January 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC] Fix for conflict of HDMI and SPDIF IEC958 controls
by Takashi Iwai 12 Feb '13

12 Feb '13

Hi, there is a long-standing problem in HD-audio regarding IEC958 controls. When both an SPDIF and an HDMI are created on the same card (e.g. one from analog codec and one from graphics chip), the driver assigns the IEC958 controls just with new indices, 0, 1, 2... The problem is that there is no way to connect between this index and the actual PCM device. Currently, alsa-lib HDA-Intel.conf has a fixed configuration: spdif -> IEC958 xxx index=0, PCM dev=1 hdmi,0 -> IEC958 xxx index=0, PCM dev=3 hdmi,1 -> IEC958 xxx index=1, PCM dev=7 hdmi,2 -> IEC958 xxx index=2, PCM dev=8 hdmi,3 -> IEC958 xxx index=3, PCM dev=9 So obviously spdif and the first hdmi conflict. Basically this can be fixed by reassigning each IEC958 control with the same device number corresponding to the PCM device. That is, for SPDIF, assign a control element with device=1, for HDMI, device=3,7... However, this obviously breaks the old configuration unless user upgrades the alsa-lib configuration. Thus this is no-go. Now here is a compromise: the IEC958 control for SPDIF is reassigned to device=1 *only* when SPDIF and HDMI coexist. Since the configuration is anyway broken as is now in such a case, it's no big deal to fix one side in an incompatible way. (The reason why SPDIF is re-assigned is that I guess majority of user require more HDMI than SPDIF in such a configuration.) In addition, we need a fix in alsa-lib. Also not for breaking the compatibility with older kernel, we need some fallback check. I did a quick hack to alsa-lib conf code and added "skip_rest" option to the hook element, so that only the first matching element is taken. Long story short, I cooked up three patches. One patch is for kernel, to add the workaround above, and the two are for alsa-lib, one clean up and one to introduce the skip_rest option (and application to HDA-Intel.conf). My plan is to merge this to 3.8 tree, so it's no urgent issue. But it's of course always good to fix something. thanks, Takashi

5 29

[alsa-devel] [PATCH RFC 00/20] More patches for HD-audio generic auto-parser
by Takashi Iwai 11 Feb '13

11 Feb '13

Hi, this is a series of patches for more enhancements of HD-audio generic parser I posted shortly ago. All patches are found in sound-unstable git tree test/hda-gen-parser branch (and merged to master branch, too). git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Takashi

2 22

[alsa-devel] PandaBoard ES Audio Problems
by Brent Weatherall 06 Feb '13

06 Feb '13

Hello. While developing with OpenMax and ALSA on the PandaBoard ES (OMAP 4460 armhf), I ran into a strange problem. At a high level, when configuring the built in audio device (3.5 mm jack), the device accepts ANY rate requested (via snd_pcm_set_rate_near). it accepts unsupported rates without alteration. If the rate is actually not supported, then 'snd_pcm_hw_params' fails with error '-22'. After this happens, it is not possible to reconfigure the rate without closing the device and starting over to completely re-configure from the start (calling snd_pcm_open, etc...). It is not possible to simply try another rate. Technical details below: When 'snd_pcm_set_rate_near' is called, during the min and max rate probing, the code reaches interval_inline.h:71 INTERVAL_INLINE int snd_interval_min(const snd_interval_t *i) During this call, line '74 return i->min;' executes. This function ALWAYS returns the same rate value requested in 'snd_pcm_set_rate_near' as the minimum possible rate. If I request a rate near 44100, the minimum rates comes back as 44100. If I request a rate near 8000, the minimum is returned as 8000 and so forth. To recap, the 'snd_pcm_set_rate_near' call ALWAYS succeeds without altering the rate passed to it the first time. The only way to find out if the rate is actually supported is to call 'snd_pcm_hw_params' and then see if that fails with err == -22 (invalid argument). Furthermore, once the failure occurs, another rate value cannot be tried unless the PCM device is closed and the entire configuration process is repeated. After the failure, subsequent calls to 'snd_pcm_set_rate_near' with ANY different value always get set back to the original bad value. Example, I request 44100 and it fails. So I try 8000 and it returns 44100 (and of course fails again if I try 'snd_pcm_hw_params'). I am not sure yet, but is the value at 'interval_inline.h:74 return i->min;' determined directly by the hardware? It seems like it is, but I don't know the code well enough yet to make that assertion. I also haven't yet examined the 'snd_interval_max' command, but my gut tells me it is behaving the same way, since after a bad rate is set, any higher or lower rate setting attempts ALWAYS come back as the original rate. How would i determine if it is hardware or maybe something bad with the hardware.software interface? I checked the PCM state value and it was correct the entire time. Here are short snippets of two gdb sessions where I found the issue and below that is test program output showing the state values as I try to configure two different rates: Code snippet in ALSA lib file interval_inline.h: 71 INTERVAL_INLINE int snd_interval_min(const snd_interval_t *i) 72 { 73 assert(!snd_interval_empty(i)); 74 return i->min; 75 } Run #1 with 44100 'snd_pcm_set_rate_near' call: (gdb) s 74 return i->min; (gdb) print i->min $6 = 44100 Run #2 with 8000 'snd_pcm_set_rate_near' call: (gdb) s 74 return i->min; (gdb) print i->min $8 = 8000 Here is a log of the high level test program: snd_pcm_hw_params_set_rate_near succeeded. Rate set to 44100. snd_pcm_hw_params_set_channels succeeded. cannot set buffer time (Invalid argument) cannot set buffer period time (Invalid argument) Before commit, state is: SND_PCM_STATE_OPEN cannot set parameters (Invalid argument) After failure, state is: SND_PCM_STATE_OPEN Trying to re-configure rate param. Before calling 'snd_pcm_drop', state is: SND_PCM_STATE_OPEN cannot drop (Input/output error) After calling 'snd_pcm_drop', state is: SND_PCM_STATE_OPEN Trying rate 8000. snd_pcm_hw_params_set_rate_near succeeded. Rate set to 44100. snd_pcm_hw_params_set_buffer_time_near succeeded. Buffer size actually set to 22050. cannot set buffer period time (Invalid argument) cannot set parameters (Invalid argument) cannot prepare audio interface for use (Input/output error) Rate 44100 supported: FAIL Please let me know your thoughts on this and if there is a good way to prove if this is a hardware error or a software error. I am running 'Linux version 3.4.0-1487-omap4' and using the standard repositories with apt-get to install ALSA. I used 'apt-get source' to obtain 'alsa-lib-1.0.25' source files and I am running 'libasound2-dbg' to enable GDB debugging. Regards, Brent

