July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH V2 0/2] ALSA: ASoC for Au1000/1500/1100
by Manuel Lauss 22 Jul '11

22 Jul '11

Hello, These 2 patches implement ASoC drivers for the AC97 and I2S controllers found on early Alchemy chips. They are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (2): ALSA: Alchemy AC97C/I2SC audio support ALSA: delete MIPS au1x00 driver arch/mips/alchemy/devboards/db1x00/platform.c | 37 ++ arch/mips/include/asm/mach-au1x00/au1000.h | 61 --- sound/mips/Kconfig | 8 - sound/mips/Makefile | 2 - sound/mips/au1x00.c | 695 ------------------------- sound/soc/au1x/Kconfig | 28 + sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 398 ++++++++++++++ sound/soc/au1x/db1000.c | 75 +++ sound/soc/au1x/dma.c | 470 +++++++++++++++++ sound/soc/au1x/i2sc.c | 353 +++++++++++++ sound/soc/au1x/psc.h | 31 +- 12 files changed, 1393 insertions(+), 775 deletions(-) delete mode 100644 sound/mips/au1x00.c create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 9

Re: [alsa-devel] asihpi - update from HPI 4.06 to 4.08 - Firmware
by Takashi Iwai 22 Jul '11

22 Jul '11

At Fri, 22 Jul 2011 16:05:43 +1200, Eliot Blennerhassett wrote: > > On 22/07/11 15:52, linux(a)audioscience.com wrote: > > From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> > > > > Main visible change is the addition of mute switch to volume controls. > > The rest are relatively minor changes and cleanups to HPI code. > > I also forgot to mention that the associated firmware/asihpi should be > updated with the contents of > > http://www.audioscience.com/internet/download/drivers/released/v4/08/02/asi… OK, now updated alsa-firmware tree, too. thanks, Takashi

1 0

[alsa-devel] [PATCH 00/15] ALSA: asihpi - update from HPI 4.06 to 4.08
by linux＠audioscience.com 22 Jul '11

22 Jul '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Main visible change is the addition of mute switch to volume controls. The rest are relatively minor changes and cleanups to HPI code. patch#6 reverts commit 4a122c10f "check adapter index in hpi_ioctl" The associated 4.08 firmware release adds support for new cards ASI5544/5541/5540/5520 Eliot Blennerhassett (15): ALSA: asihpi - Give more meaningful name to hpi request message type ALSA: asihpi - Increase request and response buffer sizes ALSA: asihpi - Remove controlex structs and associated special data transfer code ALSA: asihpi - DSP code loader API now independent of OS ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro ALSA: asihpi - Remove spurious adapter index check ALSA: asihpi - Remove unused structures, macros and functions ALSA: asihpi - Fix minor typos and spelling ALSA: asihpi - Make local function static ALSA: asihpi - Add new node and message defines ALSA: asihpi - Explicitly include mutex.h ALSA: asihpi - Use size_t for sizeof result ALSA: asihpi - Control name updates ALSA: asihpi - Add volume mute controls ALSA: asihpi - HPI version 4.08 sound/pci/asihpi/asihpi.c | 87 ++++++++++++++++++---- sound/pci/asihpi/hpi.h | 24 ++++-- sound/pci/asihpi/hpi6000.c | 11 +-- sound/pci/asihpi/hpi6205.c | 52 ++++--------- sound/pci/asihpi/hpi6205.h | 25 +++++-- sound/pci/asihpi/hpi_internal.h | 155 ++++++++++---------------------------- sound/pci/asihpi/hpicmn.c | 17 ++-- sound/pci/asihpi/hpidspcd.c | 137 ++++++++++++++-------------------- sound/pci/asihpi/hpidspcd.h | 72 ++++++++++-------- sound/pci/asihpi/hpifunc.c | 86 ++++++++++----------- sound/pci/asihpi/hpimsginit.c | 4 +- sound/pci/asihpi/hpimsgx.c | 6 +- sound/pci/asihpi/hpioctl.c | 10 +-- sound/pci/asihpi/hpios.c | 8 -- sound/pci/asihpi/hpios.h | 1 + 15 files changed, 319 insertions(+), 376 deletions(-)

3 34

[alsa-devel] Alsa Driver Compilation Problem Report.
by Georges Jahchan 21 Jul '11

21 Jul '11

Attached is the ALSA driver compile script log on Fedora Core 13 x86_64 on an HP Pavilion DV7-1223CA with AMD Turion X2 64-bit CPU, ATI HDMI Audio and Intel PCI Audio. uname -a Linux hostname 2.6.34.9-69.fc13.x86_64 #1 SMP Tue May 3 09:23:03 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux

1 0

[alsa-devel] [REPOST][PATCH v4 3/4] ASoC: SAMSUNG: Modify I2S driver to support idma
by Sangbeom Kim 21 Jul '11

21 Jul '11

Previously, I2S driver only can support system dma. In this patch, i2s driver can support internal dma too. IDMA h/w configuration is initialized on idma.c Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- As Mark brown request, I repost i2s v4 patch sound/soc/samsung/i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 1568eea..c086b78 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -21,6 +21,7 @@ #include <plat/audio.h> #include "dma.h" +#include "idma.h" #include "i2s.h" #include "i2s-regs.h" @@ -60,6 +61,7 @@ struct i2s_dai { /* DMA parameters */ struct s3c_dma_params dma_playback; struct s3c_dma_params dma_capture; + struct s3c_dma_params idma_playback; u32 quirks; u32 suspend_i2smod; u32 suspend_i2scon; @@ -877,6 +879,10 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (i2s->quirks & QUIRK_NEED_RSTCLR) writel(CON_RSTCLR, i2s->addr + I2SCON); + if (i2s->quirks & QUIRK_SEC_DAI) + idma_reg_addr_init((void *)i2s->addr, + i2s->sec_dai->idma_playback.dma_addr); + probe_exit: /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; @@ -1077,6 +1083,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) sec_dai->dma_playback.dma_size = 4; sec_dai->base = regs_base; sec_dai->quirks = quirks; + sec_dai->idma_playback.dma_addr = i2s_cfg->idma_addr; sec_dai->pri_dai = pri_dai; pri_dai->sec_dai = sec_dai; } -- 1.7.1

3 5

[alsa-devel] [PATCH] mfd: Acknowledge WM8994 IRQs before reporting
by Mark Brown 21 Jul '11

21 Jul '11

This ensures we never have a window where we've handled an interrupt but not told the hardware about it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm8994-irq.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/drivers/mfd/wm8994-irq.c b/drivers/mfd/wm8994-irq.c index 71c6e8f..d682f7b 100644 --- a/drivers/mfd/wm8994-irq.c +++ b/drivers/mfd/wm8994-irq.c @@ -231,12 +231,6 @@ static irqreturn_t wm8994_irq_thread(int irq, void *data) status[i] &= ~wm8994->irq_masks_cur[i]; } - /* Report */ - for (i = 0; i < ARRAY_SIZE(wm8994_irqs); i++) { - if (status[wm8994_irqs[i].reg - 1] & wm8994_irqs[i].mask) - handle_nested_irq(wm8994->irq_base + i); - } - /* Ack any unmasked IRQs */ for (i = 0; i < ARRAY_SIZE(status); i++) { if (status[i]) @@ -244,6 +238,12 @@ static irqreturn_t wm8994_irq_thread(int irq, void *data) status[i]); } + /* Report */ + for (i = 0; i < ARRAY_SIZE(wm8994_irqs); i++) { + if (status[wm8994_irqs[i].reg - 1] & wm8994_irqs[i].mask) + handle_nested_irq(wm8994->irq_base + i); + } + return IRQ_HANDLED; } -- 1.7.5.4

1 0

[alsa-devel] [PATCH 0/2] ALSA: ASoC for Au1000/1500/1100
by Manuel Lauss 21 Jul '11

21 Jul '11

Hello, These 2 patches implement ASoC drivers for the AC97 controller found on early Alchemy chips. They are largely based on the old mips/au1x00.c driver which they replace. Tested on a Db1500 development board. Manuel Lauss (2): ALSA: Alchemy AC97C audio support ALSA: delete MIPS au1x00 driver arch/mips/alchemy/devboards/db1x00/platform.c | 37 ++ sound/mips/Kconfig | 8 - sound/mips/Makefile | 2 - sound/mips/au1x00.c | 695 ------------------------- sound/soc/au1x/Kconfig | 25 + sound/soc/au1x/Makefile | 9 + sound/soc/au1x/ac97c.c | 398 ++++++++++++++ sound/soc/au1x/db1000.c | 75 +++ sound/soc/au1x/dma.c | 470 +++++++++++++++++ sound/soc/au1x/psc.h | 31 +- 10 files changed, 1036 insertions(+), 714 deletions(-) delete mode 100644 sound/mips/au1x00.c create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 3

[alsa-devel] [PATCH V2 0/4] Add Samsung internal dma driver support
by Sangbeom Kim 21 Jul '11

21 Jul '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. Original code is made by jassi brar on SMDKV210 After idma driver merge, I will submit low power audio support with idma driver and samsung dsp driver. Changes since v1: 1. Change idma_new by new pcm_new api 2. Add handling different idma address 3. As jassi comments, Modify the idma driver This patchset contains followings o To Jassi Brar, Mark Brown and Liam Girdwood, [PATCH V2 1/4] ASoC: SAMSUNG: Modify I2S driver to support idma [PATCH V2 2/4] ASoC: SAMSUNG: Add I2S0 internal dma driver o To Kukjin Kim [PATCH V2 3/4] ARM: SAMSUNG: Add platform device for idma [PATCH V2 4/4] ARM: SAMSUNG: Add IRQ_I2S0 definition Thanks SB Kim

4 8

Re: [alsa-devel] [PATCH REPOST] sound: codec: tlv320aic3x: remove mono controls
by Mark Brown 21 Jul '11

21 Jul '11

On Fri, Jul 15, 2011 at 05:40:54PM +0200, Jan Weitzel wrote: > 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. This is clearly not what your subject line says... > 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 */ > + This is really not great legibility and it's going to get terribly confusing if someone comes up with some other device that also needs custom init. A similar issue applied to some of the other stuff but this is the most obvious example. I'd suggest restructuring this to have all the mono-specific code in an if statement that postively selects for the feature. > 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)); This if ... else should really be a switch.

2 3

[alsa-devel] [PATCH] ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
by Liam Girdwood 20 Jul '11

20 Jul '11

In preparation for ASoC Dynamic PCM (AKA DSP) support. Provide convenience methods to retrieve the soc_card or snd_card from a DAPM context. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4205a97..aaef0ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,6 +124,36 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +/* get snd_card from DAPM context */ +static inline struct snd_card *dapm_get_snd_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card->snd_card; + else if (dapm->platform) + return dapm->platform->card->snd_card; + else + BUG(); + + /* unreachable */ + return NULL; +} + +/* get soc_card from DAPM context */ +static inline struct snd_soc_card *dapm_get_soc_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card; + else if (dapm->platform) + return dapm->platform->card; + else + BUG(); + + /* unreachable */ + return NULL; +} + static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) { if (w->codec) -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: dapm - add DAPM macro for external enum widgets
by Liam Girdwood 20 Jul '11

20 Jul '11

Add a convenience macro for external enumerated widgets. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 596387b..655acc5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -266,6 +266,12 @@ .get = snd_soc_dapm_get_enum_virt, \ .put = snd_soc_dapm_put_enum_virt, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = xget, \ + .put = xput, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_VALUE_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ -- 1.7.4.1

2 1

[alsa-devel] [PATCH v2 2/2] ASoC: davinci: add missing break statement
by Rajashekhara, Sudhakar 20 Jul '11

20 Jul '11

In davinci_vcif_trigger() function, a break() statement was missing causing the davinci_vcif_stop() function to be called as a fallback after calling davinci_vcif_start(). Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- sound/soc/davinci/davinci-vcif.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index c957e9e..1f11525 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -159,6 +159,7 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_vcif_start(substream); + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -- 1.7.1

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: davinci: fix codec start and stop functions
by Rajashekhara, Sudhakar 20 Jul '11

20 Jul '11

According to DM365 voice codec data sheet at [1], before starting recording or playback, ADC/DAC modules should follow a reset and enable cycle. Writing a 1 to the ADC/DAC bit in the register resets the module and clearing the bit to 0 will enable the module. But the driver seems to be doing the reverse of it. [1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- sound/soc/davinci/davinci-vcif.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9259f1f..c957e9e 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -62,9 +62,9 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream) w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -80,9 +80,9 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) /* Reset transmitter/receiver and sample rate/frame sync generators */ w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } -- 1.7.1

3 2

[alsa-devel] [PATCH v4 00/10] ARM: mxs: add audio support
by Dong Aisheng 20 Jul '11

