and clock lines */ + reg &= ~(FM801_GPIO_GD(TEA_256PCPR_WRITE_ENABLE) | + FM801_GPIO_GD(TEA_256PCPR_BUS_CLOCK) | + FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK)); + + for (i = 0; i < 24; i++) { + reg &= ~FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + reg |= FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + val <<= 1; + if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_256PCPR_DATA)) + val |= 1; + } + + spin_unlock_irq(&chip->reg_lock); - return (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 | - (reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0; + return val; } -static void snd_fm801_tea575x_set_direction(struct snd_tea575x *tea, bool output) +/* 64PCR GPIO numbers */ +#define TEA_64PCR_BUS_CLOCK 0 +#define TEA_64PCR_WRITE_ENABLE 1 /* inverted */ +#define TEA_64PCR_DATA 2 + +static void snd_fm801_tea575x_64pcr_write(struct snd_tea575x *tea, unsigned int val) { struct fm801 *chip = tea->private_data; - unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); - struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; + unsigned short reg; + int i = 25; + spin_lock_irq(&chip->reg_lock); + reg = inw(FM801_REG(chip, GPIO_CTRL)); /* use GPIO lines and set write enable bit */ - reg |= FM801_GPIO_GS(gpio.data) | - FM801_GPIO_GS(gpio.wren) | - FM801_GPIO_GS(gpio.clk) | - FM801_GPIO_GS(gpio.most); - if (output) { - /* all of lines are in the write direction */ - /* clear data and clock lines */ - reg &= ~(FM801_GPIO_GD(gpio.data) | - FM801_GPIO_GD(gpio.wren) | - FM801_GPIO_GD(gpio.clk) | - FM801_GPIO_GP(gpio.data) | - FM801_GPIO_GP(gpio.clk) | - FM801_GPIO_GP(gpio.wren)); - } else { - /* use GPIO lines, set data direction to input */ - reg |= FM801_GPIO_GD(gpio.data) | - FM801_GPIO_GD(gpio.most) | - FM801_GPIO_GP(gpio.data) | - FM801_GPIO_GP(gpio.most) | - FM801_GPIO_GP(gpio.wren); - /* all of lines are in the write direction, except data */ - /* clear data, write enable and clock lines */ - reg &= ~(FM801_GPIO_GD(gpio.wren) | - FM801_GPIO_GD(gpio.clk) | - FM801_GPIO_GP(gpio.clk)); + reg |= FM801_GPIO_GS(TEA_64PCR_DATA) | + FM801_GPIO_GS(TEA_64PCR_WRITE_ENABLE) | + FM801_GPIO_GS(TEA_64PCR_BUS_CLOCK); + /* all of lines are in the write direction */ + /* clear data and clock lines */ + reg &= ~(FM801_GPIO_GD(TEA_64PCR_DATA) | + FM801_GPIO_GD(TEA_64PCR_WRITE_ENABLE) | + FM801_GPIO_GD(TEA_64PCR_BUS_CLOCK) | + FM801_GPIO_GP(TEA_64PCR_DATA) | + FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK) | + FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE)); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + + while (i--) { + if (val & (1 << i)) + reg |= FM801_GPIO_GP(TEA_64PCR_DATA); + else + reg &= ~FM801_GPIO_GP(TEA_64PCR_DATA); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + reg |= FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + reg &= ~FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + } + + /* and reset the write enable bit */ + reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE) | + FM801_GPIO_GP(TEA_64PCR_DATA); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + spin_unlock_irq(&chip->reg_lock); +} + +static unsigned int snd_fm801_tea575x_64pcr_read(struct snd_tea575x *tea) +{ + struct fm801 *chip = tea->private_data; + unsigned short reg; + unsigned int val = 0; + int i; + + spin_lock_irq(&chip->reg_lock); + reg = inw(FM801_REG(chip, GPIO_CTRL)); + /* use GPIO lines, set data direction to input */ + reg |= FM801_GPIO_GS(TEA_64PCR_DATA) | + FM801_GPIO_GS(TEA_64PCR_WRITE_ENABLE) | + FM801_GPIO_GS(TEA_64PCR_BUS_CLOCK) | + FM801_GPIO_GD(TEA_64PCR_DATA) | + FM801_GPIO_GP(TEA_64PCR_DATA) | + FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); + /* all of lines are in the write direction, except data */ + /* clear data, write enable and clock lines */ + reg &= ~(FM801_GPIO_GD(TEA_64PCR_WRITE_ENABLE) | + FM801_GPIO_GD(TEA_64PCR_BUS_CLOCK) | + FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK)); + + for (i = 0; i < 24; i++) { + reg &= ~FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + reg |= FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK); + outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + val <<= 1; + if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_64PCR_DATA)) + val |= 1; } + spin_unlock_irq(&chip->reg_lock); + + return val; +} + +static void snd_fm801_tea575x_64pcr_mute(struct snd_tea575x *tea, + unsigned int mute) +{ + struct fm801 *chip = tea->private_data; + unsigned short reg; + + spin_lock_irq(&chip->reg_lock); + + reg = inw(FM801_REG(chip, GPIO_CTRL)); + if (mute) + /* 0xf800 (mute) */ + reg &= ~FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); + else + /* 0xf802 (unmute) */ + reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); outw(reg, FM801_REG(chip, GPIO_CTRL)); + udelay(1); + + spin_unlock_irq(&chip->reg_lock); } -static struct snd_tea575x_ops snd_fm801_tea_ops = { - .set_pins = snd_fm801_tea575x_set_pins, - .get_pins = snd_fm801_tea575x_get_pins, - .set_direction = snd_fm801_tea575x_set_direction, +static struct snd_tea575x_ops snd_fm801_tea_ops[3] = { + { + /* 1 = MediaForte 256-PCS */ + .write = snd_fm801_tea575x_256pcs_write, + .read = snd_fm801_tea575x_256pcs_read, + }, + { + /* 2 = MediaForte 256-PCPR */ + .write = snd_fm801_tea575x_256pcpr_write, + .read = snd_fm801_tea575x_256pcpr_read, + }, + { + /* 3 = MediaForte 64-PCR */ + .write = snd_fm801_tea575x_64pcr_write, + .read = snd_fm801_tea575x_64pcr_read, + .mute = snd_fm801_tea575x_64pcr_mute, + } }; #endif @@ -1149,7 +1371,7 @@ static int snd_fm801_free(struct fm801 *chip) outw(cmdw, FM801_REG(chip, IRQ_MASK)); __end_hw: -#ifdef CONFIG_SND_FM801_TEA575X_BOOL +#ifdef TEA575X_RADIO snd_tea575x_exit(&chip->tea); #endif if (chip->irq >= 0) @@ -1228,34 +1450,16 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); -#ifdef CONFIG_SND_FM801_TEA575X_BOOL - chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops; - sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); +#ifdef TEA575X_RADIO if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && (tea575x_tuner & TUNER_TYPE_MASK) < 4) { - if (snd_tea575x_init(&chip->tea)) { - snd_printk(KERN_ERR "TEA575x radio not found\n"); - snd_fm801_free(chip); - return -ENODEV; - } - } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) { - /* autodetect tuner connection */ - for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) { - chip->tea575x_tuner = tea575x_tuner; - if (!snd_tea575x_init(&chip->tea)) { - snd_printk(KERN_INFO "detected TEA575x radio type %s\n", - snd_fm801_tea575x_gpios[tea575x_tuner - 1].name); - break; - } - } - if (tea575x_tuner == 4) { - snd_printk(KERN_ERR "TEA575x radio not found\n"); - snd_fm801_free(chip); - return -ENODEV; - } + chip->tea.dev_nr = tea575x_tuner >> 16; + chip->tea.card = card; + chip->tea.freq_fixup = 10700; + chip->tea.private_data = chip; + chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & TUNER_TYPE_MASK) - 1]; + snd_tea575x_init(&chip->tea); } - strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card)); #endif *rchip = chip; diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index f3353b4..27709f0 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -235,8 +235,8 @@ static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = { { PCI_VDEVICE(NVIDIA, 0x0069), DEVICE_NFORCE }, /* NFORCE2 */ { PCI_VDEVICE(NVIDIA, 0x0089), DEVICE_NFORCE }, /* NFORCE2s */ { PCI_VDEVICE(NVIDIA, 0x00d9), DEVICE_NFORCE }, /* NFORCE3 */ - { PCI_VDEVICE(AMD, 0x746e), DEVICE_INTEL }, /* AMD8111 */ #if 0 + { PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL }, /* AMD8111 */ { PCI_VDEVICE(AL, 0x5455), DEVICE_ALI }, /* Ali5455 */ #endif { 0, } @@ -1261,9 +1261,9 @@ static struct shortname_table { { PCI_DEVICE_ID_NVIDIA_MCP2_MODEM, "NVidia nForce2" }, { PCI_DEVICE_ID_NVIDIA_MCP2S_MODEM, "NVidia nForce2s" }, { PCI_DEVICE_ID_NVIDIA_MCP3_MODEM, "NVidia nForce3" }, - { 0x746e, "AMD AMD8111" }, #if 0 { 0x5455, "ALi M5455" }, + { 0x746d, "AMD AMD8111" }, #endif { 0 }, }; -- 1.7.5.3 -- Michal Hocko SUSE Labs SUSE LINUX s.r.o. Lihovarska 1060/12 190 00 Praha 9 Czech Republic

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[alsa-devel] snd_pcm_link:Function not implemented
by ashwanik 10 Jun '11

10 Jun '11

Hi All, I have download and compile latency.c application from http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.ht…. This application is working fine on Linux PC. But when I am using this application on my omap4 board(using ubuntu10.10) is not working. I am getting"Streams link error: Function not implemented" error. I debug the application and found this error coming from the *snd_pcm_link* function. I am using ALSA Driver Version 1.0.23. please find pointers. Thanks in Advance Ashwani

1 0

[alsa-devel] [PATCH] ASoC: pcm - rename snd_codec_close() to snd_pcm_close().
by Liam Girdwood 10 Jun '11

10 Jun '11

Make sure we follow naming convention for all PCM ops. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f4864b0..b575939 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -268,7 +268,7 @@ static void close_delayed_work(struct work_struct *work) * freed here. The cpu DAI, codec DAI, machine and platform are also * shutdown. */ -static int soc_codec_close(struct snd_pcm_substream *substream) +static int soc_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; @@ -567,7 +567,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) /* ASoC PCM operations */ static struct snd_pcm_ops soc_pcm_ops = { .open = soc_pcm_open, - .close = soc_codec_close, + .close = soc_pcm_close, .hw_params = soc_pcm_hw_params, .hw_free = soc_pcm_hw_free, .prepare = soc_pcm_prepare, -- 1.7.4.1

3 2

[alsa-devel] [PATCH v3 00/12] MFD/ASoC/Input: TWL4030/TWL60X0 changes
by Peter Ujfalusi 10 Jun '11

10 Jun '11

Hello, Changes since v2: - TWL_COMMON_PDATA_AUDIO is used from the beginning, reducing changes later - twl6040 thermal shutdown handling added - pmic-common.c/h renamed as twl-common.c/h. This will allows new twl related features to be added, not directly related to pmic. - Rebased on top of linux-3.0-rc2 (l-o next, asoc next, mfd next) Updated intro message from the v2 series: The series will do four major things, and they are in one series, because they pretty much depending on each other, so it is easier to handle them together. 0. Move the common pmic related configurations out from the board files. New twl-common.c/h file has been created for this purpose. OMAP3 board files received two patches (pdata, and regualtor changes), OMAP4 board files has been changed with one patch. 1. Rename the twl4030-codec MFD driver to twl4030-audio Since the ASoC multicomponent introduction (2.6.37) the twl4030-codec MFD dirver had been using twl4030-audio to register (twl4030-codec is the ASoC codec driver). The documentation refers to this part of twl as audio block, so it is better to rename the driver as well to avoid confusion. 2. Introduction of MFD driver for twl6040. The TWL6040 audio IC has codec and vibra functionality. Convert the existing ASoC codec driver to use the new MFD driver's interface. 3. Vibrator driver for TWL6040 (Input/ForceFeedback driver) The driver itself, and support for the vibrators in SDP4430 board. Regards, Peter --- Misael Lopez Cruz (3): mfd: twl6040: Add initial support ASoC: twl6040: Convert into TWL6040 MFD child input: Add initial support for TWL6040 vibrator Peter Ujfalusi (9): OMAP: New twl-common for common TWL configuration OMAP4: Move common twl6030 configuration to twl-common OMAP3: Move common twl configuration to twl-common OMAP3: Move common regulator configuration to twl-common MFD: twl4030-codec: Rename internals from codec to audio MFD: twl4030-codec -> twl4030-audio: Rename the driver MFD: twl4030-audio: Rename platform data MFD: twl6040: Change platform data for soc codec driver OMAP4: SDP4430: Add twl6040 vibrator platform support arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/board-3430sdp.c | 89 +---- arch/arm/mach-omap2/board-4430sdp.c | 183 +++------ arch/arm/mach-omap2/board-cm-t35.c | 48 +-- arch/arm/mach-omap2/board-devkit8000.c | 37 +-- arch/arm/mach-omap2/board-igep0020.c | 44 +-- arch/arm/mach-omap2/board-ldp.c | 15 +- arch/arm/mach-omap2/board-omap3beagle.c | 60 +--- arch/arm/mach-omap2/board-omap3evm.c | 67 +--- arch/arm/mach-omap2/board-omap3pandora.c | 59 +--- arch/arm/mach-omap2/board-omap3stalker.c | 70 +--- arch/arm/mach-omap2/board-omap3touchbook.c | 57 +-- arch/arm/mach-omap2/board-omap4panda.c | 149 +------ arch/arm/mach-omap2/board-overo.c | 59 +--- arch/arm/mach-omap2/board-rm680.c | 8 +- arch/arm/mach-omap2/board-rx51-peripherals.c | 44 +-- arch/arm/mach-omap2/board-zoom-peripherals.c | 88 +---- arch/arm/mach-omap2/common-board-devices.c | 21 - arch/arm/mach-omap2/common-board-devices.h | 26 +- arch/arm/mach-omap2/twl-common.c | 304 +++++++++++++ arch/arm/mach-omap2/twl-common.h | 59 +++ arch/arm/plat-omap/include/plat/irqs.h | 12 +- drivers/input/misc/Kconfig | 13 +- drivers/input/misc/Makefile | 1 + drivers/input/misc/twl4030-vibra.c | 12 +- drivers/input/misc/twl6040-vibra.c | 428 ++++++++++++++++++ drivers/mfd/Kconfig | 8 +- drivers/mfd/Makefile | 3 +- drivers/mfd/twl-core.c | 13 +- drivers/mfd/twl4030-audio.c | 277 ++++++++++++ drivers/mfd/twl4030-codec.c | 277 ------------ drivers/mfd/twl6040-core.c | 601 ++++++++++++++++++++++++++ drivers/mfd/twl6040-irq.c | 205 +++++++++ include/linux/i2c/twl.h | 21 +- include/linux/mfd/twl4030-audio.h | 272 ++++++++++++ include/linux/mfd/twl4030-codec.h | 272 ------------ include/linux/mfd/twl6040.h | 261 +++++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl4030.c | 22 +- sound/soc/codecs/twl6040.c | 426 ++++--------------- sound/soc/codecs/twl6040.h | 118 ----- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/sdp4430.c | 2 + sound/soc/omap/zoom2.c | 2 +- 44 files changed, 2729 insertions(+), 2011 deletions(-) create mode 100644 arch/arm/mach-omap2/twl-common.c create mode 100644 arch/arm/mach-omap2/twl-common.h create mode 100644 drivers/input/misc/twl6040-vibra.c create mode 100644 drivers/mfd/twl4030-audio.c delete mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 drivers/mfd/twl6040-core.c create mode 100644 drivers/mfd/twl6040-irq.c create mode 100644 include/linux/mfd/twl4030-audio.h delete mode 100644 include/linux/mfd/twl4030-codec.h create mode 100644 include/linux/mfd/twl6040.h -- 1.7.5.3

2 14

[alsa-devel] [PATCH] ALSA: HDA: Add jack detection for HDMI
by David Henningsson 09 Jun '11

09 Jun '11

Just as for headphones and microphone jacks, this patch adds reporting of HDMI jack status through the input layer. I considered making additional SND_JACK_* constants for HDMI and Displayport instead of going with SND_JACK_VIDEOOUT, but it didn't seem worth the additions, and breakage of compiling with old kernels, etc. Let me know if you think otherwise and I'll prepare a second patch for that. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

7 33

Re: [alsa-devel] [PATCH 1/1] ASoC: codecs: max98088: Combined DAI operations for DAI1 and DAI2.
by Peter Hsiang 09 Jun '11

09 Jun '11

On Thu, Jun 09, 2011, Jin Park wrote: > Combined DAI operations for DAI1 and DAI2. > For this, added id into DAI runtime data to identify each DAI. > > Signed-off-by: Jin Park <jinyoungp(a)nvidia.com> > @@ -1262,87 +1262,43 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, > > - cdata = &max98088->dai[1]; > + cdata = &max98088->dai[dai->id - 1]; Please add BUG_ON protection in case later someone changes the id usage. Mark, what's your thought on the use of id for this purpose here? Would it be better to have 2 wrapper DAI functions passing in the DAI number? > +static int max98088_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) > { > + unsigned short fmt_reg; > + unsigned short clk_reg; > + unsigned short clkcfg_hi_reg; > + unsigned short clkcfg_lo_reg; > + u8 clk_val; > + u8 fmt_val = 0; > + > + switch (dai->id) { > + case 1: > + fmt_reg = M98088_REG_14_DAI1_FORMAT; > + clk_reg = M98088_REG_15_DAI1_CLOCK; > + clkcfg_hi_reg = M98088_REG_12_DAI1_CLKCFG_HI; > + clkcfg_lo_reg = M98088_REG_13_DAI1_CLKCFG_LO; > + break; > + case 2: > + fmt_reg = M98088_REG_1C_DAI2_FORMAT; > + clk_reg = M98088_REG_1D_DAI2_CLOCK; > + clkcfg_hi_reg = M98088_REG_1A_DAI2_CLKCFG_HI; > + clkcfg_lo_reg = M98088_REG_1B_DAI2_CLKCFG_LO; > + break; > + default: > + return -EINVAL; > + } These added local variables and their initialization repeats. It may be better to put them in struct max98088_cdata[]. > @@ -1470,127 +1439,54 @@ static int max98088_dai1_set_fmt(struct snd_soc_dai *codec_dai, > - reg15val = M98088_DAI_BSEL64; > + clk_val = M98088_DAI_BSEL64; > if (max98088->digmic) > - reg15val |= M98088_DAI_OSR64; > - snd_soc_write(codec, M98088_REG_15_DAI1_CLOCK, reg15val); > + clk_val |= M98088_DAI_OSR64; > + snd_soc_write(codec, clk_reg, clk_val); DAI2 does not have an OSR bit in the _CLOCK register. In the combined function, this should be conditionally for DAI1 only. Peter

2 2

[alsa-devel] Proposed changes to soc core to allow multiple codecs bound together on a single bus
by Caleb Crome 09 Jun '11

09 Jun '11

Hi all, I finally got my multi-codec board up and running, but by hacking the codec driver in a very nasty way. It would be much preferable to modify the soc core so that multiple codecs can be easily instantiated on a single bus. Problem overview ----------------------------- The problem is this: a single CPU DAI is connected to multiple codecs. Each codec is assigned a different TDM slot on the TDM bus. All codecs are bound together and expected to start at the same time and have the same clocks, etc. Either the CPU or one of the codecs can be the master, all other devices are slaves. Each codec, as it transmits, must tri-state its output when not transmitting valid data, so that the other codecs may transmit in their given time slot. I think from user-space, these codecs should really be bound pretty tightly together, so they may be opened as a single 'sound card' with the sum of the channels available. My card is an 8 codec (2 channel/codec) card, so it should appear as a 16-channel sound card that opens/closes all in one go. Currently, I don't see a way in the soc core to handle this kind of situation. It seems that when you add multiple codecs onto one host DAI, you may only open one codec at a time. The others are not available. There are a few variations on this general theme: * 1 cpu DAI, many codecs, all sharing the same data in and data out pins (TDM/network mode) (like the McBSP) * 1 cpu DAI, many codecs, but the cpu dai may have more than 1 data pin (like the McASP on TI parts). Shared clocks. * multiple cpu DAIs bound together on the same clocks, multiple codecs. What is the proper way to add this functionality to the SOC API? Perhaps a specification like this: static char *my_codec_dais[] = { "tlv320aic3x-hifi.0", <---- in this case, the '.0' "tlv320aic3x-hifi.1", ---- means that it's slot 0 "tlv320aic3x-hifi.2", ---- on the TDM bus. "tlv320aic3x-hifi.3", }; static char *my_codec_names[] = { "tlv320aic3x-codec.2-0018", <--- codec on i2c bus 2, addr 0x18 "tlv320aic3x-codec.2-0019", <--- codec on i2c bus 2, addr 0x19 "tlv320aic3x-codec.2-001a", <--- codec on i2c bus 2, addr 0x1a "tlv320aic3x-codec.2-001b", <--- codec on i2c bus 2, addr 0x1b }; static struct snd_soc_dai_link my_dai[] = { .name = "my_dai_name", .stream_name = "my_stream_name", .cpu_dai_name = "omap-mcbsp-dai.2", .platform_name= "omap-pcm_audio", .codec_dai_name = my_codec_dais, .num_codec = ARRAY_SIZE(my_codec_dais), .codec_name = my_codec_names, .ops = &my_ops, }; The core will parse the dai name for the slot order, and pass it on to the codec during hw_params. Then the codec can properly set the TDM slot on the hardware interface. It will also pass the cpu dai the number of slices that are configured, so it knows how many slots to be expecting. This means that the snd_soc_pcm_runtime, and probably a bunch of other places, will need to keep track of multiple codecs, and multiple dai's, and functions like 'hw_params' will need to loop through multiple codec_dai's, so each codec and cpu dai can be configured correctly. I suspect this would be a pretty major change to the SOC core, and non-trivial. Any thoughts on this, or if I'm totally barking up the wrong tree here? Thanks, -Caleb

3 3

[alsa-devel] [PATCH] ASoC: core - PCM mutex per rtd
by Liam Girdwood 09 Jun '11

09 Jun '11

In preparation for the new ASoC Dynamic PCM support (AKA DSP support). The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed at runtime between the PCM device end (or Frontend - FE) and the physical DAI (Backend - BE) using regular kcontrols (just like a hardware CODEC routes audio in the analog domain). The Dynamic PCM core therefore must be able to call PCM operations for both the Frontend and Backend(s) DAIs at the same time. Currently we have a global pcm_mutex that is used to serialise the ASoC PCM operations. This patch removes the global mutex and adds a mutex per RTD allowing the PCM operations to be reentrant and allow control of more than one DAI at at time. e.g. a frontend PCM hw_params() could configure multiple backend DAI hw_params() with similar or different hw parameters at the same time. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 8 ++++++++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 28 ++++++++++++++-------------- 3 files changed, 23 insertions(+), 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d5db87e..4334ab2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -269,6 +269,11 @@ enum snd_soc_compress_type { SND_SOC_RBTREE_COMPRESSION }; +enum snd_soc_pcm_subclass { + SND_SOC_PCM_CLASS_PCM = 0, + SND_SOC_PCM_CLASS_BE = 1, +}; + int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, @@ -809,6 +814,9 @@ struct snd_soc_pcm_runtime { struct device dev; struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; + struct mutex pcm_mutex; + enum snd_soc_pcm_subclass pcm_subclass; + struct snd_pcm_ops ops; unsigned int complete:1; unsigned int dev_registered:1; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 32d7d2f..32bc503 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1032,6 +1032,7 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd->dev.parent = card->dev; rtd->dev.release = rtd_release; rtd->dev.init_name = name; + mutex_init(&rtd->pcm_mutex); ret = device_register(&rtd->dev); if (ret < 0) { dev_err(card->dev, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9bebee8..f4864b0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -81,7 +81,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; - mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* startup the audio subsystem */ if (cpu_dai->driver->ops->startup) { @@ -211,7 +211,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) cpu_dai->active++; codec_dai->active++; rtd->codec->active++; - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return 0; config_err: @@ -230,7 +230,7 @@ platform_err: if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -245,7 +245,7 @@ static void close_delayed_work(struct work_struct *work) container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dai; - mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); pr_debug("pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, @@ -260,7 +260,7 @@ static void close_delayed_work(struct work_struct *work) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); } /* @@ -276,7 +276,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; @@ -321,7 +321,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return 0; } @@ -338,7 +338,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; - mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); @@ -391,7 +391,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(codec_dai, 0); out: - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -409,7 +409,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; - mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); @@ -449,7 +449,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, rtd->rate = params_rate(params); out: - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return ret; platform_err: @@ -464,7 +464,7 @@ codec_err: if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return ret; } @@ -479,7 +479,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* apply codec digital mute */ if (!codec->active) @@ -500,7 +500,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return 0; } -- 1.7.4.1

2 1

[alsa-devel] Problem capturing from sound board
by André Prado 09 Jun '11

09 Jun '11

Hello i am trying to do a simple program in order to capture the input from the microphone in my OnBoard HDA Intel sound board. However it doesnt work and gives me a segmentation fault. Any help is appreciated. #include <stdio.h> #include <string.h> #include <alsa/asoundlib.h> int main(int argc, char** argv) { register int err; int cardNum, totalCards; totalCards = 0; cardNum = -1; short buf[128]; while(1) { snd_ctl_t *cardHandle; printf("\n***********\n"); //Number of boards if ((err = snd_card_next(&cardNum)) < 0) { printf("%s\n", snd_strerror(err)); break; } if(cardNum<0) { break; } ++totalCards; char str[64]; sprintf(str, "hw:%i", cardNum); (IS THIS THE RIGHT WAY TO NAME MY DEVICE?) snd_pcm_t *capture_handle; snd_pcm_hw_params_t *hw_params; if(err = snd_pcm_open( &capture_handle, str, SND_PCM_STREAM_CAPTURE, 0 ) < 0 ) { printf(" %s\n", snd_strerror(err)); break; } snd_pcm_hw_params_malloc (&hw_params); snd_pcm_hw_params_any(capture_handle, hw_params); if(err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) printf("Nao e setar o acesso: %s\n", snd_strerror(err)); unsigned int num_canais = 0; int ret; if(err = snd_pcm_hw_params_get_channels_max(hw_params, &num_canais) < 0) printf(" Nao e conseguir o numero de canais: %s\n", snd_strerror(err)); printf("Numero de canais: %i\n",num_canais); // Configurando hardware // Resampling printf("Fim preparacao"); unsigned int resample = 1; if (err = snd_pcm_hw_params_set_rate_resample(capture_handle, hw_params, resample)<0) printf("Erro setando o sample: %s\n", snd_strerror(err)); // Formato Little endian 16 bits snd_pcm_hw_params_set_format (capture_handle, hw_params, SND_PCM_FORMAT_S16_LE); // Stereo if (err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, 2) < 0) printf("Erro setando o numero de canais: %s\n", snd_strerror(err)); // Playback Rate unsigned int actualRate = 44100; if (err = snd_pcm_hw_params_set_rate_near (capture_handle, hw_params, &actualRate, 0) < 0) printf("Erro setando o rate de playback: %s\n", snd_strerror(err)); // Setando os valores configurados no hardware. if (err = snd_pcm_hw_params (capture_handle, hw_params) < 0 ) printf("Erro setando o hardware: %s\n", snd_strerror(err)); // Tamanho do buffer recebido apos configurar o hardware snd_pcm_uframes_t bufferSize; snd_pcm_hw_params_get_buffer_size( hw_params, &bufferSize ); //printf( "Init: Buffer size = %s", bufferSize); //_playbackFrames = bufferSize; printf( "\nBits significantes para amostra linear = %i\n", snd_pcm_hw_params_get_sbits(hw_params)); // Preparando para leitura int i=0; int d=0; if ((err = snd_pcm_prepare (capture_handle)) < 0) printf("cannot prepare audio interface for use (%s)\n",snd_strerror(err)); snd_pcm_hw_params_free (hw_params); // if ((err = snd_pcm_readi (capture_handle, buf, 128)) != 128) // printf( "read from audio interface failed (%s)\n", snd_strerror (err)); } snd_config_update_free_global(); } -- Att André

1 0

[alsa-devel] dsnoop
by Lorenzo Fundaró 09 Jun '11

09 Jun '11

Hi folks, I want to use dsnoop in my .asoundrc, but this plugin does not seem to be available to me as shown by `arecord -L` . Does the availability depend on the sound card or is it just a sort of module I have to load ? Thanks, Lorenzo Fundaró García Université de Technologie de Compiègne Génie Informatique <%2B33%206%2086%2069%2056%2011>

1 0

[alsa-devel] [PATCHv2] ASoC: core - Separate out PCM operations into new file.
by Liam Girdwood 09 Jun '11