6 17

[alsa-devel] [PATCH 00/15] ARM build regressions in v3.8
by Arnd Bergmann 06 Feb '13

06 Feb '13

I know this comes late, but we have a number of broken configurations in ARM in v3.8 that were still building in v3.7, and I'd like to get them all fixed in the final 3.8 release. It would be nice if the respective maintainers could have a look at these patches and apply them directly when they are happy with them. The first patch in the series is strictly speaking not a build error but just a warning, but it is a particularly annoying one that came in through the latest binutils release rather than a kernel change. The same binutils update also broke the samsung and w90x900 platforms. A few of the other changes are the result of the imx multiplatform conversion. I'm not really fixing those here, just picking up the pieces. It would be much nicer if we could actually get those drivers to work again with CONFIG_MULTIPLATFORM enabled rather than just disabling them, but it may be much too late for that. At least the drivers don't seem to be too essential, as they are only built in allyesconfig but not in any of the defconfigs. Arnd Arnd Bergmann (15): ARM: compressed/head.S: work around new binutils warning ARM: mvebu: build coherency_ll.S for arch=armv7-a ARM: samsung: fix assembly syntax for new gas ARM: w90x900: fix legacy assembly syntax ASoC: fsl: fiq and dma cannot both be modules clk: export __clk_get_name drm/exynos: don't include plat/gpio-cfg.h drm/exynos: fimd and ipp are broken on multiplatform media: coda: don't build on multiplatform mfd/vexpress: export vexpress_config_func_{put,get} mtd: davinci_nand: fix OF support USB: gadget/freescale: disable non-multiplatform drivers USB: ehci: make orion and mxc bus glues coexist samples/seccomp: be less stupid about cross compiling staging/omapdrm: don't build on multiplatform arch/arm/boot/compressed/Makefile | 2 +- arch/arm/boot/compressed/head.S | 12 ++++++++++++ arch/arm/mach-mvebu/coherency_ll.S | 1 + arch/arm/mach-s3c24xx/include/mach/debug-macro.S | 12 ++++++------ arch/arm/mach-s3c24xx/include/mach/entry-macro.S | 4 ++-- arch/arm/mach-s3c24xx/pm-h1940.S | 2 +- arch/arm/mach-s3c24xx/sleep-s3c2410.S | 12 ++++++------ arch/arm/mach-s3c24xx/sleep-s3c2412.S | 12 ++++++------ arch/arm/mach-w90x900/include/mach/entry-macro.S | 4 ++-- arch/arm/plat-samsung/include/plat/debug-macro.S | 18 +++++++++--------- drivers/clk/clk.c | 1 + drivers/gpu/drm/exynos/Kconfig | 4 ++-- drivers/gpu/drm/exynos/exynos_hdmi.c | 1 - drivers/media/platform/Kconfig | 2 +- drivers/mfd/vexpress-config.c | 3 ++- drivers/mtd/nand/davinci_nand.c | 2 +- drivers/staging/omapdrm/Kconfig | 2 +- drivers/usb/gadget/Kconfig | 3 ++- drivers/usb/host/ehci-hcd.c | 16 +++++++++++++++- samples/seccomp/Makefile | 2 ++ sound/soc/fsl/Kconfig | 3 +++ 21 files changed, 76 insertions(+), 42 deletions(-) -- 1.7.10.4 Cc: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: David Airlie <airlied(a)linux.ie> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: James Morris <james.l.morris(a)oracle.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> Cc: Mike Turquette <mturquette(a)linaro.org> Cc: Rob Clark <rob(a)ti.com> Cc: Russell King <rmk+kernel(a)arm.linux.org.uk> Cc: Shawn Guo <shawn.guo(a)linaro.org> Cc: alsa-devel(a)alsa-project.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-media(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org