20 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Code is written based on the branch imx-for-3.1 in sascha's tree http://git.pengutronix.de/git/imx/linux-2.6.git Tested on MX28EVK. For the ASoC-patches (1-4): Tested-by: Wolfram Sang <w.sang(a)pengutronix.de> For the mach-mxs-Patches (5-10): Reviewed-by: Wolfram Sang <w.sang(a)pengutronix.de> Changes since v3: This version only has one patch changed with another two for rebase the changed one. * [PATCH v4 06/10] ARM: mxs: add saif device Drop DIGCTRL-handling based on Wolfram's comment * [PATCH v4 07/10] ARM: mxs: add sgtl5000 i2c device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device * [PATCH v4 08/10] ARM: mxs: add mxs-sgtl5000 device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device changes since v2: * mxs-pcm - fix wrong periods_max. - For robustness replace the channels_min check with a check for the relevant substream being present * mxs-saif - Add IRQ_NONE handle - improve mxs_saif_set_clk to make it more easy to read - correct return value - request_irq sequence changed to after ioremap * mxs-sgtl5000 - clean up a few unused code * Kconfig&Makefile - SND_SOC_MXS_SGTL5000 should depend on I2C in Kconfig * mxs - Select MXS_I2C in kconfig for mxs-sgtl5000 changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 2 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 3 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 ++ arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 35 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 359 ++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 677 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++ sound/soc/mxs/mxs-sgtl5000.c | 165 ++++++ 18 files changed, 1530 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

4 15

[alsa-devel] [PATCH 1/2] ASoC: Acknowledge WM8996 interrupts before acting on them
by Mark Brown 20 Jul '11

20 Jul '11

This closes the small race between a status being read in response to an interrupt and clearing the interrupt, meaning that if the status changes between those periods we might not get a reassertion of the interrupt. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 9 ++++----- 1 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 00f9ace..ab8e9d1 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2404,6 +2404,8 @@ static irqreturn_t wm8996_irq(int irq, void *data) } irq_val &= ~snd_soc_read(codec, WM8996_INTERRUPT_STATUS_2_MASK); + snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); + if (irq_val & (WM8996_DCS_DONE_01_EINT | WM8996_DCS_DONE_23_EINT)) { dev_dbg(codec->dev, "DC servo IRQ\n"); complete(&wm8996->dcs_done); @@ -2420,13 +2422,10 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); - if (irq_val) { - snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); - + if (irq_val) return IRQ_HANDLED; - } else { + else return IRQ_NONE; - } } static irqreturn_t wm8996_edge_irq(int irq, void *data) -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: davinci: bug fixes for DM365 voice codec driver
by Rajashekhara, Sudhakar 20 Jul '11

20 Jul '11

This patch fixes the following bugs found in DaVinci voice codec driver. a. According to DM365 voice codec data sheet at [1], before starting recording or playback, ADC/DAC modules should follow a reset and enable cycle. Writing a 1 to the ADC/DAC bit in the register resets the module and clearing the bit to 0 will enable the module. But the driver seems to be doing the reverse of it. b. In davinci_vcif_trigger() function, a break() statement was missing causing the davinci_vcif_stop() function to be called as a fallback after calling davinci_vcif_start(). [1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- sound/soc/davinci/davinci-vcif.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9259f1f..1f11525 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -62,9 +62,9 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream) w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -80,9 +80,9 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) /* Reset transmitter/receiver and sample rate/frame sync generators */ w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -159,6 +159,7 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_vcif_start(substream); + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: dapm - Add DAPM stream completion event.
by Liam Girdwood 20 Jul '11

20 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. This adds a callback function to be called at the completion of a DAPM stream event. This can be used by DSP components to perform calculations based on DAPM graphs after completion of stream events. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 2 ++ sound/soc/soc-dapm.c | 4 ++++ 4 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 198997d..596387b 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -524,6 +524,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int (*stream_event)(struct snd_soc_dapm_context *dapm); + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a..4c42f02 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -634,6 +634,9 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + /* codec stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; @@ -669,6 +672,9 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + /* platform stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f..159f144 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3140,6 +3140,7 @@ int snd_soc_register_platform(struct device *dev, platform->driver = platform_drv; platform->dapm.dev = dev; platform->dapm.platform = platform; + platform->dapm.stream_event = platform_drv->stream_event; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); @@ -3252,6 +3253,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.dev = dev; codec->dapm.codec = codec; codec->dapm.seq_notifier = codec_drv->seq_notifier; + codec->dapm.stream_event = codec_drv->stream_event; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aaef0ef..931dd7a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2764,6 +2764,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, } dapm_power_widgets(dapm, event); + + /* do we need to notify any clients that DAPM stream is complete */ + if (dapm->stream_event) + dapm->stream_event(dapm); } /** -- 1.7.4.1

1 0

Re: [alsa-devel] [patch][saa7134] do not change mute state for capturing audio
by Stas Sergeev 20 Jul '11

20 Jul '11

15.07.2011 05:38, Mauro Carvalho Chehab wrote: > If you want, feel free to propose a patch fixing that logic at saa7134, instead > of just removing it. Hi, I've just verified that pulseaudio indeed does the sound capturing on startup: --- saa7134[0]/alsa: saa7134[0] at 0xfe8fb800 irq 22 registered as card 2 saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: irq: field oops [even] --- So your proposal is not going to fix anything at all. Can we get back to discussing/applying mine then? And if the other drivers has that autounmute logic, then I suggest removing it there as well. You have not named any use-case for it, so I think there is none. I also think that the whole auto-unmute logic in your drivers is entirely flawed: for instance, I don't think recording from the sound card will automatically unmute its line-in or something else, so you are probably not following the generic alsa style here. I am adding alsa-devel to CC to find out what they think about that whole auto-unmute question.

5 23

[alsa-devel] [PATCH v3 00/10] ARM: mxs: add audio support
by Dong Aisheng 20 Jul '11

20 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Code is written based on the branch imx-for-3.1 in sascha's tree http://git.pengutronix.de/git/imx/linux-2.6.git Tested on MX28EVK. changes since v2: * mxs-pcm - fix wrong periods_max. - For robustness replace the channels_min check with a check for the relevant substream being present * mxs-saif - Add IRQ_NONE handle - improve mxs_saif_set_clk to make it more easy to read - correct return value - request_irq sequence changed to after ioremap * mxs-sgtl5000 - clean up a few unused code * Kconfig&Makefile - SND_SOC_MXS_SGTL5000 should depend on I2C in Kconfig * mxs - Select MXS_I2C in kconfig for mxs-sgtl5000 changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 2 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 3 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 ++ arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 40 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 359 ++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 677 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++ sound/soc/mxs/mxs-sgtl5000.c | 165 ++++++ 18 files changed, 1535 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

5 22

[alsa-devel] [PATCH 0/3] make the sgtl5000-codec work
by Wolfram Sang 19 Jul '11

19 Jul '11

These are three patches I developed while trying to make the sgtl5000 work on an mx28evk with the mxs-support provided by Dong Aisheng. The chip used to lock up before, because a wrong value was written to the power control. The first patch removes code duplication The second one is for user-comfort The third one makes the device work Aisheng: This should make your hack for the sgtl5000 obsolete. Could you please test it and add your Tested-by-tag if it also works for you? (Or Reviewed-by if you want to review the code). Zeng Zhaoming: Did this code really work for you? Regards, Wolfram Wolfram Sang (3): ASoC: sgtl5000: refactor registering internal ldo ASoC: sgtl5000: guide user when regulator support is needed ASoC: sgtl5000: fix cache handling sound/soc/codecs/sgtl5000.c | 198 +++++++++++++++---------------------------- 1 files changed, 67 insertions(+), 131 deletions(-) -- 1.7.5.4

6 15

[alsa-devel] [PATCH] ASoC: core: make comments fit the code
by Wolfram Sang 19 Jul '11

19 Jul '11

In one comment, cpu_dai was mentioned although codec_dai was used in the code. Also, fix the name for the card dai list which has no seperation into card_dai and codec_dai. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d75043e..ba95444 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1631,7 +1631,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } } cpu_dai->probed = 1; - /* mark cpu_dai as probed and add to card cpu_dai list */ + /* mark cpu_dai as probed and add to card dai list */ list_add(&cpu_dai->card_list, &card->dai_dev_list); } @@ -1672,7 +1672,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } } - /* mark cpu_dai as probed and add to card cpu_dai list */ + /* mark codec_dai as probed and add to card dai list */ codec_dai->probed = 1; list_add(&codec_dai->card_list, &card->dai_dev_list); } -- 1.7.5.4

3 2

[alsa-devel] [PATCH v5 4/4] ASoC: SAMSUNG: Add I2S0 internal dma driver
by Sangbeom Kim 19 Jul '11