09 Jun '11

In preparation for Dynamic PCM support (AKA DSP support). There will be future patches that add support to allow PCMs to be dynamically routed to multiple DAIs at startup and also during stream runtime. This patch moves the ASoC core PCM operaitions into a new file called soc-pcm.c. This will in simplify the ASoC core features into distinct files. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/Makefile | 2 +- sound/soc/soc-core.c | 612 +----------------------------------------------- sound/soc/soc-pcm.c | 639 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 641 insertions(+), 612 deletions(-) create mode 100644 sound/soc/soc-pcm.c diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1ed61c5..adb5719 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o soc-pcm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index df5608d..32d7d2f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -44,7 +44,6 @@ #define NAME_SIZE 32 -static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS @@ -58,7 +57,7 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -485,552 +484,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && - !rtd->dai_link->symmetric_rates) - return 0; - - /* This can happen if multiple streams are starting simultaneously - - * the second can need to get its constraints before the first has - * picked a rate. Complain and allow the application to carry on. - */ - if (!rtd->rate) { - dev_warn(&rtd->dev, - "Not enforcing symmetric_rates due to race\n"); - return 0; - } - - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } - - return 0; -} - -/* - * Called by ALSA when a PCM substream is opened, the runtime->hw record is - * then initialized and any private data can be allocated. This also calls - * startup for the cpu DAI, platform, machine and codec DAI. - */ -static int soc_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; - int ret = 0; - - mutex_lock(&pcm_mutex); - - /* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { - ret = cpu_dai->driver->ops->startup(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->open) { - ret = platform->driver->ops->open(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); - goto platform_err; - } - } - - if (codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); - goto codec_dai_err; - } - } - - if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { - ret = rtd->dai_link->ops->startup(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); - goto machine_err; - } - } - - /* Check that the codec and cpu DAIs are compatible */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = - max(codec_dai_drv->playback.rate_min, - cpu_dai_drv->playback.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->playback.rate_max, - cpu_dai_drv->playback.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->playback.channels_min, - cpu_dai_drv->playback.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->playback.channels_max, - cpu_dai_drv->playback.channels_max); - runtime->hw.formats = - codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; - runtime->hw.rates = - codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; - if (codec_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->playback.rates; - if (cpu_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->playback.rates; - } else { - runtime->hw.rate_min = - max(codec_dai_drv->capture.rate_min, - cpu_dai_drv->capture.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->capture.rate_max, - cpu_dai_drv->capture.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->capture.channels_min, - cpu_dai_drv->capture.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->capture.channels_max, - cpu_dai_drv->capture.channels_max); - runtime->hw.formats = - codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; - runtime->hw.rates = - codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; - if (codec_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->capture.rates; - if (cpu_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->capture.rates; - } - - ret = -EINVAL; - snd_pcm_limit_hw_rates(runtime); - if (!runtime->hw.rates) { - printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.formats) { - printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.channels_min || !runtime->hw.channels_max || - runtime->hw.channels_min > runtime->hw.channels_max) { - printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - - /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); - if (ret != 0) - goto config_err; - } - - pr_debug("asoc: %s <-> %s info:\n", - codec_dai->name, cpu_dai->name); - pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); - pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, - runtime->hw.channels_max); - pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, - runtime->hw.rate_max); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active++; - codec_dai->playback_active++; - } else { - cpu_dai->capture_active++; - codec_dai->capture_active++; - } - cpu_dai->active++; - codec_dai->active++; - rtd->codec->active++; - mutex_unlock(&pcm_mutex); - return 0; - -config_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - -machine_err: - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - -codec_dai_err: - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - -platform_err: - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Power down the audio subsystem pmdown_time msecs after close is called. - * This is to ensure there are no pops or clicks in between any music tracks - * due to DAPM power cycling. - */ -static void close_delayed_work(struct work_struct *work) -{ - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - mutex_lock(&pcm_mutex); - - pr_debug("pop wq checking: %s status: %s waiting: %s\n", - codec_dai->driver->playback.stream_name, - codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); - - /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); -} - -/* - * Called by ALSA when a PCM substream is closed. Private data can be - * freed here. The cpu DAI, codec DAI, machine and platform are also - * shutdown. - */ -static int soc_codec_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active--; - codec_dai->playback_active--; - } else { - cpu_dai->capture_active--; - codec_dai->capture_active--; - } - - cpu_dai->active--; - codec_dai->active--; - codec->active--; - - /* Muting the DAC suppresses artifacts caused during digital - * shutdown, for example from stopping clocks. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); - - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); - - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - cpu_dai->runtime = NULL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); - } else { - /* capture streams can be powered down now */ - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); - return 0; -} - -/* - * Called by ALSA when the PCM substream is prepared, can set format, sample - * rate, etc. This function is non atomic and can be called multiple times, - * it can refer to the runtime info. - */ -static int soc_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { - ret = rtd->dai_link->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: machine prepare error\n"); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->prepare) { - ret = platform->driver->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: platform prepare error\n"); - goto out; - } - } - - if (codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: codec DAI prepare error\n"); - goto out; - } - } - - if (cpu_dai->driver->ops->prepare) { - ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: cpu DAI prepare error\n"); - goto out; - } - } - - /* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; - cancel_delayed_work(&rtd->delayed_work); - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_START); - - snd_soc_dai_digital_mute(codec_dai, 0); - -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Called by ALSA when the hardware params are set by application. This - * function can also be called multiple times and can allocate buffers - * (using snd_pcm_lib_* ). It's non-atomic. - */ -static int soc_pcm_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_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { - ret = rtd->dai_link->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: machine hw_params failed\n"); - goto out; - } - } - - if (codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); - goto codec_err; - } - } - - if (cpu_dai->driver->ops->hw_params) { - ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); - goto interface_err; - } - } - - if (platform->driver->ops && platform->driver->ops->hw_params) { - ret = platform->driver->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); - goto platform_err; - } - } - - rtd->rate = params_rate(params); - -out: - mutex_unlock(&pcm_mutex); - return ret; - -platform_err: - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - -interface_err: - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - -codec_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Frees resources allocated by hw_params, can be called multiple times - */ -static int soc_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - /* apply codec digital mute */ - if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); - - /* free any machine hw params */ - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - /* free any DMA resources */ - if (platform->driver->ops && platform->driver->ops->hw_free) - platform->driver->ops->hw_free(substream); - - /* now free hw params for the DAIs */ - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - - mutex_unlock(&pcm_mutex); - return 0; -} - -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; - } - - if (platform->driver->ops && platform->driver->ops->trigger) { - ret = platform->driver->ops->trigger(substream, cmd); - if (ret < 0) - return ret; - } - - if (cpu_dai->driver->ops->trigger) { - ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; - } - return 0; -} - -/* - * soc level wrapper for pointer callback - * If cpu_dai, codec_dai, platform driver has the delay callback, than - * the runtime->delay will be updated accordingly. - */ -static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t offset = 0; - snd_pcm_sframes_t delay = 0; - - if (platform->driver->ops && platform->driver->ops->pointer) - offset = platform->driver->ops->pointer(substream); - - if (cpu_dai->driver->ops->delay) - delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - - if (codec_dai->driver->ops->delay) - delay += codec_dai->driver->ops->delay(substream, codec_dai); - - if (platform->driver->delay) - delay += platform->driver->delay(substream, codec_dai); - - runtime->delay = delay; - - return offset; -} - -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_codec_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - #ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ int snd_soc_suspend(struct device *dev) @@ -1692,9 +1145,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (order != SND_SOC_COMP_ORDER_LAST) return 0; - /* DAPM dai link stream work */ - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); - ret = soc_post_component_init(card, codec, num, 0); if (ret) return ret; @@ -2114,66 +1564,6 @@ static struct platform_driver soc_driver = { .remove = soc_remove, }; -/* create a new pcm */ -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_pcm *pcm; - char new_name[64]; - int ret = 0, playback = 0, capture = 0; - - /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - - if (codec_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min) - capture = 1; - - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); - ret = snd_pcm_new(rtd->card->snd_card, new_name, - num, playback, capture, &pcm); - if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); - return ret; - } - - rtd->pcm = pcm; - pcm->private_data = rtd; - if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; - } - - if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); - - if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); - - if (platform->driver->pcm_new) { - ret = platform->driver->pcm_new(rtd); - if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); - return ret; - } - } - - pcm->private_free = platform->driver->pcm_free; - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); - return ret; -} - /** * snd_soc_codec_volatile_register: Report if a register is volatile. * diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c new file mode 100644 index 0000000..9bebee8 --- /dev/null +++ b/sound/soc/soc-pcm.c @@ -0,0 +1,639 @@ +/* + * soc-pcm.c -- ALSA SoC PCM + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. + * + * Authors: Liam Girdwood <lrg(a)ti.com> + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> + +static DEFINE_MUTEX(pcm_mutex); + +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (!codec_dai->driver->symmetric_rates && + !cpu_dai->driver->symmetric_rates && + !rtd->dai_link->symmetric_rates) + return 0; + + /* This can happen if multiple streams are starting simultaneously - + * the second can need to get its constraints before the first has + * picked a rate. Complain and allow the application to carry on. + */ + if (!rtd->rate) { + dev_warn(&rtd->dev, + "Not enforcing symmetric_rates due to race\n"); + return 0; + } + + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + rtd->rate, rtd->rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; + } + + return 0; +} + +/* + * Called by ALSA when a PCM substream is opened, the runtime->hw record is + * then initialized and any private data can be allocated. This also calls + * startup for the cpu DAI, platform, machine and codec DAI. + */ +static int soc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; + int ret = 0; + + mutex_lock(&pcm_mutex); + + /* startup the audio subsystem */ + if (cpu_dai->driver->ops->startup) { + ret = cpu_dai->driver->ops->startup(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open interface %s\n", + cpu_dai->name); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->open) { + ret = platform->driver->ops->open(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + goto platform_err; + } + } + + if (codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open codec %s\n", + codec_dai->name); + goto codec_dai_err; + } + } + + if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + ret = rtd->dai_link->ops->startup(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); + goto machine_err; + } + } + + /* Check that the codec and cpu DAIs are compatible */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.rate_min = + max(codec_dai_drv->playback.rate_min, + cpu_dai_drv->playback.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->playback.rate_max, + cpu_dai_drv->playback.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->playback.channels_min, + cpu_dai_drv->playback.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->playback.channels_max, + cpu_dai_drv->playback.channels_max); + runtime->hw.formats = + codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; + runtime->hw.rates = + codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; + if (codec_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->playback.rates; + if (cpu_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->playback.rates; + } else { + runtime->hw.rate_min = + max(codec_dai_drv->capture.rate_min, + cpu_dai_drv->capture.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->capture.rate_max, + cpu_dai_drv->capture.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->capture.channels_min, + cpu_dai_drv->capture.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->capture.channels_max, + cpu_dai_drv->capture.channels_max); + runtime->hw.formats = + codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; + runtime->hw.rates = + codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; + if (codec_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->capture.rates; + if (cpu_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->capture.rates; + } + + ret = -EINVAL; + snd_pcm_limit_hw_rates(runtime); + if (!runtime->hw.rates) { + printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.formats) { + printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.channels_min || !runtime->hw.channels_max || + runtime->hw.channels_min > runtime->hw.channels_max) { + printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + + /* Symmetry only applies if we've already got an active stream. */ + if (cpu_dai->active || codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream); + if (ret != 0) + goto config_err; + } + + pr_debug("asoc: %s <-> %s info:\n", + codec_dai->name, cpu_dai->name); + pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); + pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, + runtime->hw.channels_max); + pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, + runtime->hw.rate_max); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active++; + codec_dai->playback_active++; + } else { + cpu_dai->capture_active++; + codec_dai->capture_active++; + } + cpu_dai->active++; + codec_dai->active++; + rtd->codec->active++; + mutex_unlock(&pcm_mutex); + return 0; + +config_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + +machine_err: + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + +codec_dai_err: + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + +platform_err: + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); +out: + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Power down the audio subsystem pmdown_time msecs after close is called. + * This is to ensure there are no pops or clicks in between any music tracks + * due to DAPM power cycling. + */ +static void close_delayed_work(struct work_struct *work) +{ + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + mutex_lock(&pcm_mutex); + + pr_debug("pop wq checking: %s status: %s waiting: %s\n", + codec_dai->driver->playback.stream_name, + codec_dai->playback_active ? "active" : "inactive", + codec_dai->pop_wait ? "yes" : "no"); + + /* are we waiting on this codec DAI stream */ + if (codec_dai->pop_wait == 1) { + codec_dai->pop_wait = 0; + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&pcm_mutex); +} + +/* + * Called by ALSA when a PCM substream is closed. Private data can be + * freed here. The cpu DAI, codec DAI, machine and platform are also + * shutdown. + */ +static int soc_codec_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock(&pcm_mutex); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active--; + codec_dai->playback_active--; + } else { + cpu_dai->capture_active--; + codec_dai->capture_active--; + } + + cpu_dai->active--; + codec_dai->active--; + codec->active--; + + /* Muting the DAC suppresses artifacts caused during digital + * shutdown, for example from stopping clocks. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_digital_mute(codec_dai, 1); + + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); + + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + cpu_dai->runtime = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* start delayed pop wq here for playback streams */ + codec_dai->pop_wait = 1; + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); + } else { + /* capture streams can be powered down now */ + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&pcm_mutex); + return 0; +} + +/* + * Called by ALSA when the PCM substream is prepared, can set format, sample + * rate, etc. This function is non atomic and can be called multiple times, + * it can refer to the runtime info. + */ +static int soc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock(&pcm_mutex); + + if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + ret = rtd->dai_link->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: machine prepare error\n"); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->prepare) { + ret = platform->driver->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: platform prepare error\n"); + goto out; + } + } + + if (codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: codec DAI prepare error\n"); + goto out; + } + } + + if (cpu_dai->driver->ops->prepare) { + ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: cpu DAI prepare error\n"); + goto out; + } + } + + /* cancel any delayed stream shutdown that is pending */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + codec_dai->pop_wait) { + codec_dai->pop_wait = 0; + cancel_delayed_work(&rtd->delayed_work); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_START); + + snd_soc_dai_digital_mute(codec_dai, 0); + +out: + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Called by ALSA when the hardware params are set by application. This + * function can also be called multiple times and can allocate buffers + * (using snd_pcm_lib_* ). It's non-atomic. + */ +static int soc_pcm_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_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock(&pcm_mutex); + + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + ret = rtd->dai_link->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: machine hw_params failed\n"); + goto out; + } + } + + if (codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't set codec %s hw params\n", + codec_dai->name); + goto codec_err; + } + } + + if (cpu_dai->driver->ops->hw_params) { + ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: interface %s hw params failed\n", + cpu_dai->name); + goto interface_err; + } + } + + if (platform->driver->ops && platform->driver->ops->hw_params) { + ret = platform->driver->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: platform %s hw params failed\n", + platform->name); + goto platform_err; + } + } + + rtd->rate = params_rate(params); + +out: + mutex_unlock(&pcm_mutex); + return ret; + +platform_err: + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + +interface_err: + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + +codec_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Frees resources allocated by hw_params, can be called multiple times + */ +static int soc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock(&pcm_mutex); + + /* apply codec digital mute */ + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); + + /* free any machine hw params */ + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + /* free any DMA resources */ + if (platform->driver->ops && platform->driver->ops->hw_free) + platform->driver->ops->hw_free(substream); + + /* now free hw params for the DAIs */ + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + + mutex_unlock(&pcm_mutex); + return 0; +} + +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); + if (ret < 0) + return ret; + } + + if (platform->driver->ops && platform->driver->ops->trigger) { + ret = platform->driver->ops->trigger(substream, cmd); + if (ret < 0) + return ret; + } + + if (cpu_dai->driver->ops->trigger) { + ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); + if (ret < 0) + return ret; + } + return 0; +} + +/* + * soc level wrapper for pointer callback + * If cpu_dai, codec_dai, platform driver has the delay callback, than + * the runtime->delay will be updated accordingly. + */ +static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t offset = 0; + snd_pcm_sframes_t delay = 0; + + if (platform->driver->ops && platform->driver->ops->pointer) + offset = platform->driver->ops->pointer(substream); + + if (cpu_dai->driver->ops->delay) + delay += cpu_dai->driver->ops->delay(substream, cpu_dai); + + if (codec_dai->driver->ops->delay) + delay += codec_dai->driver->ops->delay(substream, codec_dai); + + if (platform->driver->delay) + delay += platform->driver->delay(substream, codec_dai); + + runtime->delay = delay; + + return offset; +} + +/* ASoC PCM operations */ +static struct snd_pcm_ops soc_pcm_ops = { + .open = soc_pcm_open, + .close = soc_codec_close, + .hw_params = soc_pcm_hw_params, + .hw_free = soc_pcm_hw_free, + .prepare = soc_pcm_prepare, + .trigger = soc_pcm_trigger, + .pointer = soc_pcm_pointer, +}; + +/* create a new pcm */ +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm *pcm; + char new_name[64]; + int ret = 0, playback = 0, capture = 0; + + /* check client and interface hw capabilities */ + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); + ret = snd_pcm_new(rtd->card->snd_card, new_name, + num, playback, capture, &pcm); + if (ret < 0) { + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + return ret; + } + + /* DAPM dai link stream work */ + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + + rtd->pcm = pcm; + pcm->private_data = rtd; + if (platform->driver->ops) { + soc_pcm_ops.mmap = platform->driver->ops->mmap; + soc_pcm_ops.pointer = platform->driver->ops->pointer; + soc_pcm_ops.ioctl = platform->driver->ops->ioctl; + soc_pcm_ops.copy = platform->driver->ops->copy; + soc_pcm_ops.silence = platform->driver->ops->silence; + soc_pcm_ops.ack = platform->driver->ops->ack; + soc_pcm_ops.page = platform->driver->ops->page; + } + + if (playback) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + + if (capture) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd); + if (ret < 0) { + pr_err("asoc: platform pcm constructor failed\n"); + return ret; + } + } + + pcm->private_free = platform->driver->pcm_free; + printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, + cpu_dai->name); + return ret; +} -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: snd_soc_new_{mixer, mux, pga} make sure to use right DAPM context
by Lars-Peter Clausen 09 Jun '11

09 Jun '11

Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a DAPM context not matching the widgets context. This can lead to a wrong prefix_len calculation, which will result in undefined behaviour. To avoid this always use the DAPM context from the widget itself. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org --- sound/soc/soc-dapm.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 34106bc..fd2d774 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -337,9 +337,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, } /* create new dapm mixer control */ -static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mixer(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; int i, ret = 0; size_t name_len, prefix_len; struct snd_soc_dapm_path *path; @@ -437,9 +437,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, } /* create new dapm mux control */ -static int dapm_new_mux(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mux(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; @@ -522,8 +522,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, } /* create new dapm volume control */ -static int dapm_new_pga(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_pga(struct snd_soc_dapm_widget *w) { if (w->num_kcontrols) dev_err(w->dapm->dev, @@ -1838,13 +1837,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: w->power_check = dapm_generic_check_power; - dapm_new_mixer(dapm, w); + dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; - dapm_new_mux(dapm, w); + dapm_new_mux(w); break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: @@ -1857,7 +1856,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: w->power_check = dapm_generic_check_power; - dapm_new_pga(dapm, w); + dapm_new_pga(w); break; case snd_soc_dapm_input: case snd_soc_dapm_output: -- 1.7.2.5

3 2

Re: [alsa-devel] [PATCH 0/1] Submitting MAX98088 codec patch
by Mark Brown 09 Jun '11

09 Jun '11

On Thu, Jun 09, 2011 at 05:50:16PM +0900, Jin Park wrote: > The max98088 codec driver had unnecessary duplicated codes on DAI operations. > For example, max98088_dai1_hw_params and max98088_dai2_hw_params, > they are almost same, except only register offsets. > So I have combined DAI operations for DAI1 and DAI2. > Please review my patch. Note that there's no need to have a cover letter for a single patch - any useful information should be in the patch itself.

2 1

[alsa-devel] [PATCH 1/2] ASoC: p1022ds: fix incorrect referencing of device tree properties
by Timur Tabi 09 Jun '11

09 Jun '11

Device tree integer properties are encoded in big-endian format, but some of the Freescale ASoC drivers were assuming that the host is in big-endian format as well. Although this is true, it's better to use endian-safe accessors. Also add a check for a failed ioremap() call in the SSI driver. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_ssi.c | 9 +++++++-- sound/soc/fsl/mpc8610_hpcd.c | 10 +++++----- sound/soc/fsl/p1022_ds.c | 10 +++++----- 4 files changed, 18 insertions(+), 13 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 50b5df8..3872598 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -940,7 +940,7 @@ static int __devinit fsl_soc_dma_probe(struct platform_device *pdev) iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL); if (iprop) - dma->ssi_fifo_depth = *iprop; + dma->ssi_fifo_depth = be32_to_cpup(iprop); else /* Older 8610 DTs didn't have the fifo-depth property */ dma->ssi_fifo_depth = 8; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 313e0cc..d48afea 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -678,7 +678,12 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) kfree(ssi_private); return ret; } - ssi_private->ssi = ioremap(res.start, 1 + res.end - res.start); + ssi_private->ssi = of_iomap(np, 0); + if (!ssi_private->ssi) { + dev_err(&pdev->dev, "could not map device resources\n"); + kfree(ssi_private); + return -ENOMEM; + } ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); @@ -691,7 +696,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Determine the FIFO depth. */ iprop = of_get_property(np, "fsl,fifo-depth", NULL); if (iprop) - ssi_private->fifo_depth = *iprop; + ssi_private->fifo_depth = be32_to_cpup(iprop); else /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index c16c6b2..a192979 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -233,7 +233,7 @@ static int get_parent_cell_index(struct device_node *np) if (!iprop) return -1; - return *iprop; + return be32_to_cpup(iprop); } /** @@ -258,7 +258,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (!iprop) return -EINVAL; - addr = *iprop; + addr = be32_to_cpup(iprop); bus = get_parent_cell_index(np); if (bus < 0) @@ -305,7 +305,7 @@ static int get_dma_channel(struct device_node *ssi_np, return -EINVAL; } - *dma_channel_id = *iprop; + *dma_channel_id = be32_to_cpup(iprop); *dma_id = get_parent_cell_index(dma_channel_np); of_node_put(dma_channel_np); @@ -379,7 +379,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - machine_data->ssi_id = *iprop; + machine_data->ssi_id = be32_to_cpup(iprop); /* Get the serial format and clock direction. */ sprop = of_get_property(np, "fsl,mode", NULL); @@ -405,7 +405,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - machine_data->clk_frequency = *iprop; + machine_data->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { machine_data->dai_format = SND_SOC_DAIFMT_I2S; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 66e0b68..8fa4d5f 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -232,7 +232,7 @@ static int get_parent_cell_index(struct device_node *np) iprop = of_get_property(parent, "cell-index", NULL); if (iprop) - ret = *iprop; + ret = be32_to_cpup(iprop); of_node_put(parent); @@ -261,7 +261,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (!iprop) return -EINVAL; - addr = *iprop; + addr = be32_to_cpup(iprop); bus = get_parent_cell_index(np); if (bus < 0) @@ -308,7 +308,7 @@ static int get_dma_channel(struct device_node *ssi_np, return -EINVAL; } - *dma_channel_id = *iprop; + *dma_channel_id = be32_to_cpup(iprop); *dma_id = get_parent_cell_index(dma_channel_np); of_node_put(dma_channel_np); @@ -379,7 +379,7 @@ static int p1022_ds_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - mdata->ssi_id = *iprop; + mdata->ssi_id = be32_to_cpup(iprop); /* Get the serial format and clock direction. */ sprop = of_get_property(np, "fsl,mode", NULL); @@ -405,7 +405,7 @@ static int p1022_ds_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - mdata->clk_frequency = *iprop; + mdata->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { mdata->dai_format = SND_SOC_DAIFMT_I2S; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; -- 1.7.3.4

5 7

[alsa-devel] [PATCH] ASoC: core - Separate out PCM operations into new file.
by Liam Girdwood 09 Jun '11

09 Jun '11

In preparation for Dynamic PCM support (AKA DSP support). There will be future patches that add support to allow PCMs to be dynamically routed to multiple DAIs at startup and also during stream runtime. This patch moves the ASoC core PCM operaitions into a new file called soc-pcm.c. This will in simplify the ASoC core features into distinct files. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/Makefile | 2 +- sound/soc/soc-core.c | 612 +----------------------------------------------- sound/soc/soc-pcm.c | 645 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 647 insertions(+), 612 deletions(-) create mode 100644 sound/soc/soc-pcm.c diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1ed61c5..adb5719 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o soc-pcm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index df5608d..32d7d2f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -44,7 +44,6 @@ #define NAME_SIZE 32 -static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS @@ -58,7 +57,7 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -485,552 +484,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && - !rtd->dai_link->symmetric_rates) - return 0; - - /* This can happen if multiple streams are starting simultaneously - - * the second can need to get its constraints before the first has - * picked a rate. Complain and allow the application to carry on. - */ - if (!rtd->rate) { - dev_warn(&rtd->dev, - "Not enforcing symmetric_rates due to race\n"); - return 0; - } - - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } - - return 0; -} - -/* - * Called by ALSA when a PCM substream is opened, the runtime->hw record is - * then initialized and any private data can be allocated. This also calls - * startup for the cpu DAI, platform, machine and codec DAI. - */ -static int soc_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; - int ret = 0; - - mutex_lock(&pcm_mutex); - - /* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { - ret = cpu_dai->driver->ops->startup(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->open) { - ret = platform->driver->ops->open(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); - goto platform_err; - } - } - - if (codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); - goto codec_dai_err; - } - } - - if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { - ret = rtd->dai_link->ops->startup(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); - goto machine_err; - } - } - - /* Check that the codec and cpu DAIs are compatible */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = - max(codec_dai_drv->playback.rate_min, - cpu_dai_drv->playback.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->playback.rate_max, - cpu_dai_drv->playback.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->playback.channels_min, - cpu_dai_drv->playback.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->playback.channels_max, - cpu_dai_drv->playback.channels_max); - runtime->hw.formats = - codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; - runtime->hw.rates = - codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; - if (codec_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->playback.rates; - if (cpu_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->playback.rates; - } else { - runtime->hw.rate_min = - max(codec_dai_drv->capture.rate_min, - cpu_dai_drv->capture.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->capture.rate_max, - cpu_dai_drv->capture.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->capture.channels_min, - cpu_dai_drv->capture.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->capture.channels_max, - cpu_dai_drv->capture.channels_max); - runtime->hw.formats = - codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; - runtime->hw.rates = - codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; - if (codec_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->capture.rates; - if (cpu_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->capture.rates; - } - - ret = -EINVAL; - snd_pcm_limit_hw_rates(runtime); - if (!runtime->hw.rates) { - printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.formats) { - printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.channels_min || !runtime->hw.channels_max || - runtime->hw.channels_min > runtime->hw.channels_max) { - printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - - /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); - if (ret != 0) - goto config_err; - } - - pr_debug("asoc: %s <-> %s info:\n", - codec_dai->name, cpu_dai->name); - pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); - pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, - runtime->hw.channels_max); - pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, - runtime->hw.rate_max); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active++; - codec_dai->playback_active++; - } else { - cpu_dai->capture_active++; - codec_dai->capture_active++; - } - cpu_dai->active++; - codec_dai->active++; - rtd->codec->active++; - mutex_unlock(&pcm_mutex); - return 0; - -config_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - -machine_err: - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - -codec_dai_err: - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - -platform_err: - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Power down the audio subsystem pmdown_time msecs after close is called. - * This is to ensure there are no pops or clicks in between any music tracks - * due to DAPM power cycling. - */ -static void close_delayed_work(struct work_struct *work) -{ - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - mutex_lock(&pcm_mutex); - - pr_debug("pop wq checking: %s status: %s waiting: %s\n", - codec_dai->driver->playback.stream_name, - codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); - - /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); -} - -/* - * Called by ALSA when a PCM substream is closed. Private data can be - * freed here. The cpu DAI, codec DAI, machine and platform are also - * shutdown. - */ -static int soc_codec_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active--; - codec_dai->playback_active--; - } else { - cpu_dai->capture_active--; - codec_dai->capture_active--; - } - - cpu_dai->active--; - codec_dai->active--; - codec->active--; - - /* Muting the DAC suppresses artifacts caused during digital - * shutdown, for example from stopping clocks. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); - - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); - - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - cpu_dai->runtime = NULL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); - } else { - /* capture streams can be powered down now */ - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); - return 0; -} - -/* - * Called by ALSA when the PCM substream is prepared, can set format, sample - * rate, etc. This function is non atomic and can be called multiple times, - * it can refer to the runtime info. - */ -static int soc_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { - ret = rtd->dai_link->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: machine prepare error\n"); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->prepare) { - ret = platform->driver->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: platform prepare error\n"); - goto out; - } - } - - if (codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: codec DAI prepare error\n"); - goto out; - } - } - - if (cpu_dai->driver->ops->prepare) { - ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: cpu DAI prepare error\n"); - goto out; - } - } - - /* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; - cancel_delayed_work(&rtd->delayed_work); - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_START); - - snd_soc_dai_digital_mute(codec_dai, 0); - -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Called by ALSA when the hardware params are set by application. This - * function can also be called multiple times and can allocate buffers - * (using snd_pcm_lib_* ). It's non-atomic. - */ -static int soc_pcm_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_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { - ret = rtd->dai_link->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: machine hw_params failed\n"); - goto out; - } - } - - if (codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); - goto codec_err; - } - } - - if (cpu_dai->driver->ops->hw_params) { - ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); - goto interface_err; - } - } - - if (platform->driver->ops && platform->driver->ops->hw_params) { - ret = platform->driver->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); - goto platform_err; - } - } - - rtd->rate = params_rate(params); - -out: - mutex_unlock(&pcm_mutex); - return ret; - -platform_err: - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - -interface_err: - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - -codec_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Frees resources allocated by hw_params, can be called multiple times - */ -static int soc_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - /* apply codec digital mute */ - if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); - - /* free any machine hw params */ - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - /* free any DMA resources */ - if (platform->driver->ops && platform->driver->ops->hw_free) - platform->driver->ops->hw_free(substream); - - /* now free hw params for the DAIs */ - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - - mutex_unlock(&pcm_mutex); - return 0; -} - -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; - } - - if (platform->driver->ops && platform->driver->ops->trigger) { - ret = platform->driver->ops->trigger(substream, cmd); - if (ret < 0) - return ret; - } - - if (cpu_dai->driver->ops->trigger) { - ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; - } - return 0; -} - -/* - * soc level wrapper for pointer callback - * If cpu_dai, codec_dai, platform driver has the delay callback, than - * the runtime->delay will be updated accordingly. - */ -static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t offset = 0; - snd_pcm_sframes_t delay = 0; - - if (platform->driver->ops && platform->driver->ops->pointer) - offset = platform->driver->ops->pointer(substream); - - if (cpu_dai->driver->ops->delay) - delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - - if (codec_dai->driver->ops->delay) - delay += codec_dai->driver->ops->delay(substream, codec_dai); - - if (platform->driver->delay) - delay += platform->driver->delay(substream, codec_dai); - - runtime->delay = delay; - - return offset; -} - -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_codec_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - #ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ int snd_soc_suspend(struct device *dev) @@ -1692,9 +1145,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (order != SND_SOC_COMP_ORDER_LAST) return 0; - /* DAPM dai link stream work */ - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); - ret = soc_post_component_init(card, codec, num, 0); if (ret) return ret; @@ -2114,66 +1564,6 @@ static struct platform_driver soc_driver = { .remove = soc_remove, }; -/* create a new pcm */ -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_pcm *pcm; - char new_name[64]; - int ret = 0, playback = 0, capture = 0; - - /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - - if (codec_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min) - capture = 1; - - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); - ret = snd_pcm_new(rtd->card->snd_card, new_name, - num, playback, capture, &pcm); - if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); - return ret; - } - - rtd->pcm = pcm; - pcm->private_data = rtd; - if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; - } - - if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); - - if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); - - if (platform->driver->pcm_new) { - ret = platform->driver->pcm_new(rtd); - if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); - return ret; - } - } - - pcm->private_free = platform->driver->pcm_free; - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); - return ret; -} - /** * snd_soc_codec_volatile_register: Report if a register is volatile. * diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c new file mode 100644 index 0000000..7b725be --- /dev/null +++ b/sound/soc/soc-pcm.c @@ -0,0 +1,645 @@ +/* + * soc-pcm.c -- ALSA SoC PCM + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. + * + * Authors: Liam Girdwood <lrg(a)ti.com> + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * 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/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/debugfs.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/ac97_codec.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> + +static DEFINE_MUTEX(pcm_mutex); + +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (!codec_dai->driver->symmetric_rates && + !cpu_dai->driver->symmetric_rates && + !rtd->dai_link->symmetric_rates) + return 0; + + /* This can happen if multiple streams are starting simultaneously - + * the second can need to get its constraints before the first has + * picked a rate. Complain and allow the application to carry on. + */ + if (!rtd->rate) { + dev_warn(&rtd->dev, + "Not enforcing symmetric_rates due to race\n"); + return 0; + } + + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + rtd->rate, rtd->rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; + } + + return 0; +} + +/* + * Called by ALSA when a PCM substream is opened, the runtime->hw record is + * then initialized and any private data can be allocated. This also calls + * startup for the cpu DAI, platform, machine and codec DAI. + */ +static int soc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; + int ret = 0; + + mutex_lock(&pcm_mutex); + + /* startup the audio subsystem */ + if (cpu_dai->driver->ops->startup) { + ret = cpu_dai->driver->ops->startup(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open interface %s\n", + cpu_dai->name); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->open) { + ret = platform->driver->ops->open(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + goto platform_err; + } + } + + if (codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open codec %s\n", + codec_dai->name); + goto codec_dai_err; + } + } + + if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + ret = rtd->dai_link->ops->startup(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); + goto machine_err; + } + } + + /* Check that the codec and cpu DAIs are compatible */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.rate_min = + max(codec_dai_drv->playback.rate_min, + cpu_dai_drv->playback.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->playback.rate_max, + cpu_dai_drv->playback.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->playback.channels_min, + cpu_dai_drv->playback.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->playback.channels_max, + cpu_dai_drv->playback.channels_max); + runtime->hw.formats = + codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; + runtime->hw.rates = + codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; + if (codec_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->playback.rates; + if (cpu_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->playback.rates; + } else { + runtime->hw.rate_min = + max(codec_dai_drv->capture.rate_min, + cpu_dai_drv->capture.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->capture.rate_max, + cpu_dai_drv->capture.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->capture.channels_min, + cpu_dai_drv->capture.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->capture.channels_max, + cpu_dai_drv->capture.channels_max); + runtime->hw.formats = + codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; + runtime->hw.rates = + codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; + if (codec_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->capture.rates; + if (cpu_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->capture.rates; + } + + ret = -EINVAL; + snd_pcm_limit_hw_rates(runtime); + if (!runtime->hw.rates) { + printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.formats) { + printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.channels_min || !runtime->hw.channels_max || + runtime->hw.channels_min > runtime->hw.channels_max) { + printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + + /* Symmetry only applies if we've already got an active stream. */ + if (cpu_dai->active || codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream); + if (ret != 0) + goto config_err; + } + + pr_debug("asoc: %s <-> %s info:\n", + codec_dai->name, cpu_dai->name); + pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); + pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, + runtime->hw.channels_max); + pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, + runtime->hw.rate_max); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active++; + codec_dai->playback_active++; + } else { + cpu_dai->capture_active++; + codec_dai->capture_active++; + } + cpu_dai->active++; + codec_dai->active++; + rtd->codec->active++; + mutex_unlock(&pcm_mutex); + return 0; + +config_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + +machine_err: + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + +codec_dai_err: + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + +platform_err: + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); +out: + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Power down the audio subsystem pmdown_time msecs after close is called. + * This is to ensure there are no pops or clicks in between any music tracks + * due to DAPM power cycling. + */ +static void close_delayed_work(struct work_struct *work) +{ + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + mutex_lock(&pcm_mutex); + + pr_debug("pop wq checking: %s status: %s waiting: %s\n", + codec_dai->driver->playback.stream_name, + codec_dai->playback_active ? "active" : "inactive", + codec_dai->pop_wait ? "yes" : "no"); + + /* are we waiting on this codec DAI stream */ + if (codec_dai->pop_wait == 1) { + codec_dai->pop_wait = 0; + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&pcm_mutex); +} + +/* + * Called by ALSA when a PCM substream is closed. Private data can be + * freed here. The cpu DAI, codec DAI, machine and platform are also + * shutdown. + */ +static int soc_codec_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock(&pcm_mutex); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active--; + codec_dai->playback_active--; + } else { + cpu_dai->capture_active--; + codec_dai->capture_active--; + } + + cpu_dai->active--; + codec_dai->active--; + codec->active--; + + /* Muting the DAC suppresses artifacts caused during digital + * shutdown, for example from stopping clocks. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_digital_mute(codec_dai, 1); + + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); + + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + cpu_dai->runtime = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* start delayed pop wq here for playback streams */ + codec_dai->pop_wait = 1; + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); + } else { + /* capture streams can be powered down now */ + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&pcm_mutex); + return 0; +} + +/* + * Called by ALSA when the PCM substream is prepared, can set format, sample + * rate, etc. This function is non atomic and can be called multiple times, + * it can refer to the runtime info. + */ +static int soc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock(&pcm_mutex); + + if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + ret = rtd->dai_link->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: machine prepare error\n"); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->prepare) { + ret = platform->driver->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: platform prepare error\n"); + goto out; + } + } + + if (codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: codec DAI prepare error\n"); + goto out; + } + } + + if (cpu_dai->driver->ops->prepare) { + ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: cpu DAI prepare error\n"); + goto out; + } + } + + /* cancel any delayed stream shutdown that is pending */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + codec_dai->pop_wait) { + codec_dai->pop_wait = 0; + cancel_delayed_work(&rtd->delayed_work); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_START); + + snd_soc_dai_digital_mute(codec_dai, 0); + +out: + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Called by ALSA when the hardware params are set by application. This + * function can also be called multiple times and can allocate buffers + * (using snd_pcm_lib_* ). It's non-atomic. + */ +static int soc_pcm_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_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock(&pcm_mutex); + + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + ret = rtd->dai_link->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: machine hw_params failed\n"); + goto out; + } + } + + if (codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't set codec %s hw params\n", + codec_dai->name); + goto codec_err; + } + } + + if (cpu_dai->driver->ops->hw_params) { + ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: interface %s hw params failed\n", + cpu_dai->name); + goto interface_err; + } + } + + if (platform->driver->ops && platform->driver->ops->hw_params) { + ret = platform->driver->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: platform %s hw params failed\n", + platform->name); + goto platform_err; + } + } + + rtd->rate = params_rate(params); + +out: + mutex_unlock(&pcm_mutex); + return ret; + +platform_err: + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + +interface_err: + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + +codec_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + mutex_unlock(&pcm_mutex); + return ret; +} + +/* + * Frees resources allocated by hw_params, can be called multiple times + */ +static int soc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock(&pcm_mutex); + + /* apply codec digital mute */ + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); + + /* free any machine hw params */ + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + /* free any DMA resources */ + if (platform->driver->ops && platform->driver->ops->hw_free) + platform->driver->ops->hw_free(substream); + + /* now free hw params for the DAIs */ + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + + mutex_unlock(&pcm_mutex); + return 0; +} + +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); + if (ret < 0) + return ret; + } + + if (platform->driver->ops && platform->driver->ops->trigger) { + ret = platform->driver->ops->trigger(substream, cmd); + if (ret < 0) + return ret; + } + + if (cpu_dai->driver->ops->trigger) { + ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); + if (ret < 0) + return ret; + } + return 0; +} + +/* + * soc level wrapper for pointer callback + * If cpu_dai, codec_dai, platform driver has the delay callback, than + * the runtime->delay will be updated accordingly. + */ +static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t offset = 0; + snd_pcm_sframes_t delay = 0; + + if (platform->driver->ops && platform->driver->ops->pointer) + offset = platform->driver->ops->pointer(substream); + + if (cpu_dai->driver->ops->delay) + delay += cpu_dai->driver->ops->delay(substream, cpu_dai); + + if (codec_dai->driver->ops->delay) + delay += codec_dai->driver->ops->delay(substream, codec_dai); + + if (platform->driver->delay) + delay += platform->driver->delay(substream, codec_dai); + + runtime->delay = delay; + + return offset; +} + +/* ASoC PCM operations */ +static struct snd_pcm_ops soc_pcm_ops = { + .open = soc_pcm_open, + .close = soc_codec_close, + .hw_params = soc_pcm_hw_params, + .hw_free = soc_pcm_hw_free, + .prepare = soc_pcm_prepare, + .trigger = soc_pcm_trigger, + .pointer = soc_pcm_pointer, +}; + +/* create a new pcm */ +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm *pcm; + char new_name[64]; + int ret = 0, playback = 0, capture = 0; + + /* check client and interface hw capabilities */ + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); + ret = snd_pcm_new(rtd->card->snd_card, new_name, + num, playback, capture, &pcm); + if (ret < 0) { + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + return ret; + } + + /* DAPM dai link stream work */ + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + + rtd->pcm = pcm; + pcm->private_data = rtd; + if (platform->driver->ops) { + soc_pcm_ops.mmap = platform->driver->ops->mmap; + soc_pcm_ops.pointer = platform->driver->ops->pointer; + soc_pcm_ops.ioctl = platform->driver->ops->ioctl; + soc_pcm_ops.copy = platform->driver->ops->copy; + soc_pcm_ops.silence = platform->driver->ops->silence; + soc_pcm_ops.ack = platform->driver->ops->ack; + soc_pcm_ops.page = platform->driver->ops->page; + } + + if (playback) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + + if (capture) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd); + if (ret < 0) { + pr_err("asoc: platform pcm constructor failed\n"); + return ret; + } + } + + pcm->private_free = platform->driver->pcm_free; + printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, + cpu_dai->name); + return ret; +} -- 1.7.4.1