3 8

[alsa-devel] [PATCH 001/002] ASoC: Replace max98090 Device Driver
by Jerry Wong 05 Feb '13

05 Feb '13

From: Jerry Wong <jerry.wong(a)maximintegrated.com> This patch removes the existing max98090 driver prior to installing a more complete one. --- The original author is okay to replace his recently submitted driver with this one. "... Would you mind if we began this now by replacing the existing driver with the one attached? ..." - Jerry "... But of course I agree to replace it to your code. Please send it to upstream... " - Morimoto-san sound/soc/codecs/Kconfig | 4 - sound/soc/codecs/Makefile | 2 - sound/soc/codecs/max98090.c | 577 ------------------------------------------- 3 files changed, 0 insertions(+), 583 deletions(-) delete mode 100644 sound/soc/codecs/max98090.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a84782..0e368d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,7 +44,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C - select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -268,9 +267,6 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate -config SND_SOC_MAX98090 - tristate - config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36..aa56312 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,7 +34,6 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o -snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -158,7 +157,6 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o -obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c deleted file mode 100644 index c9772ca..0000000 --- a/sound/soc/codecs/max98090.c +++ /dev/null @@ -1,577 +0,0 @@ -/* - * max98090.c -- MAX98090 ALSA SoC Audio driver - * based on Rev0p8 datasheet - * - * Copyright (C) 2012 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * Based on - * - * max98095.c - * Copyright 2011 Maxim Integrated Products - * - * https://github.com/hardkernel/linux/commit/\ - * 3417d7166b17113b3b33b0a337c74d1c7cc313df#sound/soc/codecs/max98090.c - * Copyright 2011 Maxim Integrated Products - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/i2c.h> -#include <linux/module.h> -#include <linux/regmap.h> -#include <sound/soc.h> -#include <sound/tlv.h> - -/* - * - * MAX98090 Registers Definition - * - */ - -/* RESET / STATUS / INTERRUPT REGISTERS */ -#define MAX98090_0x00_SW_RESET 0x00 -#define MAX98090_0x01_INT_STS 0x01 -#define MAX98090_0x02_JACK_STS 0x02 -#define MAX98090_0x03_INT_MASK 0x03 - -/* QUICK SETUP REGISTERS */ -#define MAX98090_0x04_SYS_CLK 0x04 -#define MAX98090_0x05_SAMPLE_RATE 0x05 -#define MAX98090_0x06_DAI_IF 0x06 -#define MAX98090_0x07_DAC_PATH 0x07 -#define MAX98090_0x08_MIC_TO_ADC 0x08 -#define MAX98090_0x09_LINE_TO_ADC 0x09 -#define MAX98090_0x0A_ANALOG_MIC_LOOP 0x0A -#define MAX98090_0x0B_ANALOG_LINE_LOOP 0x0B - -/* ANALOG INPUT CONFIGURATION REGISTERS */ -#define MAX98090_0x0D_INPUT_CONFIG 0x0D -#define MAX98090_0x0E_LINE_IN_LVL 0x0E -#define MAX98090_0x0F_LINI_IN_CFG 0x0F -#define MAX98090_0x10_MIC1_IN_LVL 0x10 -#define MAX98090_0x11_MIC2_IN_LVL 0x11 - -/* MICROPHONE CONFIGURATION REGISTERS */ -#define MAX98090_0x12_MIC_BIAS_VOL 0x12 -#define MAX98090_0x13_DIGITAL_MIC_CFG 0x13 -#define MAX98090_0x14_DIGITAL_MIC_MODE 0x14 - -/* ADC PATH AND CONFIGURATION REGISTERS */ -#define MAX98090_0x15_L_ADC_MIX 0x15 -#define MAX98090_0x16_R_ADC_MIX 0x16 -#define MAX98090_0x17_L_ADC_LVL 0x17 -#define MAX98090_0x18_R_ADC_LVL 0x18 -#define MAX98090_0x19_ADC_BIQUAD_LVL 0x19 -#define MAX98090_0x1A_ADC_SIDETONE 0x1A - -/* CLOCK CONFIGURATION REGISTERS */ -#define MAX98090_0x1B_SYS_CLK 0x1B -#define MAX98090_0x1C_CLK_MODE 0x1C -#define MAX98090_0x1D_ANY_CLK1 0x1D -#define MAX98090_0x1E_ANY_CLK2 0x1E -#define MAX98090_0x1F_ANY_CLK3 0x1F -#define MAX98090_0x20_ANY_CLK4 0x20 -#define MAX98090_0x21_MASTER_MODE 0x21 - -/* INTERFACE CONTROL REGISTERS */ -#define MAX98090_0x22_DAI_IF_FMT 0x22 -#define MAX98090_0x23_DAI_TDM_FMT1 0x23 -#define MAX98090_0x24_DAI_TDM_FMT2 0x24 -#define MAX98090_0x25_DAI_IO_CFG 0x25 -#define MAX98090_0x26_FILTER_CFG 0x26 -#define MAX98090_0x27_DAI_PLAYBACK_LVL 0x27 -#define MAX98090_0x28_EQ_PLAYBACK_LVL 0x28 - -/* HEADPHONE CONTROL REGISTERS */ -#define MAX98090_0x29_L_HP_MIX 0x29 -#define MAX98090_0x2A_R_HP_MIX 0x2A -#define MAX98090_0x2B_HP_CTR 0x2B -#define MAX98090_0x2C_L_HP_VOL 0x2C -#define MAX98090_0x2D_R_HP_VOL 0x2D - -/* SPEAKER CONFIGURATION REGISTERS */ -#define MAX98090_0x2E_L_SPK_MIX 0x2E -#define MAX98090_0x2F_R_SPK_MIX 0x2F -#define MAX98090_0x30_SPK_CTR 0x30 -#define MAX98090_0x31_L_SPK_VOL 0x31 -#define MAX98090_0x32_R_SPK_VOL 0x32 - -/* ALC CONFIGURATION REGISTERS */ -#define MAX98090_0x33_ALC_TIMING 0x33 -#define MAX98090_0x34_ALC_COMPRESSOR 0x34 -#define MAX98090_0x35_ALC_EXPANDER 0x35 -#define MAX98090_0x36_ALC_GAIN 0x36 - -/* RECEIVER AND LINE_OUTPUT REGISTERS */ -#define MAX98090_0x37_RCV_LOUT_L_MIX 0x37 -#define MAX98090_0x38_RCV_LOUT_L_CNTL 0x38 -#define MAX98090_0x39_RCV_LOUT_L_VOL 0x39 -#define MAX98090_0x3A_LOUT_R_MIX 0x3A -#define MAX98090_0x3B_LOUT_R_CNTL 0x3B -#define MAX98090_0x3C_LOUT_R_VOL 0x3C - -/* JACK DETECT AND ENABLE REGISTERS */ -#define MAX98090_0x3D_JACK_DETECT 0x3D -#define MAX98090_0x3E_IN_ENABLE 0x3E -#define MAX98090_0x3F_OUT_ENABLE 0x3F -#define MAX98090_0x40_LVL_CTR 0x40 -#define MAX98090_0x41_DSP_FILTER_ENABLE 0x41 - -/* BIAS AND POWER MODE CONFIGURATION REGISTERS */ -#define MAX98090_0x42_BIAS_CTR 0x42 -#define MAX98090_0x43_DAC_CTR 0x43 -#define MAX98090_0x44_ADC_CTR 0x44 -#define MAX98090_0x45_DEV_SHUTDOWN 0x45 - -/* REVISION ID REGISTER */ -#define MAX98090_0xFF_REV_ID 0xFF - -#define MAX98090_REG_MAX_CACHED 0x45 -#define MAX98090_REG_END 0xFF - -/* - * - * MAX98090 Registers Bit Fields - * - */ - -/* MAX98090_0x06_DAI_IF */ -#define MAX98090_DAI_IF_MASK 0x3F -#define MAX98090_RJ_M (1 << 5) -#define MAX98090_RJ_S (1 << 4) -#define MAX98090_LJ_M (1 << 3) -#define MAX98090_LJ_S (1 << 2) -#define MAX98090_I2S_M (1 << 1) -#define MAX98090_I2S_S (1 << 0) - -/* MAX98090_0x45_DEV_SHUTDOWN */ -#define MAX98090_SHDNRUN (1 << 7) - -/* codec private data */ -struct max98090_priv { - struct regmap *regmap; -}; - -static const struct reg_default max98090_reg_defaults[] = { - /* RESET / STATUS / INTERRUPT REGISTERS */ - {MAX98090_0x00_SW_RESET, 0x00}, - {MAX98090_0x01_INT_STS, 0x00}, - {MAX98090_0x02_JACK_STS, 0x00}, - {MAX98090_0x03_INT_MASK, 0x04}, - - /* QUICK SETUP REGISTERS */ - {MAX98090_0x04_SYS_CLK, 0x00}, - {MAX98090_0x05_SAMPLE_RATE, 0x00}, - {MAX98090_0x06_DAI_IF, 0x00}, - {MAX98090_0x07_DAC_PATH, 0x00}, - {MAX98090_0x08_MIC_TO_ADC, 0x00}, - {MAX98090_0x09_LINE_TO_ADC, 0x00}, - {MAX98090_0x0A_ANALOG_MIC_LOOP, 0x00}, - {MAX98090_0x0B_ANALOG_LINE_LOOP, 0x00}, - - /* ANALOG INPUT CONFIGURATION REGISTERS */ - {MAX98090_0x0D_INPUT_CONFIG, 0x00}, - {MAX98090_0x0E_LINE_IN_LVL, 0x1B}, - {MAX98090_0x0F_LINI_IN_CFG, 0x00}, - {MAX98090_0x10_MIC1_IN_LVL, 0x11}, - {MAX98090_0x11_MIC2_IN_LVL, 0x11}, - - /* MICROPHONE CONFIGURATION REGISTERS */ - {MAX98090_0x12_MIC_BIAS_VOL, 0x00}, - {MAX98090_0x13_DIGITAL_MIC_CFG, 0x00}, - {MAX98090_0x14_DIGITAL_MIC_MODE, 0x00}, - - /* ADC PATH AND CONFIGURATION REGISTERS */ - {MAX98090_0x15_L_ADC_MIX, 0x00}, - {MAX98090_0x16_R_ADC_MIX, 0x00}, - {MAX98090_0x17_L_ADC_LVL, 0x03}, - {MAX98090_0x18_R_ADC_LVL, 0x03}, - {MAX98090_0x19_ADC_BIQUAD_LVL, 0x00}, - {MAX98090_0x1A_ADC_SIDETONE, 0x00}, - - /* CLOCK CONFIGURATION REGISTERS */ - {MAX98090_0x1B_SYS_CLK, 0x00}, - {MAX98090_0x1C_CLK_MODE, 0x00}, - {MAX98090_0x1D_ANY_CLK1, 0x00}, - {MAX98090_0x1E_ANY_CLK2, 0x00}, - {MAX98090_0x1F_ANY_CLK3, 0x00}, - {MAX98090_0x20_ANY_CLK4, 0x00}, - {MAX98090_0x21_MASTER_MODE, 0x00}, - - /* INTERFACE CONTROL REGISTERS */ - {MAX98090_0x22_DAI_IF_FMT, 0x00}, - {MAX98090_0x23_DAI_TDM_FMT1, 0x00}, - {MAX98090_0x24_DAI_TDM_FMT2, 0x00}, - {MAX98090_0x25_DAI_IO_CFG, 0x00}, - {MAX98090_0x26_FILTER_CFG, 0x80}, - {MAX98090_0x27_DAI_PLAYBACK_LVL, 0x00}, - {MAX98090_0x28_EQ_PLAYBACK_LVL, 0x00}, - - /* HEADPHONE CONTROL REGISTERS */ - {MAX98090_0x29_L_HP_MIX, 0x00}, - {MAX98090_0x2A_R_HP_MIX, 0x00}, - {MAX98090_0x2B_HP_CTR, 0x00}, - {MAX98090_0x2C_L_HP_VOL, 0x1A}, - {MAX98090_0x2D_R_HP_VOL, 0x1A}, - - /* SPEAKER CONFIGURATION REGISTERS */ - {MAX98090_0x2E_L_SPK_MIX, 0x00}, - {MAX98090_0x2F_R_SPK_MIX, 0x00}, - {MAX98090_0x30_SPK_CTR, 0x00}, - {MAX98090_0x31_L_SPK_VOL, 