19 Jul '11

I2S in Exynos4 and S5PC110(S5PV210) has a internal dma. It can be used low power audio mode and 2nd channel transfer. This patch can support idma. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- V2: change idma_new by new pcm_new api add handling different idma address as jassi comments, Modify the idma driver V3: merge idma_reg_int and idma_addr_init V4: move idma setting in i2s.c to idma_hw_params in idma.c move idma_reg_addr_init calling modify idma address handling error modify module information delete useless dev_dbg V5: delete redundant error print and check routine modify irq handler as jassi comment fix the indent error sound/soc/samsung/Makefile | 2 + sound/soc/samsung/idma.c | 453 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/idma.h | 26 +++ 3 files changed, 481 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/idma.c create mode 100644 sound/soc/samsung/idma.h diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 00ba557..41becdd 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o +snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c new file mode 100644 index 0000000..ebde074 --- /dev/null +++ b/sound/soc/samsung/idma.c @@ -0,0 +1,453 @@ +/* + * sound/soc/samsung/idma.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * I2S0's Internal DMA driver + * + * 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/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "idma.h" +#include "dma.h" +#include "i2s-regs.h" + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +static const struct snd_pcm_hardware idma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_U24_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_IDMA_BUFFER, + .period_bytes_min = 128, + .period_bytes_max = MAX_IDMA_PERIOD, + .periods_min = 1, + .periods_max = 2, +}; + +struct idma_ctrl { + spinlock_t lock; + int state; + dma_addr_t start; + dma_addr_t pos; + dma_addr_t end; + dma_addr_t period; + dma_addr_t periodsz; + void *token; + void (*cb)(void *dt, int bytes_xfer); +}; + +static struct idma_info { + spinlock_t lock; + void __iomem *regs; + dma_addr_t lp_tx_addr; +} idma; + +static void idma_getpos(dma_addr_t *src) +{ + *src = idma.lp_tx_addr + + (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; +} + +static int idma_enqueue(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 val; + + spin_lock(&prtd->lock); + prtd->token = (void *) substream; + spin_unlock(&prtd->lock); + + /* Internal DMA Level0 Interrupt Address */ + val = idma.lp_tx_addr + prtd->periodsz; + writel(val, idma.regs + I2SLVL0ADDR); + + /* Start address0 of I2S internal DMA operation. */ + val = idma.lp_tx_addr; + writel(val, idma.regs + I2SSTR0); + + /* + * Transfer block size for I2S internal DMA. + * Should decide transfer size before start dma operation + */ + val = readl(idma.regs + I2SSIZE); + val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); + val |= (((runtime->dma_bytes >> 2) & + I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); + writel(val, idma.regs + I2SSIZE); + + val = readl(idma.regs + I2SAHB); + val |= AHB_INTENLVL0; + writel(val, idma.regs + I2SAHB); + + return 0; +} + +static void idma_setcallbk(struct snd_pcm_substream *substream, + void (*cb)(void *, int)) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + spin_lock(&prtd->lock); + prtd->cb = cb; + spin_unlock(&prtd->lock); +} + +static void idma_control(int op) +{ + u32 val = readl(idma.regs + I2SAHB); + + spin_lock(&idma.lock); + + switch (op) { + case LPAM_DMA_START: + val |= (AHB_INTENLVL0 | AHB_DMAEN); + break; + case LPAM_DMA_STOP: + val &= ~(AHB_INTENLVL0 | AHB_DMAEN); + break; + default: + spin_unlock(&idma.lock); + return; + } + + writel(val, idma.regs + I2SAHB); + spin_unlock(&idma.lock); +} + +static void idma_done(void *id, int bytes_xfer) +{ + struct snd_pcm_substream *substream = id; + struct idma_ctrl *prtd = substream->runtime->private_data; + + if (prtd && (prtd->state & ST_RUNNING)) + snd_pcm_period_elapsed(substream); +} + +static int idma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 mod = readl(idma.regs + I2SMOD); + u32 ahb = readl(idma.regs + I2SAHB); + + ahb |= (AHB_DMARLD | AHB_INTMASK); + mod |= MOD_TXS_IDMA; + writel(ahb, idma.regs + I2SAHB); + writel(mod, idma.regs + I2SMOD); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->start = prtd->pos = runtime->dma_addr; + prtd->period = params_periods(params); + prtd->periodsz = params_period_bytes(params); + prtd->end = runtime->dma_addr + runtime->dma_bytes; + + idma_setcallbk(substream, idma_done); + + return 0; +} + +static int idma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int idma_prepare(struct snd_pcm_substream *substream) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + prtd->pos = prtd->start; + + /* flush the DMA channel */ + idma_control(LPAM_DMA_STOP); + idma_enqueue(substream); + + return 0; +} + +static int idma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + idma_control(LPAM_DMA_START); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + idma_control(LPAM_DMA_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t + idma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + dma_addr_t src; + unsigned long res; + + spin_lock(&prtd->lock); + + idma_getpos(&src); + res = src - prtd->start; + + spin_unlock(&prtd->lock); + + return bytes_to_frames(substream->runtime, res); +} + +static int idma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long size, offset; + int ret; + + /* From snd_pcm_lib_mmap_iomem */ + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_flags |= VM_IO; + size = vma->vm_end - vma->vm_start; + offset = vma->vm_pgoff << PAGE_SHIFT; + ret = io_remap_pfn_range(vma, vma->vm_start, + (runtime->dma_addr + offset) >> PAGE_SHIFT, + size, vma->vm_page_prot); + + return ret; +} + +static irqreturn_t iis_irq(int irqno, void *dev_id) +{ + struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; + u32 iiscon, iisahb, val, addr; + + iisahb = readl(idma.regs + I2SAHB); + iiscon = readl(idma.regs + I2SCON); + + val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; + + if (val) { + iisahb |= val; + writel(iisahb, idma.regs + I2SAHB); + + addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; + addr += prtd->periodsz; + addr %= (prtd->end - prtd->start); + addr += idma.lp_tx_addr; + + writel(addr, idma.regs + I2SLVL0ADDR); + + if (prtd->cb) + prtd->cb(prtd->token, prtd->period); + } + + return IRQ_HANDLED; +} + +static int idma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &idma_hardware); + + prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + if (ret < 0) { + pr_err("fail to claim i2s irq , ret = %d\n", ret); + kfree(prtd); + return ret; + } + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int idma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + + free_irq(IRQ_I2S0, prtd); + + if (!prtd) + pr_err("idma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static struct snd_pcm_ops idma_ops = { + .open = idma_open, + .close = idma_close, + .ioctl = snd_pcm_lib_ioctl, + .trigger = idma_trigger, + .pointer = idma_pointer, + .mmap = idma_mmap, + .hw_params = idma_hw_params, + .hw_free = idma_hw_free, + .prepare = idma_prepare, +}; + +static void idma_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) + return; + + buf = &substream->dma_buffer; + if (!buf->area) + return; + + iounmap(buf->area); + + buf->area = NULL; + buf->addr = 0; +} + +static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Assign PCM buffer pointers */ + buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; + buf->addr = idma.lp_tx_addr; + buf->bytes = idma_hardware.buffer_bytes_max; + buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + + return 0; +} + +static u64 idma_mask = DMA_BIT_MASK(32); + +static int idma_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) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) + ret = preallocate_idma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + + return ret; +} + +void idma_reg_addr_init(void *regs, dma_addr_t addr) +{ + spin_lock_init(&idma.lock); + idma.regs = regs; + idma.lp_tx_addr = addr; +} + +struct snd_soc_platform_driver asoc_idma_platform = { + .ops = &idma_ops, + .pcm_new = idma_new, + .pcm_free = idma_free, +}; + +static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); +} + +static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_idma_driver = { + .driver = { + .name = "samsung-idma", + .owner = THIS_MODULE, + }, + + .probe = asoc_idma_platform_probe, + .remove = __devexit_p(asoc_idma_platform_remove), +}; + +static int __init asoc_idma_init(void) +{ + return platform_driver_register(&asoc_idma_driver); +} +module_init(asoc_idma_init); + +static void __exit asoc_idma_exit(void) +{ + platform_driver_unregister(&asoc_idma_driver); +} +module_exit(asoc_idma_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); +MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h new file mode 100644 index 0000000..4827321 --- /dev/null +++ b/sound/soc/samsung/idma.h @@ -0,0 +1,26 @@ +/* + * sound/soc/samsung/idma.h + * + * 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. + * + */ + +#ifndef __SND_SOC_SAMSUNG_IDMA_H_ +#define __SND_SOC_SAMSUNG_IDMA_H_ + +extern void idma_reg_addr_init(void *regs, dma_addr_t addr); + +/* dma_state */ +#define LPAM_DMA_STOP 0 +#define LPAM_DMA_START 1 + +#define MAX_IDMA_PERIOD (128 * 1024) +#define MAX_IDMA_BUFFER (160 * 1024) + +#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- 1.7.1

6 11

Re: [alsa-devel] HP-path handling on VIA codecs
by alex dot baldacchino dot alsasub at gmail dot com 18 Jul '11

18 Jul '11

2011/7/8 Takashi Iwai <tiwai(a)suse.de>: > Hi, > > while testing with the gathered alsa-info.sh outputs, now I'm > considering how to improve the handling of the headphone (and speaker) > output paths. > > On VIA codecs, it looks like that the loopback mixer always contains > the input from the front DAC, and the front DAC is the only DAC that > can be mixed with loopback. (It might be not always, but on many > codecs, at least). > > I mean, some codecs of other vendors have a connection like: > Input-pin(s) -> Mix1 -> Mix2 -> Output1 > DAC1 -> Mix2 -> Output1 > Input-pin(s) -> Mix3 -> Mix4 -> Output2 > DAC2 -> Mix4 -> Output2 > .... > For such devices, you can use DAC as the volume control of each output > channel (HP, line-out, speaker, etc). So I started implementing in > that way for VIA, too. It's found as two parsed output paths, > spec->hp_out, and spec->hp_dep_out, > > But later I found that VIA has only > Input-pin(s) -> Mix1 -> Output1 > DAC1 -> Mix1 -> Output1 > or > DAC1 -> Output1 > > Input-pin(s) -> Mix1 -> Output1 > DAC1 -> Mix1 -> Output2 > or > DAC2 -> Output2 > ... > > Suppose the output2 is HP, it means that you can't have an individual > volume control via DAC2 if you want to mix the loopback. > Now we face a dilemma: usually an individual volume control is really > nice. But, if we take the individual control, it looses the loopback, > as these are mutual-exclusive for HP (or speaker) outputs. > > Thus we need to decide a policy: either > > A. HP has no volume unless independent mode > > B. Don't mix loopback to HP or others (but front-out) > > C. Allow to turn on/off aa-loopback mode; when loopback is enabled, > switch the path to hp_dep without volume control (only for front). > When loopback is disabled, switch to the direct DAC path. > > A is the old driver behavior, I guess. B is somewhat similar to now. > C would be smart, but need more implementation and add yet another > mixer element. > > What do you think? > > > thanks, > > Takashi > Personally, if I'm not misunderstanding it, I'd vote for C. I had tried something similar for old implementation, and can tell it worked and was easy enough to implement, AFAICT (I can resend the code if found of any interest, since it seems to me my first mail about it went lost, at least that's not in this mailing list archives, though that's based on the old code, and could be useful only to exchange ideas). It was quite simple: I used a few new fields in struct via_spec and two input mux controls, one for hp and one for front (to have a fine grained control and disable/enable aa path independently for those lines - and test if it could work this way), called "Direct Output" and "HP Direct Output" (I chose names thinking about the sound coming straight from analog inputs and going to analog outputs through the AA mixer, but could be, for instance, "({Front, HP}) Output Mode" (or "Output Source") with items labeled "Mixed with input" (or "Stereo mixer") and "Playback only", or the like). Corresponding controls callbacks worked on the mixer connected to each output jack and getting input from a dac (through an audio selector) and from the aa mixer, choosing the right mixer basing on the control they were invoked for, and the corresponding values in via_spec new fields: .get() returned the (negated) value of the mute bit of aa mixer connection in the chosen mixer connection list, and .put() muted/un-muted such a connection if in a different state (a better behavior would be to alternatively mute/unmute the connection to the dac and the connection to the loopback mixer, taking care of hp state (working on aa mixer connection only for hp mixer while in independent mode), and, of course, there could be a single switch/mux control to enable/disable the aa path globally). My reason to test such a solution was some noise I constantly got (and still get with newer code, but for Independent HP in latest version) from my headset microphone to each such line out (front and hp), but I can think, for instance, of someone wishing to listen to something while recording something else from line-in and not aiming to mix both sounds in output. But I guess some problems could arise in actual implementation, unless I'm overvaluing/misunderstanding them. First of all, now there's a pair of PCM Playback controls (switch and volume) attached to aa mixer and I'm not sure if muting connections to it could affect those controls (specially the volume). In such a case, they could be moved to the front dac, but this problem, if existing, could be re-raised for hp path in independent mode (in which case front dac and attached controls would be out of the game). An alternative could be leaving those controls where they're now, but reimplementing callbacks so that they could use a different nid (and work on more than one nid at a time). This way: - the .put callback for turning aa-loopback on/off for a certain nid (or all affected nids at once, if only one such control existed) would be responsible: 1. to record somewhere (accessible by via_spec) the nid(s) to work on (quite easy with separate controls attached to each affected mixer: the nid to gather would be just the actual one being unmuted in the two-entries connection list of passed-in kcontrol nid or the non loopback connection for hp mixer in independent mode), 2. to set the right volume (not just un-muting) for the non-loopback path(s) when switching to it (basing on pcm volume setting) and muting/unmuting the connection(s) to aa mixer as needed; - the .put callback for adjusting pcm volume setting would work as now for aa mixer nid (both to let it work as expected when used as source for output and to gather correct values from it, but also for correct handling of capture from stereo mixer), while performing same operations for each alternative nid, gathered from a proper list and being non-null and different from aa mixer nid; - the .get callback for "PCM Playback Volume" could be left untouched (to always gather infos from the 'base' nid the control is attacthed to). Similar considerations could/should be done for the pcm switch, that (now, in my case) has the effect to mute/unmute front output (only). However, I might be missing the point since I'm not sure how those controls are expected to work properly. Previously, with old code, I had no PCM Playback Switch and a "standalone" PCM Playback Volume, detached from any nid, and such seemed to work for all outputs, as far as I remember (I should re-install old implementation to make a test); alsa-info.sh reported following output for alsactl: control.31 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } Now I have both controls attached to 0x21, they seem to have effect on front path only (throughout aa mixer - previously its connection to front dac, for my codec, was just un-muted with an initialization verb) and alsa-info.sh reports: control.9 { iface MIXER name 'PCM Playback Volume' value.0 22 value.1 22 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -150 dbvalue.1 -150 } } control.10 { iface MIXER name 'PCM Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } Similar concerns could be raised for Master Playback controls, given some codecs (but not mine - vt2020) holds them via their aa mixer nid. Another, possibly minor, problem, as pointed out in Lydia's answer, could be the somewhat inconsistence introduced between newer codecs supporting solution C and older ones unable to mute their aa mixer without loosing their front dac (because their aa mixer is just between their front dac and their rear green jack, but also between front dac and hp jack when in redirected mode). Such could be simply disregarded, taken as a consequence of a different set of supported features; therefore, it should be possible to choose automatically whether to create the proper control(s) for turning on/off aa-loopback or not while parsing output paths, looking at the connection lists of nids in a certain path, starting from an output jack and moving towards a dac, and looking if an element (a mixer or the jack nid itself) is found with at least two connections, one of which should be the aa mixer nid and the other one either the front dac nid, or an audio selector connected to it. For hp path the non-loopback connection shouldn't be hp dac directly (not passing through a selector), or the implementation would conflict with redirected output if such a dac were shared, which is the case for several codecs, specially older ones (newer ones seem to use an audio selector, instead, for front and hp), though I'm not sure if that's the same for all such codecs - hp_indep_shared could tell that, so that a few older codecs could support solution C for hp only, but in such a case solution C would behave mostly as B in current implementation, but for redirected mode, given front stream would be 'cloned' for hp dac if not shared, so hp could get loopback mixed in or not at whim, even if such would be inconsistent with front being always mixed with loopback). Regards, Alex

2 5

Re: [alsa-devel] model needed for Clevo P150HM (Realtek ALC892)
by Takashi Iwai 18 Jul '11