1 1

Re: [alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Takashi Iwai 09 Jun '11

09 Jun '11

At Thu, 9 Jun 2011 11:28:26 +0200, Michal Hocko wrote: > > [1 <text/plain; us-ascii (7bit)>] > On Thu 09-06-11 11:20:04, Takashi Iwai wrote: > > At Thu, 9 Jun 2011 11:06:07 +0200, > > Michal Hocko wrote: > > > > > > On Thu 09-06-11 10:56:16, Michal Hocko wrote: > > > > On Thu 09-06-11 08:09:18, Takashi Iwai wrote: > > > > > At Wed, 8 Jun 2011 19:48:55 +0200, > > > [...] > > > > > Thanks, that's helpful. > > > > > As a quirk solution, could you try to pass model=auto option to > > > > > snd-hda-intel module? > > > > > > > > Headphones work now but speakers are quite... I have tried to play > > > > with Headphone and Speaker controls in alsamixer but no changes. > > > > > > > > Here is what I get in logs when the module is loaded: > > > > [ 5320.360277] HDA Intel 0000:00:1b.0: PCI INT A disabled > > > > [ 5326.920247] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > > > [ 5326.920262] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > > > [ 5326.920279] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 > > > > [ 5326.920375] HDA Intel 0000:00:1b.0: irq 42 for MSI/MSI-X > > > > [ 5326.920405] HDA Intel 0000:00:1b.0: setting latency timer to 64 > > > > [ 5326.981085] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input18 > > > > [ 5326.981262] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input19 > > > > [ 5326.987934] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input20 > > > > > > > > [maybe not important but input numbers are increasing for every new load] > > > > > > > > There is another strange thing. After I stop mplayer and wait for some > > > > time (~1min) I can hear a "click" sound and then there is no sound at > > > > all... If I start mplayer in the meantime it just works... > > > > If I try to remove the module and load it again then things work and stop > > > > working after first playback again. > > > > > > And I have just found out that I do not have to start any playback to > > > see this. It is sufficient to load the module and wait a minute and > > > there is no sound at all (both from speakers or headphones). > > > > OK, could you give alsa-info.sh outputs at both working and > > non-working states? > > Sure. Attached. The only change seems to be: > Node 0x02 [Audio Output] wcaps 0x11: Stereo > Device: name="ALC262 Analog", type="Audio", device=0 > - Converter: stream=5, channel=0 > + Converter: stream=0, channel=0 This is OK. It shows that the converter was assigned once to this stream when you played back a sound before power-save. As long as the codec registers show the same value, it's not obviously what went wrong. The symptom sounds like a resume issue. If so, S3/S4 might show the same problem. BTW, does my patch fix the problem when you load without model=auto? If yes, does the problem appear with it? I.e. is the silence problem specific to model=auto or not? thanks, Takashi

1 0

Re: [alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Takashi Iwai 09 Jun '11

09 Jun '11

At Thu, 9 Jun 2011 11:06:07 +0200, Michal Hocko wrote: > > On Thu 09-06-11 10:56:16, Michal Hocko wrote: > > On Thu 09-06-11 08:09:18, Takashi Iwai wrote: > > > At Wed, 8 Jun 2011 19:48:55 +0200, > [...] > > > Thanks, that's helpful. > > > As a quirk solution, could you try to pass model=auto option to > > > snd-hda-intel module? > > > > Headphones work now but speakers are quite... I have tried to play > > with Headphone and Speaker controls in alsamixer but no changes. > > > > Here is what I get in logs when the module is loaded: > > [ 5320.360277] HDA Intel 0000:00:1b.0: PCI INT A disabled > > [ 5326.920247] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > [ 5326.920262] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > > [ 5326.920279] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 > > [ 5326.920375] HDA Intel 0000:00:1b.0: irq 42 for MSI/MSI-X > > [ 5326.920405] HDA Intel 0000:00:1b.0: setting latency timer to 64 > > [ 5326.981085] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input18 > > [ 5326.981262] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input19 > > [ 5326.987934] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input20 > > > > [maybe not important but input numbers are increasing for every new load] > > > > There is another strange thing. After I stop mplayer and wait for some > > time (~1min) I can hear a "click" sound and then there is no sound at > > all... If I start mplayer in the meantime it just works... > > If I try to remove the module and load it again then things work and stop > > working after first playback again. > > And I have just found out that I do not have to start any playback to > see this. It is sufficient to load the module and wait a minute and > there is no sound at all (both from speakers or headphones). OK, could you give alsa-info.sh outputs at both working and non-working states? thanks, Takashi

1 0

Re: [alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Takashi Iwai 09 Jun '11

09 Jun '11

At Thu, 9 Jun 2011 10:56:16 +0200, Michal Hocko wrote: > > On Thu 09-06-11 08:09:18, Takashi Iwai wrote: > > At Wed, 8 Jun 2011 19:48:55 +0200, > > Michal Hocko wrote: > > > > > > On Wed 08-06-11 18:03:24, Takashi Iwai wrote: > > > > At Tue, 7 Jun 2011 20:05:58 +0200, > > > > Michal Hocko wrote: > > > > > > > > > > Hi, > > > > > before 3.0.0-rc1, when I plugged anything into headphone output of my > > > > > laptop, speakers were disabled and the sound has been routed to the > > > > > headphone output. This doesn't seem to work anymore starting with > > > > > 3.0.0-rc1. Speakers stay on even if I plug something to the headphone > > > > > output. > > > > > Is this a bug or a new behavior that userspace has to cope > > > > > with? > > > > > > > > > > I am sorry for the lack of information but I have no idea what to > > > > > provide. > > > > > > > > Please give alsa-info.sh output at best. Run it with --no-upload > > > > option, and attach the output file. This will contain most of > > > > necessary information. > > > > > > See attached. I have added also .39 output where things work as expected > > > (just in case it could be useful). > > > > Thanks, that's helpful. > > As a quirk solution, could you try to pass model=auto option to > > snd-hda-intel module? > > Headphones work now but speakers are quite... I have tried to play > with Headphone and Speaker controls in alsamixer but no changes. > > Here is what I get in logs when the module is loaded: > [ 5320.360277] HDA Intel 0000:00:1b.0: PCI INT A disabled > [ 5326.920247] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > [ 5326.920262] HDA Intel 0000:00:1b.0: power state changed by ACPI to D0 > [ 5326.920279] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 > [ 5326.920375] HDA Intel 0000:00:1b.0: irq 42 for MSI/MSI-X > [ 5326.920405] HDA Intel 0000:00:1b.0: setting latency timer to 64 > [ 5326.981085] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input18 > [ 5326.981262] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input19 > [ 5326.987934] input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input20 > > [maybe not important but input numbers are increasing for every new load] No bug, it's the implementation issue of input layer :) > There is another strange thing. After I stop mplayer and wait for some > time (~1min) I can hear a "click" sound and then there is no sound at > all... If I start mplayer in the meantime it just works... This is possibly the powersave. It means that the resume doesn't work properly after powersave. > If I try to remove the module and load it again then things work and stop > working after first playback again. > > > You may need to adjust the mixer after changing > > this, but it should provide the full (even better) functionality. > > It'll support also the auto-mic jack switching. > > $ amixer | grep -i "auto.*mic" > What is the control to look at? There is no control. Simply plugging the mic jack will switch the recording source automatically. Takashi

1 0

[alsa-devel] Retrieving the number of input channels on a sound card
by André Prado 09 Jun '11

09 Jun '11

Hello. I am new to Alsa programming and i want to get the number of input channels on an audio device. I am reading a lot of tutorials and documentations but i am struggling to do this. Here is a code i've saw to Lists the hardware names of MIDI output device/sub-devices. I've understood it. But i want to get the number of input channels. Any help is appreciated. THanks // listmidi.c // Lists the hardware names of MIDI output device/sub-devices // upon each ALSA sound card in the system. // // Compile as: // gcc -o listmidi listmidi.c -lasound #include <stdio.h> #include <string.h> #include <alsa/asoundlib.h> int main(int argc, char **argv) { register int err; int cardNum; // Start with first card cardNum = -1; for (;;) { snd_ctl_t *cardHandle; // Get next sound card's card number. When "cardNum" == -1, then ALSA // fetches the first card if ((err = snd_card_next(&cardNum)) < 0) { printf("Can't get the next card number: %s\n", snd_strerror(err)); break; } // No more cards? ALSA sets "cardNum" to -1 if so if (cardNum < 0) break; // Open this card's control interface. We specify only the card number -- not // any device nor sub-device too { char str[64]; sprintf(str, "hw:%i", cardNum); if ((err = snd_ctl_open(&cardHandle, str, 0)) < 0) { printf("Can't open card %i: %s\n", cardNum, snd_strerror(err)); continue; } } { int devNum; // Start with the first MIDI device on this card devNum = -1; for (;;) { snd_rawmidi_info_t *rawMidiInfo; register int subDevCount, i; // Get the number of the next MIDI device on this card if ((err = snd_ctl_rawmidi_next_device(cardHandle, &devNum)) < 0) { printf("Can't get next MIDI device number: %s\n", snd_strerror(err)); break; } // No more MIDI devices on this card? ALSA sets "devNum" to -1 if so. // NOTE: It's possible that this sound card may have no MIDI devices on it // at all, for example if it's only a digital audio card if (devNum < 0) break; // To get some info about the subdevices of this MIDI device (on the card), we need a // snd_rawmidi_info_t, so let's allocate one on the stack snd_rawmidi_info_alloca(&rawMidiInfo); memset(rawMidiInfo, 0, snd_rawmidi_info_sizeof()); // Tell ALSA which device (number) we want info about snd_rawmidi_info_set_device(rawMidiInfo, devNum); // Get info on the MIDI outs of this device snd_rawmidi_info_set_stream(rawMidiInfo, SND_RAWMIDI_STREAM_OUTPUT); i = -1; subDevCount = 1; // More subdevices? while (++i < subDevCount) { // Tell ALSA to fill in our snd_rawmidi_info_t with info on this subdevice snd_rawmidi_info_set_subdevice(rawMidiInfo, i); if ((err = snd_ctl_rawmidi_info(cardHandle, rawMidiInfo)) < 0) { printf("Can't get info for MIDI output subdevice hw:%i,%i,%i: %s\n", cardNum, devNum, i, snd_strerror(err)); continue; } // Print out how many subdevices (once only) if (!i) { subDevCount = snd_rawmidi_info_get_subdevices_count(rawMidiInfo); printf("\nFound %i MIDI output subdevices on card %i\n", subDevCount, cardNum); } // NOTE: If there's only one subdevice, then the subdevice number is immaterial, // and can be omitted when you specify the hardware name printf((subDevCount > 1 ? " hw:%i,%i,%i\n" : " hw:%i,%i\n"), cardNum, devNum, i); } } } // Close the card's control interface after we're done with it snd_ctl_close(cardHandle); } snd_config_update_free_global(); } -- Att André

2 2

Re: [alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Takashi Iwai 09 Jun '11

09 Jun '11

At Wed, 8 Jun 2011 19:48:55 +0200, Michal Hocko wrote: > > On Wed 08-06-11 18:03:24, Takashi Iwai wrote: > > At Tue, 7 Jun 2011 20:05:58 +0200, > > Michal Hocko wrote: > > > > > > Hi, > > > before 3.0.0-rc1, when I plugged anything into headphone output of my > > > laptop, speakers were disabled and the sound has been routed to the > > > headphone output. This doesn't seem to work anymore starting with > > > 3.0.0-rc1. Speakers stay on even if I plug something to the headphone > > > output. > > > Is this a bug or a new behavior that userspace has to cope > > > with? > > > > > > I am sorry for the lack of information but I have no idea what to > > > provide. > > > > Please give alsa-info.sh output at best. Run it with --no-upload > > option, and attach the output file. This will contain most of > > necessary information. > > See attached. I have added also .39 output where things work as expected > (just in case it could be useful). Thanks, that's helpful. As a quirk solution, could you try to pass model=auto option to snd-hda-intel module? You may need to adjust the mixer after changing this, but it should provide the full (even better) functionality. It'll support also the auto-mic jack switching. If auto-parser works with this hardware, we don't need to debug the old code but just switch to the auto-parser statically in the driver. thanks, Takashi

1 1

[alsa-devel] [PATCH] ASoC: WM8804 does not support sample rates below 32kHz
by Mark Brown 08 Jun '11

08 Jun '11

Reported-by: Kieran O'Leary <Kieran.O'Leary(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8804.c | 9 +++++++-- 1 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6785688..9a5e67c 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = { #define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) +#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + static struct snd_soc_dai_driver wm8804_dai = { .name = "wm8804-spdif", .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = WM8804_RATES, .formats = WM8804_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = WM8804_RATES, .formats = WM8804_FORMATS, }, .ops = &wm8804_dai_ops, -- 1.7.5.3

2 1

[alsa-devel] [PATCH] ASoC: Fix mismerge of Speyside set_bias_level_post()
by Mark Brown 08 Jun '11

08 Jun '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 2515dba..d6dee4d 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -301,6 +301,7 @@ static struct snd_soc_card speyside = { .num_configs = ARRAY_SIZE(speyside_codec_conf), .set_bias_level = speyside_set_bias_level, + .set_bias_level_post = speyside_set_bias_level_post, .controls = controls, .num_controls = ARRAY_SIZE(controls), -- 1.7.5.3

2 1

[alsa-devel] [PATCH] UCM: get: Implement 'exact' value retrieval
by Stephen Warren 08 Jun '11

08 Jun '11

[=]<NAME>[/[<modifier>|</device>][/<verb>]] - value identifier <NAME> - Search starts at given modifier or device if any, else at a verb - Search starts at given verb if any, else current verb - Searches modifier/device, then verb, then defaults - Specify a leading "=" to search only the exact device/modifier/verb specified, and not search through each object in turn. - Examples: "PlaybackPCM/Play Music" "CapturePCM/SPDIF" From ValueDefaults only: "=Variable" From current active verb: "=Variable//" From verb "Verb": "=Variable//Verb" From "Modifier" in current active verb: "=Variable/Modifier/" From "Modifier" in "Verb": "=Variable/Modifier/Verb" Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- include/use-case.h | 30 +++++++++++--- src/ucm/main.c | 110 +++++++++++++++++++++++++++++++++++++-------------- 2 files changed, 104 insertions(+), 36 deletions(-) diff --git a/include/use-case.h b/include/use-case.h index b346877..c37c842 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -230,12 +230,30 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * NULL - return current card * _verb - return current verb * - * <NAME>[/<modifier>|</device>] - * - value identifier <NAME> for - * given modifier or device - * - Examples: "PlaybackPCM/Play Music", - * "CapturePCM/SPDIF" - * + * [=]<NAME>[/[<modifier>|</device>][/<verb>]] + * - value identifier <NAME> + * - Search starts at given modifier or device if any, + * else at a verb + * - Search starts at given verb if any, + * else current verb + * - Searches modifier/device, then verb, then defaults + * - Specify a leading "=" to search only the exact + * device/modifier/verb specified, and not search + * through each object in turn. + * - Examples: + * "PlaybackPCM/Play Music" + * "CapturePCM/SPDIF" + * From ValueDefaults only: + * "=Variable" + * From current active verb: + * "=Variable//" + * From verb "Verb": + * "=Variable//Verb" + * From "Modifier" in current active verb: + * "=Variable/Modifier/" + * From "Modifier" in "Verb": + * "=Variable/Modifier/Verb" + * * Recommended names for values: * TQ - Tone Quality * PlaybackPCM - full PCM playback device name diff --git a/src/ucm/main.c b/src/ucm/main.c index 13ea1ed..8e9a85d 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1174,35 +1174,60 @@ static int get_value3(const char **value, static int get_value(snd_use_case_mgr_t *uc_mgr, const char *identifier, const char **value, - const char *item) + const char *mod_dev_name, + const char *verb_name, + int exact) { + struct use_case_verb *verb; struct use_case_modifier *mod; struct use_case_device *dev; int err; - if (!uc_mgr->active_verb) - return -ENOENT; - - if (item != NULL) { - mod = find_modifier(uc_mgr, uc_mgr->active_verb, item, 0); - if (mod != NULL) { - err = get_value1(value, &mod->value_list, identifier); - if (err >= 0 || err != -ENOENT) - return err; + if (mod_dev_name || verb_name || !exact) { + if (verb_name && strlen(verb_name)) { + verb = find_verb(uc_mgr, verb_name); + } else { + verb = uc_mgr->active_verb; } - dev = find_device(uc_mgr, uc_mgr->active_verb, item, 0); - if (dev != NULL) { - err = get_value1(value, &dev->value_list, identifier); + if (verb) { + if (mod_dev_name) { + mod = find_modifier(uc_mgr, verb, + mod_dev_name, 0); + if (mod) { + err = get_value1(value, + &mod->value_list, + identifier); + if (err >= 0 || err != -ENOENT) + return err; + } + + dev = find_device(uc_mgr, verb, + mod_dev_name, 0); + if (dev) { + err = get_value1(value, + &dev->value_list, + identifier); + if (err >= 0 || err != -ENOENT) + return err; + } + + if (exact) + return -ENOENT; + } + + err = get_value1(value, &verb->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } + + if (exact) + return -ENOENT; } - err = get_value1(value, &uc_mgr->active_verb->value_list, identifier); - if (err >= 0 || err != -ENOENT) - return err; + err = get_value1(value, &uc_mgr->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; + return -ENOENT; } @@ -1220,7 +1245,9 @@ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, const char *identifier, const char **value) { - char *str, *str1; + const char *slash1, *slash2, *mod_dev_after; + const char *ident, *mod_dev, *verb; + int exact = 0; int err; pthread_mutex_lock(&uc_mgr->mutex); @@ -1246,19 +1273,42 @@ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, err = -ENOENT; goto __end; } else { - str1 = strchr(identifier, '/'); - if (str1) { - str = strdup(str1 + 1); - if (str == NULL) { - err = -ENOMEM; - goto __end; - } - } else { - str = NULL; - } - err = get_value(uc_mgr, identifier, value, str); - if (str) - free(str); + if (identifier[0] == '=') { + exact = 1; + identifier++; + } + + slash1 = strchr(identifier, '/'); + if (slash1) { + ident = strndup(identifier, slash1 - identifier); + + slash2 = strchr(slash1 + 1, '/'); + if (slash2) { + mod_dev_after = slash2; + verb = slash2 + 1; + } + else { + mod_dev_after = slash1 + strlen(slash1); + verb = NULL; + } + + if (mod_dev_after == slash1 + 1) + mod_dev = NULL; + else + mod_dev = strndup(slash1 + 1, + mod_dev_after - (slash1 + 1)); + } + else { + ident = identifier; + mod_dev = NULL; + verb = NULL; + } + + err = get_value(uc_mgr, ident, value, mod_dev, verb, exact); + if (ident != identifier) + free((void *)ident); + if (mod_dev) + free((void *)mod_dev); } __end: pthread_mutex_unlock(&uc_mgr->mutex); -- 1.7.0.4

3 5

[alsa-devel] Headphone output doesn't work with 3.0.0-rc* anymore
by Michal Hocko 08 Jun '11

08 Jun '11

Hi, before 3.0.0-rc1, when I plugged anything into headphone output of my laptop, speakers were disabled and the sound has been routed to the headphone output. This doesn't seem to work anymore starting with 3.0.0-rc1. Speakers stay on even if I plug something to the headphone output. Is this a bug or a new behavior that userspace has to cope with? I am sorry for the lack of information but I have no idea what to provide. Let's add dmesg and config at least. Btw. I have noticed the following message which might be related: input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 I do not see anything like this in .39 kernel. I am using Intel HDA integrated soundcard: $ lspci -vv [...] 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) Subsystem: Fujitsu Limited. Device 1397 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 42 Region 0: Memory at f0640000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4189 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=1 ArbSelect=Fixed TC/VC=80 Status: NegoPending- InProgress- Capabilities: [130 v1] Root Complex Link Desc: PortNumber=0f ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c000 Kernel driver in use: HDA Intel [...] $ lsmod | grep snd snd_hda_codec_realtek 236509 1 snd_hda_intel 23614 1 snd_hda_codec 66591 2 snd_hda_codec_realtek,snd_hda_intel snd_pcm_oss 36561 0 snd_mixer_oss 17908 1 snd_pcm_oss snd_pcm 64165 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_oss 29025 0 snd_seq_midi_event 14076 1 snd_seq_oss snd_seq 42319 4 snd_seq_oss,snd_seq_midi_event snd_timer 23709 2 snd_pcm,snd_seq snd_seq_device 13797 2 snd_seq_oss,snd_seq snd 49969 12 snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_seq_device snd_page_alloc 13709 2 snd_hda_intel,snd_pcm $ grep -I . /sys/devices/pci0000\:00/0000\:00\:1b.0/* /sys/devices/pci0000:00/0000:00:1b.0/broken_parity_status:0 /sys/devices/pci0000:00/0000:00:1b.0/class:0x040300 /sys/devices/pci0000:00/0000:00:1b.0/consistent_dma_mask_bits:64 /sys/devices/pci0000:00/0000:00:1b.0/device:0x27d8 /sys/devices/pci0000:00/0000:00:1b.0/dma_mask_bits:64 /sys/devices/pci0000:00/0000:00:1b.0/enable:1 /sys/devices/pci0000:00/0000:00:1b.0/irq:42 /sys/devices/pci0000:00/0000:00:1b.0/local_cpulist:0-7 /sys/devices/pci0000:00/0000:00:1b.0/local_cpus:ff /sys/devices/pci0000:00/0000:00:1b.0/modalias:pci:v00008086d000027D8sv000010CFsd00001397bc04sc03i00 /sys/devices/pci0000:00/0000:00:1b.0/resource:0x00000000f0640000 0x00000000f0643fff 0x0000000000140204 /sys/devices/pci0000:00/0000:00:1b.0/resource:0x0000000000000000 0x0000000000000000 0x0000000000000000 /sys/devices/pci0000:00/0000:00:1b.0/resource:0x0000000000000000 0x0000000000000000 0x0000000000000000 /sys/devices/pci0000:00/0000:00:1b.0/resource:0x0000000000000000 0x0000000000000000 0x0000000000000000 /sys/devices/pci0000:00/0000:00:1b.0/resource:0x0000000000000000 0x0000000000000000 0x0000000000000000 /sys/devices/pci0000:00/0000:00:1b.0/resource:0x0000000000000000 0x0000000000000000 0x0000000000000000 /sys/devices/pci0000:00/0000:00:1b.0/resource:0x0000000000000000 0x0000000000000000 0x0000000000000000 /sys/devices/pci0000:00/0000:00:1b.0/subsystem_device:0x1397 /sys/devices/pci0000:00/0000:00:1b.0/subsystem_vendor:0x10cf /sys/devices/pci0000:00/0000:00:1b.0/uevent:DRIVER=HDA Intel /sys/devices/pci0000:00/0000:00:1b.0/uevent:PCI_CLASS=40300 /sys/devices/pci0000:00/0000:00:1b.0/uevent:PCI_ID=8086:27D8 /sys/devices/pci0000:00/0000:00:1b.0/uevent:PCI_SUBSYS_ID=10CF:1397 /sys/devices/pci0000:00/0000:00:1b.0/uevent:PCI_SLOT_NAME=0000:00:1b.0 /sys/devices/pci0000:00/0000:00:1b.0/uevent:MODALIAS=pci:v00008086d000027D8sv000010CFsd00001397bc04sc03i00 /sys/devices/pci0000:00/0000:00:1b.0/vendor:0x8086 Let me know if you need any further information. -- Michal Hocko SUSE Labs SUSE LINUX s.r.o. Lihovarska 1060/12 190 00 Praha 9 Czech Republic