0x2C}, - {MAX98090_0x32_R_SPK_VOL, 0x2C}, - - /* ALC CONFIGURATION REGISTERS */ - {MAX98090_0x33_ALC_TIMING, 0x00}, - {MAX98090_0x34_ALC_COMPRESSOR, 0x00}, - {MAX98090_0x35_ALC_EXPANDER, 0x00}, - {MAX98090_0x36_ALC_GAIN, 0x00}, - - /* RECEIVER AND LINE_OUTPUT REGISTERS */ - {MAX98090_0x37_RCV_LOUT_L_MIX, 0x00}, - {MAX98090_0x38_RCV_LOUT_L_CNTL, 0x00}, - {MAX98090_0x39_RCV_LOUT_L_VOL, 0x15}, - {MAX98090_0x3A_LOUT_R_MIX, 0x00}, - {MAX98090_0x3B_LOUT_R_CNTL, 0x00}, - {MAX98090_0x3C_LOUT_R_VOL, 0x15}, - - /* JACK DETECT AND ENABLE REGISTERS */ - {MAX98090_0x3D_JACK_DETECT, 0x00}, - {MAX98090_0x3E_IN_ENABLE, 0x00}, - {MAX98090_0x3F_OUT_ENABLE, 0x00}, - {MAX98090_0x40_LVL_CTR, 0x00}, - {MAX98090_0x41_DSP_FILTER_ENABLE, 0x00}, - - /* BIAS AND POWER MODE CONFIGURATION REGISTERS */ - {MAX98090_0x42_BIAS_CTR, 0x00}, - {MAX98090_0x43_DAC_CTR, 0x00}, - {MAX98090_0x44_ADC_CTR, 0x06}, - {MAX98090_0x45_DEV_SHUTDOWN, 0x00}, -}; - -static const unsigned int max98090_hp_tlv[] = { - TLV_DB_RANGE_HEAD(5), - 0x0, 0x6, TLV_DB_SCALE_ITEM(-6700, 400, 0), - 0x7, 0xE, TLV_DB_SCALE_ITEM(-4000, 300, 0), - 0xF, 0x15, TLV_DB_SCALE_ITEM(-1700, 200, 0), - 0x16, 0x1B, TLV_DB_SCALE_ITEM(-400, 100, 0), - 0x1C, 0x1F, TLV_DB_SCALE_ITEM(150, 50, 0), -}; - -static struct snd_kcontrol_new max98090_snd_controls[] = { - SOC_DOUBLE_R_TLV("Headphone Volume", MAX98090_0x2C_L_HP_VOL, - MAX98090_0x2D_R_HP_VOL, 0, 31, 0, max98090_hp_tlv), -}; - -/* Left HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x29_L_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x29_L_HP_MIX, 0, 1, 0), -}; - -/* Right HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x2A_R_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x2A_R_HP_MIX, 0, 1, 0), -}; - -static struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - /* Output */ - SND_SOC_DAPM_OUTPUT("HPL"), - SND_SOC_DAPM_OUTPUT("HPR"), - - /* PGA */ - SND_SOC_DAPM_PGA("HPL Out", MAX98090_0x3F_OUT_ENABLE, 7, 0, NULL, 0), - SND_SOC_DAPM_PGA("HPR Out", MAX98090_0x3F_OUT_ENABLE, 6, 0, NULL, 0), - - /* Mixer */ - SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, - max98090_left_hp_mixer_controls, - ARRAY_SIZE(max98090_left_hp_mixer_controls)), - - SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, - max98090_right_hp_mixer_controls, - ARRAY_SIZE(max98090_right_hp_mixer_controls)), - - /* DAC */ - SND_SOC_DAPM_DAC("DACL", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 0, 0), - SND_SOC_DAPM_DAC("DACR", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 1, 0), -}; - -static struct snd_soc_dapm_route max98090_audio_map[] = { - /* Output */ - {"HPL", NULL, "HPL Out"}, - {"HPR", NULL, "HPR Out"}, - - /* PGA */ - {"HPL Out", NULL, "HPL Mixer"}, - {"HPR Out", NULL, "HPR Mixer"}, - - /* Mixer*/ - {"HPL Mixer", "DACR Switch", "DACR"}, - {"HPL Mixer", "DACL Switch", "DACL"}, - - {"HPR Mixer", "DACR Switch", "DACR"}, - {"HPR Mixer", "DACL Switch", "DACL"}, -}; - -static bool max98090_volatile(struct device *dev, unsigned int reg) -{ - if ((reg == MAX98090_0x01_INT_STS) || - (reg == MAX98090_0x02_JACK_STS) || - (reg > MAX98090_REG_MAX_CACHED)) - return true; - - return false; -} - -static int max98090_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; - unsigned int val; - - switch (params_rate(params)) { - case 96000: - val = 1 << 5; - break; - case 32000: - val = 1 << 4; - break; - case 48000: - val = 1 << 3; - break; - case 44100: - val = 1 << 2; - break; - case 16000: - val = 1 << 1; - break; - case 8000: - val = 1 << 0; - break; - default: - dev_err(codec->dev, "unsupported rate\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x05_SAMPLE_RATE, 0x03F, val); - - return 0; -} - -static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned int val; - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, 0); - - switch (freq) { - case 26000000: - val = 1 << 7; - break; - case 19200000: - val = 1 << 6; - break; - case 13000000: - val = 1 << 5; - break; - case 12288000: - val = 1 << 4; - break; - case 12000000: - val = 1 << 3; - break; - case 11289600: - val = 1 << 2; - break; - default: - dev_err(codec->dev, "Invalid master clock