18 Jul '11

At Thu, 30 Jun 2011 08:30:10 +0200, Julian Sikorski wrote: > > W dniu 30.06.2011 07:26, Takashi Iwai pisze: > > At Wed, 29 Jun 2011 21:29:38 +0200, > > Julian Sikorski wrote: > >> > >> I also tried merging sound-2.6/master with Linus's tree, but this also > >> did not help. > > > > It's really strange. Are you sure that this is the kernel you > > modified properly? Before further debugging, check two things: > > > > - Run "git log sound/pci/hda/patch_realtek.c" and the first few > > entries, whether they come from sound git tree. For example, > > check the commit below is contained: > > > > commit 7ec9c6ccc6007b14a916021d4ba7ffbcc7822ae3 > > Author: Takashi Iwai <tiwai(a)suse.de> > > Date: Mon Jun 27 15:53:38 2011 +0200 > > > > ALSA: hda - Fix volume-init for ALC259 with invalid widget caps > > commit 0b0c8490670a77b937f502491ae9d5236aa4054a > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Tue Jun 28 12:45:47 2011 +0200 > > ALSA: hda - Create snd_hda_get_conn_index() helper function > > Create snd_hda_get_conn_index() helper function for obtaining the > connection index of the widget. Replaced the similar codes used in > several codec-drivers with this common helper. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > commit 4f574b7b1a1cc8aac617e938459e8f03a641e678 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Mon Jun 27 16:17:07 2011 +0200 > > ALSA: hda - More volume-init fixes for ALC267 codec > > More similar fixes like previous commits: handle the exceptional case > like ALC267 where no volume amp is found in ADC widget but in the > capsrc widget instead. > > Also minor checks for avoiding possible erros: no connection-select > when the pin has a single selection, and add beep verbs only when the > 0x1d is used for beep. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > To be even more sure, I re-synced your branch again and built a new > kernel. I am attaching patch_realtek.c file being built. Fresh > alsa-info.sh is here: > http://www.alsa-project.org/db/?f=5051b2cb788b1db4273e06e98a78783dd0ec679f OK, then it's correct. > > > > - Confirm the running kernel is what you really built and installed. > > You can modify the suffix via $CONFIG_LOCALVERSION in .config file. > > Name it uniquely, rebuild, re-install and reboot. The new suffix > > should appear in the output of "uname -r". > > belegdol4 is the suffix I added: > $ uname -r > 3.0-0.rc5.git0.1.belegdol4.fc15.x86_64 > > > - Once when these are correct, check /proc/asound/card0/codec#0: > > the widget NID 0x02 should contain "Speaker Playback Volume" instead > > of "Headphone Playback Volume" with the latest driver. > > Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out > Control: name="Headphone Playback Volume", index=0, device=0 > > Perhaps I need to specify a correct model to enable this code? No, you shouldn't pass any model option. Could you apply the patch below to add some debug prints, and give the relevant kernel messages you get? Just grep lines with 'XXX'. Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b96dce..3134555 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18770,8 +18770,10 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) { hda_nid_t sel = alc_go_down_to_selector(codec, pin); - if (snd_hda_get_conn_list(codec, sel, NULL) == 1) + if (snd_hda_get_conn_list(codec, sel, NULL) == 1) { + printk(KERN_DEBUG "XXX pin %x coupled with a single DAC\n", pin); return alc_auto_look_for_dac(codec, pin); + } return 0; } @@ -18783,6 +18785,7 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec) bool redone; int i; + printk(KERN_DEBUG "XXX alc662_auto_fill_dac_nids called\n"); again: spec->multiout.num_dacs = 0; spec->multiout.hp_nid = 0; @@ -18813,6 +18816,7 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec) * checking the hard-wired DACs */ redone = true; + printk(KERN_DEBUG "XXX redoing...\n"); goto again; } } @@ -18826,6 +18830,17 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec) sizeof(hda_nid_t) * (cfg->line_outs - i - 1)); } + printk(KERN_DEBUG "XXX filled dacs: %x/%x/%x/%x/%x, hp %x, ext %x/%x/%x\n", + spec->private_dac_nids[0], + spec->private_dac_nids[1], + spec->private_dac_nids[2], + spec->private_dac_nids[3], + spec->private_dac_nids[4], + spec->multiout.hp_nid, + spec->multiout.extra_out_nid[0], + spec->multiout.extra_out_nid[1], + spec->multiout.extra_out_nid[2]); + return 0; } @@ -18877,6 +18892,7 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, mix = alc_auto_dac_to_mix(codec, pin, nid); if (!mix) continue; + printk(KERN_DEBUG "XXX create mixer ch %d pin %x dac %x mix %x\n", i, pin, nid, mix); name = alc_get_line_out_pfx(spec, i, true, &index); if (!name) { /* Center/LFE */ @@ -19026,6 +19042,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, dac = alc_auto_look_for_dac(codec, nid); if (!dac) continue; + printk(KERN_DEBUG "XXX added multi-io pin %x, dac %x\n", nid, dac); spec->multi_io[num_pins].pin = nid; spec->multi_io[num_pins].dac = dac; num_pins++; @@ -19128,6 +19145,7 @@ static int alc_auto_add_multi_channel_mode(struct hda_codec *codec, if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && cfg->hp_outs == 1) { /* use HP as primary out */ + printk(KERN_DEBUG "XXX swap primary out to HP\n"); cfg->speaker_outs = cfg->line_outs; memcpy(cfg->speaker_pins, cfg->line_out_pins, sizeof(cfg->speaker_pins));

3 37

[alsa-devel] [PATCH] ASoC: Mark cache as dirty when suspending
by Mark Brown 18 Jul '11

18 Jul '11

Since quite a few drivers are not managing to flag the cache as needing to be resynced after suspend and it's a reasonable thing to do flag the cache as needing sync automatically when suspending. The expectation is that systems will mainly only keep the CODEC powered when doing audio through the CODEC so we won't actually suspend the device anyway; drivers which want to can override this behaviour when they resume. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f..93109a4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -577,6 +577,7 @@ int snd_soc_suspend(struct device *dev) case SND_SOC_BIAS_OFF: codec->driver->suspend(codec, PMSG_SUSPEND); codec->suspended = 1; + codec->cache_sync = 1; break; default: dev_dbg(codec->dev, "CODEC is on over suspend\n"); -- 1.7.5.4

2 1

[alsa-devel] [RESEND][PATCH v5 4/4] ASoC: SAMSUNG: Add I2S0 internal dma driver
by Sangbeom Kim 18 Jul '11

18 Jul '11

I2S in Exynos4 and S5PC110(S5PV210) has a internal dma. It can be used low power audio mode and 2nd channel transfer. This patch can support idma. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- V2: change idma_new by new pcm_new api add handling different idma address as jassi comments, Modify the idma driver V3: merge idma_reg_int and idma_addr_init V4: move idma setting in i2s.c to idma_hw_params in idma.c move idma_reg_addr_init calling modify idma address handling error modify module information delete useless dev_dbg V5: delete redundant error print and check routine modify irq handler as jassi comment fix the indent error modify period_bytes_max value to set half size of max buffer sound/soc/samsung/Makefile | 2 + sound/soc/samsung/idma.c | 453 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/idma.h | 26 +++ 3 files changed, 481 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/idma.c create mode 100644 sound/soc/samsung/idma.h diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 00ba557..41becdd 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o +snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c new file mode 100644 index 0000000..ebde074 --- /dev/null +++ b/sound/soc/samsung/idma.c @@ -0,0 +1,453 @@ +/* + * sound/soc/samsung/idma.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * I2S0's Internal DMA driver + * + * 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/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "idma.h" +#include "dma.h" +#include "i2s-regs.h" + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +static const struct snd_pcm_hardware idma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_U24_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_IDMA_BUFFER, + .period_bytes_min = 128, + .period_bytes_max = MAX_IDMA_BUFFER/2, + .periods_min = 1, + .periods_max = 2, +}; + +struct idma_ctrl { + spinlock_t lock; + int state; + dma_addr_t start; + dma_addr_t pos; + dma_addr_t end; + dma_addr_t period; + dma_addr_t periodsz; + void *token; + void (*cb)(void *dt, int bytes_xfer); +}; + +static struct idma_info { + spinlock_t lock; + void __iomem *regs; + dma_addr_t lp_tx_addr; +} idma; + +static void idma_getpos(dma_addr_t *src) +{ + *src = idma.lp_tx_addr + + (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; +} + +static int idma_enqueue(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 val; + + spin_lock(&prtd->lock); + prtd->token = (void *) substream; + spin_unlock(&prtd->lock); + + /* Internal DMA Level0 Interrupt Address */ + val = idma.lp_tx_addr + prtd->periodsz; + writel(val, idma.regs + I2SLVL0ADDR); + + /* Start address0 of I2S internal DMA operation. */ + val = idma.lp_tx_addr; + writel(val, idma.regs + I2SSTR0); + + /* + * Transfer block size for I2S internal DMA. + * Should decide transfer size before start dma operation + */ + val = readl(idma.regs + I2SSIZE); + val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); + val |= (((runtime->dma_bytes >> 2) & + I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); + writel(val, idma.regs + I2SSIZE); + + val = readl(idma.regs + I2SAHB); + val |= AHB_INTENLVL0; + writel(val, idma.regs + I2SAHB); + + return 0; +} + +static void idma_setcallbk(struct snd_pcm_substream *substream, + void (*cb)(void *, int)) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + spin_lock(&prtd->lock); + prtd->cb = cb; + spin_unlock(&prtd->lock); +} + +static void idma_control(int op) +{ + u32 val = readl(idma.regs + I2SAHB); + + spin_lock(&idma.lock); + + switch (op) { + case LPAM_DMA_START: + val |= (AHB_INTENLVL0 | AHB_DMAEN); + break; + case LPAM_DMA_STOP: + val &= ~(AHB_INTENLVL0 | AHB_DMAEN); + break; + default: + spin_unlock(&idma.lock); + return; + } + + writel(val, idma.regs + I2SAHB); + spin_unlock(&idma.lock); +} + +static void idma_done(void *id, int bytes_xfer) +{ + struct snd_pcm_substream *substream = id; + struct idma_ctrl *prtd = substream->runtime->private_data; + + if (prtd && (prtd->state & ST_RUNNING)) + snd_pcm_period_elapsed(substream); +} + +static int idma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 mod = readl(idma.regs + I2SMOD); + u32 ahb = readl(idma.regs + I2SAHB); + + ahb |= (AHB_DMARLD | AHB_INTMASK); + mod |= MOD_TXS_IDMA; + writel(ahb, idma.regs + I2SAHB); + writel(mod, idma.regs + I2SMOD); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->start = prtd->pos = runtime->dma_addr; + prtd->period = params_periods(params); + prtd->periodsz = params_period_bytes(params); + prtd->end = runtime->dma_addr + runtime->dma_bytes; + + idma_setcallbk(substream, idma_done); + + return 0; +} + +static int idma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int idma_prepare(struct snd_pcm_substream *substream) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + prtd->pos = prtd->start; + + /* flush the DMA channel */ + idma_control(LPAM_DMA_STOP); + idma_enqueue(substream); + + return 0; +} + +static int idma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + idma_control(LPAM_DMA_START); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + idma_control(LPAM_DMA_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t + idma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + dma_addr_t src; + unsigned long res; + + spin_lock(&prtd->lock); + + idma_getpos(&src); + res = src - prtd->start; + + spin_unlock(&prtd->lock); + + return bytes_to_frames(substream->runtime, res); +} + +static int idma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long size, offset; + int ret; + + /* From snd_pcm_lib_mmap_iomem */ + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_flags |= VM_IO; + size = vma->vm_end - vma->vm_start; + offset = vma->vm_pgoff << PAGE_SHIFT; + ret = io_remap_pfn_range(vma, vma->vm_start, + (runtime->dma_addr + offset) >> PAGE_SHIFT, + size, vma->vm_page_prot); + + return ret; +} + +static irqreturn_t iis_irq(int irqno, void *dev_id) +{ + struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; + u32 iiscon, iisahb, val, addr; + + iisahb = readl(idma.regs + I2SAHB); + iiscon = readl(idma.regs + I2SCON); + + val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; + + if (val) { + iisahb |= val; + writel(iisahb, idma.regs + I2SAHB); + + addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; + addr += prtd->periodsz; + addr %= (prtd->end - prtd->start); + addr += idma.lp_tx_addr; + + writel(addr, idma.regs + I2SLVL0ADDR); + + if (prtd->cb) + prtd->cb(prtd->token, prtd->period); + } + + return IRQ_HANDLED; +} + +static int idma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &idma_hardware); + + prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + if (ret < 0) { + pr_err("fail to claim i2s irq , ret = %d\n", ret); + kfree(prtd); + return ret; + } + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int idma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + + free_irq(IRQ_I2S0, prtd); + + if (!prtd) + pr_err("idma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static struct snd_pcm_ops idma_ops = { + .open = idma_open, + .close = idma_close, + .ioctl = snd_pcm_lib_ioctl, + .trigger = idma_trigger, + .pointer = idma_pointer, + .mmap = idma_mmap, + .hw_params = idma_hw_params, + .hw_free = idma_hw_free, + .prepare = idma_prepare, +}; + +static void idma_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) + return; + + buf = &substream->dma_buffer; + if (!buf->area) + return; + + iounmap(buf->area); + + buf->area = NULL; + buf->addr = 0; +} + +static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Assign PCM buffer pointers */ + buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; + buf->addr = idma.lp_tx_addr; + buf->bytes = idma_hardware.buffer_bytes_max; + buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + + return 0; +} + +static u64 idma_mask = DMA_BIT_MASK(32); + +static int idma_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) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) + ret = preallocate_idma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + + return ret; +} + +void idma_reg_addr_init(void *regs, dma_addr_t addr) +{ + spin_lock_init(&idma.lock); + idma.regs = regs; + idma.lp_tx_addr = addr; +} + +struct snd_soc_platform_driver asoc_idma_platform = { + .ops = &idma_ops, + .pcm_new = idma_new, + .pcm_free = idma_free, +}; + +static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); +} + +static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_idma_driver = { + .driver = { + .name = "samsung-idma", + .owner = THIS_MODULE, + }, + + .probe = asoc_idma_platform_probe, + .remove = __devexit_p(asoc_idma_platform_remove), +}; + +static int __init asoc_idma_init(void) +{ + return platform_driver_register(&asoc_idma_driver); +} +module_init(asoc_idma_init); + +static void __exit asoc_idma_exit(void) +{ + platform_driver_unregister(&asoc_idma_driver); +} +module_exit(asoc_idma_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); +MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h new file mode 100644 index 0000000..4827321 --- /dev/null +++ b/sound/soc/samsung/idma.h @@ -0,0 +1,26 @@ +/* + * sound/soc/samsung/idma.h + * + * 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. + * + */ + +#ifndef __SND_SOC_SAMSUNG_IDMA_H_ +#define __SND_SOC_SAMSUNG_IDMA_H_ + +extern void idma_reg_addr_init(void *regs, dma_addr_t addr); + +/* dma_state */ +#define LPAM_DMA_STOP 0 +#define LPAM_DMA_START 1 + + +#define MAX_IDMA_BUFFER (128 * 1024) + +#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- 1.7.1