2 1

[alsa-devel] [PATCH 1/6] ASoC: Fix WM8962 headphone volume update for use of advanced caches
by Mark Brown 08 Jun '11

08 Jun '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index f90ae42..5e05eed 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA) + if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA) return snd_soc_write(codec, WM8962_HPOUTL_VOLUME, reg_cache[WM8962_HPOUTL_VOLUME]); /* ...otherwise the right. The VU is stereo. */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA) + if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA) return snd_soc_write(codec, WM8962_HPOUTR_VOLUME, reg_cache[WM8962_HPOUTR_VOLUME]); -- 1.7.5.3

2 6

[alsa-devel] Something broke in snd_soc_pcm_stream
by Timur Tabi 08 Jun '11

08 Jun '11

Mark, I'm testing your for-3.1 branch on the P1022DS, and I'm getting a kernel panic because a pointer is NULL where I don't expect it to be. In fsl_dma_new(), I have this code: if (dai->driver->capture.channels_min) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { dev_err(card->dev, "can't alloc capture dma buffer\n"); snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); return ret; } } I test for channels_min because the playback stream is registered separately from the capture stream, so I expect fsl_dma_new() to be called twice: once to initialize playback, and again to initialize capture. In the past, if dai->driver->capture.channels_min was equal to some non-zero value, then pcm->streams[1].substream was also non-NULL. This appears no longer to be true. Now it appears that playback.channels_min and capture.channels_min are both set to 2 before fsl_dma_new() is called the first time. I can change my code to do this: if (pcm->streams[1].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { dev_err(card->dev, "can't alloc capture dma buffer\n"); snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); return ret; } } But before I do that, I'd like to know if this is a bug in ASoC, or just a new behavior that I need to handle. -- Timur Tabi Linux kernel developer at Freescale

2 3

[alsa-devel] [PATCH V4 0/5] Adding ASoC drivers for SPEAr13XX platform
by Rajeev Kumar 08 Jun '11

08 Jun '11

This patchset contains five patches which add support for ASoC sound drivers on ST's SPEAr13XX platform. Details of the SPEAr13XX platforms can be seen here: http://www.st.com/internet/mcu/product/250658.jsp The ARCH and PLATFORM specific code for SPEAr13XX are already under review in Russell King's ARM mailing list(Linux-kernel-version 2.6.38-rc4). This patchset has been tested for linux-kernel-version 2.6.38-rc4 and on ARM platform. Please review the same and consider for mainline inclusion. Changes in machine driver code since V3 1.Use of snd_soc_register_card() rather than using the soc-audio device. Changes in codec code since V3: 1. use standard cache sync implementation i.e. snd_soc_cache_sync() Changes in spear Kconfig and Makefile since V2: 1.giving all the the modules snd-soc- prefixes to their names Changes in codec code since V2: 1. use SOC_DOUBLE for volume control 2. remove default configuration for codec .offer them as runtime controls. 3. restore the register cache in resume changes in platform code since V1: 1. Spilt machine driver from platform driver. 2. Updated rate supported. 3. Moved clock specific part from "spear13xx_i2s_set_dai_sysclk". function to platform code. This function is removed now. 4. Function "spear13xx_i2s_startup" removed. 5. Function "spear13xx_i2s_shutdown" removed. Changes in codec code since V1: 1. Removed sta529 version number. 2. Removed "sta529_read_reg_cache". 3. Removed function "sta529_write_reg_cache". 4. Removed function "sta529_write". 5. Moved sta529.h file contents to sta529.c file. 6. Move clock specific part from "spear_sta_set_dai_sysclk". function to platform code. This function is removed now. 7. Renamed function names from spear_sta_* to sta_*. 8. Modified probe() function to avoid rewriting default chip values to hardware. Rajeev Kumar (5): sound: asoc: Adding support for STA529 Audio Codec sound: asoc: Adding support for SPEAr13XX ASoC platform driver sound: asoc: Adding support for SPEAr13XX ASoC machine driver sound: asoc: Adding Kconfig and Makefile to support SPEAr13XX ASoC driver sound: asoc: Adding support for SPEAr13XX in soc sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta529.c | 374 ++++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 19 ++ sound/soc/spear/Makefile | 12 + sound/soc/spear/evb_sta529.c | 121 +++++++++ sound/soc/spear/spear13xx-i2s.c | 524 +++++++++++++++++++++++++++++++++++++++ sound/soc/spear/spear13xx-i2s.h | 19 ++ sound/soc/spear/spear13xx-pcm.c | 500 +++++++++++++++++++++++++++++++++++++ sound/soc/spear/spear13xx-pcm.h | 50 ++++ 12 files changed, 1628 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/evb_sta529.c create mode 100644 sound/soc/spear/spear13xx-i2s.c create mode 100644 sound/soc/spear/spear13xx-i2s.h create mode 100644 sound/soc/spear/spear13xx-pcm.c create mode 100644 sound/soc/spear/spear13xx-pcm.h

5 16

[alsa-devel] [PATCH v2 00/12] MFD/ASoC/Input: TWL4030/TWL60X0 changes
by Peter Ujfalusi 08 Jun '11

08 Jun '11

Hello, The second version gowned a bit since v1... Changes since v1: - Common pmic related configuration moved out from board files - Rest of the series has been rebased on top of this change Not changed since v1: - TWL6040 MFD driver has not been converted to i2c driver I have looked at this, and while it can be done, I feel that the changes needed in twl-core will be not accepted, since IMHO the twl-core need to be cleaned first. The solution, which can be done with the current code might look a bit hackish. Updated intro message from the v1 series: The series will do four major things, and they are in one series, because they pretty much depending on each other, so it is easier to handle them together. 0. Move the common pmic related configurations out from the board files. New pmic-common.c/h file has been created for this purpose. OMAP3 board files received two patches (pdata, and regualtor changes), OMAP4 board files has been changed with one patch. 1. Rename the twl4030-codec MFD driver to twl4030-audio Since the ASoC multicomponent introduction (2.6.37) the twl4030-codec MFD dirver had been using twl4030-audio to register (twl4030-codec is the ASoC codec driver). The documentation refers to this part of twl as audio block, so it is better to rename the driver as well to avoid confusion. 2. Introduction of MFD driver for twl6040. The TWL6040 audio IC has codec and vibra functionality. Convert the existing ASoC codec driver to use the new MFD driver's interface. 3. Vibrator driver for TWL6040 (Input/ForceFeedback driver) The driver itself, and support for the vibrators in SDP4430 board. The series has been tested on top of merged omap-next, asoc-next, and mfd-next. It applies on top of linux-next as well. Regards, Peter --- Misael Lopez Cruz (3): mfd: twl6040: Add initial support ASoC: twl6040: Convert into TWL6040 MFD child input: Add initial support for TWL6040 vibrator Peter Ujfalusi (9): OMAP: New pmic-common for common TWL configuration OMAP4: Move common twl6030 configuration to pmic-common OMAP3: Move common pmic configuration to pmic-common OMAP3: Move common regulator configuration to pmic-common MFD: twl4030-codec: Rename internals from codec to audio MFD: twl4030-codec -> twl4030-audio: Rename the driver MFD: twl4030-audio: Rename platform data MFD: twl6040: Change platform data for soc codec driver OMAP4: SDP4430: Add twl6040 vibrator platform support arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/board-3430sdp.c | 89 +---- arch/arm/mach-omap2/board-4430sdp.c | 183 +++------ arch/arm/mach-omap2/board-cm-t35.c | 48 +-- arch/arm/mach-omap2/board-devkit8000.c | 37 +-- arch/arm/mach-omap2/board-igep0020.c | 44 +-- arch/arm/mach-omap2/board-ldp.c | 15 +- arch/arm/mach-omap2/board-omap3beagle.c | 60 +--- arch/arm/mach-omap2/board-omap3evm.c | 67 +--- arch/arm/mach-omap2/board-omap3pandora.c | 59 +--- arch/arm/mach-omap2/board-omap3stalker.c | 70 +--- arch/arm/mach-omap2/board-omap3touchbook.c | 57 +--- arch/arm/mach-omap2/board-omap4panda.c | 149 +------ arch/arm/mach-omap2/board-overo.c | 59 +--- arch/arm/mach-omap2/board-rm680.c | 8 +- arch/arm/mach-omap2/board-rx51-peripherals.c | 44 +-- arch/arm/mach-omap2/board-zoom-peripherals.c | 88 +---- arch/arm/mach-omap2/common-board-devices.c | 21 - arch/arm/mach-omap2/common-board-devices.h | 26 +- arch/arm/mach-omap2/pmic-common.c | 304 +++++++++++++ arch/arm/mach-omap2/pmic-common.h | 59 +++ arch/arm/plat-omap/include/plat/irqs.h | 12 +- drivers/input/misc/Kconfig | 13 +- drivers/input/misc/Makefile | 1 + drivers/input/misc/twl4030-vibra.c | 12 +- drivers/input/misc/twl6040-vibra.c | 428 +++++++++++++++++++ drivers/mfd/Kconfig | 8 +- drivers/mfd/Makefile | 3 +- drivers/mfd/twl-core.c | 13 +- drivers/mfd/twl4030-audio.c | 278 ++++++++++++ drivers/mfd/twl4030-codec.c | 277 ------------ drivers/mfd/twl6040-core.c | 588 ++++++++++++++++++++++++++ drivers/mfd/twl6040-irq.c | 205 +++++++++ include/linux/i2c/twl.h | 21 +- include/linux/mfd/twl4030-audio.h | 272 ++++++++++++ include/linux/mfd/twl4030-codec.h | 272 ------------ include/linux/mfd/twl6040.h | 260 ++++++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl4030.c | 22 +- sound/soc/codecs/twl6040.c | 426 ++++--------------- sound/soc/codecs/twl6040.h | 118 ----- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/sdp4430.c | 2 + sound/soc/omap/zoom2.c | 2 +- 44 files changed, 2716 insertions(+), 2011 deletions(-) create mode 100644 arch/arm/mach-omap2/pmic-common.c create mode 100644 arch/arm/mach-omap2/pmic-common.h create mode 100644 drivers/input/misc/twl6040-vibra.c create mode 100644 drivers/mfd/twl4030-audio.c delete mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 drivers/mfd/twl6040-core.c create mode 100644 drivers/mfd/twl6040-irq.c create mode 100644 include/linux/mfd/twl4030-audio.h delete mode 100644 include/linux/mfd/twl4030-codec.h create mode 100644 include/linux/mfd/twl6040.h -- 1.7.5.3

2 14

[alsa-devel] [PATCH 0/6] HDMI: Implement pcm-per-pin
by Stephen Warren 08 Jun '11

08 Jun '11

This patchset implements a PCM-per-pin-widget model for all HDMI codecs that use the generic codec support. For most existing codecs, there is 1 pin and 1 converter per codec, so this change should have no functional impact. For some codecs (e.g. GeForce 520, Intel Ibex Peak) there are fewer converters than pins, and the default routing is for all pins to be mux'd from a single codec, so this patch allows: a) Multiple PCMs to be used at once (1 per converter). Previously, only 1 PCM was created due to the default muxing. b) A direct mapping from PCM device to pin, and hence to ELD data and connected monitor. This should enable simpler/better decisions re: which audio formats are supported for that PCM, since only 1 ELD has to be considered (previously, I think the ELDs other than from the first pin were simply ignored) The very first change in the series is actually somewhat unrelated, but I noticed it while working on this, and I believe it's a correct change. Stephen Warren (6): ALSA: hda: Gate ELD usage only by whether ELD is valid ALSA: hda: Allow multple SPDIF controls per codec ALSA: hda: Virtualize SPDIF out controls ALSA: hda: Separate generic and non-generic implementations ALSA: hda: hdmi_eld_update_pcm_info: update a stream in place ALSA: hda: HDMI: Support codecs with fewer cvts than pins sound/pci/hda/hda_codec.c | 139 ++++++--- sound/pci/hda/hda_codec.h | 16 +- sound/pci/hda/hda_eld.c | 46 ++-- sound/pci/hda/hda_intel.c | 5 +- sound/pci/hda/hda_local.h | 8 +- sound/pci/hda/patch_analog.c | 4 +- sound/pci/hda/patch_ca0110.c | 3 +- sound/pci/hda/patch_cirrus.c | 3 +- sound/pci/hda/patch_cmedia.c | 4 +- sound/pci/hda/patch_conexant.c | 1 + sound/pci/hda/patch_hdmi.c | 704 ++++++++++++++++++++++++---------------- sound/pci/hda/patch_realtek.c | 1 + sound/pci/hda/patch_sigmatel.c | 4 +- sound/pci/hda/patch_via.c | 13 +- 14 files changed, 594 insertions(+), 357 deletions(-)

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Re: [alsa-devel] alsa-driver build problem : no lola dir
by Takashi Iwai 08 Jun '11

08 Jun '11

At Tue, 07 Jun 2011 11:26:50 +1200, Eliot Blennerhassett wrote: > > Trying to build alsa-driver git fails: > > make[2]: Leaving directory > `/home/eliot/src/alsagit/alsa-driver/pci/korg1212' > make: Entering an unknown directory > make: *** lola: No such file or directory. Stop. > make: Leaving an unknown directory > make[1]: *** [prepare] Error 1 > make[1]: Leaving directory `/home/eliot/src/alsagit/alsa-driver/pci' > make: *** [dep] Error 1 > ~/src/alsagit/alsa-driver$ > > my alsa-kernel links point to a current sound-2.6 git tree. Always use my alsa-driver-build.git if you are using sound-2.6 git or the latest kernel: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-driver-build.git The repos in alsa-project.org are still based on old versions. thanks, Takashi

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[alsa-devel] Two capture inputs from Cirrus CS4207
by Kattwinkel, Bryan (SA-1) 07 Jun '11

07 Jun '11

I have a requirement to concurrently read both analog stereo inputs from a Cirrus CS4207 codec on a custom board. I had hoped to find a way to do this in userspace with a configuration of plugins. But like most ALSA drivers, pci/hda/patch_cirrus.c exposes just one analog stereo input, and relies on the mixer and/or mic detect to choose which hardware pin and which of the two hardware ADC channels to connect to the single capture input. One possible solution would be to add an alternate capture device to the driver, and there seems to be an example of this approach in the patch_realtek driver. My questions are what approach should be used to solve this kind of problem, and is there a better example source to look at instead of or in addition to patch_realtek.c? Thanks, Bryan NOTICE: This e-mail transmission (and/or the attachments accompanying it) may contain confidential or proprietary information belonging to DRS Technologies or the sender. The information is only for the use of the intended recipient. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or the taking of any action in reliance on the contents of this information is strictly prohibited. Any unauthorized interception of this transmission is illegal under the law. If you have received this transmission in error, please promptly notify the sender by reply e-mail, and then destroy all copies of the transmission. "This (document/presentation) may contain technical data as defined in the International Traffic In Arms Regulations (ITAR) 22 CFR 120.10. Export of this material is restricted by the Arms Export Control Act (22 U.S.C. 2751 et seq.) and may not be exported to foreign persons without prior written approval from the U.S. Department of State."

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[alsa-devel] [PATCH] ASoC: core - Optimise and refactor pcm_new() to pass only rtd
by Liam Girdwood 07 Jun '11

07 Jun '11

Currently pcm_new() passes in 3 arguments :- card, pcm and DAI. Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains card, pcm and DAI along with other members too that are useful too. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 3 +-- sound/soc/atmel/atmel-pcm.c | 6 ++++-- sound/soc/au1x/dbdma2.c | 7 ++++--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 ++++-- sound/soc/blackfin/bf5xx-i2s-pcm.c | 6 ++++-- sound/soc/blackfin/bf5xx-tdm-pcm.c | 6 ++++-- sound/soc/davinci/davinci-pcm.c | 6 ++++-- sound/soc/ep93xx/ep93xx-pcm.c | 6 ++++-- sound/soc/fsl/fsl_dma.c | 6 ++++-- sound/soc/fsl/mpc5200_dma.c | 7 ++++--- sound/soc/imx/imx-pcm-fiq.c | 8 +++++--- sound/soc/imx/imx-ssi.c | 7 ++++--- sound/soc/imx/imx-ssi.h | 3 +-- sound/soc/jz4740/jz4740-pcm.c | 6 ++++-- sound/soc/kirkwood/kirkwood-dma.c | 6 ++++-- sound/soc/mid-x86/sst_platform.c | 5 +++-- sound/soc/nuc900/nuc900-pcm.c | 7 +++++-- sound/soc/omap/omap-pcm.c | 6 ++++-- sound/soc/pxa/pxa2xx-pcm.c | 6 ++++-- sound/soc/s6000/s6000-pcm.c | 7 ++++--- sound/soc/samsung/dma.c | 6 ++++-- sound/soc/sh/dma-sh7760.c | 6 +++--- sound/soc/sh/fsi.c | 6 +++--- sound/soc/sh/siu_pcm.c | 5 +++-- sound/soc/soc-core.c | 3 +-- sound/soc/tegra/tegra_pcm.c | 6 ++++-- sound/soc/txx9/txx9aclc.c | 5 +++-- 27 files changed, 96 insertions(+), 61 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index baaba6a..d5db87e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -638,8 +638,7 @@ struct snd_soc_platform_driver { int (*resume)(struct snd_soc_dai *dai); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, - struct snd_pcm *); + int (*pcm_new)(struct snd_soc_pcm_runtime *); void (*pcm_free)(struct snd_pcm *); /* diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 51dde4e..f81d4c3 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -364,9 +364,11 @@ static struct snd_pcm_ops atmel_pcm_ops = { \*--------------------------------------------------------------------------*/ static u64 atmel_pcm_dmamask = 0xffffffff; -static int atmel_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 10fdd28..20bb53a 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -319,10 +319,11 @@ static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int au1xpsc_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1); diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 98b44b3..9e59f68 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -418,9 +418,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; pr_debug("%s enter\n", __func__); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index b5101ef..96d0d90 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -248,9 +248,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; pr_debug("%s enter\n", __func__); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 07cfc7a..c95cc03 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -283,9 +283,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index c9e0320..a49e667 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -833,9 +833,11 @@ static void davinci_pcm_free(struct snd_pcm *pcm) static u64 davinci_pcm_dmamask = 0xffffffff; -static int davinci_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret; if (!card->dev->dma_mask) diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index a456e49..e27c417 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -266,9 +266,11 @@ static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 ep93xx_pcm_dmamask = 0xffffffff; -static int ep93xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 15dac0f..50b5df8 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -294,9 +294,11 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * Regardless of where the memory is actually allocated, since the device can * technically DMA to any 36-bit address, we do need to set the DMA mask to 36. */ -static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); int ret; diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index fff695c..19ad0c1 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -299,10 +299,11 @@ static struct snd_pcm_ops psc_dma_ops = { }; static u64 psc_dma_dmamask = 0xffffffff; -static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); size_t size = psc_dma_hardware.buffer_bytes_max; int rc = 0; diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 413b78d..309c59e 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -238,12 +238,14 @@ static struct snd_pcm_ops imx_pcm_ops = { static int ssi_irq = 0; -static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret; - ret = imx_pcm_new(card, dai, pcm); + ret = imx_pcm_new(rtd); if (ret) return ret; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 5b13fec..158a91c 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -388,10 +388,11 @@ static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); -int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) { - + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index dc8a875..0a84cec 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -225,8 +225,7 @@ struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, struct imx_ssi *ssi); int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); -int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm); +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); void imx_pcm_free(struct snd_pcm *pcm); /* diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c index fb1483f..a7c9578 100644 --- a/sound/soc/jz4740/jz4740-pcm.c +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -299,9 +299,11 @@ static void jz4740_pcm_free(struct snd_pcm *pcm) static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); -int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index e13c6ce..cd33de1 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -312,9 +312,11 @@ static int kirkwood_dma_preallocate_dma_buffer(struct snd_pcm *pcm, return 0; } -static int kirkwood_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret; if (!card->dev->dma_mask) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 5a946b4..3e78260 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -402,9 +402,10 @@ static void sst_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int retval = 0; pr_debug("sst_pcm_new called\n"); diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index 8263f56..d589ef1 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -315,9 +315,12 @@ static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm) } static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); -static int nuc900_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + if (!card->dev->dma_mask) card->dev->dma_mask = &nuc900_pcm_dmamask; if (!card->dev->coherent_dma_mask) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index e6a6b99..b2f5751 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -366,9 +366,11 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 2ce0b2d..d73d6f6 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -85,9 +85,11 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = { static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); -static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index ab3ccae..80c85fd 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -443,10 +443,11 @@ static void s6000_pcm_free(struct snd_pcm *pcm) static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); -static int s6000_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) { - struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct snd_card *card = runtime->card->snd_card; + struct snd_soc_dai *dai = runtime->cpu_dai; + struct snd_pcm *pcm = runtime->pcm; struct s6000_pcm_dma_params *params; int res; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 5cb3b88..9465588 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -425,9 +425,11 @@ static void dma_free_dma_buffers(struct snd_pcm *pcm) static u64 dma_mask = DMA_BIT_MASK(32); -static int dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index c326d29..db74005 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -327,10 +327,10 @@ static void camelot_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int camelot_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_pcm *pcm = rtd->pcm; + /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel * in MMAP mode (i.e. aplay -M) */ diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index d2f17ce..8e112cc 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1168,10 +1168,10 @@ static void fsi_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int fsi_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_pcm *pcm = rtd->pcm; + /* * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel * in MMAP mode (i.e. aplay -M) diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index a423bab..f8f6816 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -527,10 +527,11 @@ static snd_pcm_uframes_t siu_pcm_pointer_dma(struct snd_pcm_substream *ss) return bytes_to_frames(ss->runtime, ptr); } -static int siu_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int siu_pcm_new(struct snd_soc_pcm_runtime *rtd) { /* card->dev == socdev->dev, see snd_soc_new_pcms() */ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; struct siu_info *info = siu_i2s_data; struct platform_device *pdev = to_platform_device(card->dev); int ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ae4c5e4..df5608d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2161,8 +2161,7 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); if (platform->driver->pcm_new) { - ret = platform->driver->pcm_new(rtd->card->snd_card, - codec_dai, pcm); + ret = platform->driver->pcm_new(rtd); if (ret < 0) { pr_err("asoc: platform pcm constructor failed\n"); return ret; diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 3c271f9..ff86e5e 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -322,9 +322,11 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) static u64 tegra_dma_mask = DMA_BIT_MASK(32); -static int tegra_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index f4aa4e0..34aa972 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -288,9 +288,10 @@ static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; struct platform_device *pdev = to_platform_device(dai->platform->dev); struct txx9aclc_soc_device *dev; struct resource *r; -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: core - Allow components to probe/remove in sequence.
by Liam Girdwood 07 Jun '11

07 Jun '11

Some ASoC components depend on other ASoC components to provide clocks and power resources in order to probe() and vice versa for remove(). Allow components to be ordered so that components can be probed() and removed() in sequences that conform to their dependencies. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dai.h | 4 +++ include/sound/soc.h | 18 ++++++++++++++ sound/soc/soc-core.c | 60 ++++++++++++++++++++++++++++++---------------- 3 files changed, 61 insertions(+), 21 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1bafe95..5ad5f3a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -209,6 +209,10 @@ struct snd_soc_dai_driver { struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; unsigned int symmetric_rates:1; + + /* probe ordering - for components with runtime dependencies */ + int probe_order; + int remove_order; }; /* diff --git a/include/sound/soc.h b/include/sound/soc.h index f55efc1..baaba6a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -203,6 +203,16 @@ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) /* + * Component probe and remove ordering levels for components with runtime + * dependencies. + */ +#define SND_SOC_COMP_ORDER_FIRST -2 +#define SND_SOC_COMP_ORDER_EARLY -1 +#define SND_SOC_COMP_ORDER_NORMAL 0 +#define SND_SOC_COMP_ORDER_LATE 1 +#define SND_SOC_COMP_ORDER_LAST 2 + +/* * Bias levels * * @ON: Bias is fully on for audio playback and capture operations. @@ -613,6 +623,10 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + + /* probe ordering - for components with runtime dependencies */ + int probe_order; + int remove_order; }; /* SoC platform interface */ @@ -637,6 +651,10 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + + /* probe ordering - for components with runtime dependencies */ + int probe_order; + int remove_order; }; struct snd_soc_platform { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c53f5d5..ae4c5e4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1394,7 +1394,7 @@ static void soc_remove_codec(struct snd_soc_codec *codec) module_put(codec->dev->driver->owner); } -static void soc_remove_dai_link(struct snd_soc_card *card, int num) +static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec = rtd->codec; @@ -1411,7 +1411,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC DAI */ - if (codec_dai && codec_dai->probed) { + if (codec_dai && codec_dai->probed && + codec_dai->driver->remove_order == order) { if (codec_dai->driver->remove) { err = codec_dai->driver->remove(codec_dai); if (err < 0) @@ -1422,7 +1423,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the platform */ - if (platform && platform->probed) { + if (platform && platform->probed && + platform->driver->remove_order == order) { if (platform->driver->remove) { err = platform->driver->remove(platform); if (err < 0) @@ -1434,11 +1436,13 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC */ - if (codec && codec->probed) + if (codec && codec->probed && + codec->driver->remove_order == order) soc_remove_codec(codec); /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed) { + if (cpu_dai && cpu_dai->probed && + cpu_dai->driver->remove_order == order) { if (cpu_dai->driver->remove) { err = cpu_dai->driver->remove(cpu_dai); if (err < 0) @@ -1452,11 +1456,13 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) static void soc_remove_dai_links(struct snd_soc_card *card) { - int i; - - for (i = 0; i < card->num_rtd; i++) - soc_remove_dai_link(card, i); + int dai, order; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (dai = 0; dai < card->num_rtd; dai++) + soc_remove_dai_link(card, dai, order); + } card->num_rtd = 0; } @@ -1597,7 +1603,7 @@ static int soc_post_component_init(struct snd_soc_card *card, return 0; } -static int soc_probe_dai_link(struct snd_soc_card *card, int num) +static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; @@ -1606,7 +1612,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; int ret; - dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num); + dev_dbg(card->dev, "probe %s dai link %d late %d\n", + card->name, num, order); /* config components */ codec_dai->codec = codec; @@ -1618,7 +1625,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) rtd->pmdown_time = pmdown_time; /* probe the cpu_dai */ - if (!cpu_dai->probed) { + if (!cpu_dai->probed && + cpu_dai->driver->probe_order == order) { if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; @@ -1637,14 +1645,16 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* probe the CODEC */ - if (!codec->probed) { + if (!codec->probed && + codec->driver->probe_order == order) { ret = soc_probe_codec(card, codec); if (ret < 0) return ret; } /* probe the platform */ - if (!platform->probed) { + if (!platform->probed && + platform->driver->probe_order == order) { if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; @@ -1663,7 +1673,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* probe the CODEC DAI */ - if (!codec_dai->probed) { + if (!codec_dai->probed && codec_dai->driver->probe_order == order) { if (codec_dai->driver->probe) { ret = codec_dai->driver->probe(codec_dai); if (ret < 0) { @@ -1678,6 +1688,10 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) list_add(&codec_dai->card_list, &card->dai_dev_list); } + /* complete DAI probe during last probe */ + if (order != SND_SOC_COMP_ORDER_LAST) + return 0; + /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); @@ -1818,7 +1832,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; - int ret, i; + int ret, i, order; mutex_lock(&card->mutex); @@ -1896,12 +1910,16 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) goto card_probe_error; } - for (i = 0; i < card->num_links; i++) { - ret = soc_probe_dai_link(card, i); - if (ret < 0) { - pr_err("asoc: failed to instantiate card %s: %d\n", + /* early DAI link probe */ + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_dai_link(card, i, order); + if (ret < 0) { + pr_err("asoc: failed to instantiate card %s: %d\n", card->name, ret); - goto probe_dai_err; + goto probe_dai_err; + } } } -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: twl6040 - According to TWL6040 specification, gain start at 6dB and not -6dB.
by Liam Girdwood 07 Jun '11

07 Jun '11

Signed-off-by: Ricardo Neri <ricardo.neri(a)ti.com> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4c33663..cd63bba 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -954,9 +954,9 @@ static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); /* * MICGAIN volume control: - * from -6 to 30 dB in 6 dB steps + * from 6 to 30 dB in 6 dB steps */ -static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0); +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); /* * AFMGAIN volume control: -- 1.7.4.1

2 2

[alsa-devel] MIDI issue with RME HDSPM MADI
by Jörn Nettingsmeier 07 Jun '11

07 Jun '11

Hi *! The HDSPM MADI driver seems to have an issue with its MIDI implementation. It is rock-solid for audio under Jack at 64 frames/period, but as soon as I start using the MIDI over MADI interface (to remote-control my Micstasy preamps), I'm getting xruns every ten seconds or so. I've tried to debug the issue with the help of faberman, and he came up with a midi drain() implementation for the driver, but it didn't help with the xrun issue. It did help with busy MIDI programs in that they are no longer stuck in D state almost all the time, but sleeping, like any good program should. This is a 2.6.39-rc7 kernel, and faberman has confirmed the issue with .39. The attached test program rawmiditest.c (build with -lasound) will reproducibly cause xruns as soon as there is a mictasy on the MADI ring so that there's actual traffic. With no-one listening, there will be no xruns. (Might that be a hint that the problem is at the receiving end?) This is really totally over my head, but I found one place in the driver (snd_hdspm_midi_input_trigger) where a midi routine locks the entire hdspm structure. it's flushing the midi input with the hdspm spinlock held - could this be taking so long as to cause xruns? I'm waiting for a MIDI cable to arrive so that I can test if the same problem occurs when using the normal MIDI interfaces of the card (i.e. not the MIDI over MADI one)... For the record, faberman suspected that there is a bit too much locking going on wrt the hmidi struct. OTOH, when we removed all hmidi spinlocks, I was able to lock up the machine twice... maybe there is some middle ground? Meanwhile, any insights would be appreciated. Best, Jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister (VDT) http://stackingdwarves.net

2 5

[alsa-devel] [RFC PATCH] Add combined clock support to Atmel SoC audio
by Ryan Mallon 07 Jun '11