frequency\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x04_SYS_CLK, 0xFD, val); - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - dev_dbg(dai->dev, "sysclk is %uHz\n", freq); - - return 0; -} - -static int max98090_dai_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct snd_soc_codec *codec = dai->codec; - int is_master; - u8 val; - - /* master/slave mode */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - is_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - is_master = 0; - break; - default: - dev_err(codec->dev, "unsupported clock\n"); - return -EINVAL; - } - - /* format */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_I2S: - val = (is_master) ? MAX98090_I2S_M : MAX98090_I2S_S; - break; - case SND_SOC_DAIFMT_RIGHT_J: - val = (is_master) ? MAX98090_RJ_M : MAX98090_RJ_S; - break; - case SND_SOC_DAIFMT_LEFT_J: - val = (is_master) ? MAX98090_LJ_M : MAX98090_LJ_S; - break; - default: - dev_err(codec->dev, "unsupported format\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x06_DAI_IF, - MAX98090_DAI_IF_MASK, val); - - return 0; -} - -#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 -#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) - -static struct snd_soc_dai_ops max98090_dai_ops = { - .set_sysclk = max98090_dai_set_sysclk, - .set_fmt = max98090_dai_set_fmt, - .hw_params = max98090_dai_hw_params, -}; - -static struct snd_soc_dai_driver max98090_dai = { - .name = "max98090-Hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = MAX98090_RATES, - .formats = MAX98090_FORMATS, - }, - .ops = &max98090_dai_ops, -}; - -static int max98090_probe(struct snd_soc_codec *codec) -{ - struct max98090_priv *priv = snd_soc_codec_get_drvdata(codec); - struct device *dev = codec->dev; - int ret; - - codec->control_data = priv->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - /* Device active */ - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - return 0; -} - -static int max98090_remove(struct snd_soc_codec *codec) -{ - return 0; -} - -static struct snd_soc_codec_driver soc_codec_dev_max98090 = { - .probe = max98090_probe, - .remove = max98090_remove, - .controls = max98090_snd_controls, - .num_controls = ARRAY_SIZE(max98090_snd_controls), - .dapm_widgets = max98090_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(max98090_dapm_widgets), - .dapm_routes = max98090_audio_map, - .num_dapm_routes = ARRAY_SIZE(max98090_audio_map), -}; - -static const struct regmap_config max98090_regmap = { - .reg_bits = 8, - .val_bits = 8, - .max_register = MAX98090_REG_END, - .volatile_reg = max98090_volatile, - .cache_type = REGCACHE_RBTREE, - .reg_defaults = max98090_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(max98090_reg_defaults), -}; - -static int max98090_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct max98090_priv *priv; - struct device *dev = &i2c->dev; - unsigned int val; - int ret; - - priv = devm_kzalloc(dev, sizeof(struct max98090_priv), - GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); - if (IS_ERR(priv->regmap)) { - ret = PTR_ERR(priv->regmap); - dev_err(dev, "Failed to init regmap: %d\n", ret); - return ret; - } - - i2c_set_clientdata(i2c, priv); - - ret = regmap_read(priv->regmap, MAX98090_0xFF_REV_ID, &val); - if (ret < 0) { - dev_err(dev, "Failed to read device revision: %d\n", ret); - return ret; - } - dev_info(dev, "revision 0x%02x\n", val); - - ret = snd_soc_register_codec(dev, - &soc_codec_dev_max98090, - &max98090_dai, 1); - - return ret; -} - -static int max98090_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id max98090_i2c_id[] = { - { "max98090", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); - -static struct i2c_driver max98090_i2c_driver = { - .driver = { - .name = "max98090", - .owner = THIS_MODULE, - }, - .probe = max98090_i2c_probe, - .remove = max98090_i2c_remove, - .id_table = max98090_i2c_id, -}; -module_i2c_driver(max98090_i2c_driver); - -MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); -MODULE_AUTHOR("Peter Hsiang, Kuninori Morimoto"); -MODULE_LICENSE("GPL"); -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: soc-compress: Add support for not memory mapped DSPs
by Charles Keepax 05 Feb '13