2 3

[alsa-devel] ASoC updates for 3.1
by Mark Brown 18 Jul '11

18 Jul '11

The following changes since commit e320bc42be746b8f006d163b6e340e5fb79ce7b2: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-09 11:43:04 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Axel Lin (1): ASoC: wm8900: fix a memory leak if wm8900_set_fll fails Dimitris Papastamos (1): ASoC: WM8983: Initial driver Giridhar Maruthy (1): ASoC: SAMSUNG: 24-bit audio playback on Exynos4210 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Johannes Stezenbach (2): ASoC: STA32x: Add mixer controls for biquad coefficients ASoC: STA32x: Preserve reserved register bits Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (19): Merge branch 'for-3.0' into for-3.1 Merge branch 'for-3.0' into for-3.1 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-3.1 ASoC: Conditionalize the enable of WM8994 ADC TDM mode ASoC: Use late enable handling for direct voice, speaker and headphone ASoC: Implement DC servo completion IRQ handling for wm_hubs devices ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958 ASoC: Use WM8994 FLL lock interrupt ASoC: Don't warn on low WM8994/58 AIFnCLKs ASoC: Log WM8994 FIFO errors from the interrupt ASoC: Fix shift in WM8958 accessory detection default implementation Merge branch 'for-3.0' into for-3.1 ASoC: Handle spurious wm_hubs DC servo done interrupts ASoC: Handle failed WM8994 FLL lock waits ASoC: Report an error for unknown adav80x formats ASoC: Reduce power consumption for idle DAIs in WM8994 ASoC: Don't use -1 to boostrap subseq so it can be used by drivers ASoC: Correct WM8994 MICBIAS supply widget hookup Merge branch 'for-3.0' into for-3.1 Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration Takashi Iwai (4): Merge branch 'for-3.0' of git://git.kernel.org/.../lrg/asoc-2.6 into fix/asoc Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc ASoC: Don't set invalid name string to snd_card->driver field Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/sta32x.c | 140 +++++ sound/soc/codecs/tlv320aic26.c | 14 +- sound/soc/codecs/tlv320aic3x.c | 9 +- sound/soc/codecs/wm8900.c | 1 + sound/soc/codecs/wm8983.c | 1203 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8983.h | 1029 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 156 +++++- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_hubs.c | 33 +- sound/soc/codecs/wm_hubs.h | 8 + sound/soc/samsung/smdk_wm8994.c | 5 +- sound/soc/sh/fsi-ak4642.c | 16 +- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 +- sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 2 +- 20 files changed, 2593 insertions(+), 49 deletions(-) create mode 100644 sound/soc/codecs/wm8983.c create mode 100644 sound/soc/codecs/wm8983.h

2 1

[alsa-devel] ASoC updates for 3.0
by Mark Brown 18 Jul '11

18 Jul '11

The following changes since commit 18361bbe3180eca62796188d62aefac1519f4c83: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-07-09 09:44:09 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (2): ASoC: Fix shift in WM8958 accessory detection default implementation ASoC: Correct WM8994 MICBIAS supply widget hookup sound/soc/codecs/wm8994.c | 9 +++++---- sound/soc/sh/fsi-ak4642.c | 16 ++++++++-------- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 ++-- 4 files changed, 16 insertions(+), 15 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: Report an error for unknown adav80x formats
by Mark Brown 16 Jul '11

16 Jul '11

Not only fixes error handling but also some uninitialized variable warnings. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/adav80x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index e30fba6..300c04b 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -456,7 +456,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, val = ADAV80X_CAPTURE_WORD_LEN24; break; default: - break; + return -EINVAL; } snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], @@ -488,7 +488,7 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; break; default: - break; + return -EINVAL; } snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: Don't use -1 to boostrap subseq so it can be used by drivers
by Mark Brown 16 Jul '11

16 Jul '11

Makes life a little easier if you want to add subsequences to an existing driver as you can use -1 to put things at the start of sequences. 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 54fa2e5..fbfcda0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -989,7 +989,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&pending); cur_sort = -1; - cur_subseq = -1; + cur_subseq = INT_MIN; cur_reg = SND_SOC_NOPM; cur_dapm = NULL; } -- 1.7.5.4

1 0

[alsa-devel] [PATCH V2 00/10]
by Dong Aisheng 15 Jul '11

15 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Code is written based on the branch imx-for-3.1 in sascha's tree http://git.pengutronix.de/git/imx/linux-2.6.git Tested on MX28EVK. Main changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 1 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 3 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 ++ arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 39 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 19 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 367 +++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 656 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++ sound/soc/mxs/mxs-sgtl5000.c | 175 ++++++ 18 files changed, 1529 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

7 51

Re: [alsa-devel] [PATCH] ASoC: WM8983: Initial driver
by Mark Brown 15 Jul '11

15 Jul '11

On Fri, Jul 15, 2011 at 01:51:30PM +0100, Dimitris Papastamos wrote: > The WM8983 is a low power, high quality stereo CODEC > designed for portable multimedia applications. Highly flexible > analogue mixing functions enable new application features, > combining hi-fi quality audio with voice communication. I've applied this but there's one issue you should fix up later: > + /* mute all outputs and set PGAs to minimum gain */ > + for (i = WM8983_LOUT1_HP_VOLUME_CTRL; > + i <= WM8983_OUT4_MONO_MIX_CTRL; ++i) > + snd_soc_update_bits(codec, i, 0x40, 0x40); I know you're doing this to minimise artifacts on initial bias raise but we should really be doing this without affecting the default register settings.

2 2

[alsa-devel] [PATCH] ASoC: Fix shift in WM8958 accessory detection default implementation
by Mark Brown 15 Jul '11

15 Jul '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bc62b6e..bfb7e66 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2834,7 +2834,7 @@ static void wm8958_default_micdet(u16 status, void *data) report = SND_JACK_MICROPHONE; /* Everything else is buttons; just assign slots */ - if (status & 0x1c0) + if (status & 0x1c) report |= SND_JACK_BTN_0; done: -- 1.7.5.4

1 0

Re: [alsa-devel] [PATCH] sound: codec: tlv320aic3x: remove mono controls
by Mark Brown 15 Jul '11

15 Jul '11

On Fri, Jul 15, 2011 at 12:59:15PM +0200, Jan Weitzel wrote: > Any rework needed? It'd be easier to review the patch if you sent me a copy...

1 0

[alsa-devel] [PATCH] ASoC: Handle spurious wm_hubs DC servo done interrupts
by Mark Brown 15 Jul '11

15 Jul '11

Don't assume the first fire indicates that we're done. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 30 ++++++++++++++++-------------- 1 files changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ced41e8..4de6794 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -66,8 +66,8 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); unsigned int reg; int count = 0; + int timeout; unsigned int val; - unsigned long timeout; val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1; @@ -76,21 +76,23 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) dev_dbg(codec->dev, "Waiting for DC servo...\n"); - if (hubs->dcs_done_irq) { - timeout = wait_for_completion_timeout(&hubs->dcs_done, - msecs_to_jiffies(500)); - if (timeout == 0) - dev_warn(codec->dev, "No DC servo interrupt\n"); + if (hubs->dcs_done_irq) + timeout = 4; + else + timeout = 400; - reg = snd_soc_read(codec, WM8993_DC_SERVO_0); - } else { - do { - count++; + do { + count++; + + if (hubs->dcs_done_irq) + wait_for_completion_timeout(&hubs->dcs_done, + msecs_to_jiffies(250)); + else msleep(1); - reg = snd_soc_read(codec, WM8993_DC_SERVO_0); - dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & op && count < 400); - } + + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); + dev_dbg(codec->dev, "DC servo: %x\n", reg); + } while (reg & op && count < timeout); if (reg & op) dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", -- 1.7.5.4

1 0

[alsa-devel] Major reworks on patch_via.c (TESTERS WANTED)
by Takashi Iwai 15 Jul '11

15 Jul '11

Hi, as there have many problems reported for VIA codecs, I started looking at the driver code, and decided to rework on it. The current development is found in sound git tree topic/via-cleanup branch: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git You can pull into the latest Linus tree from topic/via-cleanup branch: # this is the vanilla tree % cd linux-2.6 % git pull \ git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git \ topic/via-cleanup A good thing is that a fair amount of code reduction is done there: % git diff HEAD..topic/via-cleanup --stat sound/pci/hda/patch_via.c sound/pci/hda/patch_via.c | 5892 +++++++++++---------------------------------- 1 files changed, 1453 insertions(+), 4439 deletions(-) But, the driver isn't tested with real machines, yet. That is, I need testers with real machines. Could VIA machine owners take a look at these commits and test it if possible? Especially, I'd like to know whether the headphone auto-mute works in this form or not. FWIW, the branch was merged to sound-unstable tree, so you can build from alsa-driver-unstable snapshot tarball if you'd like to build external modules without the kernel tree and git-merge. ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… Also, if anyone has alsa-info.sh outputs of VIA codecs, please let me know. It'd be very helpful for debugging, since I can check it via hda-emu with alsa-info.sh output. For now, I have the files of only VT1702, VT1708S, VT1718S, VT1828S and VT2020 codecs. The alsa-info.sh outputs of other VIA codecs would be greatly appreciated. thanks, Takashi

5 18

[alsa-devel] [PATCH] ALSA: intel8x0: Apply headphones+mute LED quirk for Dell Inspiron 9300
by Dan Chen 15 Jul '11

15 Jul '11

Hi, Attached is a patch against sound-2.6 master HEAD resolving the original reporter's volume keys issue. Please consider applying. Cheers, -Dan

2 1

[alsa-devel] [PATCH 11/20] ARM i.MX dma: Fix burstsize settings
by Sascha Hauer 14 Jul '11

14 Jul '11

From: Wolfram Sang <w.sang(a)pengutronix.de> Already upstreamed. dmaengine expects the maxburst parameter in words, not bytes. The imxdma driver and its users do this wrong. Fix this. As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work with imx-dma. This broke the driver with imx-sdma support which correctly takes the maxburst parameter in words. This patch puts the sdma based sound back to work. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Chris Ball <cjb(a)laptop.org> Cc: linux-mmc(a)vger.kernel.org Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- drivers/dma/imx-dma.c | 3 ++- sound/soc/imx/imx-pcm-dma-mx2.c | 4 ++-- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/dma/imx-dma.c b/drivers/dma/imx-dma.c index e18eaab..d99f71c 100644 --- a/drivers/dma/imx-dma.c +++ b/drivers/dma/imx-dma.c @@ -135,7 +135,8 @@ static int imxdma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, if (ret) return ret; - imx_dma_config_burstlen(imxdmac->imxdma_channel, imxdmac->watermark_level); + imx_dma_config_burstlen(imxdmac->imxdma_channel, + imxdmac->watermark_level * imxdmac->word_size); return 0; default: diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 4173b3d..43fdc24f 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.direction = DMA_TO_DEVICE; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; - slave_config.dst_maxburst = dma_params->burstsize * buswidth; + slave_config.dst_maxburst = dma_params->burstsize; } else { slave_config.direction = DMA_FROM_DEVICE; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; - slave_config.src_maxburst = dma_params->burstsize * buswidth; + slave_config.src_maxburst = dma_params->burstsize; } ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); -- 1.7.5.4