07 Jun '11

The following patch is one that has been floating around in various forms in our own internal trees for a while. The Atmel SSC peripheral has seperate TX and RX clocks which use separate pins from the the micro. TF (frame) and TK (clock) for transmit and RF and RK for receive. Not all codecs have separate frame and bit clocks for transmit and receive so we want to be able to do both playback and capture using a single set of pins. This patch introduces a combined clock mode for the Atmel SSC peripheral. Which allows playback and capture to use a single set of pins. Currently combined clock is only supported on the TF/TK pins (some incomplete support exists for using RF/RK). I have tested this patch on our AT91SAM9G45 + TLV320AIC26 platform. Playback and capture work individually. Simultaneous playback and capture have been tested by connecting a loopback cable on the linein and lineout jacks and then doing: arecord -c 2 -f S16_LE -r 44100 > recording.wav & aplay 500hz_sine.wav This patch is posted as RFC since the approach is incomplete and a bit hackish. I am mostly interested in knowing if this is a sensible approach, and could be cleaned up for mainline inclusion, or if there is a better way to do this. Signed-off-by: Ryan Mallon <ryan(a)bluewatersys.com> --- diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 5d230ce..ee00172 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -43,6 +43,7 @@ #include <sound/soc.h> #include <mach/hardware.h> +#include <asm/atomic.h> #include "atmel-pcm.h" #include "atmel_ssc_dai.h" @@ -246,7 +247,18 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; if (dma_params != NULL) { - ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + if (ssc_p->combined_clock) { + /* + * When using a combined clock we only disable the + * clock once all substreams have completed + */ + if (atomic_dec_and_test(&ssc_p->substreams_running)) + ssc_writel(ssc_p->ssc->regs, CR, + dma_params->mask->ssc_disable); + } else { + ssc_writel(ssc_p->ssc->regs, CR, + dma_params->mask->ssc_disable); + } pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", (dir ? "receive" : "transmit"), ssc_readl(ssc_p->ssc->regs, SR)); @@ -328,6 +340,21 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, return 0; } +void atmel_ssc_setup_combined_clock(struct atmel_ssc_info *ssc_p, + int combined_clock) +{ + int i; + + ssc_p->combined_clock = combined_clock; + for (i = 0; i < ARRAY_SIZE(ssc_dma_params); i++) { + if (ssc_p->combined_clock == ATMEL_SSC_CLOCK_TX_ON_RX) + ssc_dma_params[i][0].mask->ssc_enable |= SSC_BIT(CR_RXEN); + else if (ssc_p->combined_clock == ATMEL_SSC_CLOCK_RX_ON_TX) + ssc_dma_params[i][1].mask->ssc_enable |= SSC_BIT(CR_TXEN); + } +} +EXPORT_SYMBOL(atmel_ssc_setup_combined_clock); + /* * Configure the SSC. */ @@ -548,6 +575,34 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, ssc_p->daifmt); return -EINVAL; } + + if (ssc_p->combined_clock == ATMEL_SSC_CLOCK_RX_ON_TX) { + /* RX clock is always running */ + rcmr &= ~SSC_BF(RCMR_CKO, 0x7); + rcmr |= SSC_BF(RCMR_CKO, SSC_CKO_CONTINUOUS); + + /* TX clock is always running */ + tcmr &= ~SSC_BF(TCMR_CKO, 0x7); + tcmr |= SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS); + + /* RX clock is sourced from TK pin */ + rcmr &= ~SSC_BF(RCMR_CKS, 0x7); + rcmr |= SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + /* Start RX on TX start */ + rcmr &= ~SSC_BF(RCMR_START, 0xf); + rcmr |= SSC_BF(RCMR_START, SSC_START_TX_RX); + + if (dir == 1) { + /* + * Set the TX clock period to the RX clock period + * FIXME - Is this okay if we are already doing TX? + */ + tcmr &= 0x00ffffff; + tcmr |= rcmr & 0xff000000; + } + } + pr_debug("atmel_ssc_hw_params: " "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); @@ -581,6 +636,9 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, return ret; } + if (ssc_p->combined_clock) + atomic_set(&ssc_p->substreams_running, 0); + ssc_p->initialized = 1; } @@ -616,6 +674,21 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + if (ssc_p->combined_clock) { + atomic_inc(&ssc_p->substreams_running); + + if (ssc_p->combined_clock == ATMEL_SSC_CLOCK_RX_ON_TX && + dir == 1) { + /* + * Jump start the dma for capture by loading the + * transmit holding register with a dummy value. + */ + pr_debug("Jump starting SSC RX DMA\n"); + if (!(ssc_readl(ssc_p->ssc->regs, SR) & (1 << 16))) + ssc_writel(ssc_p->ssc->regs, THR, 0); + } + } + pr_debug("%s enabled SSC_SR=0x%08x\n", dir ? "receive" : "transmit", ssc_readl(ssc_p->ssc->regs, SR)); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index 5d4f0f9..6b20ee6 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -102,6 +102,8 @@ struct atmel_ssc_state { u32 ssc_imr; }; +#define ATMEL_SSC_CLOCK_RX_ON_TX 1 +#define ATMEL_SSC_CLOCK_TX_ON_RX 2 struct atmel_ssc_info { char *name; @@ -115,8 +117,14 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + + /* For combined clocks */ + int combined_clock; + atomic_t substreams_running; }; int atmel_ssc_set_audio(int ssc); +extern void atmel_ssc_setup_combined_clock(struct atmel_ssc_info *ssc_p, + int combined_clock); #endif /* _AT91_SSC_DAI_H */

3 6

[alsa-devel] [PATCH] RME HDSPM MADI lock fix
by Jörn Nettingsmeier 07 Jun '11

07 Jun '11

hi *! while we were trying to debug the midi-related xruns in the current driver, i came across what looks like an unrelated locking bug to me: static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi) ... spin_lock_irqsave (&hmidi->lock, flags); <--------------- n_pending = snd_hdspm_midi_input_available (hmidi->hdspm, hmidi->id); if (n_pending > 0) { if (hmidi->input) { if (n_pending > (int)sizeof (buf)) n_pending = sizeof (buf); for (i = 0; i < n_pending; ++i) buf[i] = snd_hdspm_midi_read_byte (hmidi->hdspm, hmidi->id); if (n_pending) snd_rawmidi_receive (hmidi->input, buf, n_pending); } else { /* flush the MIDI input FIFO */ while (n_pending--) snd_hdspm_midi_read_byte (hmidi->hdspm, hmidi->id); } } hmidi->pending = 0; hmidi->hdspm->control_register |= hmidi->ie;<------- !!! hdspm_write(hmidi->hdspm, HDSPM_controlRegister, hmidi->hdspm->control_register); spin_unlock_irqrestore (&hmidi->lock, flags);<--------- return snd_hdspm_midi_output_write (hmidi); } if i understand this whole business correctly, we should be holding the hdspm lock here, not the hmidi one. please review the attached patch and apply if you think it's ok. best, jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister (VDT) http://stackingdwarves.net

3 7

[alsa-devel] [PATCH] ASoC: Blackfin: bf5xx-ad1836: Fix codec device name
by Lars-Peter Clausen 07 Jun '11

07 Jun '11

Fix the codec_name field of the dai_link to match the actual device name of the codec. Otherwise the card won't be instantiated. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ad1836.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index ea4951c..f79d165 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.0", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, { @@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.1", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, }; -- 1.7.2.5

3 2

[alsa-devel] [PATCH] ASoC: AD1836: Fix build error
by Lars-Peter Clausen 07 Jun '11

07 Jun '11

Commit f97d0c6d5f94 ("ASoC: AD1836: Add input gain control for ADC2") contained a typo in the register name, causing a build error. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1836.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 1e9a801..e3a9493 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -127,7 +127,7 @@ static const struct snd_soc_dapm_route ad183x_dapm_routes[] = { static const DECLARE_TLV_DB_SCALE(ad1836_in_tlv, 0, 300, 0); static const struct snd_kcontrol_new ad1836_controls[] = { - SOC_DOUBLE_TLV("ADC2 Capture Volume", AD183X_ADC_CTRL1, 3, 0, 4, 0, + SOC_DOUBLE_TLV("ADC2 Capture Volume", AD1836_ADC_CTRL1, 3, 0, 4, 0, ad1836_in_tlv), }; -- 1.7.2.5

3 2

Re: [alsa-devel] [PATCH] wm8940: remove unecessary if statement
by Jonathan Cameron 07 Jun '11

07 Jun '11

On 06/06/11 01:47, Greg Dietsche wrote: > the code always returns ret regardless, so if(ret) check is unecessary. Good point, though please spell check your commit messages. unecessary -> unnecessary Also if you want to do this sort of cleanup, please also fix the equivalent in wm8940_resume and wm8940_add_widgets. Ack is for what is here, plus those if you do them. Just as an aside, there is no earthly point in cc'ing lkml for a simple cleanup like this. Just adds to already huge amount of noise! > > Signed-off-by: Greg Dietsche <Gregory.Dietsche(a)cuw.edu> Acked-by: Jonathan Cameron <jic23(a)cam.ac.uk> > --- > sound/soc/codecs/wm8940.c | 3 --- > 1 files changed, 0 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c > index 25580e3..1ada6d3 100644 > --- a/sound/soc/codecs/wm8940.c > +++ b/sound/soc/codecs/wm8940.c > @@ -730,9 +730,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) > if (ret) > return ret; > ret = wm8940_add_widgets(codec); > - if (ret) > - return ret; > - > return ret; > } >

4 8

[alsa-devel] [PATCH] hdspm.c: make driver name generic to avoid user confusion
by Jörn Nettingsmeier 07 Jun '11

07 Jun '11

hi adi, *! here's a trivial patch to change the driver name of hdspm.c to avoid irritating users who don't have the MADI version (just happened on the LAU list). iiuc, no external software should make any assumptions about this string, so there should be no side effects, but i have been wrong before :) please review and apply. best, jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister (VDT) http://stackingdwarves.net

1 0

[alsa-devel] [PATCH] ALSA: hda: Fix quirk for Dell Inspiron 910
by Daniel Chen 07 Jun '11

07 Jun '11

Hi, Attached is a patch against sound-2.6 git HEAD that resolves inaudible audio from internal speakers on the reporter's Dell Inspiron 910. Please apply. Cheers, -Dan

2 1

[alsa-devel] alsa-driver build problem : no lola dir
by Eliot Blennerhassett 07 Jun '11

07 Jun '11

(resend from address subscribed to alsa-devel) Trying to build alsa-driver git fails: make[2]: Leaving directory `/home/eliot/src/alsagit/alsa-driver/pci/korg1212' make: Entering an unknown directory make: *** lola: No such file or directory. Stop. make: Leaving an unknown directory make[1]: *** [prepare] Error 1 make[1]: Leaving directory `/home/eliot/src/alsagit/alsa-driver/pci' make: *** [dep] Error 1 ~/src/alsagit/alsa-driver$ my alsa-kernel links point to a current sound-2.6 git tree. -- Eliot Blennerhassett AudioScience Inc.

1 0

[alsa-devel] [PATCH 1/6] ASoC: AD1836: Fix setting the PCM format
by Lars-Peter Clausen 06 Jun '11

06 Jun '11

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This patch doesn't need to go into stable since there are currently no users of PCM formats other than S32, which is the default. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1836.c | 14 +++++++------- sound/soc/codecs/ad1836.h | 6 ++++++ 2 files changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index ab63d52..754c496 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - word_len = 3; + word_len = AD1836_WORD_LEN_16; break; case SNDRV_PCM_FORMAT_S20_3LE: - word_len = 1; + word_len = AD1836_WORD_LEN_20; break; case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S32_LE: - word_len = 0; + word_len = AD1836_WORD_LEN_24; break; } - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, - AD1836_DAC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK, + word_len << AD1836_DAC_WORD_LEN_OFFSET); - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, - AD1836_ADC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK, + word_len << AD1836_ADC_WORD_OFFSET); return 0; } diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 8455967..9d6a3f8 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -25,6 +25,7 @@ #define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) #define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) #define AD1836_DAC_WORD_LEN_MASK 0x18 +#define AD1836_DAC_WORD_LEN_OFFSET 3 #define AD1836_DAC_CTRL2 1 #define AD1836_DACL1_MUTE 0 @@ -51,6 +52,7 @@ #define AD1836_ADCL2_MUTE 2 #define AD1836_ADCR2_MUTE 3 #define AD1836_ADC_WORD_LEN_MASK 0x30 +#define AD1836_ADC_WORD_OFFSET 5 #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) @@ -60,4 +62,8 @@ #define AD1836_NUM_REGS 16 +#define AD1836_WORD_LEN_24 0x0 +#define AD1836_WORD_LEN_20 0x1 +#define AD1836_WORD_LEN_16 0x2 + #endif -- 1.7.2.5

3 8

[alsa-devel] [PATCH] ASoC: Allow WM8915 BCLK calculation outside hw_params()
by Mark Brown 06 Jun '11

06 Jun '11

Allow more dynamic management of the device clocking by allowing BCLK to be calculated when we set SYSCLK. This means that if the system is idle when hw_params() runs then we don't try to use the SYSCLK used in that case to set up the BCLK dividers, we can instead wait until a later point such as bias level configuration. This makes it easier to manage low power modes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8915.c | 82 +++++++++++++++++++++++++++++---------------- 1 files changed, 53 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index a1d8618..423baa9 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -75,6 +75,7 @@ struct wm8915_priv { struct wm8915_pdata pdata; int rx_rate[WM8915_AIFS]; + int bclk_rate[WM8915_AIFS]; /* Platform dependant ReTune mobile configuration */ int num_retune_mobile_texts; @@ -1562,6 +1563,50 @@ static int wm8915_reset(struct snd_soc_codec *codec) return snd_soc_write(codec, WM8915_SOFTWARE_RESET, 0x8915); } +static const int bclk_divs[] = { + 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 +}; + +static void wm8915_update_bclk(struct snd_soc_codec *codec) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int aif, best, cur_val, bclk_rate, bclk_reg, i; + + /* Don't bother if we're in a low frequency idle mode that + * can't support audio. + */ + if (wm8915->sysclk < 64000) + return; + + for (aif = 0; aif < WM8915_AIFS; aif++) { + switch (aif) { + case 0: + bclk_reg = WM8915_AIF1_BCLK; + break; + case 1: + bclk_reg = WM8915_AIF2_BCLK; + break; + } + + bclk_rate = wm8915->bclk_rate[aif]; + + /* Pick a divisor for BCLK as close as we can get to ideal */ + best = 0; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate; + if (cur_val < 0) /* BCLK table is sorted */ + break; + best = i; + } + bclk_rate = wm8915->sysclk / bclk_divs[best]; + dev_dbg(codec->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", + bclk_divs[best], bclk_rate); + + snd_soc_update_bits(codec, bclk_reg, + WM8915_AIF1_BCLK_DIV_MASK, best); + } +} + static int wm8915_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1714,10 +1759,6 @@ static int wm8915_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static const int bclk_divs[] = { - 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 -}; - static const int dsp_divs[] = { 48000, 32000, 16000, 8000 }; @@ -1728,17 +1769,11 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); - int bits, i, bclk_rate, best, cur_val; + int bits, i, bclk_rate; int aifdata = 0; - int bclk = 0; int lrclk = 0; int dsp = 0; - int aifdata_reg, bclk_reg, lrclk_reg, dsp_shift; - - if (!wm8915->sysclk) { - dev_err(codec->dev, "SYSCLK not configured\n"); - return -EINVAL; - } + int aifdata_reg, lrclk_reg, dsp_shift; switch (dai->id) { case 0: @@ -1750,7 +1785,6 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, aifdata_reg = WM8915_AIF1TX_DATA_CONFIGURATION_1; lrclk_reg = WM8915_AIF1_TX_LRCLK_1; } - bclk_reg = WM8915_AIF1_BCLK; dsp_shift = 0; break; case 1: @@ -1762,7 +1796,6 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, aifdata_reg = WM8915_AIF2TX_DATA_CONFIGURATION_1; lrclk_reg = WM8915_AIF2_TX_LRCLK_1; } - bclk_reg = WM8915_AIF2_BCLK; dsp_shift = WM8915_DSP2_DIV_SHIFT; break; default: @@ -1776,6 +1809,9 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, return bclk_rate; } + wm8915->bclk_rate[dai->id] = bclk_rate; + wm8915->rx_rate[dai->id] = params_rate(params); + /* Needs looking at for TDM */ bits = snd_pcm_format_width(params_format(params)); if (bits < 0) @@ -1793,18 +1829,7 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, } dsp |= i << dsp_shift; - /* Pick a divisor for BCLK as close as we can get to ideal */ - best = 0; - for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { - cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate; - if (cur_val < 0) /* BCLK table is sorted */ - break; - best = i; - } - bclk_rate = wm8915->sysclk / bclk_divs[best]; - dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", - bclk_divs[best], bclk_rate); - bclk |= best; + wm8915_update_bclk(codec); lrclk = bclk_rate / params_rate(params); dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", @@ -1814,14 +1839,11 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, WM8915_AIF1TX_WL_MASK | WM8915_AIF1TX_SLOT_LEN_MASK, aifdata); - snd_soc_update_bits(codec, bclk_reg, WM8915_AIF1_BCLK_DIV_MASK, bclk); snd_soc_update_bits(codec, lrclk_reg, WM8915_AIF1RX_RATE_MASK, lrclk); snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_2, WM8915_DSP1_DIV_SHIFT << dsp_shift, dsp); - wm8915->rx_rate[dai->id] = params_rate(params); - return 0; } @@ -1882,6 +1904,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, return -EINVAL; } + wm8915_update_bclk(codec); + snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_SRC_MASK | WM8915_SYSCLK_DIV_MASK, src << WM8915_SYSCLK_SRC_SHIFT | ratediv); -- 1.7.5.3

2 1

[alsa-devel] [PATCH 1/6] ASoC: Indentation fix for null loop operation
by Mark Brown 06 Jun '11

06 Jun '11

More with the legibility. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6be6546..988cdff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -209,7 +209,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val, item, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; + ; val = snd_soc_read(w->codec, e->reg); item = (val >> e->shift_l) & (bitmask - 1); -- 1.7.5.3

2 6

[alsa-devel] [PATCH 1/2] ASoC: Specify target bias state directly as a bias state
by Mark Brown 06 Jun '11

06 Jun '11

Rather than a simple flag to say if we want the DAPM context to be at full power specify the target bias state. This should have no current effect but is a bit more direct and so makes it easier to change our decisions about the which bias state to go into in future. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 8 +++--- sound/soc/soc-dapm.c | 60 +++++++++++++++++++++++++-------------------- 3 files changed, 38 insertions(+), 32 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c46e7d8..7415878 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,7 +510,7 @@ struct snd_soc_dapm_context { struct snd_soc_card *card; /* parent card */ /* used during DAPM updates */ - int dev_power; + enum snd_soc_bias_level target_bias_level; struct list_head list; #ifdef CONFIG_DEBUG_FS diff --git a/include/sound/soc.h b/include/sound/soc.h index f1de3e0..0f29700 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -214,10 +214,10 @@ * @OFF: Power Off. No restrictions on transition times. */ enum snd_soc_bias_level { - SND_SOC_BIAS_OFF, - SND_SOC_BIAS_STANDBY, - SND_SOC_BIAS_PREPARE, - SND_SOC_BIAS_ON, + SND_SOC_BIAS_OFF = 0, + SND_SOC_BIAS_STANDBY = 1, + SND_SOC_BIAS_PREPARE = 2, + SND_SOC_BIAS_ON = 3, }; struct snd_jack; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 776e6f4..0bbded4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1042,16 +1042,17 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) struct snd_soc_dapm_context *d = data; int ret; - if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) { + /* If we're off and we're not supposed to be go into STANDBY */ + if (d->bias_level == SND_SOC_BIAS_OFF && + d->target_bias_level != SND_SOC_BIAS_OFF) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, "Failed to turn on bias: %d\n", ret); } - /* If we're changing to all on or all off then prepare */ - if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) || - (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) { + /* Prepare for a STADDBY->ON or ON->STANDBY transition */ + if (d->bias_level != d->target_bias_level) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); if (ret != 0) dev_err(d->dev, @@ -1068,7 +1069,9 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) int ret; /* If we just powered the last thing off drop to standby bias */ - if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) { + if (d->bias_level == SND_SOC_BIAS_PREPARE && + (d->target_bias_level == SND_SOC_BIAS_STANDBY || + d->target_bias_level == SND_SOC_BIAS_OFF)) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, "Failed to apply standby bias: %d\n", @@ -1076,14 +1079,16 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } /* If we're in standby and can support bias off then do that */ - if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) { + if (d->bias_level == SND_SOC_BIAS_STANDBY && + d->target_bias_level == SND_SOC_BIAS_OFF) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF); if (ret != 0) dev_err(d->dev, "Failed to turn off bias: %d\n", ret); } /* If we just powered up then move to active bias */ - if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) { + if (d->bias_level == SND_SOC_BIAS_PREPARE && + d->target_bias_level == SND_SOC_BIAS_ON) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON); if (ret != 0) dev_err(d->dev, "Failed to apply active bias: %d\n", @@ -1108,13 +1113,19 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) LIST_HEAD(up_list); LIST_HEAD(down_list); LIST_HEAD(async_domain); + enum snd_soc_bias_level bias; int power; trace_snd_soc_dapm_start(card); - list_for_each_entry(d, &card->dapm_list, list) - if (d->n_widgets || d->codec == NULL) - d->dev_power = 0; + list_for_each_entry(d, &card->dapm_list, list) { + if (d->n_widgets || d->codec == NULL) { + if (d->idle_bias_off) + d->target_bias_level = SND_SOC_BIAS_OFF; + else + d->target_bias_level = SND_SOC_BIAS_STANDBY; + } + } /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. @@ -1137,7 +1148,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) else power = 1; if (power) - w->dapm->dev_power = 1; + w->dapm->target_bias_level = SND_SOC_BIAS_ON; if (w->power == power) continue; @@ -1161,24 +1172,19 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) switch (event) { case SND_SOC_DAPM_STREAM_START: case SND_SOC_DAPM_STREAM_RESUME: - dapm->dev_power = 1; + dapm->target_bias_level = SND_SOC_BIAS_ON; break; case SND_SOC_DAPM_STREAM_STOP: - dapm->dev_power = !!dapm->codec->active; + if (dapm->codec->active) + dapm->target_bias_level = SND_SOC_BIAS_ON; + else + dapm->target_bias_level = SND_SOC_BIAS_STANDBY; break; case SND_SOC_DAPM_STREAM_SUSPEND: - dapm->dev_power = 0; + dapm->target_bias_level = SND_SOC_BIAS_STANDBY; break; case SND_SOC_DAPM_STREAM_NOP: - switch (dapm->bias_level) { - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - dapm->dev_power = 0; - break; - default: - dapm->dev_power = 1; - break; - } + dapm->target_bias_level = dapm->bias_level; break; default: break; @@ -1186,12 +1192,12 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } /* Force all contexts in the card to the same bias state */ - power = 0; + bias = SND_SOC_BIAS_OFF; list_for_each_entry(d, &card->dapm_list, list) - if (d->dev_power) - power = 1; + if (d->target_bias_level > bias) + bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - d->dev_power = power; + d->target_bias_level = bias; /* Run all the bias changes in parallel */ -- 1.7.5.3

2 2

Re: [alsa-devel] [PATCH] snd-usb-audio with multiple audio functions
by Clemens Ladisch 06 Jun '11

06 Jun '11

Rolf Meeser wrote: > On 06/06/2011 01:44 PM, Clemens Ladisch wrote: > > Many audio devices with multiple vendor-specific interfaces do not have > > a control interface. This code would split them into multiple cards. > > Breaking existing setups would indeed be bad. > Are that cards with audio interfaces not following the Audio Class spec? Yes. > What would be the easiest way to identify them, so we could assign a new > card index only in case of a compliant card? Non-compliant devices don't have a control interface. In theory, the USB core probes a device's interfaces in order, so the control interface usually is probed first. However, I'm not sure if this is really true for all devices. It's probably safest to check explicitly if the device has any interface marked as audio control 1/2. Regards, Clemens

1 0

[alsa-devel] [PATCH] snd-usb-audio with multiple audio functions
by Rolf Meeser 06 Jun '11

06 Jun '11

Hello, This patch fixes the currently broken support for multiple audio functions (AF) in snd-usb-audio. The patch is against the stable 2.6.39.1 kernel. It addresses three issues: 1. Controls of the second and any further AF are currently not accessibly (controls of the first AF are duplicated!) 2. Multiple AF's are currently incorrectly combined into one sound card index. 3. Module usage count of snd-usb-audio does not drop to zero after disconnecting a multiple AF device. All three issues are fixed by assigning a new card index for each AF in a USB device. Rationale: - A USB audio function (AF) consists of a control interface and zero or more streaming interfaces. USB devices can have multiple AF, where the individual AF are completely independent of each other. In particular, it cannot be assumed that the controls of one AF can be manipulated through the control interface of another (the first) AF. - Older kernels (up to 2.6.32? at least before the major rewrite of USB audio code) supported multiple AF's. However, they combined the streams into one sound card: For instance hw:1,0 for the stream interface of the first AF, and hw:1,1 for the stream interface of the second AF. I believe this is incorrect, as different AF's must be treated as an independent entity. Each AF must get its own card index, for the example this would be hw:1,0 and hw:2,0. This is also the way THE OTHER OS implements it. - Newer kernels have broken multiple AF support. In a dual AF setup these kernels detect the second AF, but they duplicate the controls of the first AF, and ignore the controls of the second one! - The patch also fixes a problem with module usage: When you unplug a multiple AF device, the snd-usb-audio module usage count currently stays at 1, while you could expect it to go to zero. This is the first time I'm sending a patch to this list. Please be patient with me if I did something in a wrong way :-) Regards, Rolf

4 4

[alsa-devel] Fwd: Re: [PATCH] snd-usb-audio with multiple audio functions
by Rolf Meeser 06 Jun '11

06 Jun '11

Hi Clemens, On 06/06/2011 01:44 PM, Clemens Ladisch wrote: > Rolf Meeser wrote: >> + if (usb_chip[i]&& >> + (usb_chip[i]->dev == dev)&& >> + (usb_chip[i]->ctrl_intf)&& >> + (usb_chip[i]->ctrl_intf->desc.bInterfaceNumber == ifnum)) { > Many audio devices with multiple vendor-specific interfaces do not have > a control interface. This code would split them into multiple cards. > > > Regards, > Clemens > Breaking existing setups would indeed be bad! Can you see an easy way to identify such cards here? Then we could assign a new card index only in case of a class compliant card, where this is desirable. Regards, Rolf

1 0

[alsa-devel] [PATCH] ASoC: Check for NULL register bank in snd_soc_get_cache_val()
by Mark Brown 06 Jun '11

06 Jun '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d82ab42..9a88a27 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -463,6 +463,9 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx, static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, unsigned int word_size) { + if (!base) + return -1; + switch (word_size) { case 1: { const u8 *cache = base; -- 1.7.5.3

2 1

[alsa-devel] [PATCH] ASoC: Suppress restore of default register values for rbtree cache sync
by Mark Brown 06 Jun '11

06 Jun '11

Currently the rbtree code will write out the entire register map when doing a cache sync which is wasteful and will slow things down. Check to see if the value we're about to write is the default and don't bother restoring it if it is, either the value will have been retained or the device will have been reset and holds the value already. We should really store the defaults in the nodes but this resolves the immediate issue. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d8ce34c..d82ab42 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -607,7 +607,7 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) struct rb_node *node; struct snd_soc_rbtree_node *rbnode; unsigned int regtmp; - unsigned int val; + unsigned int val, def; int ret; int i; @@ -619,6 +619,11 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) WARN_ON(codec->writable_register && codec->writable_register(codec, regtmp)); val = snd_soc_rbtree_get_register(rbnode, i); + def = snd_soc_get_cache_val(codec->reg_def_copy, i, + rbnode->word_size); + if (val == def) + continue; + codec->cache_bypass = 1; ret = snd_soc_write(codec, regtmp, val); codec->cache_bypass = 0; -- 1.7.5.3

3 5

[alsa-devel] [PATCH] ASoC: Update omap/ams-delta for TTY API change
by Janusz Krzysztofik 06 Jun '11

06 Jun '11

Following the receive_buf() change to return the number of bytes received, the cx20442 CODEC driver was updated to report that we accepted all the data that was offered to us. Do the same with the corresponding Amstrad Delta sound card level wrapper function, which is actually registered as the tty line discipline callback instead of the the CODEC function, which it then calls. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- sound/soc/omap/ams-delta.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) --- git/sound/soc/omap/ams-delta.c.orig 2011-04-09 00:39:54.000000000 +0200 +++ git/sound/soc/omap/ams-delta.c 2011-05-12 18:50:33.000000000 +0200 @@ -331,7 +331,7 @@ static int cx81801_hangup(struct tty_str } /* Line discipline .recieve_buf() */ -static void cx81801_receive(struct tty_struct *tty, +static unsigned int cx81801_receive(struct tty_struct *tty, const unsigned char *cp, char *fp, int count) { struct snd_soc_codec *codec = tty->disc_data; @@ -347,7 +347,7 @@ static void cx81801_receive(struct tty_s /* Initialize timer used for config pulse generation */ setup_timer(&cx81801_timer, cx81801_timeout, 0); - v253_ops.receive_buf(tty, cp, fp, count); + ret = v253_ops.receive_buf(tty, cp, fp, count); /* Link hook switch to DAPM pins */ ret = snd_soc_jack_add_pins(&ams_delta_hook_switch, @@ -358,10 +358,10 @@ static void cx81801_receive(struct tty_s "Failed to link hook switch to DAPM pins, " "will continue with hook switch unlinked.\n"); - return; + return count; } - v253_ops.receive_buf(tty, cp, fp, count); + ret = v253_ops.receive_buf(tty, cp, fp, count); for (c = &cp[count - 1]; c >= cp; c--) { if (*c != '\r') @@ -381,6 +381,7 @@ static void cx81801_receive(struct tty_s AMS_DELTA_LATCH2_MODEM_CODEC); break; } + return count; } /* Line discipline .write_wakeup() */

4 10

[alsa-devel] UCM: Possible snd_use_case_get enhancements
by Stephen Warren 06 Jun '11

06 Jun '11

Liam, The recent patches I posted allow querying a modifier/device's supported/ conflicting device list from a modifier/device either within the currently active verb (if any) or an explicitly defined verb. I figured a similar enhancement to snd_use_case_get might be useful; i.e. to allow querying within a specific verb, and also to indicate that one shouldn't fall back to a verb, or overall values. I'd propose something like this: NAME[[=]/<modifier>|<device>[/[=]<verb>]] The "=" would prevent fallback to a "higher" object if present. So for example: Current verb, not from device or modifier, with fallback to verb and ValueDefaults: PlaybackVolume Current verb, from device/modifier with fallback to verb and ValueDefaults: PlaybackVolume/Play Music Current verb, from dev/mod without fallback to verb or ValueDefaults: PlaybackVolume/=Play Music Specify a verb and dev/mod, but allow fallback to verb or ValueDefaults: PlaybackVolume/Play Music/HiFi Specify a verb and dev/mod, but no fallback to verb or ValueDefaults: PlaybackVolume/=Play Music/=HiFi One can also query from just a verb, with/without fallback: PlaybackVolume//HiFi PlaybackVolume//=HiFi Or current verb with no fallback: PlaybackVolume//= Does this seem useful, and a sane query string syntax? An alternative would be to drop the added "=" stuff, and never auto- fallback. However, that'd be a regression relative to current functionality, force clients to implement fallback themselves, and make querying ValueDefaults hard. -- nvpublic

2 1

[alsa-devel] [PATCH 0/8] MFD/ASoC: TWL4030/TWL6040 changes
by Peter Ujfalusi 06 Jun '11