05 Feb '13

The ASoC compressed API did not implement the copy callback in its compressed ops which is required for DSPs that are not memory mapped. This patch adds a second set of compressed ops which does implement the copy callback and uses that when copy is defined in the platform compressed ops, ie. when the DSP is not memory mapped. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 36 +++++++++++++++++++++++++++++++++++- 1 files changed, 35 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3ea7956..f9fb112 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -302,6 +302,22 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, return 0; } +static int soc_compr_copy(struct snd_compr_stream *cstream, + const char __user *buf, size_t count) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + ret = platform->driver->compr_ops->copy(cstream, buf, count); + + mutex_unlock(&rtd->pcm_mutex); + return ret; +} + /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, @@ -315,10 +331,25 @@ static struct snd_compr_ops soc_compr_ops = { .get_codec_caps = soc_compr_get_codec_caps }; +/* ASoC Compress operations for non memory mapped DSPs */ +static struct snd_compr_ops soc_compr_nomap_ops = { + .open = soc_compr_open, + .free = soc_compr_free, + .set_params = soc_compr_set_params, + .get_params = soc_compr_get_params, + .trigger = soc_compr_trigger, + .pointer = soc_compr_pointer, + .copy = soc_compr_copy, + .ack = soc_compr_ack, + .get_caps = soc_compr_get_caps, + .get_codec_caps = soc_compr_get_codec_caps +}; + /* create a new compress */ int soc_new_compress(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_compr *compr; @@ -335,7 +366,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -ENOMEM; } - compr->ops = &soc_compr_ops; + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + compr->ops = &soc_compr_nomap_ops; + else + compr->ops = &soc_compr_ops; mutex_init(&compr->lock); ret = snd_compress_new(rtd->card->snd_card, num, direction, compr); if (ret < 0) { -- 1.7.2.5

3 11

[alsa-devel] [PATCH 0/4] ASoC: atmel: fixes for working as module
by Bo Shen 04 Feb '13

04 Feb '13

The following patches fixes the audio part when working as module Bo Shen (4): ASoC: atmel_pcm: make it buildable as module ASoC: atmel_ssc_dai: remove error set private data ASoC: atmel_ssc_dai: correct sequence when unload ASoC: sam9g20_wm8731: disable clock and correct sequence when unload sound/soc/atmel/atmel-pcm.h | 6 ++++-- sound/soc/atmel/atmel_ssc_dai.c | 12 +----------- sound/soc/atmel/sam9g20_wm8731.c | 6 +++--- 3 files changed, 8 insertions(+), 16 deletions(-) -- 1.7.9.5

2 5

[alsa-devel] [RESEND PATCH v3 0/5] ARM: OMAP4+: HDMI: Update platform devices for audio
by Ricardo Neri 04 Feb '13