1 1

[alsa-devel] [PATCH] ALSA: hda - Disable automute feature by default
by Ben Hutchings 14 Jul '11

14 Jul '11

The default behaviour should be the same as before this feature was introduced, to avoid user confusion. Reported-by: Toni Mueller <support(a)oeko.net> References: http://bugs.debian.org/631963 Signed-off-by: Ben Hutchings <ben(a)decadent.org.uk> --- The code is rather inconsistent in whether 'mode' means the behaviour or mechanism of automute. I didn't touch that but you might want to consider some renaming. Ben. sound/pci/hda/patch_realtek.c | 74 ++--------------------------------------- 1 files changed, 3 insertions(+), 71 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b48fb43..1247a04 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -306,7 +306,7 @@ struct alc_multi_io { }; enum { - ALC_AUTOMUTE_PIN, /* change the pin control */ + ALC_AUTOMUTE_PIN = 1, /* change the pin control */ ALC_AUTOMUTE_AMP, /* mute/unmute the pin AMP */ ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; @@ -1599,10 +1599,9 @@ static void alc_init_auto_hp(struct hda_codec *codec) snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } - if (spec->automute && cfg->line_out_pins[0] && + if (spec->automute_mode && cfg->line_out_pins[0] && cfg->line_out_pins[0] != cfg->hp_pins[0] && cfg->line_out_pins[0] != cfg->speaker_pins[0]) { for (i = 0; i < cfg->line_outs; i++) { @@ -1619,7 +1618,7 @@ static void alc_init_auto_hp(struct hda_codec *codec) spec->automute_lines = spec->detect_line; } - if (spec->automute) { + if (spec->automute_mode) { /* create a control for automute mode */ alc_add_automute_mode_enum(codec); spec->unsol_event = alc_sku_unsol_event; @@ -2241,7 +2240,6 @@ static void alc889_automute_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x17; spec->autocfg.speaker_pins[3] = 0x19; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2259,7 +2257,6 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1b; /* hp */ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ spec->autocfg.speaker_pins[1] = 0x15; /* bass */ - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2543,7 +2540,6 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2555,7 +2551,6 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2567,7 +2562,6 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2579,7 +2573,6 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -3667,7 +3660,6 @@ static void alc880_uniwill_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -3699,7 +3691,6 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -3969,7 +3960,6 @@ static void alc880_lg_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -4055,7 +4045,6 @@ static void alc880_lg_lw_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -4128,7 +4117,6 @@ static void alc880_medion_rim_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -6277,7 +6265,6 @@ static void alc260_hp_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x0f; spec->autocfg.speaker_pins[0] = 0x10; spec->autocfg.speaker_pins[1] = 0x11; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -6308,7 +6295,6 @@ static void alc260_hp_3013_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x10; spec->autocfg.speaker_pins[1] = 0x11; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -6352,7 +6338,6 @@ static void alc260_hp_3012_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x0f; spec->autocfg.speaker_pins[1] = 0x11; spec->autocfg.speaker_pins[2] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -8887,7 +8872,6 @@ static void alc885_imac24_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8901,7 +8885,6 @@ static void alc885_mba21_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8913,7 +8896,6 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8924,7 +8906,6 @@ static void alc885_imac91_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8958,7 +8939,6 @@ static void alc882_targa_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9405,7 +9385,6 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1a; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9559,7 +9538,6 @@ static void alc883_mitac_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9725,7 +9703,6 @@ static void alc888_3st_hp_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x18; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9786,7 +9763,6 @@ static void alc888_lenovo_ms7195_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9797,7 +9773,6 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9811,7 +9786,6 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9841,7 +9815,6 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9851,7 +9824,6 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9862,7 +9834,6 @@ static void alc883_lenovo_101e_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->detect_line = 1; spec->automute_lines = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; @@ -9876,7 +9847,6 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9907,7 +9877,6 @@ static void alc888_6st_dell_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x15; spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9921,7 +9890,6 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9932,7 +9900,6 @@ static void alc883_vaiott_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9959,7 +9926,6 @@ static void alc883_mode2_setup(struct hda_codec *codec) spec->ext_mic.mux_idx = 0; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11470,7 +11436,6 @@ static void alc262_hp_bpc_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11480,7 +11445,6 @@ static void alc262_hp_wildwest_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11560,7 +11524,6 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11674,7 +11637,6 @@ static void alc262_hippo_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11684,7 +11646,6 @@ static void alc262_hippo1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11748,7 +11709,6 @@ static void alc262_tyan_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11893,7 +11853,6 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec) spec->int_mic.pin = 0x12; spec->int_mic.mux_idx = 9; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11998,7 +11957,6 @@ static void alc262_fujitsu_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.hp_pins[1] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -12045,7 +12003,6 @@ static void alc262_lenovo_3000_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -13255,7 +13212,6 @@ static void alc268_acer_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -13339,7 +13295,6 @@ static void alc268_acer_lc_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -13379,7 +13334,6 @@ static void alc268_dell_setup(struct hda_codec *codec) spec->int_mic.pin = 0x19; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -13411,7 +13365,6 @@ static void alc267_quanta_il1_setup(struct hda_codec *codec) spec->int_mic.pin = 0x19; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -14415,7 +14368,6 @@ static void alc269_quanta_fl1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14437,7 +14389,6 @@ static void alc269_lifebook_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1a; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; } @@ -14510,7 +14461,6 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14525,7 +14475,6 @@ static void alc269_laptop_dmic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14540,7 +14489,6 @@ static void alc269vb_laptop_amic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14555,7 +14503,6 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -16878,7 +16825,6 @@ static void alc861vd_lenovo_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -16953,7 +16899,6 @@ static void alc861vd_dallas_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -18222,7 +18167,6 @@ static void alc662_lenovo_101e_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->detect_line = 1; spec->automute_lines = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; @@ -18246,7 +18190,6 @@ static void alc662_eeepc_ep20_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -18256,7 +18199,6 @@ static void alc663_m51va_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18272,7 +18214,6 @@ static void alc663_mode1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18287,7 +18228,6 @@ static void alc662_mode2_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18303,7 +18243,6 @@ static void alc663_mode3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18321,7 +18260,6 @@ static void alc663_mode4_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x16; spec->automute_mixer_nid[0] = 0x0c; spec->automute_mixer_nid[1] = 0x0e; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18339,7 +18277,6 @@ static void alc663_mode5_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x16; spec->automute_mixer_nid[0] = 0x0c; spec->automute_mixer_nid[1] = 0x0e; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18356,7 +18293,6 @@ static void alc663_mode6_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18373,7 +18309,6 @@ static void alc663_mode7_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18390,7 +18325,6 @@ static void alc663_mode8_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18405,7 +18339,6 @@ static void alc663_g71v_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.line_out_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; spec->detect_line = 1; spec->automute_lines = 1; @@ -19818,7 +19751,6 @@ static void alc680_base_setup(struct hda_codec *codec) spec->autocfg.inputs[0].type = AUTO_PIN_MIC; spec->autocfg.inputs[1].pin = 0x19; spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ASoC: Correct WM8994 MICBIAS supply widget hookup
by Mark Brown 14 Jul '11

14 Jul '11

The WM8994 and WM8958 series of devices have two MICBIAS supplies rather than one, the current widget actually manages the microphone detection control register bit (which is managed separately by the relevant API). Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2 widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 79dee89..414f817 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1206,7 +1206,6 @@ SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), SND_SOC_DAPM_INPUT("Clock"), -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0), SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, SND_SOC_DAPM_PRE_PMU), @@ -1517,8 +1516,10 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF2DACDAT", NULL, "AIF1DACDAT" }, { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, - { "MICBIAS", NULL, "CLK_SYS" }, - { "MICBIAS", NULL, "MICBIAS Supply" }, + { "MICBIAS1", NULL, "CLK_SYS" }, + { "MICBIAS1", NULL, "MICBIAS Supply" }, + { "MICBIAS2", NULL, "CLK_SYS" }, + { "MICBIAS2", NULL, "MICBIAS Supply" }, }; static const struct snd_soc_dapm_route wm8994_intercon[] = { -- 1.7.5.4

1 0

[alsa-devel] [RFC PATCH] ASoC: codecs: add codec driver for MFD mc13783
by Jürgen Lambrecht 14 Jul '11