06 Jun '11

Hello, The series will do three major things, and they are in one series, because they pretty much depending on each other, so it is easier to handle them together. 1. Rename the twl4030-codec MFD driver to twl4030-audio Since the ASoC multicomponent introduction (2.6.37) the twl4030-codec MFD dirver had been using twl4030-audio to register (twl4030-codec is the ASoC codec driver). The documentation refers to this part of twl as audio block, so it is better to rename the driver as well to avoid confusion. 2. Introduction of MFD driver for twl6040. The TWL6040 audio IC has codec and vibra functionality. Convert the existing ASoC codec driver to use the new MFD driver's interface. 3. Vibrator driver for TWL6040 (Input/ForceFeedback driver) The driver itself, and support for the vibrators in SDP4430 board. The series has been tested on top of merged omap-next, asoc-next, and mfd-next. It applies on top of linux-next as well. Regards, Peter --- Misael Lopez Cruz (3): mfd: twl6040: Add initial support ASoC: twl6040: Convert into TWL6040 MFD child input: Add initial support for TWL6040 vibrator Peter Ujfalusi (5): MFD: twl4030-codec: Rename internals from codec to audio MFD: twl4030-codec -> twl4030-audio: Rename the driver MFD: twl4030-audio: Rename platform data MFD: twl6040: Change platform data for soc codec driver OMAP4: SDP4430: Add twl6040 vibrator platform support arch/arm/mach-omap2/board-3430sdp.c | 8 +- arch/arm/mach-omap2/board-4430sdp.c | 50 +++- arch/arm/mach-omap2/board-devkit8000.c | 8 +- arch/arm/mach-omap2/board-igep0020.c | 8 +- arch/arm/mach-omap2/board-omap3beagle.c | 8 +- arch/arm/mach-omap2/board-omap3evm.c | 8 +- arch/arm/mach-omap2/board-omap3pandora.c | 8 +- arch/arm/mach-omap2/board-omap3stalker.c | 8 +- arch/arm/mach-omap2/board-omap3touchbook.c | 8 +- arch/arm/mach-omap2/board-overo.c | 8 +- arch/arm/mach-omap2/board-rx51-peripherals.c | 6 +- arch/arm/mach-omap2/board-zoom-peripherals.c | 14 +- arch/arm/plat-omap/include/plat/irqs.h | 12 +- drivers/input/misc/Kconfig | 13 +- drivers/input/misc/Makefile | 1 + drivers/input/misc/twl4030-vibra.c | 12 +- drivers/input/misc/twl6040-vibra.c | 428 +++++++++++++++++++ drivers/mfd/Kconfig | 8 +- drivers/mfd/Makefile | 3 +- drivers/mfd/twl-core.c | 13 +- drivers/mfd/twl4030-audio.c | 278 ++++++++++++ drivers/mfd/twl4030-codec.c | 277 ------------ drivers/mfd/twl6040-core.c | 588 ++++++++++++++++++++++++++ drivers/mfd/twl6040-irq.c | 205 +++++++++ include/linux/i2c/twl.h | 21 +- include/linux/mfd/twl4030-audio.h | 272 ++++++++++++ include/linux/mfd/twl4030-codec.h | 272 ------------ include/linux/mfd/twl6040.h | 260 ++++++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl4030.c | 22 +- sound/soc/codecs/twl6040.c | 426 ++++--------------- sound/soc/codecs/twl6040.h | 118 ----- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/sdp4430.c | 2 + sound/soc/omap/zoom2.c | 2 +- 35 files changed, 2290 insertions(+), 1090 deletions(-) create mode 100644 drivers/input/misc/twl6040-vibra.c create mode 100644 drivers/mfd/twl4030-audio.c delete mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 drivers/mfd/twl6040-core.c create mode 100644 drivers/mfd/twl6040-irq.c create mode 100644 include/linux/mfd/twl4030-audio.h delete mode 100644 include/linux/mfd/twl4030-codec.h create mode 100644 include/linux/mfd/twl6040.h -- 1.7.5.3

4 20

[alsa-devel] [PATCH 1/4] ASoC: Suppress noop SYSCLK updates in WM8915
by Mark Brown 06 Jun '11

06 Jun '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8915.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 5a59ef7..c61bc32 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -55,6 +55,7 @@ struct wm8915_priv { int ldo1ena; int sysclk; + int sysclk_src; int fll_src; int fll_fref; @@ -1834,6 +1835,9 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, int src; int old; + if (freq == wm8915->sysclk && clk_id == wm8915->sysclk_src) + return 0; + /* Disable SYSCLK while we reconfigure */ old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1); snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, @@ -1885,6 +1889,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_ENA, old); + wm8915->sysclk_src = clk_id; + return 0; } -- 1.7.5.3

2 4

[alsa-devel] [PATCH 1/5] UCM: Deprecate index on Section{Device, Modifier}
by Stephen Warren 06 Jun '11

06 Jun '11

The previous supported "legacy" syntax was: SectionDevice."Speaker".0 { SectionModifier."Capture Voice".0 { This change supports new syntax: SectionDevice."Speaker" { SectionModifier."Capture Voice" { ... but also allows the old syntax, iff the index is exactly "0". If an index is present, but not exactly "0", parsing will appear to succeed, but produce an empty device or modifier. When naming devices and modifiers, even if the legacy format is used, any index is not included in the name; i.e. both sets of syntax above name the device just "Speaker". The SupportedDevice list syntax still also accepts either "x" or "x.0", but internally strips ".0" from the tail of any device name. Any other name including "." is disallowed. Finally, when comparing device or modifier names, a simple exact string compare is now used, since no index data is ever present in device or modifier names. The one functional change introduced here is that a SupportedDevice entry of just "x" will now only ever match a single device. It previously acted as a wildcard for any device named "x.foo". Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- src/ucm/main.c | 43 ++++----------- src/ucm/parser.c | 156 ++++++++++++++++++++++++++++++++++++------------------ 2 files changed, 114 insertions(+), 85 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index fb4aa91..5acb5a8 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -484,33 +484,18 @@ static int is_modifier_supported(snd_use_case_mgr_t *uc_mgr, struct dev_list *device; struct use_case_device *adev; struct list_head *pos, *pos1; - char *cpos; - int dlen, len; list_for_each(pos, &modifier->dev_list) { device = list_entry(pos, struct dev_list, list); - cpos = strchr(device->name, '.'); - if (cpos) { - if (find(&uc_mgr->active_devices, - struct use_case_device, active_list, - name, device->name)) - return 1; - } else { - dlen = strlen(device->name); - list_for_each(pos1, &uc_mgr->active_devices) { - adev = list_entry(pos1, struct use_case_device, - active_list); - cpos = strchr(adev->name, '.'); - if (cpos) - len = cpos - adev->name; - else - len = strlen(adev->name); - if (len != dlen) - continue; - if (memcmp(adev->name, device->name, len)) - continue; - return 1; - } + + list_for_each(pos1, &uc_mgr->active_devices) { + adev = list_entry(pos1, struct use_case_device, + active_list); + + if (strcmp(adev->name, device->name)) + continue; + + return 1; } } return 0; @@ -528,19 +513,11 @@ static struct use_case_modifier * struct use_case_modifier *modifier; struct use_case_verb *verb = uc_mgr->active_verb; struct list_head *pos; - char name[64], *cpos; list_for_each(pos, &verb->modifier_list) { modifier = list_entry(pos, struct use_case_modifier, list); - strncpy(name, modifier->name, sizeof(name)); - name[sizeof(name)-1] = '\0'; - cpos = strchr(name, '.'); - if (!cpos) - continue; - *cpos= '\0'; - - if (strcmp(name, modifier_name)) + if (strcmp(modifier->name, modifier_name)) continue; if (is_modifier_supported(uc_mgr, modifier)) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index f7de9bd..b0bebe0 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -59,7 +59,7 @@ int parse_string(snd_config_t *n, char **res) /* * Parse safe ID */ -int parse_is_name_safe(char *name) +int parse_is_name_safe(const char *name) { if (strchr(name, '.')) { uc_error("char '.' not allowed in '%s'", name); @@ -75,7 +75,9 @@ int parse_get_safe_id(snd_config_t *n, const char **id) err = snd_config_get_id(n, id); if (err < 0) return err; - return parse_is_name_safe((char *)(*id)); + if (!parse_is_name_safe((char *)(*id))) + return -EINVAL; + return 0; } /* @@ -151,7 +153,7 @@ static int parse_compound(snd_use_case_mgr_t *uc_mgr, snd_config_t *cfg, n = snd_config_iterator_entry(i); if (snd_config_get_type(cfg) != SND_CONFIG_TYPE_COMPOUND) { - uc_error("compound type expected for %s", id); + uc_error("compound type expected for %s, is %d", id, snd_config_get_type(cfg)); return -EINVAL; } @@ -163,6 +165,19 @@ static int parse_compound(snd_use_case_mgr_t *uc_mgr, snd_config_t *cfg, return 0; } +static int strip_legacy_dev_index(char *name) +{ + char *dot = strchr(name, '.'); + if (!dot) + return 0; + if (dot[1] != '0' || dot[2] != '\0') { + uc_error("device name %s contains a '.'," + " and is not legacy foo.0 format", name); + return -EINVAL; + } + *dot = '\0'; + return 0; +} /* * Parse transition @@ -199,6 +214,12 @@ static int parse_supported_device(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, free(sdev); return err; } + err = strip_legacy_dev_index(sdev->name); + if (err < 0) { + free(sdev->name); + free(sdev); + return err; + } list_add(&sdev->list, dlist); } return 0; @@ -389,12 +410,12 @@ static int parse_value(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, * Parse Modifier Use cases * * # Each modifier is described in new section. N modifiers are allowed - * SectionModifier."Capture Voice".0 { + * SectionModifier."Capture Voice" { * * Comment "Record voice call" * SupportedDevice [ - * "x" # all x device instances - * "y.0" # device y instance 0 only + * "x" + * "y" * ] * * EnableSequence [ @@ -426,14 +447,21 @@ static int parse_modifier(snd_use_case_mgr_t *uc_mgr, { struct use_case_verb *verb = data1; struct use_case_modifier *modifier; - char *name = data2; - const char *id; + const char *name; snd_config_iterator_t i, next; snd_config_t *n; int err; - if (!parse_is_name_safe(name)) - return -EINVAL; + if (data2) { + name = data2; + if (!parse_is_name_safe(name)) + return -EINVAL; + } + else { + if (parse_get_safe_id(cfg, &name) < 0) + return -EINVAL; + } + /* allocate modifier */ modifier = calloc(1, sizeof(*modifier)); if (modifier == NULL) @@ -444,15 +472,7 @@ static int parse_modifier(snd_use_case_mgr_t *uc_mgr, INIT_LIST_HEAD(&modifier->dev_list); INIT_LIST_HEAD(&modifier->value_list); list_add_tail(&modifier->list, &verb->modifier_list); - err = parse_get_safe_id(cfg, &id); - if (err < 0) - return err; - modifier->name = malloc(strlen(name) + strlen(id) + 2); - if (modifier->name == NULL) - return -ENOMEM; - strcpy(modifier->name, name); - strcat(modifier->name, "."); - strcat(modifier->name, id); + modifier->name = strdup(name); snd_config_for_each(i, next, cfg) { const char *id; @@ -530,7 +550,7 @@ static int parse_modifier(snd_use_case_mgr_t *uc_mgr, * Parse Device Use Cases * *# Each device is described in new section. N devices are allowed - *SectionDevice."Headphones".0 { + *SectionDevice."Headphones" { * Comment "Headphones connected to 3.5mm jack" * * EnableSequence [ @@ -551,21 +571,28 @@ static int parse_modifier(snd_use_case_mgr_t *uc_mgr, * } * } */ -static int parse_device_index(snd_use_case_mgr_t *uc_mgr, - snd_config_t *cfg, - void *data1, - void *data2) +static int parse_device(snd_use_case_mgr_t *uc_mgr, + snd_config_t *cfg, + void *data1, + void *data2) { struct use_case_verb *verb = data1; - char *name = data2; + const char *name; struct use_case_device *device; - const char *id; snd_config_iterator_t i, next; snd_config_t *n; int err; - - if (!parse_is_name_safe(name)) - return -EINVAL; + + if (data2) { + name = data2; + if (!parse_is_name_safe(name)) + return -EINVAL; + } + else { + if (parse_get_safe_id(cfg, &name) < 0) + return -EINVAL; + } + device = calloc(1, sizeof(*device)); if (device == NULL) return -ENOMEM; @@ -574,14 +601,7 @@ static int parse_device_index(snd_use_case_mgr_t *uc_mgr, INIT_LIST_HEAD(&device->transition_list); INIT_LIST_HEAD(&device->value_list); list_add_tail(&device->list, &verb->device_list); - if (parse_get_safe_id(cfg, &id) < 0) - return -EINVAL; - device->name = malloc(strlen(name) + strlen(id) + 2); - if (device->name == NULL) - return -ENOMEM; - strcpy(device->name, name); - strcat(device->name, "."); - strcat(device->name, id); + device->name = strdup(name); snd_config_for_each(i, next, cfg) { const char *id; @@ -643,34 +663,66 @@ static int parse_device_index(snd_use_case_mgr_t *uc_mgr, return 0; } -static int parse_device_name(snd_use_case_mgr_t *uc_mgr, - snd_config_t *cfg, - void *data1, - void *data2 ATTRIBUTE_UNUSED) +static int parse_compound_check_legacy(snd_use_case_mgr_t *uc_mgr, + snd_config_t *cfg, + int (*fcn)(snd_use_case_mgr_t *, snd_config_t *, void *, void *), + void *data1) { - const char *id; + const char *id, *idchild; + int child_ctr = 0, legacy_format = 1; + snd_config_iterator_t i, next; + snd_config_t *child; int err; err = snd_config_get_id(cfg, &id); if (err < 0) return err; - return parse_compound(uc_mgr, cfg, parse_device_index, - data1, (void *)id); + + snd_config_for_each(i, next, cfg) { + child_ctr++; + if (child_ctr > 1) { + break; + } + + child = snd_config_iterator_entry(i); + + if (snd_config_get_type(cfg) != SND_CONFIG_TYPE_COMPOUND) { + legacy_format = 0; + break; + } + + if (snd_config_get_id(child, &idchild) < 0) + return -EINVAL; + + if (strcmp(idchild, "0")) { + legacy_format = 0; + break; + } + } + if (child_ctr != 1) { + legacy_format = 0; + } + + if (legacy_format) + return parse_compound(uc_mgr, cfg, fcn, data1, (void *)id); + else + return fcn(uc_mgr, cfg, data1, NULL); } -static int parse_modifier_name(snd_use_case_mgr_t *uc_mgr, +static int parse_device_name(snd_use_case_mgr_t *uc_mgr, snd_config_t *cfg, void *data1, void *data2 ATTRIBUTE_UNUSED) { - const char *id; - int err; + return parse_compound_check_legacy(uc_mgr, cfg, parse_device, data1); +} - err = snd_config_get_id(cfg, &id); - if (err < 0) - return err; - return parse_compound(uc_mgr, cfg, parse_modifier, - data1, (void *)id); +static int parse_modifier_name(snd_use_case_mgr_t *uc_mgr, + snd_config_t *cfg, + void *data1, + void *data2 ATTRIBUTE_UNUSED) +{ + return parse_compound_check_legacy(uc_mgr, cfg, parse_modifier, data1); } /* -- 1.7.0.4

2 5

[alsa-devel] [alsa-lib] A suggestion on file cards.c
by alex dot baldacchino dot alsasub at gmail dot com 05 Jun '11

05 Jun '11

Hello, first of all, let me point out documentation for snd_card_get_index is incorrect: it is stated that zero is returned in case of success, instead, a value greater than or equal to zero is the real success result. In second place, let me suggest a different implementation. There's a number of functions (to get index, name, longname of a card; other ones could be added eventually, as the interface could evolve) resorting to a round of calls and pointer dereferencing to snd_ctl_hw_open() and snd_ctl_card_info()/snd_ctl_hw_card_info() to perform about the same operations as snd_card_load2() (eventually combined with snd_card_load1() ), potentially in a more dangerous fashion and possibly causing snd_card_load2() to be called anyway. Thus, I'm wondering if such calls could be avoided with just a few (not excessively invasive) changes to snd_card_load2() ( + snd_card_load1(), if the case, as I think), and I've made an attempt to implement those functions this way. Let's consider a few things. Function snd_ctl_hw_open() is called with the only purpose to allocate and populate an object of type snd_ctl_t, whose pointer is then passed to snd_ctl_card_info(); the latter one accesses that object fields to end up calling snd_ctl_hw_card_info(), whose purpose is to call the same ioctl as snd_card_load2(), in order to populate an object of type snd_ctl_card_info_t, therefore a call to a (not too much) modified version of snd_card_load2 could be enough for the same goals of successive calls to snd_ctl_hw_open() and snd_ctl_card_info(). This would rise following constraints: 1) Creating an object of type snd_ctl_card_info_t accessible by the caller after the function (snd_card_load2) returns. This is easy to accomplish: the caller uses a local variable of that type, thus let's just add an argument of type snd_ctl_card_info_t * and either use it in place of the internal (automatic variable) one used by snd_card_load2(), or choose whether to use the pointer argument (if not null) or an automatic variable (I've chosen this path). 2) Accessing a device special file corresponding to an integer card number. More reasoning needed here, since snd_ctl_hw_open() does it on its own, whereas snd_card_load2 requires a char* argument. Translating card number to device path can be done in three ways, essentially: - building a proper path name the same way as snd_ctl_hw_open() does it (that is, about the same way as snd_card_load1() does it - see below) in each function calling snd_card_load2(); feasible, yet not very good for maintenance; - doing so inside snd_card_load2() if passed char pointer is null; feasible, but it would still be duplicating (at least part of the) code from snd_card_load1(); - calling snd_card_load2() via snd_card_load1() (as it's done right now, but for one case - within snd_card_get_index - which would remain about the same), by just furnishing snd_card_load1() with a pointer-to-snd_ctl_card_info_t parameter to be passed straightforward to snd_card_load2() (that could be null when specific card infos other than card number aren't needed by the caller). After a few considerations, the latter one seems feasible, and have been my choice. Function snd_ctl_hw_open() builds the path name for a control* device the same way it's done by snd_card_load1() (apparently aload* devices aren't taken into account, but please, read further), then snd_open_device is called, as it's done by snd_card_load2() BUT with an important difference: the latter function opens devices read-only, whereas snd_ctl_hw_open() can (and, as actually called by snd_card_* stuff, it does) open devices read-write, which is both needless and potentially dangerous. In case of failure opening chosen device, snd_ctl_hw_open calls snd_card_load() and retries: this means that snd_card_load1() and snd_card_load2() (via snd_card_load()) can be called anyway, so that some operations are performed more than once needlessly (and there's no need to call snd_card_load() from within snd_card_get_index() anyway), but also that aload stuff, if compiled as supported, is somehow taken into account silently even when it would seem to be ignored (but perhaps shouldn't be?), as in snd_card_get_name/longname. This should make snd_card_load1() suitable to translate card number to device path and call snd_card_load2. 3) Printing out error messages. In some cases both snd_ctl_hw_open() and snd_ctl_hw_card_info() can produce error messages, whereas snd_card_load#() functions are mute in same scenarios. Whenever they're called directly (in current implementation), such a behaviour should remain as is, but when replacing snd_ctl_hw_* stuff same errors/messages should be printed out. This is possible by duplicating those (small) pieces of code from snd_ctl_hw_* functions and executing them conditionally, availing of a further integer argument (used as a boolean flag). To avoid complicating calls to snd_card_load1/2(), a few macros can be defined to both mask out the "printing flag" and give them an alias name appropriate for the context they're being called from (which could add expressiveness to the implementation). Anyway, take this as a simple suggestion, just give it a look. =================================================== src/control/cards.c - attached ===================================================

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[alsa-devel] [PATCH] speaker-test.1: Clarify man page by giving examples
by Daniel Chen 05 Jun '11

05 Jun '11

Hi, Here's a forwarded patch from a Debian bug report for the man page of the speaker-test utility. It gives examples for using speaker-test alongside aplay. Please consider applying. Cheers, -Dan

2 1

[alsa-devel] AudioArduino - an ALSA soundcard driver for FTDI-based Arduinos
by sdaau 04 Jun '11

04 Jun '11

Hi all, (cross-posting to alsa-devel(a)alsa-project.org and gmane.linux.usb.general/linux-usb(a)vger.kernel.org; apologies if it's inappropriate) I'd just like to inform that the AudioArduino project has been posted online: http://imi.aau.dk/~sd/phd/index.php?title=AudioArduino It contains links to open-source Arduino microcontroller code, and corresponding ALSA driver source code - which can demonstrate mono, full-duplex, 8-bit / 44.1 kHz soundcard operation on an FTDI-based Arduino. There is also a related post on the (new) Arduino Forum: http://arduino.cc/forum/index.php/topic,62582.0.html The driver essentially represents an ALSA extension to ftdi-sio (which is why I thought it would be relevant to both alsa-devel, and linux-usb). Many thanks to the community for critical help, like in: ftdi_sio, and usb_serial->interface->dev->driver_data being null http://comments.gmane.org/gmane.linux.usb.general/30455 Help with dummy.c (where/how to write?) http://comments.gmane.org/gmane.linux.alsa.devel/74487 Looking forward to any feedback regarding this project, Cheers!

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[alsa-devel] Need general idea how to architect driver for multi-codec board
by Caleb Crome 04 Jun '11

04 Jun '11

Hello All, I'm building a board with multiple TLV320AIC33 codecs on board, and I don't know how I should architect the interface. The board is set up to use TDM (16 channels on 1 TDM PORT), it has 16 input and 16 output channels, all bonded together -- the hardware does not allow different codecs to run at different speeds. The board also includes several other widgets, like an I2C mux (to address all 8 stereo codecs), a couple of power amplifiers, and a limited number of ways the audio can be routed in or out. My question is this: I'm not sure how to architect the driver: 1) Simply instantiate multiple TLV320IAC33 codec drivers, as the AIC33 driver is already written. This seems like it won't work because I'd end up with multiple user-space 'cards', yet I want the card to appear as a single 16-channel sound card. Also, when you start one PCM interface for one codec, they would all start automatically, which I imagine the linux system wouldn't get. Also, the PCM interface needs to be configured differently for each codec (the TDM offset must be 0, 16, 32, 48, etc. bits). 2) rewrite the AIC33 driver to include all 8 codecs, including all the mixer controls for each codec. This is going to get REALLY big and ugly in the user space mixer control -- there will be a zillion controls, most of which are not really useful for this board. 3) rewrite the AIC33 driver to include all 8 codecs, but pare down the mixer controls to just the ones that I care about. If the card changes later, add in the changes to the driver. I don't have any huge desire to get the driver into the mainstream kernel (though I would not be upset if it was popular enough to warrant that), it's just not a priority. Any thoughts would be most welcome. Thanks, -Caleb

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[alsa-devel] unable to compile manually Alsa 1.0.24 on Kubuntu 11.04
by Thomas CROUX 04 Jun '11

04 Jun '11

Hello , I need to compile manually Alsa Driver because I made a mistake with my previous audio driver. My sound card is integrated into the mother board Asus P5K-SE. Following the product description coming from Asus, the chipset is : Realtek ALC 883 8 -Channel High Definition Audio CODEC Coaxial / Optical S/PDIF out ports at back I/O ASUS Noise Filter Support Jack-Sensing, Enumeration, Multi-streaming My Linux installation is Kubuntu 11.04, with kernel 2.6.38-8-generic. lspci gives the following : --------------------------- 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 02) . hwinfo returns : ---------------- > hal.1: read hal dataprocess 2298: arguments to dbus_move_error() were incorrect, assertion "(dest) == NULL || !dbus_error_is_set ((dest))" failed in file dbus-errors.c line 280. This is normally a bug in some application using the D-Bus library. libhal.c 3483 : Error unsubscribing to signals, error=The name org.freedesktop.Hal was not provided by any .service files 14: PCI 1b.0: 0403 Audio device [Created at pci.318] Unique ID: u1Nb.c9J1KXzTGCE SysFS ID: /devices/pci0000:00/0000:00:1b.0 SysFS BusID: 0000:00:1b.0 Hardware Class: sound Model: "Intel 82801I (ICH9 Family) HD Audio Controller" Vendor: pci 0x8086 "Intel Corporation" Device: pci 0x293e "82801I (ICH9 Family) HD Audio Controller" SubVendor: pci 0x1043 "ASUSTeK Computer Inc." SubDevice: pci 0x829f Revision: 0x02 Memory Range: 0xf9ff8000-0xf9ffbfff (rw,non-prefetchable) IRQ: 15 (no events) Module Alias: "pci:v00008086d0000293Esv00001043sd0000829Fbc04sc03i00" Driver Info #0: Driver Status: snd_hda_intel is not active Driver Activation Cmd: "modprobe snd_hda_intel" Config Status: cfg=new, avail=yes, need=no, active=unknown So now, the following lines are the one used to compile manually latest Alsa Drivers : -------------------------------------------------------------------------------------- mkdir src cd src mkdir alsa cd alsa sudo apt-get install build-essential linux-headers-$(uname -r) wget ftp://ftp.alsa-project.org/pub/driver/a … 24.tar.bz2 tar xvjf alsa-driver-1.0.24.tar.bz2 cd alsa-driver-1.0.24 sudo ./configure --with-kernel=/usr/src/linux-headers-$(uname -r) --with-cards="hda-intel" --with-oss=yes sudo make sudo make install And just before ending the sudo make command, I have the following error : -------------------------------------------------------------------------- make -C /usr/src/linux-headers-2.6.38-8-generic SUBDIRS=/home/thomas/src/alsa/alsa-driver-1.0.24 CPP="gcc -E" CC="gcc" modules make[1]: entrant dans le répertoire « /usr/src/linux-headers-2.6.38-8-generic » Makefile:556: /usr/src/linux-headers-2.6.38-8-generic/arch//Makefile: Aucun fichier ou dossier de ce type make[1]: *** Pas de règle pour fabriquer la cible « /usr/src/linux-headers-2.6.38-8-generic/arch//Makefile ». Arrêt. make[1]: quittant le répertoire « /usr/src/linux-headers-2.6.38-8-generic » make: *** [compile] Erreur 2 And I get the same error when using sudo ./alsa-compile.sh --driver --kmodules (with the script downloaded from Alsa servers). ------------------------------------------------------------------------------------------------------------------------------ make -C /lib/modules/2.6.38-8-generic/build SUBDIRS=/tmp/alsa-compile-script/alsa-driver-1.0.24.13.g953fe.286.ge539e CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/linux-headers-2.6.38-8-generic' Makefile:556: /usr/src/linux-headers-2.6.38-8-generic/arch//Makefile: No such file or directory make[1]: *** No rule to make target `/usr/src/linux-headers-2.6.38-8-generic/arch//Makefile'. Stop. make[1]: Leaving directory `/usr/src/linux-headers-2.6.38-8-generic' make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.24.13.g953fe.286.ge539e. Compilation of alsa-driver failed. Report this problem to the mailing list. So, at this time, I'm really down in the mouth. I'm totally unable to correct the problem I created by installing drivers from realtek web site. Could you help me please in trying to compile Alsa ? Thank you very much, Cheers, Thomas.