04 Feb '13

Hi Mark, Tomi, Liam, Tony, I am resending this series as Mark and possibly other people missed it. This set aims to be the version 3 of my previous submission[1] and aims to address the comments that Mark and Tomi kindly provided on such submission. The creation of the platform device for the HDMI audio interface from within the OMAPDSS HDMI driver that was previously submitted[2] is resubmitted to be complemented with code to relocate to arch/arm/mach-omap2/display.c the creation of the platform devices for the HDMI ASoC codec and card drivers. This series does not break the HDMI audio functionality in any patch. Also, the names of the platform devices are changed to give them more logical and more descriptive names. As the commit commit 14840b9a83c6a56629db2ba0ec247503e975f143 Author: Ricardo Neri <ricardo.neri(a)ti.com> Date: Tue Nov 6 00:19:17 2012 -0600 OMAPDSS: HDMI: Create platform device for audio support is reverted in Tomi's git://gitorious.org/linux-omap-dss2/linux.git master branch, this series applies cleanly. Changes from v1: *Put in a single series all the patches related to platform device updates. *Now HDMI audio works correctly in every patch. *Remove reference to the TPD12S015 HDMI companion chip as the ASoC drivers are not aware of this and other chips could be used in the future. Changes from v2: *Split in two patches the renaming and the relocation of the platform devices. *Create a separate patch to pass only the address offset of the DMA data port to the audio interface platform device. *Keep the name hdmi-audio-codec to not refer to explicitly to OMAP. The codec can be made generic in a different patch series submitted to alsa-devel. BR, Ricardo [1]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg80785.html [2]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg79795.html Ricardo Neri (5): ARM: OMAP4+: HDMI: Rename platform devices for ASoC drivers ARM: OMAP4: Assign IDs to DSS HDMI devices ARM4: OMAP4+: HDMI: Relocate devices for audio codec and card ARM: OMAP4+: HDMI: Relocate the device for audio interface ARM: OMAP4+: HDMI: Refine the DMA port resource for audio arch/arm/mach-omap2/board-4430sdp.c | 9 ++--- arch/arm/mach-omap2/board-omap4panda.c | 9 ++--- arch/arm/mach-omap2/devices.c | 31 ---------------- arch/arm/mach-omap2/display.c | 31 ++++++++++++++++ drivers/video/omap2/dss/hdmi.c | 62 ++++++++++++++++++++++++++++++++ sound/soc/omap/omap-hdmi-card.c | 4 +-- sound/soc/omap/omap-hdmi.c | 5 ++- sound/soc/omap/omap-hdmi.h | 2 -- 8 files changed, 103 insertions(+), 50 deletions(-) -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: core: don't return uninitialized snd_compr_tstamp
by Mark Brown 04 Feb '13

04 Feb '13

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> The snd_compr_update_tstamp() can only fill in the snd_compr_tstamp if the codec implements the pointer() function. If that happened the code was previously returning uninitialized garbage in the tstamp because it wasn't initialized anywhere. This change zero-fills the tstamp in the two places it is used before calling snd_compr_update_tstamp(), and also has snd_compr_update_tstamp() return an error indication if it can't provide a tstamp. For the case of snd_compr_calc_avail() it ignores this error because we still need to return info on the available buffer space even if we can't provide tstamp info - when the tstamp is not valid all fields are now guaranteed to be zero. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..2d62068 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -144,16 +144,17 @@ static int snd_compr_free(struct inode *inode, struct file *f) return 0; } -static void snd_compr_update_tstamp(struct snd_compr_stream *stream, +static int snd_compr_update_tstamp(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp) { if (!stream->ops->pointer) - return; + return -ENOTSUPP; stream->ops->pointer(stream, tstamp); pr_debug("dsp consumed till %d total %d bytes\n", tstamp->byte_offset, tstamp->copied_total); stream->runtime->hw_pointer = tstamp->byte_offset; stream->runtime->total_bytes_transferred = tstamp->copied_total; + return 0; } static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, @@ -161,7 +162,9 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, { long avail_calc; /*this needs to be signed variable */ + memset(avail, 0, sizeof(*avail)); snd_compr_update_tstamp(stream, &avail->tstamp); + /* Still need to return avail even if tstamp can't be filled in */ /* FIXME: This needs to be different for capture stream, available is # of compressed data, for playback it's @@ -517,11 +520,14 @@ out: static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { - struct snd_compr_tstamp tstamp; + struct snd_compr_tstamp tstamp = {0}; + int ret; - snd_compr_update_tstamp(stream, &tstamp); - return copy_to_user((struct snd_compr_tstamp __user *)arg, - &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + ret = snd_compr_update_tstamp(stream, &tstamp); + if (ret == 0) + ret = copy_to_user((struct snd_compr_tstamp __user *)arg, + &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + return ret; } static int snd_compr_pause(struct snd_compr_stream *stream) -- 1.7.10.4

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[alsa-devel] [PATCH] ASOC: imx: audmux: add a integrated configuration function
by Gary Zhang 04 Feb '13

04 Feb '13

based on imx_audmux_v2_configure_port(), add a integrated interface to configure audmux port conveniently Signed-off-by: Gary Zhang <b13634(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 43 ++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 42 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 251f4d9..ebd38a0 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -1,5 +1,5 @@ /* - * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012-2013 Freescale Semiconductor, Inc. * Copyright 2012 Linaro Ltd. * Copyright 2009 Pengutronix, Sascha Hauer <s.hauer(a)pengutronix.de> * @@ -244,6 +244,47 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); +int imx_audmux_configure_port(unsigned int int_port, unsigned int ext_port) +{ + int ret; + + if ((int_port > 7) || (ext_port > 7)) + return -EINVAL; + + if (audmux_type != IMX31_AUDMUX) + return -EINVAL; + + if (!audmux_base) + return -ENOSYS; + + if (audmux_clk) + clk_prepare_enable(audmux_clk); + + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + pr_err("audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + pr_err("audmux external port setup failed\n"); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(imx_audmux_configure_port); + static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; -- 1.7.0.4

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