14 Jul '11

Sascha Hauer's driver from Pengutronix' imx-sound branch. Added some code found in a linux 2.6.33. Needs further work and cleanup. Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 781 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 29 ++ 4 files changed, 816 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..77fb7b2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_MC13783 if SPI select SND_SOC_PCM3008 select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -200,6 +201,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_MC13783 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..a129af2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-mc13783-objs := mc13783.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c new file mode 100644 index 0000000..af7603a --- /dev/null +++ b/sound/soc/codecs/mc13783.c @@ -0,0 +1,781 @@ +/* + * Copyright 2008 Juergen Beisert, kernel(a)pengutronix.de + * Copyright 2009 Sascha Hauer, s.hauer(a)pengutronix.de + * + * Initial development of this code was funded by + * Phytec Messtechnik GmbH, http://www.phytec.de + * + * 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. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ +#include <linux/device.h> +#include <linux/mfd/mc13783.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "mc13783.h" + +#define PMIC_AUDIO_RX_0 36 +#define PMIC_AUDIO_RX_1 37 +#define PMIC_AUDIO_TX 38 +#define PMIC_SSI_NETWORK 39 +#define PMIC_AUDIO_CODEC 40 +#define PMIC_AUDIO_DAC 41 + +#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2) +#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4) +#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6) +#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8) +#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10) +#define SSI_NETWORK_CDCFSDLY(x) (1 << 11) +#define SSI_NETWORK_STDCSLOT(x) (((x) & 0x3) << 12) +#define SSI_NETWORK_STDCRXSLOT(x) (((x) & 0x3) << 14) +#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16) +#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18) +#define SSI_NETWORK_STDCSUMGAIN (1 << 20) + +#define STEREO_DAC_STD_SSI_SEL (1 << 0) +#define STEREO_DAC_STD_CLK_SEL (1 << 1) +#define STEREO_DAC_STD_CSM (1 << 2) +#define STEREO_DAC_STD_BCL_INV (1 << 3) +#define STEREO_DAC_STD_CFS_INV (1 << 4) +#define STEREO_DAC_STD_CFS(x) (((x) & 0x3) << 5) +#define STEREO_DAC_STD_CLK(x) (((x) & 0x7) << 7) +#define STEREO_DAC_STD_CFS_DLY_B (1 << 10) +#define STEREO_DAC_STD_C_EN (1 << 11) +#define STEREO_DAC_STD_C_CLK_EN (1 << 12) +#define STEREO_DAC_STD_C_RESET (1 << 15) +#define STEREO_DAC_SPDIF (1 << 16) +#define STEREO_DAC_SR(x) (((x) & 0xf) << 17) + +#define RX_0_VAUDIOON 0 +#define RX_0_BIASEN 1 +#define RX_0_BIASSPEED 2 +#define RX_0_ASPEN 3 +#define RX_0_ASPSEL 4 +#define RX_0_ALSPEN 5 +#define RX_0_ALSPREF 6 +#define RX_0_ALSPSEL 7 +#define RX_0_LSPLEN 8 +#define RX_0_AHSREN 9 +#define RX_0_AHSLEN 10 +#define RX_0_AHSSEL 11 +#define RX_0_HSPGDIG 12 +#define RX_0_HSDETEN 13 +#define RX_0_HSDETAUTOB 14 +#define RX_0_ARXOUTREN 15 +#define RX_0_ARXOUTLEN 16 +#define RX_0_ARXOUTSEL 17 +#define RX_0_CDCOUTEN 18 +#define RX_0_HSLDETEN 19 +#define RX_0_ADDCDC 21 +#define RX_0_ADDSTDC 22 +#define RX_0_ADDRXIN 23 + +#define RX_1_PGASTEN 5 +#define RX_1_PGAST 6 +#define RX_1_ARXINEN 10 +#define RX_1_ARXIN 11 +#define RX_1_PGARXIN 12 + +#define MC13783_REG_CACHE_SIZE 42 + +struct mc13783_priv { + struct snd_soc_codec codec; + struct mc13783 *mc13783; + + u32 reg_cache[MC13783_REG_CACHE_SIZE]; + + int mc13783_asp_val; + int mc13783_alsp_val; + int mc13783_ahsx_val; +}; + +static unsigned int mc13783_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u32 ret; + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + if (reg >= MC13783_REG_CACHE_SIZE) { + ret = 0; + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); + } else { + ret = priv->reg_cache[reg]; + } + return ret; +} + +static int mc13783_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + if (reg >= MC13783_REG_CACHE_SIZE) { + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); + ret = -1; + } else { + mc13783_lock(priv->mc13783); + priv->reg_cache[reg] = value; + + ret = mc13783_reg_write(priv->mc13783, reg, value); + + mc13783_unlock(priv->mc13783); + } + return ret; +} + +/* Mapping between sample rates and register value */ +static unsigned int mc13783_rates[] = { + 8000, 11025, 12000, 16000, + 22050, 24000, 32000, 44100, + 48000, 64000, 96000 +}; + +static int mc13783_pcm_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; + unsigned int rate = params_rate(params); + int i; + unsigned int reg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) { + if (rate == mc13783_rates[i]) { + snd_soc_update_bits(codec, PMIC_AUDIO_DAC, + 0xf << 17, i << 17); + return 0; + } + } + } else { + if (rate == 8000) + reg = 0; + else + reg = (1 << 10); + + snd_soc_update_bits(codec, PMIC_AUDIO_CODEC, + 0x1 << 10, reg); + + return 0; + } + + return -EINVAL; +} + +static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg; + + if (dai->id == MC13783_ID_STEREO_DAC) + reg = mc13783_read(codec, PMIC_AUDIO_DAC); + else + reg = mc13783_read(codec, PMIC_AUDIO_CODEC); + + reg &= ~STEREO_DAC_STD_CFS(3); + reg &= ~STEREO_DAC_STD_BCL_INV; + reg &= ~STEREO_DAC_STD_CFS_INV; + reg &= ~STEREO_DAC_STD_CSM; + reg &= ~STEREO_DAC_STD_C_CLK_EN; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg |= STEREO_DAC_STD_CFS(2); + break; + case SND_SOC_DAIFMT_DSP_A: + reg |= STEREO_DAC_STD_CFS(1); + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + reg |= STEREO_DAC_STD_BCL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + reg |= STEREO_DAC_STD_BCL_INV; + reg |= STEREO_DAC_STD_CFS_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + reg |= STEREO_DAC_STD_CFS_INV; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg |= STEREO_DAC_STD_C_CLK_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg |= STEREO_DAC_STD_CSM; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + } + + reg |= STEREO_DAC_STD_C_EN; /* DAC power up */ + reg |= STEREO_DAC_STD_C_RESET; /* reset filter */ + + if (dai->id == MC13783_ID_STEREO_DAC) + mc13783_write(codec, PMIC_AUDIO_DAC, reg); + else + mc13783_write(codec, PMIC_AUDIO_CODEC, reg); + + return 0; +} + +static int mc13783_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + int clk; + unsigned int reg; + + if (dai->id == MC13783_ID_STEREO_DAC) + reg = mc13783_read(codec, PMIC_AUDIO_DAC); + else + reg = mc13783_read(codec, PMIC_AUDIO_CODEC); + + reg &= ~STEREO_DAC_STD_CLK(0x7); + reg &= ~STEREO_DAC_STD_CLK_SEL; + + switch (freq) { + case 13000000: + clk = 0; + break; + case 15360000: + clk = 1; + break; + case 16800000: + clk = 2; + break; + case 26000000: + clk = 4; + break; + case 12000000: + clk = 5; + break; + case 3686400: + clk = 6; + break; + case 33600000: + clk = 7; + break; + default: + return -EINVAL; + } + + if (clk_id == MC13783_CLK_CLIB) + reg |= STEREO_DAC_STD_CLK_SEL; + + reg |= STEREO_DAC_STD_CLK(clk); + + if (dai->id == MC13783_ID_STEREO_DAC) + mc13783_write(codec, PMIC_AUDIO_DAC, reg); + else + mc13783_write(codec, PMIC_AUDIO_CODEC, reg); + + return 0; +} + +static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg; + + if (slots != 4) + return -EINVAL; + + reg = mc13783_read(codec, PMIC_SSI_NETWORK); + + reg &= ~(0xfff << 0); + reg |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ + reg |= (0x01 << 4); /* secondary timeslot TX is 1 */ + reg |= (0x01 << 6); /* secondary timeslot RX is 1 */ + + mc13783_write(codec, PMIC_SSI_NETWORK, reg); + + return 0; +} + +static int mc13783_get_alsp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = priv->mc13783_alsp_val; + + return 0; +} + +static int mc13783_put_alsp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + priv->mc13783_alsp_val = ucontrol->value.integer.value[0]; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + reg &= ~((1 << RX_0_ALSPEN) | (1 << RX_0_ALSPSEL)); + + if (priv->mc13783_alsp_val) + reg |= 1 << RX_0_ALSPEN; + + if (priv->mc13783_alsp_val == 2) + reg |= 1 << RX_0_ALSPSEL; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; +} + +static int mc13783_get_ahsx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = priv->mc13783_ahsx_val; + + return 0; +} + +static int mc13783_put_ahsx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + priv->mc13783_ahsx_val = ucontrol->value.integer.value[0]; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + reg &= ~((1 << RX_0_AHSREN) | (1 << RX_0_AHSLEN) | (1 << RX_0_AHSSEL)); + + if (priv->mc13783_ahsx_val) + reg |= (1 << RX_0_AHSREN) | (1 << RX_0_AHSLEN); + + if (priv->mc13783_ahsx_val == 2) + reg |= 1 << RX_0_AHSSEL; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; + } + +static int mc13783_pcm_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = mc13783_read(codec, PMIC_AUDIO_RX_0); + ucontrol->value.enumerated.item[0] = (val >> RX_0_ADDSTDC) & 1; + printk(KERN_DEBUG "pcm_get %d\n", ucontrol->value.enumerated.item[0]); + + return 0; +} + +static int mc13783_pcm_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r36, r37; + + r36 = mc13783_read(codec, PMIC_AUDIO_RX_0); + r37 = mc13783_read(codec, PMIC_AUDIO_RX_1); + + r36 &= ~(1 << RX_0_ADDSTDC); + r37 &= ~(1 << RX_1_PGASTEN); + + if (ucontrol->value.enumerated.item[0]) { + printk(KERN_DEBUG "pcm_put on %d\n", ucontrol->value.enumerated.item[0]); + r36 |= (1 << RX_0_ADDSTDC); + r37 |= (1 << RX_1_PGASTEN); + } else { + printk(KERN_DEBUG "pcm_put off\n"); + r36 &= ~(1 << RX_0_ADDSTDC); + r37 &= ~(1 << RX_1_PGASTEN); + } + + mc13783_write(codec, PMIC_AUDIO_RX_0, r36); + mc13783_write(codec, PMIC_AUDIO_RX_1, r37); + + return 0; +} + +static int mc13783_linein_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = mc13783_read(codec, PMIC_AUDIO_RX_0); + ucontrol->value.enumerated.item[0] = (val >> RX_0_ADDRXIN) & 1; + + return 0; +} + +static int mc13783_linein_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r36, r37; + + r36 = mc13783_read(codec, PMIC_AUDIO_RX_0); + r37 = mc13783_read(codec, PMIC_AUDIO_RX_1); + + r36 &= ~(1 << RX_0_ADDRXIN); + r37 &= ~(1 << RX_1_ARXINEN); + + if (ucontrol->value.enumerated.item[0]) { + r36 |= (1 << RX_0_ADDRXIN); + r37 |= (1 << RX_1_ARXINEN); + } else { + r36 &= ~(1 << RX_0_ADDRXIN); + r37 &= ~(1 << RX_1_ARXINEN); + } + + mc13783_write(codec, PMIC_AUDIO_RX_0, r36); + mc13783_write(codec, PMIC_AUDIO_RX_1, r37); + + return 0; +} + +static int mc13783_capure_cache; + +static int mc13783_get_capture(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = mc13783_capure_cache; + return 0; +} + +#define AMC1REN (1 << 5) +#define AMC2EN (1 << 9) +#define ATXINEN (1 << 11) +#define RXINREC (1 << 13) +#define AMC1LEN (1 << 7) + +static int mc13783_put_capture(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r38, change; + + r38 = mc13783_read(codec, PMIC_AUDIO_TX); + + change = (mc13783_capure_cache != ucontrol->value.enumerated.item[0]); + mc13783_capure_cache = ucontrol->value.enumerated.item[0]; + r38 &= ~(AMC1REN | AMC2EN | ATXINEN | RXINREC | AMC1LEN); + + switch (mc13783_capure_cache) { + case 0: + break; + case 1: + r38 |= RXINREC; + break; + case 2: + r38 |= AMC1REN | AMC1LEN; + break; + case 3: + r38 |= AMC1REN; + break; + case 4: + r38 |= AMC2EN; + break; + case 5: + r38 |= AMC1LEN | AMC2EN; + break; + case 6: + r38 |= ATXINEN; + break; + case 7: + r38 |= AMC1LEN | ATXINEN; + break; + case 8: + r38 |= AMC1LEN | RXINREC; + break; + case 9: + r38 |= AMC1LEN; + break; + default: + break; + } + + mc13783_write(codec, PMIC_AUDIO_TX, r38); + + return change; +} + +static const char *mc13783_asp[] = {"Off", "Codec", "Right"}; +static const char *mc13783_alsp[] = {"Off", "Codec", "Right"}; + +static const char *mc13783_ahs[] = {"Codec", "Mixer"}; + +//static const char *mc13783_ahsx[] = {"Off", "Codec", "Right"}; + +static const char *mc13783_arxout[] = {"Codec", "Mixer"}; + +static const char *mc13783_capture[] = {"off/off", "rxinl/rxinr", + "mc1lin/mc1rin", "off/mc1rin", "off/mc2in", "mc1lin/mc2in", + "off/txin", "mc1lin/txin", "mc1lin/rxinr", "mc1lin/off"}; + +static const char *mc13783_3d_mixer[] = {"Stereo", "Phase Mix", + "Mono", "Mono Mix"}; + +//static const char *mc13783_pcms[] = {"Off", "On"}; + +static const struct soc_enum mc13783_enum_asp = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 3, ARRAY_SIZE(mc13783_asp), mc13783_asp); + +static const struct soc_enum mc13783_enum_alsp = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mc13783_alsp), mc13783_alsp); + +static const struct soc_enum mc13783_enum_ahs = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 11, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_arxout = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 17, ARRAY_SIZE(mc13783_arxout), + mc13783_arxout); + +static const struct soc_enum mc13783_enum_capture = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mc13783_capture), mc13783_capture); + +static const struct soc_enum mc13783_enum_3d_mixer = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_1, 16, ARRAY_SIZE(mc13783_3d_mixer), + mc13783_3d_mixer); + +//static const struct soc_enum mc13783_enum_ahsx = +// SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 9, ARRAY_SIZE(mc13783_ahsx), +// mc13783_ahsx); +// +//static const struct soc_enum mc13783_enum_pcms = +// SOC_ENUM_SINGLE(PMIC_AUDIO_RX_1, 5, ARRAY_SIZE(mc13783_pcms), +// mc13783_pcms); + +static struct snd_kcontrol_new mc13783_control_list[] = { + /* Output Routing */ + SOC_ENUM("Asp Source", mc13783_enum_asp), + SOC_ENUM_EXT("Alsp Source", mc13783_enum_alsp, mc13783_get_alsp, + mc13783_put_alsp), + SOC_ENUM("Ahs Source", mc13783_enum_ahs), + SOC_SINGLE("Ahsr enable", PMIC_AUDIO_RX_0, 9, 1, 0), + SOC_SINGLE("Ahsl enable", PMIC_AUDIO_RX_0, 10, 1, 0), +/* instead of the above 3 lines, could put this line: */ +// SOC_ENUM_EXT("Ahs Source", mc13783_enum_ahsx, mc13783_get_ahsx, +// mc13783_put_ahsx), +//// SOC_ENUM("Ahs Source", mc13783_enum_ahs), +//// SOC_SINGLE("Ahsr enable", PMIC_AUDIO_RX_0, 9, 2, 0), +//// SOC_SINGLE("Ahsl enable", PMIC_AUDIO_RX_0, 10, 2, 0), +//// SOC_ENUM("Ahsr enable", mc13783_enum_ahsr), +//// SOC_ENUM("Ahsl enable", mc13783_enum_ahsl), + SOC_ENUM("Arxout Source", mc13783_enum_arxout), + SOC_SINGLE("ArxoutR enable", PMIC_AUDIO_RX_0, 16, 1, 0), + SOC_SINGLE("ArxoutL enable", PMIC_AUDIO_RX_0, 15, 1, 0), + SOC_SINGLE_EXT("PCM Playback Switch", 0, 0, 1, 0, mc13783_pcm_get, +// SOC_ENUM_EXT("PCMS Playback Switch", mc13783_enum_pcms, mc13783_pcm_get, + mc13783_pcm_put), + SOC_SINGLE("PCM Playback Volume", PMIC_AUDIO_RX_1, 6, 15, 0), +// SOC_SINGLE("PCMP Playback Volume", PMIC_AUDIO_RX_1, RX_1_PGAST, 15, 0), + SOC_SINGLE_EXT("Line in Switch", 0, 0, 1, 0, mc13783_linein_get, + mc13783_linein_put), + SOC_SINGLE("Line in Volume", PMIC_AUDIO_RX_1, 12, 15, 0), +// SOC_SINGLE("Line in Volume", PMIC_AUDIO_RX_1, RX_1_PGARXIN, 15, 0), + SOC_ENUM_EXT("Capture Source", mc13783_enum_capture, mc13783_get_capture, + mc13783_put_capture), + SOC_DOUBLE("PCM Capture Volume", PMIC_AUDIO_TX, 19, 14, 31, 0), + SOC_ENUM("3D Control", mc13783_enum_3d_mixer), +}; + +static int mc13783_probe(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int i, ret = 0; + + /* these are the reset values */ + priv->reg_cache[PMIC_AUDIO_RX_0] = (1 << RX_0_HSPGDIG); //0x001000; + priv->reg_cache[PMIC_AUDIO_RX_1] = 0x00d35A; + priv->reg_cache[PMIC_AUDIO_TX] = 0x420000; + priv->reg_cache[PMIC_SSI_NETWORK] = 0x013060; + priv->reg_cache[PMIC_AUDIO_CODEC] = 0x180027; + priv->reg_cache[PMIC_AUDIO_DAC] = 0x0e0004; + + /* VAUDIOON -> supply audio part, BIAS enable */ + priv->reg_cache[PMIC_AUDIO_RX_0] |= (1 << RX_0_VAUDIOON) | (1 << RX_0_BIASEN); //0x3; + + for (i = 36; i < 42; i++) + mc13783_write(codec, i, priv->reg_cache[i]); + + snd_soc_add_controls(codec, mc13783_control_list, + ARRAY_SIZE(mc13783_control_list)); + + return ret; +} + +static int mc13783_remove(struct snd_soc_codec *codec) +{ + unsigned int reg; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + /* VAUDIOON -> switch off audio part, BIAS disable */ + reg &= ~0x3; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; +} + +#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) + +#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) +/*Other functions (soc_dai.h): + * set_pll, set_clkdiv + * set_tdm_slot, set_channel_map, set_tristate + * digital_mute + * startup, shutdown, hw_free, prepare, trigger +*/ +static struct snd_soc_dai_ops mc13783_ops = { + .hw_params = mc13783_pcm_hw_params, + .set_fmt = mc13783_set_fmt, + .set_sysclk = mc13783_set_sysclk, + .set_tdm_slot = mc13783_set_tdm_slot_codec, +}; + +static struct snd_soc_dai_driver mc13783_dai = { + .name = "mc13783-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops, +}; + +/* + * ASoC codec device structure + * + * Assign this variable to the codec_dev field of the machine driver's + * snd_soc_device structure. + */ +static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { + .probe = mc13783_probe, + .remove = mc13783_remove, +// .suspend +// .resume + .read = mc13783_read, + .write = mc13783_write, +// .display_register +// .volatile_register +// .readable_register +// .reg_cache_size = MC13783_REG_CACHE_SIZE, TODO: use codec->reg_cache instead of privately +// reg_cache_step +// .reg_word_size = sizeof(u32), TODO: else: kmemdup error in soc-core.c::snd_soc_register_codec +// .reg_cache_default = wm8960_reg, +// .compress_type +// .set_bias_level +}; + +static int mc13783_codec_probe(struct platform_device *pdev) +{ + struct mc13783 *mc13783; + struct mc13783_priv *priv; + int ret; + + mc13783 = dev_get_drvdata(pdev->dev.parent); + + priv = kzalloc(sizeof(struct mc13783_priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + priv->mc13783 = mc13783; + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, &mc13783_dai, 1); + if (ret) + goto err_register_codec; + + return 0; + +err_register_codec: + dev_err(&pdev->dev, "register codec failed with %d\n", ret); + kfree(priv); + + return ret; +} + +static int mc13783_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver mc13783_codec_driver = { + .driver = { + .name = "mc13783-codec", + .owner = THIS_MODULE, + }, + .probe = mc13783_codec_probe, + .remove = __devexit_p(mc13783_codec_remove), +}; + +static __init int mc13783_init(void) +{ + return platform_driver_register(&mc13783_codec_driver); +} + +static __exit void mc13783_exit(void) +{ + platform_driver_unregister(&mc13783_codec_driver); +} + +module_init(mc13783_init); +module_exit(mc13783_exit); + +MODULE_DESCRIPTION("ASoC MC13783 driver"); +MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer(a)pengutronix.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h new file mode 100644 index 0000000..7285370 --- /dev/null +++ b/sound/soc/codecs/mc13783.h @@ -0,0 +1,29 @@ +/* + * Copyright 2008 Juergen Beisert, kernel(a)pengutronix.de + * + * 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. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + */ + +#ifndef MC13783_MIXER_H +#define MC13783_MIXER_H + +extern int mc13783_add_ctl(struct snd_card*, void *); + +#define MC13783_CLK_CLIA 1 +#define MC13783_CLK_CLIB 2 + +#define MC13783_ID_STEREO_DAC 1 +#define MC13783_ID_STEREO_CODEC 2 + +#endif /* MC13783_MIXER_H */ -- 1.7.1