1 0

Re: [alsa-devel] [PATCH] cm6206: turn off de-emphasis in s/pdif (fwd)
by Eric Lammerts 03 Jun '11

03 Jun '11

Hi, I submitted this a while ago but nothing happened. Could you apply this patch? I rediffed against the current git. thanks, Eric ---------- Forwarded message ---------- Date: Sun, 21 Nov 2010 20:23:11 -0500 (EST) From: Eric Lammerts <alsa-devel(a)lists.lammerts.org> To: Clemens Ladisch <clemens(a)ladisch.de> Cc: Dan Allongo <gongo2k1(a)gmail.com>, alsa-devel(a)alsa-project.org Subject: Re: [alsa-devel] [PATCH] cm6206: turn off de-emphasis in s/pdif On Thu, 18 Nov 2010, Clemens Ladisch wrote: > Thanks. Please provide a Signed-off-by tag as described in > Documentation/SubmittingPatches. CM6206: Turn off de-emphasis channel status bit in S/PDIF output. Signed-off-by: Eric Lammerts <eric(a)lammerts.org> diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 355759b..e006d3c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -387,7 +387,7 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) { int err, reg; - int val[] = {0x200c, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; + int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; for (reg = 0; reg < ARRAY_SIZE(val); reg++) { err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]);

2 1

[alsa-devel] [PATCH 1/1] Added patch file and Makefile entry for CA0132 HDA codec
by Ian Minett 03 Jun '11

03 Jun '11

Attached is a patch that adds support for the CA0132 HDA codec, creating patch_ca0132.c in pci/hda. It also adds an entry to the pci/hda makefile that builds the codec when SND_HDA_CODEC_CA0132 is defined. However, we are not sure where to add the KConfig entry for SND_HDA_CODEC_CA0132, which is below, or if we should even create such an entry for this submission. config SND_HDA_CODEC_CA0132 bool "Build Creative CA0132 codec support" depends on SND_HDA_INTEL default y help Say Y here to include Creative CA0132 codec support in snd-hda-intel driver. When the HD-audio driver is built as a module, the codec support code is also built as another module, snd-hda-codec-ca0132. This module is automatically loaded at probing. Please let us know if the Makefile entry needs to be changed or if we should include the KConfig entry somewhere particular. Thank you, Ian Minett --- Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> diff --git a/pci/hda/Makefile b/pci/hda/Makefile index e7baf44..a30437f 100644 --- a/pci/hda/Makefile +++ b/pci/hda/Makefile @@ -5,6 +5,7 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf + clean-files := hda_intel.c hda_beep.c export-objs := hda_codec.o @@ -13,6 +14,13 @@ EXTRA_CFLAGS += -I$(SND_TOPDIR)/alsa-kernel/pci/hda include $(SND_TOPDIR)/alsa-kernel/pci/hda/Makefile +snd-hda-codec-ca0132-objs := patch_ca0132.o + +ifdef CONFIG_SND_HDA_CODEC_CA0132 +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o +endif + + ifdef CONFIG_SND_HDA_CODEC_HDMI snd-hda-codec-atihdmi-objs := dummy.o snd-hda-codec-intelhdmi-objs := dummy.o diff --git a/pci/hda/patch_ca0132.c b/pci/hda/patch_ca0132.c new file mode 100644 index 0000000..54aa921 --- /dev/null +++ b/pci/hda/patch_ca0132.c @@ -0,0 +1,1160 @@ +/* + * HD audio interface patch for Creative CA0132 chip + * + * Copyright (c) 2011, Creative Technology Ltd. + * + * Based on patch_ca0110.c + * Copyright (c) 2008 Takashi Iwai <tiwai(a)suse.de> + * + * This driver 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. + * + * This driver 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include "adriver.h" +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + +#define WIDGET_CHIP_CTRL 0x15 +#define WIDGET_DSP_CTRL 0x16 + +#define WUH_MEM_CONNID 10 +#define DSP_MEM_CONNID 16 + +enum hda_cmd_vendor_io { + /* for DspIO node */ + HDA_SHORT_CMD_VENDOR_DSP_IO_SCP_WRITE_DATA_LOW = 0x000, + HDA_SHORT_CMD_VENDOR_DSP_IO_SCP_WRITE_DATA_HIGH = 0x100, + + HDA_LONG_CMD_VENDOR_DSP_IO_STATUS = 0xF01, + HDA_LONG_CMD_VENDOR_DSP_IO_SCP_POST_READ_DATA = 0x702, + HDA_LONG_CMD_VENDOR_DSP_IO_SCP_READ_DATA = 0xF02, + HDA_LONG_CMD_VENDOR_DSP_IO_DSP_INIT = 0x703, + HDA_LONG_CMD_VENDOR_DSP_IO_SCP_POST_COUNT_QUERY = 0x704, + HDA_LONG_CMD_VENDOR_DSP_IO_SCP_READ_COUNT = 0xF04, + + /* for ChipIO node */ + HDA_SHORT_CMD_VENDOR_CHIP_IO_ADDRESS_LOW = 0x000, + HDA_SHORT_CMD_VENDOR_CHIP_IO_ADDRESS_HIGH = 0x100, + HDA_SHORT_CMD_VENDOR_CHIP_IO_STREAM_FORMAT = 0x200, + HDA_SHORT_CMD_VENDOR_CHIP_IO_DATA_LOW = 0x300, + HDA_SHORT_CMD_VENDOR_CHIP_IO_DATA_HIGH = 0x400, + + HDA_LONG_CMD_VENDOR_CHIP_IO_GET_PARAMETER = 0xF00, + HDA_LONG_CMD_VENDOR_CHIP_IO_STATUS = 0xF01, + HDA_LONG_CMD_VENDOR_CHIP_IO_HIC_POST_READ = 0x702, + HDA_LONG_CMD_VENDOR_CHIP_IO_HIC_READ_DATA = 0xF03, + + HDA_LONG_CMD_VENDOR_CHIP_IO_CT_EXTENSIONS_ENABLE = 0x70A, + + HDA_LONG_CMD_VENDOR_CHIP_IO_PLL_PMU_WRITE = 0x70C, + HDA_LONG_CMD_VENDOR_CHIP_IO_PLL_PMU_READ = 0xF0C, + HDA_LONG_CMD_VENDOR_CHIP_IO_8051_ADDRESS_LOW = 0x70D, + HDA_LONG_CMD_VENDOR_CHIP_IO_8051_ADDRESS_HIGH = 0x70E, + HDA_LONG_CMD_VENDOR_CHIP_IO_FLAG_SET = 0x70F, + HDA_LONG_CMD_VENDOR_CHIP_IO_FLAGS_GET = 0xF0F, + HDA_LONG_CMD_VENDOR_CHIP_IO_PARAMETER_SET = 0x710, + HDA_LONG_CMD_VENDOR_CHIP_IO_PARAMETER_GET = 0xF10, + + HDA_LONG_CMD_VENDOR_CHIP_IO_PORT_ALLOC_CONFIG_SET = 0x711, + HDA_LONG_CMD_VENDOR_CHIP_IO_PORT_ALLOC_SET = 0x712, + HDA_LONG_CMD_VENDOR_CHIP_IO_PORT_ALLOC_GET = 0xF12, + HDA_LONG_CMD_VENDOR_CHIP_IO_PORT_FREE_SET = 0x713, + + HDA_LONG_CMD_VENDOR_CHIP_IO_PARAMETER_EX_ID_GET = 0xF17, + HDA_LONG_CMD_VENDOR_CHIP_IO_PARAMETER_EX_ID_SET = 0x717, + HDA_LONG_CMD_VENDOR_CHIP_IO_PARAMETER_EX_VALUE_GET = 0xF18, + HDA_LONG_CMD_VENDOR_CHIP_IO_PARAMETER_EX_VALUE_SET = 0x718 +}; + +/* + * Control flag IDs + */ +enum control_flag_id { + /* Connection manager stream setup is bypassed/enabled */ + CONTROL_FLAG_C_MGR = 0, + /* DSP DMA is bypassed/enabled */ + CONTROL_FLAG_DMA = 1, + /* 8051 'idle' mode is disabled/enabled */ + CONTROL_FLAG_IDLE_ENABLE = 2, + /* Tracker for the SPDIF-in path is bypassed/enabled */ + CONTROL_FLAG_TRACKER = 3, + /* DigitalOut to Spdif2Out connection is disabled/enabled */ + CONTROL_FLAG_SPDIF2OUT = 4, + /* Digital Microphone is disabled/enabled */ + CONTROL_FLAG_DMIC = 5, + /* ADC_B rate is 48 kHz/96 kHz */ + CONTROL_FLAG_ADC_B_96KHZ = 6, + /* ADC_C rate is 48 kHz/96 kHz */ + CONTROL_FLAG_ADC_C_96KHZ = 7, + /* DAC rate is 48 kHz/96 kHz (affects all DACs) */ + CONTROL_FLAG_DAC_96KHZ = 8, + /* DSP rate is 48 kHz/96 kHz */ + CONTROL_FLAG_DSP_96KHZ = 9, + /* SRC clock is 98 MHz/196 MHz (196 MHz forces rate to 96 KHz) */ + CONTROL_FLAG_SRC_CLOCK_196MHZ = 10, + /* SRC rate is 48 kHz/96 kHz (48 kHz disabled when clock is 196 MHz) */ + CONTROL_FLAG_SRC_RATE_96KHZ = 11, + /* Decode Loop (DSP->SRC->DSP) is disabled/enabled */ + CONTROL_FLAG_DECODE_LOOP = 12, + /* De-emphasis filter on DAC-1 disabled/enabled */ + CONTROL_FLAG_DAC1_DEEMPHASIS = 13, + /* De-emphasis filter on DAC-2 disabled/enabled */ + CONTROL_FLAG_DAC2_DEEMPHASIS = 14, + /* De-emphasis filter on DAC-3 disabled/enabled */ + CONTROL_FLAG_DAC3_DEEMPHASIS = 15, + /* High-pass filter on ADC_B disabled/enabled */ + CONTROL_FLAG_ADC_B_HIGH_PASS = 16, + /* High-pass filter on ADC_C disabled/enabled */ + CONTROL_FLAG_ADC_C_HIGH_PASS = 17, + /* Common mode on Port_A disabled/enabled */ + CONTROL_FLAG_PORT_A_COMMON_MODE = 18, + /* Common mode on Port_D disabled/enabled */ + CONTROL_FLAG_PORT_D_COMMON_MODE = 19, + /* Impedance for ramp generator on Port_A 16 Ohm/10K Ohm */ + CONTROL_FLAG_PORT_A_10KOHM_LOAD = 20, + /* Impedance for ramp generator on Port_D, 16 Ohm/10K Ohm */ + CONTROL_FLAG_PORT_D_10K0HM_LOAD = 21, + /* ASI rate is 48kHz/96kHz */ + CONTROL_FLAG_ASI_96KHZ = 22, + /* DAC power settings able to control attached ports no/yes */ + CONTROL_FLAG_DACS_CONTROL_PORTS = 23, + /* Clock Stop OK reporting is disabled/enabled */ + CONTROL_FLAG_CONTROL_STOP_OK_ENABLE = 24, + /* Number of control flags */ + CONTROL_FLAGS_MAX = (CONTROL_FLAG_CONTROL_STOP_OK_ENABLE+1) +}; + +/* + * Control parameter IDs + */ +enum control_parameter_id { + /* 0: force HDA, 1: allow DSP if HDA Spdif1Out stream is idle */ + CONTROL_PARAMETER_SPDIF1_SOURCE = 2, + + /* Stream Control */ + + /* Select stream with the given ID */ + CONTROL_PARAMETER_STREAM_ID = 24, + /* Source connection point for the selected stream */ + CONTROL_PARAMETER_STREAM_SOURCE_CONN_POINT = 25, + /* Destination connection point for the selected stream */ + CONTROL_PARAMETER_STREAM_DEST_CONN_POINT = 26, + /* Number of audio channels in the selected stream */ + CONTROL_PARAMETER_STREAMS_CHANNELS = 27, + /*Enable control for the selected stream */ + CONTROL_PARAMETER_STREAM_CONTROL = 28, + + /* Connection Point Control */ + + /* Select connection point with the given ID */ + CONTROL_PARAMETER_CONN_POINT_ID = 29, + /* Connection point sample rate */ + CONTROL_PARAMETER_CONN_POINT_SAMPLE_RATE = 30, + + /* Node Control */ + + /* Select HDA node with the given ID */ + CONTROL_PARAMETER_NODE_ID = 31 +}; + +/* + * Queue IDs + */ +enum hda_vendor_dsp_io_scp_queue_id { + HDA_VENDOR_DSP_IO_SCP_QUEUE_8051_COMMAND = 0, + HDA_VENDOR_DSP_IO_SCP_QUEUE_HOST_COMMAND = 1, + HDA_VENDOR_DSP_IO_SCP_QUEUE_HOST_RESPONSE = 2 +}; + +/* + * Dsp Io Status codes + */ +enum hda_status_vendor_dsp_io { + /* Success */ + HDA_STATUS_VENDOR_DSP_IO_OK = 0x00, + /* Busy, unable to accept new command, the host must retry */ + HDA_STATUS_VENDOR_DSP_IO_BUSY = 0x01, + /* SCP command queue is full */ + HDA_STATUS_VENDOR_DSP_IO_SCP_COMMAND_QUEUE_FULL = 0x02, + /* SCP response queue is empty */ + HDA_STATUS_VENDOR_DSP_IO_SCP_RESPONSE_QUEUE_EMPTY = 0x03 +}; + +/* + * Chip Io Status codes + */ +enum hda_status_vendor_chip_io { + /* Success */ + HDA_STATUS_VENDOR_CHIP_IO_OK = 0x00, + /* Busy, unable to accept new command, the host must retry */ + HDA_STATUS_VENDOR_CHIP_IO_BUSY = 0x01 +}; + +/* + * CA0132 sample rate + */ +enum ca0132_sample_rate { + SR_6_000 = 0x00, + SR_8_000 = 0x01, + SR_9_600 = 0x02, + SR_11_025 = 0x03, + SR_16_000 = 0x04, + SR_22_050 = 0x05, + SR_24_000 = 0x06, + SR_32_000 = 0x07, + SR_44_100 = 0x08, + SR_48_000 = 0x09, + SR_88_200 = 0x0A, + SR_96_000 = 0x0B, + SR_144_000 = 0x0C, + SR_176_400 = 0x0D, + SR_192_000 = 0x0E, + SR_384_000 = 0x0F, + + SR_COUNT = 0x10, + + SR_RATE_UNKNOWN = 0x1F +}; + +/* + * Scp Helper function + */ +enum get_set { + IS_SET = 0, + IS_GET = 1, +}; + +/* + * Duplicated from ca0110 codec + */ + +static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) +{ + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + } + if (dac) + snd_hda_codec_write(codec, dac, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); +} + +static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) +{ + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_VREF80); + if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + } + if (adc) + snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); +} + +static char *dirstr[2] = { "Playback", "Capture" }; + +static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, + int chan, int dir) +{ + char namestr[44]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, + int chan, int dir) +{ + char namestr[44]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +#define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0) +#define add_out_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 0) +#define add_in_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 1) +#define add_in_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 1) +#define add_mono_switch(codec, nid, pfx, chan) \ + _add_switch(codec, nid, pfx, chan, 0) +#define add_mono_volume(codec, nid, pfx, chan) \ + _add_volume(codec, nid, pfx, chan, 0) +#define add_in_mono_switch(codec, nid, pfx, chan) \ + _add_switch(codec, nid, pfx, chan, 1) +#define add_in_mono_volume(codec, nid, pfx, chan) \ + _add_volume(codec, nid, pfx, chan, 1) + + +/* + * CA0132 specific + */ + +struct ca0132_spec { + struct auto_pin_cfg autocfg; + struct hda_multi_out multiout; + hda_nid_t out_pins[AUTO_CFG_MAX_OUTS]; + hda_nid_t dacs[AUTO_CFG_MAX_OUTS]; + hda_nid_t hp_dac; + hda_nid_t input_pins[AUTO_PIN_LAST]; + hda_nid_t adcs[AUTO_PIN_LAST]; + hda_nid_t dig_out; + hda_nid_t dig_in; + unsigned int num_inputs; + long curr_hp_switch; + long curr_hp_volume[2]; + long curr_speaker_switch; + struct mutex chipio_mutex; + const char *input_labels[AUTO_PIN_LAST]; + struct hda_pcm pcm_rec[2]; /* PCM information */ +}; + +/* + * CA0132 chip access stuffs + */ +static int chipio_acquire_mutex(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + return 0; +} + +static int chipio_release_mutex(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_unlock(&spec->chipio_mutex); + return 0; +} + +/* + * Write chip address through the vendor widget -- NOT protected by the Mutex! + */ +static int chipio_write_address(struct hda_codec *codec, + unsigned int chip_addx) +{ + unsigned int res = 0; + int retry = 100; + + /* send low 16 bits of the address */ + do { + res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0, + HDA_SHORT_CMD_VENDOR_CHIP_IO_ADDRESS_LOW, + chip_addx & 0xffff); + retry--; + } while (res != HDA_STATUS_VENDOR_CHIP_IO_OK && retry); + + if (!retry) + return -EIO; + + retry = 100; + /* send high 16 bits of the address */ + do { + res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0, + HDA_SHORT_CMD_VENDOR_CHIP_IO_ADDRESS_HIGH, + chip_addx >> 16); + retry--; + } while (res != HDA_STATUS_VENDOR_CHIP_IO_OK && retry); + + if (!retry) + return -EIO; + + return 0; +} + +/* + * Write data through the vendor widget -- NOT protected by the Mutex! + */ +static int chipio_write_data(struct hda_codec *codec, unsigned int data) +{ + unsigned int res = 0; + int retry = 100; + + /* send low 16 bits of the data */ + do { + res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0, + HDA_SHORT_CMD_VENDOR_CHIP_IO_DATA_LOW, + data & 0xffff); + retry--; + } while (res != HDA_STATUS_VENDOR_CHIP_IO_OK && retry); + + if (!retry) + return -EIO; + + /* send high 16 bits of the data */ + retry = 100; + do { + res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0, + HDA_SHORT_CMD_VENDOR_CHIP_IO_DATA_HIGH, + data >> 16); + retry--; + } while (res != HDA_STATUS_VENDOR_CHIP_IO_OK && retry); + + if (!retry) + return -EIO; + + return 0; +} + +/* + * Read data through the vendor widget -- NOT protected by the Mutex! + */ +static int chipio_read_data(struct hda_codec *codec, unsigned int *data) +{ + unsigned int res = 0; + int retry = 100; + + /* post read */ + do { + res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0, + HDA_LONG_CMD_VENDOR_CHIP_IO_HIC_POST_READ, + 0); + retry--; + } while (res != HDA_STATUS_VENDOR_CHIP_IO_OK && retry); + + if (!retry) + return -EIO; + + /* read status */ + retry = 100; + do { + res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0, + HDA_LONG_CMD_VENDOR_CHIP_IO_STATUS, + 0); + retry--; + } while (res != HDA_STATUS_VENDOR_CHIP_IO_OK && retry); + + if (!retry) + return -EIO; + + /* read data */ + *data = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0, + HDA_LONG_CMD_VENDOR_CHIP_IO_HIC_READ_DATA, + 0); + + return 0; +} + +/* + * Write given value to the given address through the chip I/O widget. + * protected by the Mutex + */ +static int chipio_write(struct hda_codec *codec, + unsigned int chip_addx, const unsigned int data) +{ + int err = chipio_acquire_mutex(codec); + if (err < 0) + return err; + + /* write the address, and if successful proceed to write data */ + err = chipio_write_address(codec, chip_addx); + if (err < 0) + goto exit; + + err = chipio_write_data(codec, data); + if (err < 0) + goto exit; + +exit: + chipio_release_mutex(codec); + return err; +} + +/* + * Read the given address through the chip I/O widget + * protected by the Mutex + */ +static int chipio_read(struct hda_codec *codec, + unsigned int chip_addx, unsigned int *data) +{ + int err = chipio_acquire_mutex(codec); + if (err < 0) + return err; + + /* write the address, and if successful proceed to write data */ + err = chipio_write_address(codec, chip_addx); + if (err < 0) + goto exit; + + err = chipio_read_data(codec, data); + if (err < 0) + goto exit; + +exit: + chipio_release_mutex(codec); + return err; +} + +/* + * PCM stuffs + */ +static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, + u32 stream_tag, + int channel_id, int format) +{ + unsigned int oldval, newval; + + if (!nid) + return; + + snd_printdd("ca0132_setup_stream: " + "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n", + nid, stream_tag, channel_id, format); + + /* update the format-id if changed */ + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_STREAM_FORMAT, + 0); + if (oldval != format) { + msleep(20); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, + format); + } + + oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + newval = (stream_tag << 4) | channel_id; + if (oldval != newval) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + newval); + } +} + +static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) +{ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); +} + +/* + * PCM callbacks + */ +static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); + + return 0; +} + +static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_cleanup_stream(codec, spec->dacs[0]); + + return 0; +} + +/* + * Digital out + */ +static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format); + + return 0; +} + +static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_cleanup_stream(codec, spec->dig_out); + + return 0; +} + +/* + * Analog capture + */ +static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_setup_stream(codec, spec->adcs[substream->number], + stream_tag, 0, format); + + return 0; +} + +static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_cleanup_stream(codec, spec->adcs[substream->number]); + + return 0; +} + +/* + * Digital capture + */ +static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format); + + return 0; +} + +static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_cleanup_stream(codec, spec->dig_in); + + return 0; +} + +/* + */ +static struct hda_pcm_stream ca0132_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0132_playback_pcm_prepare, + .cleanup = ca0132_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0132_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0132_capture_pcm_prepare, + .cleanup = ca0132_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0132_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0132_dig_playback_pcm_prepare, + .cleanup = ca0132_dig_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0132_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0132_dig_capture_pcm_prepare, + .cleanup = ca0132_dig_capture_pcm_cleanup + }, +}; + +static int ca0132_build_pcms(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + + codec->pcm_info = info; + codec->num_pcms = 0; + + info->name = "CA0132 Analog"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = + spec->multiout.max_channels; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; + codec->num_pcms++; + + if (!spec->dig_out && !spec->dig_in) + return 0; + + info++; + info->name = "CA0132 Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->dig_out) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + ca0132_pcm_digital_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out; + } + if (spec->dig_in) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + ca0132_pcm_digital_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in; + } + codec->num_pcms++; + + return 0; +} + +#define REG_CODEC_MUTE 0x18b014 +#define REG_CODEC_HP_VOL_L 0x18b070 +#define REG_CODEC_HP_VOL_R 0x18b074 + +static int ca0132_hp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + + *valp = spec->curr_hp_switch; + snd_printd("hp_switch_get: val=0x%lx\n", spec->curr_hp_switch); + return 0; +} + +static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + unsigned int data; + int err; + + /* any change? */ + if (spec->curr_hp_switch == *valp) + return 0; + + err = chipio_read(codec, REG_CODEC_MUTE, &data); + if (err < 0) + return err; + + /* *valp 0 is mute, 1 is unmute */ + data = (data & 0x7f) | (*valp ? 0 : 0x80); + chipio_write(codec, REG_CODEC_MUTE, data); + if (err < 0) + return err; + + spec->curr_hp_switch = *valp; + snd_printd("hp_switch_put: val=0x%lx\n", spec->curr_hp_switch); + return 1; +} + +static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + + *valp = spec->curr_speaker_switch; + snd_printd("speaker_switch_get: val=0x%lx\n", + spec->curr_speaker_switch); + return 0; +} + +static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + unsigned int data; + int err; + + /* any change? */ + if (spec->curr_speaker_switch == *valp) + return 0; + + err = chipio_read(codec, REG_CODEC_MUTE, &data); + if (err < 0) + return err; + + /* *valp 0 is mute, 1 is unmute */ + data = (data & 0xef) | (*valp ? 0 : 0x10); + chipio_write(codec, REG_CODEC_MUTE, data); + if (err < 0) + return err; + + spec->curr_speaker_switch = *valp; + snd_printd("speaker_switch_put: val=0x%lx\n", + spec->curr_speaker_switch); + return 1; +} + +static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + + *valp++ = spec->curr_hp_volume[0]; + *valp = spec->curr_hp_volume[1]; + snd_printd("hp_volume_get: val[0]=0x%lx, val[1]=0x%lx\n", + spec->curr_hp_volume[0], spec->curr_hp_volume[1]); + return 0; +} + +static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + long left_vol, right_vol; + unsigned int data; + int val; + int err; + + left_vol = *valp++; + right_vol = *valp; + + /* any change? */ + if ((spec->curr_hp_volume[0] == left_vol) && + (spec->curr_hp_volume[1] == right_vol)) + return 0; + + err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data); + if (err < 0) + return err; + + val = 31 - left_vol; + data = (data & 0xe0) | val; + chipio_write(codec, REG_CODEC_HP_VOL_L, data); + if (err < 0) + return err; + + val = 31 - right_vol; + data = (data & 0xe0) | val; + chipio_write(codec, REG_CODEC_HP_VOL_R, data); + if (err < 0) + return err; + + spec->curr_hp_volume[0] = left_vol; + spec->curr_hp_volume[1] = right_vol; + snd_printd("hp_volume_put: val[0]=0x%lx, val[1]=0x%lx\n", + spec->curr_hp_volume[0], spec->curr_hp_volume[1]); + return 1; +} + +static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Headphone Playback Switch", + nid, 1, 0, HDA_OUTPUT); + knew.get = ca0132_hp_switch_get; + knew.put = ca0132_hp_switch_put; + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +static int add_hp_volume(struct hda_codec *codec, hda_nid_t nid) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO("Headphone Playback Volume", + nid, 3, 0, HDA_OUTPUT); + knew.get = ca0132_hp_volume_get; + knew.put = ca0132_hp_volume_put; + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +static int add_speaker_switch(struct hda_codec *codec, hda_nid_t nid) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Speaker Playback Switch", + nid, 1, 0, HDA_OUTPUT); + knew.get = ca0132_speaker_switch_get; + knew.put = ca0132_speaker_switch_put; + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +static void ca0132_fix_hp_caps(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int caps; + + /* set mute-capable, 1db step, 32 steps, ofs 6 */ + caps = 0x80031f06; + snd_hda_override_amp_caps(codec, cfg->hp_pins[0], HDA_OUTPUT, caps); +} + +static int ca0132_build_controls(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, err; + + if (spec->multiout.num_dacs) { + err = add_speaker_switch(codec, spec->out_pins[0]); + if (err < 0) + return err; + } + + if (cfg->hp_outs) { + ca0132_fix_hp_caps(codec); + err = add_hp_switch(codec, cfg->hp_pins[0]); + if (err < 0) + return err; + err = add_hp_volume(codec, cfg->hp_pins[0]); + if (err < 0) + return err; + } + + for (i = 0; i < spec->num_inputs; i++) { + const char *label = spec->input_labels[i]; + + err = add_in_switch(codec, spec->adcs[i], label); + if (err < 0) + return err; + err = add_in_volume(codec, spec->adcs[i], label); + if (err < 0) + return err; + if (cfg->inputs[i].type == AUTO_PIN_MIC) { + /* add Mic-Boost */ + err = add_in_mono_volume(codec, spec->input_pins[i], + "Mic Boost", 1); + if (err < 0) + return err; + } + } + + if (spec->dig_out) { + err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out); + if (err < 0) + return err; + err = add_out_volume(codec, spec->dig_out, "IEC958"); + if (err < 0) + return err; + } + + if (spec->dig_in) { + err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); + if (err < 0) + return err; + err = add_in_volume(codec, spec->dig_in, "IEC958"); + } + return 0; +} + + +/* + * refresh widget caps that stored in cache + */ +static void ca0132_refresh_widget_caps(struct hda_codec *codec) +{ + int i; + hda_nid_t nid, start; + + start = nid = codec->start_nid; + snd_printd("ca0132_refresh_widget_caps: start_nid=%d\n", start); + for (i = 0; i < codec->num_nodes; i++, nid++) { + codec->wcaps[i] = snd_hda_param_read(codec, nid, + AC_PAR_AUDIO_WIDGET_CAP); + snd_printd("wcaps[0x%02x]=0x%x\n", i+start, codec->wcaps[i]); + } +} + +static void ca0132_set_ct_ext(struct hda_codec *codec, int enable) +{ + /* Set Creative extension */ + snd_printd("SET CREATIVE EXTENSION\n"); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + HDA_LONG_CMD_VENDOR_CHIP_IO_CT_EXTENSIONS_ENABLE, + enable); + msleep(20); +} + + +static void ca0132_config(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + + /* line-outs */ + cfg->line_outs = 1; + cfg->line_out_pins[0] = 0x0b; /* front */ + cfg->line_out_type = AUTO_PIN_LINE_OUT; + + spec->dacs[0] = 0x02; + spec->out_pins[0] = 0x0b; + spec->multiout.dac_nids = spec->dacs; + spec->multiout.num_dacs = 1; + spec->multiout.max_channels = 2; + + /* headphone */ + cfg->hp_outs = 1; + cfg->hp_pins[0] = 0x0f; + + spec->hp_dac = 0; + spec->multiout.hp_nid = 0; + + /* inputs */ + cfg->num_inputs = 2; /* Mic-in and line-in */ + cfg->inputs[0].pin = 0x12; + cfg->inputs[0].type = AUTO_PIN_MIC; + cfg->inputs[1].pin = 0x11; + cfg->inputs[1].type = AUTO_PIN_LINE_IN; + + /* Mic-in */ + spec->input_pins[0] = 0x12; + spec->input_labels[0] = "Mic-In"; + spec->adcs[0] = 0x07; + + /* Line-In */ + spec->input_pins[1] = 0x11; + spec->input_labels[1] = "Line-In"; + spec->adcs[1] = 0x08; + spec->num_inputs = 2; +} + +static void ca0132_init_chip(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_init(&spec->chipio_mutex); +} + +static void ca0132_exit_chip(struct hda_codec *codec) +{ + /* put any chip cleanup stuffs here. */ +} + +static int ca0132_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < spec->multiout.num_dacs; i++) { + init_output(codec, spec->out_pins[i], + spec->multiout.dac_nids[i]); + } + init_output(codec, cfg->hp_pins[0], spec->hp_dac); + init_output(codec, cfg->dig_out_pins[0], spec->dig_out); + + for (i = 0; i < spec->num_inputs; i++) + init_input(codec, spec->input_pins[i], spec->adcs[i]); + + init_input(codec, cfg->dig_in_pin, spec->dig_in); + + return 0; +} + + +static void ca0132_free(struct hda_codec *codec) +{ + ca0132_exit_chip(codec); + kfree(codec->spec); +} + +static struct hda_codec_ops ca0132_patch_ops = { + .build_controls = ca0132_build_controls, + .build_pcms = ca0132_build_pcms, + .init = ca0132_init, + .free = ca0132_free, +}; + + + +static int patch_ca0132(struct hda_codec *codec) +{ + struct ca0132_spec *spec; + + snd_printd("patch_ca0132\n"); + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + memset((void *)spec, 0, sizeof(*spec)); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + ca0132_init_chip(codec); + ca0132_set_ct_ext(codec, 1); + ca0132_config(codec); + + codec->patch_ops = ca0132_patch_ops; + + return 0; +} + +/* + * patch entries + */ +static struct hda_codec_preset snd_hda_preset_ca0132[] = { + { .id = 0x11020011, .name = "CA0132", .patch = patch_ca0132 }, + {} /* terminator */ +}; + +MODULE_ALIAS("snd-hda-codec-id:11020011"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Creative CA0132, CA0132 HD-audio codec"); + +static struct hda_codec_preset_list ca0132_list = { + .preset = snd_hda_preset_ca0132, + .owner = THIS_MODULE, +}; + +static int __init patch_ca0132_init(void) +{ + return snd_hda_add_codec_preset(&ca0132_list); +} + +static void __exit patch_ca0132_exit(void) +{ + snd_hda_delete_codec_preset(&ca0132_list); +} + +module_init(patch_ca0132_init) +module_exit(patch_ca0132_exit) -- 1.7.0.4

2 1

Re: [alsa-devel] [PATCH] thinkpad-acpi: Enable microphone mute LED
by Andrew Lutomirski 03 Jun '11

03 Jun '11

On Fri, May 27, 2011 at 12:48 PM, Matthew Garrett <mjg59(a)srcf.ucam.org> wrote: > On Fri, May 27, 2011 at 12:39:10PM -0400, Andrew Lutomirski wrote: > >> I don't have docs so I don't really know. But the mute LED does not >> appear to be controllable from the LED interface and when mute is lit >> there's no sound regardless of what ALSA thinks. When I light the >> micmute light, I can still record sound just fine. >> >> The mic mute button does nothing on Windows unless the special driver >> is installed, and that driver seems to mute the Windows mixer, so I >> think there's nothing special other than a little userspace helper >> going on. > > Hm. That makes things kind of tricky. While it's easy to just expose the > LED, really we want to do it in such a way that userspace knows which > LEDs it should be toggling while setting the microphone mute. One thought: call the thing laptop::micmute and teach userspace to look for an led with that name. Do any of the ALSA or PulseAudio people have any preferences? --Andy

3 2

[alsa-devel] [alsa-lib] Possible bug in function 'snd_ctl_hw_open', file control_hw.c
by alex dot baldacchino dot alsasub at gmail dot com 03 Jun '11

03 Jun '11

Hello, I've noticed at some point function snd_ctl_hw_open() calls snd_ctl_new() to allocate memory for variable snd_ctl_t *ctl; then it tries to handle an error condition, but, regardless of such an error, memory locations pointed by ctl are accessed and filled in (eventually with invalid data, since the error causes 'fd' to be closed and 'hw' to be freed). Since ctl could point a random area in case of allocation failure, perhaps the function should return if such an error occurs and avoid dereferencing it. An easy fix would be the following, unless I'm missing something. ------------------------------------------------------ control_hw.patch -------------------------------------------------------------- --- old/src/control/control_hw.c 2011-05-31 14:20:12.975999400 +0200 +++ new/src/control/control_hw.c 2011-05-31 14:21:35.559999388 +0200 @@ -414,6 +414,7 @@ if (err < 0) { close(fd); free(hw); + return err; } ctl->ops = &snd_ctl_hw_ops; ctl->private_data = hw;

2 1

[alsa-devel] Regd alsa under runs
by Thilaga 03 Jun '11

03 Jun '11

Hi All, I am just reading data from mic and giving to the PC speakers through ALSA API's. The output binary which is working fine in one PC is not working in the other PC. It gives an under run error. Could you Please help me on this. Thanks & Regards, Thilaga

1 0

Re: [alsa-devel] Creative X-Fi Titanium HD ctxfi bugs 5378 5218 4868
by Harry Butterworth 03 Jun '11

03 Jun '11

Here's a second version of this patch (against 1.0.24) which fixes the input source selection and reverts the PLL hacks. The PLL initialization for 44100Hz is still unreliable. I'm posting this so the Creative tech support can use it to debug the PLL init code. Still not to be applied - I'll do another version against the git repository once we have the PLL init sorted. Harry

1 0

[alsa-devel] [PATCH] UCM: Implement ConflictingDevices, add device list to devices
by Stephen Warren 03 Jun '11

03 Jun '11

Wherever SupportedDevice can appear, also allow ConflictingDevice. Only one or the other (or neither) may be specified. When neither is specified, allow anything. Sometimes, listing ConflictingDevices may result in a shorter list than explicitly listing all SupportedDevices. Add support for SupportedDevice and ConflictingDevice to SectionDevice. This allows representing devices which are mutually exclusive, e.g. due to a mux that switches between capturing from two different microphones, without the possibility of mixing. Enhance is_modifier_supported to allow ignoring SupportedDevice and ConflictingDevice. This is useful when querying values from a SectionModifier; there's no reason we shouldn't be able to query values just because the current configuration would prevent enabling that device. The new is_device_supported is implemented similarly. Enhance switch_device to remove the old device from the current device list before querying for the new device, and add it back immediately afterwards. This allows the query for the new device to ignore any conflicts caused solely by the old device. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Depending on decisions in the thread I recently started entitled "UCM modifier names", this patch may change, e.g. name_match may not be needed, or could be simpler. I should probably split some of this into separate patches, but I figured I'd get it out there for people to discuss first. This patch builds on the other couple of alsa-lib UCM patches I posted earlier today. src/ucm/main.c | 169 +++++++++++++++++++++++++++++++++------------------ src/ucm/parser.c | 74 ++++++++++++++++++---- src/ucm/ucm_local.h | 19 +++++- src/ucm/utils.c | 9 ++- 4 files changed, 190 insertions(+), 81 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index fb4aa91..0a3331a 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -463,6 +463,82 @@ static inline struct use_case_verb *find_verb(snd_use_case_mgr_t *uc_mgr, verb_name); } +static int name_match(const char *test, const char *against) +{ + char *pos_test, *pos_against; + int len_test, len_against; + + pos_test = strchr(test, '.'); + if (pos_test) { + if (strcmp(test, against)) + return 0; + return 1; + } + + pos_against = strchr(against, '.'); + if (!pos_against) { + if (strcmp(test, against)) + return 0; + return 1; + } + + len_test = strlen(test); + len_against = pos_against - against; + + if (len_test != len_against) + return 0; + + if (memcmp(test, against, len_test)) + return 0; + + return 1; +} + +static int is_devlist_supported(snd_use_case_mgr_t *uc_mgr, + struct dev_list *dev_list) +{ + struct dev_list_node *device; + struct use_case_device *adev; + struct list_head *pos, *pos1; + int found_ret; + + switch (dev_list->type) { + case DEVLIST_NONE: + default: + return 1; + case DEVLIST_SUPPORTED: + found_ret = 1; + break; + case DEVLIST_CONFLICTING: + found_ret = 0; + break; + } + + list_for_each(pos, &dev_list->list) { + device = list_entry(pos, struct dev_list_node, list); + + list_for_each(pos1, &uc_mgr->active_devices) { + adev = list_entry(pos1, struct use_case_device, + active_list); + if (name_match(device->name, adev->name)) + return found_ret; + } + } + return 1 - found_ret; +} + +static inline int is_modifier_supported(snd_use_case_mgr_t *uc_mgr, + struct use_case_modifier *modifier) +{ + return is_devlist_supported(uc_mgr, &modifier->dev_list); +} + +static inline int is_device_supported(snd_use_case_mgr_t *uc_mgr, + struct use_case_device *device) +{ + return is_devlist_supported(uc_mgr, &device->dev_list); +} + /** * \brief Find device * \param verb Use case verb @@ -470,52 +546,27 @@ static inline struct use_case_verb *find_verb(snd_use_case_mgr_t *uc_mgr, * \return structure on success, otherwise a NULL (not found) */ static inline struct use_case_device * - find_device(struct use_case_verb *verb, - const char *device_name) + find_device(snd_use_case_mgr_t *uc_mgr, const char *device_name, + int check_supported) { - return find(&verb->device_list, - struct use_case_device, list, name, - device_name); -} + struct use_case_device *device; + struct use_case_verb *verb = uc_mgr->active_verb; + struct list_head *pos; -static int is_modifier_supported(snd_use_case_mgr_t *uc_mgr, - struct use_case_modifier *modifier) -{ - struct dev_list *device; - struct use_case_device *adev; - struct list_head *pos, *pos1; - char *cpos; - int dlen, len; - - list_for_each(pos, &modifier->dev_list) { - device = list_entry(pos, struct dev_list, list); - cpos = strchr(device->name, '.'); - if (cpos) { - if (find(&uc_mgr->active_devices, - struct use_case_device, active_list, - name, device->name)) - return 1; - } else { - dlen = strlen(device->name); - list_for_each(pos1, &uc_mgr->active_devices) { - adev = list_entry(pos1, struct use_case_device, - active_list); - cpos = strchr(adev->name, '.'); - if (cpos) - len = cpos - adev->name; - else - len = strlen(adev->name); - if (len != dlen) - continue; - if (memcmp(adev->name, device->name, len)) - continue; - return 1; - } - } + list_for_each(pos, &verb->device_list) { + device = list_entry(pos, struct use_case_device, list); + + if (!name_match(device_name, device->name)) + continue; + + if (check_supported && + !is_device_supported(uc_mgr, device)) + continue; + + return device; } - return 0; + return NULL; } - /** * \brief Find modifier * \param verb Use case verb @@ -523,28 +574,24 @@ static int is_modifier_supported(snd_use_case_mgr_t *uc_mgr, * \return structure on success, otherwise a NULL (not found) */ static struct use_case_modifier * - find_modifier(snd_use_case_mgr_t *uc_mgr, const char *modifier_name) + find_modifier(snd_use_case_mgr_t *uc_mgr, const char *modifier_name, + int check_supported) { struct use_case_modifier *modifier; struct use_case_verb *verb = uc_mgr->active_verb; struct list_head *pos; - char name[64], *cpos; list_for_each(pos, &verb->modifier_list) { modifier = list_entry(pos, struct use_case_modifier, list); - strncpy(name, modifier->name, sizeof(name)); - name[sizeof(name)-1] = '\0'; - cpos = strchr(name, '.'); - if (!cpos) + if (!name_match(modifier_name, modifier->name)) continue; - *cpos= '\0'; - if (strcmp(name, modifier_name)) + if (check_supported && + !is_modifier_supported(uc_mgr, modifier)) continue; - if (is_modifier_supported(uc_mgr, modifier)) - return modifier; + return modifier; } return NULL; } @@ -1084,13 +1131,13 @@ static int get_value(snd_use_case_mgr_t *uc_mgr, int err; if (item != NULL) { - mod = find_modifier(uc_mgr, item); + mod = find_modifier(uc_mgr, item, 0); if (mod != NULL) { err = get_value1(value, &mod->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } - dev = find_device(uc_mgr->active_verb, item); + dev = find_device(uc_mgr, item, 0); if (dev != NULL) { err = get_value1(value, &dev->value_list, identifier); if (err >= 0 || err != -ENOENT) @@ -1309,7 +1356,7 @@ static int set_device_user(snd_use_case_mgr_t *uc_mgr, if (uc_mgr->active_verb == NULL) return -ENOENT; - device = find_device(uc_mgr->active_verb, device_name); + device = find_device(uc_mgr, device_name, 1); if (device == NULL) return -ENOENT; return set_device(uc_mgr, device, enable); @@ -1324,7 +1371,7 @@ static int set_modifier_user(snd_use_case_mgr_t *uc_mgr, if (uc_mgr->active_verb == NULL) return -ENOENT; - modifier = find_modifier(uc_mgr, modifier_name); + modifier = find_modifier(uc_mgr, modifier_name, 1); if (modifier == NULL) return -ENOENT; return set_modifier(uc_mgr, modifier, enable); @@ -1349,10 +1396,12 @@ static int switch_device(snd_use_case_mgr_t *uc_mgr, uc_error("error: device %s already enabled", new_device); return -EINVAL; } - xold = find_device(uc_mgr->active_verb, old_device); + xold = find_device(uc_mgr, old_device, 1); if (xold == NULL) return -ENOENT; - xnew = find_device(uc_mgr->active_verb, new_device); + list_del(&xold->active_list); + xnew = find_device(uc_mgr, new_device, 1); + list_add_tail(&xold->active_list, &uc_mgr->active_devices); if (xnew == NULL) return -ENOENT; err = 0; @@ -1401,10 +1450,10 @@ static int switch_modifier(snd_use_case_mgr_t *uc_mgr, uc_error("error: modifier %s already enabled", new_modifier); return -EINVAL; } - xold = find_modifier(uc_mgr, old_modifier); + xold = find_modifier(uc_mgr, old_modifier, 1); if (xold == NULL) return -ENOENT; - xnew = find_modifier(uc_mgr, new_modifier); + xnew = find_modifier(uc_mgr, new_modifier, 1); if (xnew == NULL) return -ENOENT; err = 0; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index f7de9bd..53ff035 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -165,18 +165,25 @@ static int parse_compound(snd_use_case_mgr_t *uc_mgr, snd_config_t *cfg, /* - * Parse transition + * Parse device list */ -static int parse_supported_device(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, - struct list_head *dlist, - snd_config_t *cfg) +static int parse_device_list(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, + struct dev_list *dev_list, + enum dev_list_type type, + snd_config_t *cfg) { - struct dev_list *sdev; + struct dev_list_node *sdev; const char *id; snd_config_iterator_t i, next; snd_config_t *n; int err; + if (dev_list->type != DEVLIST_NONE) { + uc_error("error: multiple supported or" + " conflicting device lists"); + return -EEXIST; + } + if (snd_config_get_id(cfg, &id) < 0) return -EINVAL; @@ -191,7 +198,7 @@ static int parse_supported_device(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, if (snd_config_get_id(n, &id) < 0) return -EINVAL; - sdev = calloc(1, sizeof(struct dev_list)); + sdev = calloc(1, sizeof(struct dev_list_node)); if (sdev == NULL) return -ENOMEM; err = parse_string(n, &sdev->name); @@ -199,8 +206,11 @@ static int parse_supported_device(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, free(sdev); return err; } - list_add(&sdev->list, dlist); + list_add(&sdev->list, &dev_list->list); } + + dev_list->type = type; + return 0; } @@ -392,11 +402,17 @@ static int parse_value(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, * SectionModifier."Capture Voice".0 { * * Comment "Record voice call" + * * SupportedDevice [ * "x" # all x device instances * "y.0" # device y instance 0 only * ] * + * ConflictingDevice [ + * "x" # all x device instances + * "y.0" # device y instance 0 only + * ] + * * EnableSequence [ * .... * ] @@ -418,6 +434,9 @@ static int parse_value(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, * } * * } + * + * SupportedDevice and ConflictingDevice cannot be specified together. + * Both are optional. */ static int parse_modifier(snd_use_case_mgr_t *uc_mgr, snd_config_t *cfg, @@ -441,7 +460,7 @@ static int parse_modifier(snd_use_case_mgr_t *uc_mgr, INIT_LIST_HEAD(&modifier->enable_list); INIT_LIST_HEAD(&modifier->disable_list); INIT_LIST_HEAD(&modifier->transition_list); - INIT_LIST_HEAD(&modifier->dev_list); + INIT_LIST_HEAD(&modifier->dev_list.list); INIT_LIST_HEAD(&modifier->value_list); list_add_tail(&modifier->list, &verb->modifier_list); err = parse_get_safe_id(cfg, &id); @@ -470,7 +489,8 @@ static int parse_modifier(snd_use_case_mgr_t *uc_mgr, } if (strcmp(id, "SupportedDevice") == 0) { - err = parse_supported_device(uc_mgr, &modifier->dev_list, n); + err = parse_device_list(uc_mgr, &modifier->dev_list, + DEVLIST_SUPPORTED, n); if (err < 0) { uc_error("error: failed to parse supported" " device list"); @@ -478,6 +498,16 @@ static int parse_modifier(snd_use_case_mgr_t *uc_mgr, } } + if (strcmp(id, "ConflictingDevice") == 0) { + err = parse_device_list(uc_mgr, &modifier->dev_list, + DEVLIST_CONFLICTING, n); + if (err < 0) { + uc_error("error: failed to parse conflicting" + " device list"); + return err; + } + } + if (strcmp(id, "EnableSequence") == 0) { err = parse_sequence(uc_mgr, &modifier->enable_list, n); if (err < 0) { @@ -518,11 +548,6 @@ static int parse_modifier(snd_use_case_mgr_t *uc_mgr, } } - if (list_empty(&modifier->dev_list)) { - uc_error("error: %s: modifier missing supported device sequence", modifier->name); - return -EINVAL; - } - return 0; } @@ -572,6 +597,7 @@ static int parse_device_index(snd_use_case_mgr_t *uc_mgr, INIT_LIST_HEAD(&device->enable_list); INIT_LIST_HEAD(&device->disable_list); INIT_LIST_HEAD(&device->transition_list); + INIT_LIST_HEAD(&device->dev_list.list); INIT_LIST_HEAD(&device->value_list); list_add_tail(&device->list, &verb->device_list); if (parse_get_safe_id(cfg, &id) < 0) @@ -598,6 +624,26 @@ static int parse_device_index(snd_use_case_mgr_t *uc_mgr, continue; } + if (strcmp(id, "SupportedDevice") == 0) { + err = parse_device_list(uc_mgr, &device->dev_list, + DEVLIST_SUPPORTED, n); + if (err < 0) { + uc_error("error: failed to parse supported" + " device list"); + return err; + } + } + + if (strcmp(id, "ConflictingDevice") == 0) { + err = parse_device_list(uc_mgr, &device->dev_list, + DEVLIST_CONFLICTING, n); + if (err < 0) { + uc_error("error: failed to parse conflicting" + " device list"); + return err; + } + } + if (strcmp(id, "EnableSequence") == 0) { uc_dbg("EnableSequence"); err = parse_sequence(uc_mgr, &device->enable_list, n); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 2ceceaa..0522bf5 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -76,11 +76,21 @@ struct transition_sequence { /* * Modifier Supported Devices. */ -struct dev_list { +enum dev_list_type { + DEVLIST_NONE, + DEVLIST_SUPPORTED, + DEVLIST_CONFLICTING +}; + +struct dev_list_node { struct list_head list; char *name; }; +struct dev_list { + enum dev_list_type type; + struct list_head list; +}; /* * Describes a Use Case Modifier and it's enable and disable sequences. @@ -100,8 +110,8 @@ struct use_case_modifier { /* modifier transition list */ struct list_head transition_list; - /* list of supported devices per modifier */ - struct list_head dev_list; + /* list of devices supported or conflicting */ + struct dev_list dev_list; /* values */ struct list_head value_list; @@ -125,6 +135,9 @@ struct use_case_device { /* device transition list */ struct list_head transition_list; + /* list of devices supported or conflicting */ + struct dev_list dev_list; + /* value list */ struct list_head value_list; }; diff --git a/src/ucm/utils.c b/src/ucm/utils.c index 85549e1..45307b0 100644 --- a/src/ucm/utils.c +++ b/src/ucm/utils.c @@ -99,13 +99,13 @@ void uc_mgr_free_value(struct list_head *base) } } -void uc_mgr_free_dev_list(struct list_head *base) +void uc_mgr_free_dev_list(struct dev_list *dev_list) { struct list_head *pos, *npos; - struct dev_list *dlist; + struct dev_list_node *dlist; - list_for_each_safe(pos, npos, base) { - dlist = list_entry(pos, struct dev_list, list); + list_for_each_safe(pos, npos, &dev_list->list) { + dlist = list_entry(pos, struct dev_list_node, list); free(dlist->name); list_del(&dlist->list); free(dlist); @@ -189,6 +189,7 @@ void uc_mgr_free_device(struct list_head *base) uc_mgr_free_sequence(&dev->enable_list); uc_mgr_free_sequence(&dev->disable_list); uc_mgr_free_transition(&dev->transition_list); + uc_mgr_free_dev_list(&dev->dev_list); uc_mgr_free_value(&dev->value_list); list_del(&dev->list); free(dev); -- 1.7.0.4

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[alsa-devel] UCM modifier names
by Stephen Warren 03 Jun '11

03 Jun '11

Liam, UCM syntax is e.g.: SectionModifier."Capture Voice".0 { This maps to a modifier->name of "Capture Voice.0". However, when find_modifier searches for modifiers, it strips off the index (".0") from modifier->name, and just compares base name. As such, the index doesn't seem useful. I propose: a) In parse_modifier, fail if index!=0. b) In parse_modifier, store just the name in modifier->name. c) In find_modifier, remove all the special handling of "." in the name, and just strcmp(modifier->name, modifier_name). Perhaps the index should be completely removed from the file syntax too? If an index was actually useful in any case, the author of the file could simply add it within the name string, i.e.: typical: SectionModifier."Capture Voice" { unusual: SectionModifier."Capture Voice 0" { Of course, that wouldn't be backwards-compatible file syntax. Perhaps we could make the index optional, but always validate it was 0 if present, as I said above. I actually wonder if the index on the SectionDevice makes any sense either; functions like is_modifier_supported would be a lot simpler if devices simply never had an index, such that is_modifier_supported was a simple strcmp too. The worst fallout from that change might be a requirement to list more entries in SupportedDevice/ConflictingDevice. Second, when find_modifier is called from get_value, it'll fail with a valid modifier name that happens not to be supported on any currently active devices. Is that intended? I'd expect to be able to retrieve values from any/all devices and modifiers irrespective of whether those devices or modifiers could actually be activated given the currently active configuration. I propose: Adding a parameter to find_modifier to indicate whether to call is_modifier_supported or not, this new parameter being false for get_value and true in other cases. This issue will also be relevant to find_device when I modify that to call new function is_device_supported. For find_device, we may also have to add a parameter for a device name that gets ignore when checking SupportedDevice/ConflictingDevice, which can be passed when switching devices. Thanks for your thoughts. -- nvpublic

2 1

[alsa-devel] [patch 07/14] i8253: Make pcsp sound driver use the shared i8253_lock.
by ralf＠linux-mips.org 03 Jun '11

03 Jun '11

Signed-off-by: Ralf Baechle <ralf(a)linux-mips.org> To: linux-kernel(a)vger.kernel.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org sound/drivers/pcsp/pcsp.h | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) Index: linux-mips/sound/drivers/pcsp/pcsp.h =================================================================== --- linux-mips.orig/sound/drivers/pcsp/pcsp.h +++ linux-mips/sound/drivers/pcsp/pcsp.h @@ -10,14 +10,8 @@ #define __PCSP_H__ #include <linux/hrtimer.h> -#include <linux/timex.h> -#if defined(CONFIG_MIPS) || defined(CONFIG_X86) -/* Use the global PIT lock ! */ #include <linux/i8253.h> -#else -#include <asm/8253pit.h> -static DEFINE_RAW_SPINLOCK(i8253_lock); -#endif +#include <linux/timex.h> #define PCSP_SOUND_VERSION 0x400 /* read 4.00 */ #define PCSP_DEBUG 0

2 1

[alsa-devel] [PATCH 1/2] alsaucm: Add list1 command for non-tuple lists
by Stephen Warren 03 Jun '11

03 Jun '11

snd_use_case_get_list returns lists of strings that are either: a) A sequence of single strings b) A sequence of pairs of strings all flattened into a single list The current list command assumes layout (b) above, and hence prints nothing when printing a single-entry list that's actually in layout (a). Add a new command "list1" to dump lists in layout (a). Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- alsaucm/usecase.c | 29 +++++++++++++++++++++-------- 1 files changed, 21 insertions(+), 8 deletions(-) diff --git a/alsaucm/usecase.c b/alsaucm/usecase.c index 83781d0..f24e63e 100644 --- a/alsaucm/usecase.c +++ b/alsaucm/usecase.c @@ -63,7 +63,8 @@ enum uc_cmd { OM_RESET, OM_RELOAD, OM_LISTCARDS, - OM_LIST, + OM_LIST2, + OM_LIST1, /* set/get */ OM_SET, @@ -87,7 +88,8 @@ static struct cmd cmds[] = { { OM_RESET, 0, 1, "reset" }, { OM_RELOAD, 0, 1, "reload" }, { OM_LISTCARDS, 0, 0, "listcards" }, - { OM_LIST, 1, 1, "list" }, + { OM_LIST1, 1, 1, "list1" }, + { OM_LIST2, 1, 1, "list" }, { OM_SET, 2, 1, "set" }, { OM_GET, 1, 1, "get" }, { OM_GETI, 1, 1, "geti" }, @@ -172,7 +174,7 @@ static int do_one(struct context *context, struct cmd *cmd, char **argv) { const char **list, *str; long lval; - int err, i; + int err, i, j, entries; if (cmd->opencard && context->uc_mgr == NULL) { fprintf(stderr, "%s: command '%s' requires an open card\n", @@ -233,7 +235,17 @@ static int do_one(struct context *context, struct cmd *cmd, char **argv) } snd_use_case_free_list(list, err); break; - case OM_LIST: + case OM_LIST1: + case OM_LIST2: + switch (cmd->code) { + case OM_LIST1: + entries = 1; + break; + case OM_LIST2: + entries = 2; + break; + } + err = snd_use_case_get_list(context->uc_mgr, argv[0], &list); @@ -246,10 +258,11 @@ static int do_one(struct context *context, struct cmd *cmd, char **argv) } if (err == 0) printf(" list is empty\n"); - for (i = 0; i < err / 2; i++) { - printf(" %i: %s\n", i, list[i*2]); - if (list[i*2+1]) - printf(" %s\n", list[i*2+1]); + for (i = 0; i < err / entries; i++) { + printf(" %i: %s\n", i, list[i*entries]); + for (j = 0; j < entries - 1; j++) + if (list[i*entries+j+1]) + printf(" %s\n", list[i*entries+j+1]); } snd_use_case_free_list(list, err); break; -- 1.7.0.4

2 2

[alsa-devel] [Trivial PATCH 0/5] treewide: Use angle brackets for system includes
by Joe Perches 03 Jun '11

03 Jun '11

Just neatening. Joe Perches (5): drbd: Use angle brackets for system includes drm: Use angle brackets for system includes aix94xx: Use angle brackets for system includes ALSA: asihpi: Use angle brackets for system includes staging: msm: Use angle brackets for system includes drivers/block/drbd/drbd_int.h | 2 +- drivers/block/drbd/drbd_nl.c | 4 ++-- drivers/gpu/drm/drm_ioctl.c | 2 +- drivers/gpu/drm/i915/i915_gem_tiling.c | 4 ++-- drivers/gpu/drm/nouveau/nouveau_drv.h | 10 +++++----- drivers/gpu/drm/ttm/ttm_agp_backend.c | 6 +++--- drivers/gpu/drm/ttm/ttm_bo.c | 6 +++--- drivers/gpu/drm/ttm/ttm_bo_manager.c | 6 +++--- drivers/gpu/drm/ttm/ttm_bo_util.c | 4 ++-- drivers/gpu/drm/ttm/ttm_execbuf_util.c | 6 +++--- drivers/gpu/drm/ttm/ttm_lock.c | 4 ++-- drivers/gpu/drm/ttm/ttm_memory.c | 6 +++--- drivers/gpu/drm/ttm/ttm_module.c | 2 +- drivers/gpu/drm/ttm/ttm_object.c | 4 ++-- drivers/gpu/drm/ttm/ttm_page_alloc.c | 4 ++-- drivers/gpu/drm/ttm/ttm_tt.c | 8 ++++---- drivers/gpu/drm/vmwgfx/vmwgfx_buffer.c | 4 ++-- drivers/gpu/drm/vmwgfx/vmwgfx_drv.c | 8 ++++---- drivers/gpu/drm/vmwgfx/vmwgfx_drv.h | 12 ++++++------ drivers/gpu/drm/vmwgfx/vmwgfx_execbuf.c | 4 ++-- drivers/gpu/drm/vmwgfx/vmwgfx_fb.c | 2 +- drivers/gpu/drm/vmwgfx/vmwgfx_fifo.c | 2 +- drivers/gpu/drm/vmwgfx/vmwgfx_gmr.c | 2 +- drivers/gpu/drm/vmwgfx/vmwgfx_gmrid_manager.c | 6 +++--- drivers/gpu/drm/vmwgfx/vmwgfx_overlay.c | 2 +- drivers/gpu/drm/vmwgfx/vmwgfx_resource.c | 4 ++-- drivers/scsi/aic94xx/aic94xx_dump.c | 2 +- drivers/staging/msm/ebi2_l2f.c | 2 +- drivers/staging/msm/ebi2_tmd20.c | 2 +- drivers/staging/msm/mddihost.h | 2 +- drivers/staging/msm/mdp_ppp.c | 2 +- drivers/staging/msm/mdp_ppp_v20.c | 2 +- drivers/staging/msm/mdp_ppp_v31.c | 2 +- drivers/staging/msm/msm_fb.h | 2 +- include/drm/ttm/ttm_bo_driver.h | 12 ++++++------ include/drm/ttm/ttm_execbuf_util.h | 2 +- include/drm/ttm/ttm_lock.h | 2 +- include/linux/drbd_tag_magic.h | 2 +- sound/pci/asihpi/hpidspcd.c | 2 +- 39 files changed, 80 insertions(+), 80 deletions(-) -- 1.7.5.rc3.dirty

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[alsa-devel] [PATCH] asoc atmel_ssc: Don't try to free ssc if request failed
by Joachim Eastwood 03 Jun '11

03 Jun '11

Hello, Patch below fixes a segmentation fault in atmel_ssc_dai if ssc_request() fails. We should only call ssc_free() when ssc_request() succeeds or bad things will happen. Signed-off-by: Joachim Eastwood <joachim.eastwood(a)jotron.com> --- sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 7fbfa05..eda955b 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -848,9 +848,10 @@ int atmel_ssc_set_audio(int ssc_id) if (IS_ERR(ssc)) pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n", PTR_ERR(ssc)); - else + else { ssc_pdev->dev.parent = &(ssc->pdev->dev); - ssc_free(ssc); + ssc_free(ssc); + } ret = platform_device_add(ssc_pdev); if (ret < 0) -- 1.7.5.2

3 3

[alsa-devel] [patch] ALSA: fm801: add error handling if auto-detect fails
by Dan Carpenter 03 Jun '11

03 Jun '11

In the original code if auto detect failed and tea575x_tuner == 4 then we copy bogus information to chip->tea.card. I've changed the autodetect code to cleanup and return -ENODEV on error instead. I made the previous chunk return on error as well. Potentially the hardware should limp along without the radio? I don't know this hardware well enough to say. Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- Compile tested only. diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index eacd490..a7ec703 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1234,9 +1234,12 @@ static int __devinit snd_fm801_create(struct snd_card *card, sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && (tea575x_tuner & TUNER_TYPE_MASK) < 4) { - if (snd_tea575x_init(&chip->tea)) + if (snd_tea575x_init(&chip->tea)) { snd_printk(KERN_ERR "TEA575x radio not found\n"); - } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) + snd_fm801_free(chip); + return -ENODEV; + } + } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) { /* autodetect tuner connection */ for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) { chip->tea575x_tuner = tea575x_tuner; @@ -1246,6 +1249,12 @@ static int __devinit snd_fm801_create(struct snd_card *card, break; } } + if (tea575x_tuner == 4) { + snd_printk(KERN_ERR "TEA575x radio not found\n"); + snd_fm801_free(chip); + return -ENODEV; + } + } strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card)); #endif

2 1

[alsa-devel] [PATCH ALSA HDA 1/2] Fix HP pin and Front pin of ad1988/ad1989 in ad198x_eapd_powerdown
by Raymond Yau 03 Jun '11

03 Jun '11

Refer to commit ea52bf260ecbb175339af3178c15788df21b7516 1) Fix HP pin and Front pin of ad1988/ad1989 in ad198x_eapd_powerdown According to ad1988a/ad1988b datasheet, Port A (0x11) is HP and Port D (0x12) is Front but both of these pins in ad1988a are not EAPD capable 2) Check pin support EAPD in ad198x_power_eapd_write

2 1

[alsa-devel] [PATCH] Sound, 6fire: Don't leak firmware in error path
by Jesper Juhl 03 Jun '11

03 Jun '11

One of the error paths in sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload() neglects to free the memory allocated for the firmware before returning, thus leaking the memory. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- firmware.c | 1 + 1 file changed, 1 insertion(+) patch against 3.0-rc1 compile tested only diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index d47beff..a91719d 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -227,6 +227,7 @@ static int usb6fire_fw_ezusb_upload( ret = usb6fire_fw_ihex_init(fw, rec); if (ret < 0) { kfree(rec); + release_firmware(fw); snd_printk(KERN_ERR PREFIX "error validating ezusb " "firmware %s.\n", fwname); return ret; -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

2 1

[alsa-devel] [PATCH 1/2] UCM: Fix typo in error message
by Stephen Warren 03 Jun '11

03 Jun '11

parse_name_is_safe disallows dots in the name, so not is the correct word to use Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- src/ucm/parser.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 38fdba0..f7de9bd 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -62,7 +62,7 @@ int parse_string(snd_config_t *n, char **res) int parse_is_name_safe(char *name) { if (strchr(name, '.')) { - uc_error("char '.' now allowed in '%s'", name); + uc_error("char '.' not allowed in '%s'", name); return 0; } return 1; -- 1.7.0.4

1 1

[alsa-devel] ASoC updates for 3.1
by Mark Brown 02 Jun '11

02 Jun '11

The following changes since commit 55922c9d1b84b89cb946c777fddccb3247e7df2c: Linux 3.0-rc1 (2011-05-29 17:43:36 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.41 Axel Lin (1): ASoC: Remove redundant freq assignment for max98095->sysclk/max98088->sysclk Ben Gardiner (7): ASoC: davinci-pcm: trivial: make ping-pong params setup symmetrical ASoC: davinci-pcm: expand the .formats ASoC: davinci-pcm: increase the maximum channels ASoC: davinci-pcm: fix audible glitch on 2nd ping-pong playback ASoC: davinci-pcm: extract period elapsed functions ASoC: davinci-pcm: convert to BATCH mode ASoC: davinci-pcm: comments for the conversion to BATCH mode Daniel Mack (1): ASoC: Fix comment in cs4270 codec driver Dimitris Papastamos (2): ASoC: soc-cache: Block based rbtree compression ASoC: soc-cache: Cache a pointer to the last accessed rbnode Kuninori Morimoto (9): ASoC: sh: fsi: tidyup parameter of fsi_stream_push ASoC: sh: fsi: add fsi_set_master_clk ASoC: sh: fsi: irq control moves to fsi_port_start/stop ASoC: sh: fsi: tidyup unclear variable naming ASoC: sh: fsi: remove pm_runtime from fsi_dai_set_fmt. ASoC: sh: fsi: make sure fsi_stream_push/pop access by spin lock ASoC: sh: fsi: remove fsi_module_init/kill ASoC: sh: fsi: cleanup suspend/resume ASoC: sh: fsi: add fsi_hw_startup/shutdown Mark Brown (6): ASoC: Convert 16x16 write to use cpu_to_be16() ASoC: Convert 7x9 write to use cpu_to_be16() ASoC: Use cpu_to_be16() in 8x16 write ASoC: Use explicit endianness conversion in snd_soc_16_8_write() ASoC: Fix wm_hubs input PGA ZC bits Merge branch 'for-2.6.40' into for-2.6.41 Nicolas Ferre (3): ASoC: trivial: typo in debug comment ASoC: trivial: typo in atmel_pcm_dma_params strucutre comment ASoC: atmel_ssc_dai: fix ssc error path Stephen Warren (3): ASoC: core: Don't schedule deferred_resume_work twice ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared ASoC: Tegra: Enable Kaen HP_MUTE at boot sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel-pcm.h | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 4 +- sound/soc/codecs/cs4270.c | 5 +- sound/soc/codecs/max98088.c | 2 - sound/soc/codecs/max98095.c | 2 - sound/soc/codecs/wm_hubs.c | 8 +- sound/soc/davinci/davinci-pcm.c | 148 ++++++----- sound/soc/sh/fsi.c | 576 ++++++++++++++++++++------------------- sound/soc/soc-cache.c | 294 +++++++++++++++----- sound/soc/soc-core.c | 19 +- sound/soc/soc-dapm.c | 5 +- sound/soc/tegra/tegra_wm8903.c | 2 +- 13 files changed, 629 insertions(+), 440 deletions(-)

2 1