4 3

[alsa-devel] [PATCH] ASoC: Reduce power consumption for idle DAIs in WM8994
by Mark Brown 14 Jul '11

14 Jul '11

If DAIs are idle but their clocks are in use for some reason (eg, as SYSCLK or for accessory detect) then set the clock dividers to the maximum to reduce slightly the power consumption of the unclocked circuits. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 29 +++++++++++++++++++++++++++++ 1 files changed, 29 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 31854dc..bc62b6e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2311,6 +2311,33 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); } +static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + int rate_reg = 0; + + switch (dai->id) { + case 1: + rate_reg = WM8994_AIF1_RATE; + break; + case 2: + rate_reg = WM8994_AIF1_RATE; + break; + default: + break; + } + + /* If the DAI is idle then configure the divider tree for the + * lowest output rate to save a little power if the clock is + * still active (eg, because it is system clock). + */ + if (rate_reg && !dai->playback_active && !dai->capture_active) + snd_soc_update_bits(codec, rate_reg, + WM8994_AIF1_SR_MASK | + WM8994_AIF1CLK_RATE_MASK, 0x9); +} + static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -2377,6 +2404,7 @@ static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, + .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2386,6 +2414,7 @@ static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, + .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, -- 1.7.5.4

1 0

Re: [alsa-devel] [PATCH] ASoC: wm8900: fix a memory leak if wm8900_set_fll fails
by Mark Brown 14 Jul '11

14 Jul '11

On Thu, Jul 14, 2011 at 06:14:46PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks

1 0

[alsa-devel] www.alsa-project.org down?
by David Henningsson 14 Jul '11

14 Jul '11

Hi, Is www.alsa-project.org down? Someone in a bug report he couldn't connect to it and here's what happens when I try: $ wget http://www.alsa-project.org --2011-07-08 11:41:36-- http://www.alsa-project.org/ Resolving www.alsa-project.org... 77.48.224.243 Connecting to www.alsa-project.org|77.48.224.243|:80... connected. HTTP request sent, awaiting response... (It's been awaiting response for a few minutes now...) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 2

Re: [alsa-devel] Specifying the default buffer-size for snd-hda-intel
by Raymond Yau 14 Jul '11

14 Jul '11

> +config SND_HDA_PREALLOC_SIZE > + int "Pre-allocated buffer size for HD-audio driver" > + range 0 32768 > + default 64 > + help > + Speficies the default pre-allocated buffer-size in kB for > + HD-audio driver. A larger buffer (e.g. 2048) is preferred > + for systems with PulseAudio. The default 64 is chosen just > + from the compatibility reason. > + + range 0 32768 - range 64 32768 It seem that the minimum range cannot be zero Does current PA server still work with sound drivers with 64K buffer ?

2 1

[alsa-devel] [PATCH] ASoC: Log WM8994 FIFO errors from the interrupt
by Mark Brown 14 Jul '11

14 Jul '11

We should spot them anyway on state changes but logging them gives us better time information about when the misconfiguration happened. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index fcd067f..d774fd7a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2894,6 +2894,15 @@ out: return IRQ_HANDLED; } +static irqreturn_t wm8994_fifo_error(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "FIFO error\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -2973,6 +2982,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, + wm8994_fifo_error, "FIFO error", codec); + ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", &wm8994->hubs); @@ -3255,6 +3267,7 @@ err_irq: &wm8994->fll_locked[i]); wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); err: kfree(wm8994); return ret; @@ -3276,6 +3289,7 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); switch (control->type) { case WM8994: -- 1.7.5.4

1 0

[alsa-devel] ASoC updates for 3.1
by Mark Brown 14 Jul '11

14 Jul '11

The following changes since commit e320bc42be746b8f006d163b6e340e5fb79ce7b2: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-09 11:43:04 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Giridhar Maruthy (1): ASoC: SAMSUNG: 24-bit audio playback on Exynos4210 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Johannes Stezenbach (2): ASoC: STA32x: Add mixer controls for biquad coefficients ASoC: STA32x: Preserve reserved register bits Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (9): Merge branch 'for-3.0' into for-3.1 Merge branch 'for-3.0' into for-3.1 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-3.1 ASoC: Conditionalize the enable of WM8994 ADC TDM mode ASoC: Use late enable handling for direct voice, speaker and headphone ASoC: Implement DC servo completion IRQ handling for wm_hubs devices ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958 ASoC: Use WM8994 FLL lock interrupt ASoC: Don't warn on low WM8994/58 AIFnCLKs Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration Takashi Iwai (4): Merge branch 'for-3.0' of git://git.kernel.org/.../lrg/asoc-2.6 into fix/asoc Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc ASoC: Don't set invalid name string to snd_card->driver field Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc sound/soc/codecs/sta32x.c | 140 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic26.c | 14 +++- sound/soc/codecs/tlv320aic3x.c | 9 ++- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 101 +++++++++++++++++++++++----- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_hubs.c | 37 ++++++++-- sound/soc/codecs/wm_hubs.h | 8 ++ sound/soc/samsung/smdk_wm8994.c | 5 +- sound/soc/sh/fsi-ak4642.c | 16 ++-- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 +- sound/soc/soc-core.c | 5 +- 13 files changed, 301 insertions(+), 45 deletions(-)

1 0

Re: [alsa-devel] hda-intel front headphone (8086:27d8 1043:83bc)
by Raymond Yau 13 Jul '11

13 Jul '11

> My knowledge of audio/alsa programming is really poor, but if someone suggest > a patch I can setup the thing and do some debug. Try Takashi 's daily snapshot tarball you should able to get "Headphone Playback Volume" after this patch "ALSA: hda - Code consolidation for ALC88x and ALC662 auto-parsers" using hda-emu and it seem the driver support 6 channels even when the pink and blue jacks are not retasked

2 3

[alsa-devel] ASoC updates for 3.0
by Mark Brown 13 Jul '11

13 Jul '11

The following changes since commit 18361bbe3180eca62796188d62aefac1519f4c83: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-07-09 09:44:09 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name sound/soc/sh/fsi-ak4642.c | 16 ++++++++-------- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 ++-- 3 files changed, 11 insertions(+), 11 deletions(-)

2 1

[alsa-devel] [PATCH 0/2] ASoC: STA32x: filter coefficients, bug fix
by Johannes Stezenbach 13 Jul '11

13 Jul '11

Hi, here are two patches for the ST STA32x ASoC codec driver. The first one adds SNDRV_CTL_ELEM_TYPE_BYTES controls for access to the biquad filter coefficients. I haven't found any other driver using SNDRV_CTL_ELEM_TYPE_BYTES in this way, but it works nicely. The coefficients can be set with amixer (although amixer has a quirk in that it prints the values in hex but can read them only as decimal). # amixer cget name='Ch1 - Biquad 1' numid=43,iface=MIXER,name='Ch1 - Biquad 1' ; type=BYTES,access=rw------,values=15 : values=0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00 # amixer cset name='Ch1 - Biquad 1' 12,34,0,0,0,0,0,0,0,0,0,0,0,255,78 numid=43,iface=MIXER,name='Ch1 - Biquad 1' ; type=BYTES,access=rw------,values=15 : values=0x0c,0x22,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0xff,0x4e Due to the way the internal coeffcient RAM is accessed, the corresponding I2C registers are volatile, thus I added the .volatile_register function in this patch. The second patch preserves reset values as required by the chip documentation, it is mainly for future compatibility and was added during hardware testing to rule out issues from this side, although it did not make a difference in my tests. However, the chip spec insists it is important. Thanks Johannes

2 3