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

[alsa-devel] ASoC fixes for v3.12
by Mark Brown 30 Oct '13

30 Oct '13

The following changes since commit 959f58544b7f20c92d5eb43d1232c96c15c01bfb: Linux 3.12-rc7 (2013-10-27 16:12:03 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v3.12-rc7 for you to fetch changes up to 8723b795aaa0c5f0b9db2099af575f5e089a63f1: Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus (2013-10-30 10:11:55 -0700) ---------------------------------------------------------------- ASoC: Fixes for v3.12 A few of the Coverity fixes from Takashi, one of which (the wm_hubs one) is particularly noticable. ---------------------------------------------------------------- Mark Brown (1): Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus Takashi Iwai (3): ASoC: dapm: Fix source list debugfs outputs ASoC: dapm: Return -ENOMEM in snd_soc_dapm_new_dai_widgets() ASoC: wm_hubs: Add missing break in hp_supply_event() sound/soc/codecs/wm_hubs.c | 1 + sound/soc/soc-dapm.c | 4 +++- 2 files changed, 4 insertions(+), 1 deletion(-)

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[alsa-devel] [PATCH 1/2 v2] ASoC: wm_adsp: Print error when regmap reads/writes fail
by Dimitris Papastamos 30 Oct '13

30 Oct '13

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e09ac94..cd007cc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -339,8 +339,8 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_write(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to write %zu bytes to %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to write %zu bytes to %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } @@ -393,8 +393,8 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to read %zu bytes from %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to read %zu bytes from %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } @@ -1227,8 +1227,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) le32_to_cpu(blk->len)); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write to %x in %s\n", - file, blocks, reg, region_name); + "%s.%d: Failed to write to %x in %s: %d\n", + file, blocks, reg, region_name, ret); } kfree(buf); -- 1.8.4.2

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[alsa-devel] Getting ELD and available chmaps without opening device?
by Anssi Hannula 30 Oct '13

30 Oct '13

Hi all, I want to read the ELD of a HDMI sink while another playback is ongoing. I have the device name (e.g. "hdmi:CARD=NVidia,DEV=0"), and normally I'd open the device, use snd_pcm_info() to get snd_pcm_info_t with which I can use snd_pcm_info_get_card() and snd_pcm_info_get_device() to get the necessary information to access the control with snd_ctl_open(), snd_hctl_open_ctl() etc. However, since another playback is ongoing I can't open the device. Is there another way I'm missing, or do I simply have to magically know the card/device numbers (which in my case is possible with some tricks since the playback is in the same application, but I'd guess that might not always be the case)? Same affects available chmaps from what I can see. -- Anssi Hannula

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[alsa-devel] [PATCH v2] ASoC: Add pinctrl PM to components of active DAIs
by Nicolin Chen 30 Oct '13

30 Oct '13

It's quite popular that more drivers are using pinctrl PM, for example: (Documentation/devicetree/bindings/arm/primecell.txt). Just like what runtime PM does, it would de-active and en-active pin group depending on whether it's being used or not. And this pinctrl PM might be also beneficial to cpu dai drivers because they might have actual pinctrl so as to sleep their pins and wake them up as needed. To achieve this goal, this patch sets pins to the default state during resume or startup; While during suspend and shutdown, it would set pins to the sleep state. As pinctrl PM would return zero if there is no such pinctrl sleep state settings, this patch would not break current ASoC subsystem directly. [ However, there is still an exception that the patch can not handle, that is, when cpu dai driver does not have pinctrl property but another device has it. (The AUDMUX <-> SSI on Freescale i.MX6 series for example. SSI as a cpu dai doesn't contain pinctrl property while AUDMUX, an Audio Multiplexer, has it). In this case, this kind of cpu dai driver needs to find a way to obtain the pinctrl property as its own, by moving property from AUDMUX to SSI, or creating a pins link/dependency between these two devices, or using a more decent way after we figure it out. ] Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v1->v2: * Use proper verbs in comments. * Add 'dai->active' condition for resume case. --- sound/soc/soc-core.c | 29 +++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 3 +++ 2 files changed, 32 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4280c70..5c972ac 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -684,6 +684,13 @@ int snd_soc_suspend(struct device *dev) if (card->suspend_post) card->suspend_post(card); + /* deactivate pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_suspend); @@ -807,6 +814,14 @@ int snd_soc_resume(struct device *dev) if (list_empty(&card->codec_dev_list)) return 0; + /* activate pins from sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + if (cpu_dai->active) + pinctrl_pm_select_default_state(cpu_dai->dev); + } + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume @@ -1929,6 +1944,13 @@ int snd_soc_poweroff(struct device *dev) snd_soc_dapm_shutdown(card); + /* deactivate pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_poweroff); @@ -3766,6 +3788,13 @@ int snd_soc_register_card(struct snd_soc_card *card) if (ret != 0) soc_cleanup_card_debugfs(card); + /* deactivate pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6..05bdba0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -183,6 +183,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; + pinctrl_pm_select_default_state(cpu_dai->dev); pm_runtime_get_sync(cpu_dai->dev); pm_runtime_get_sync(codec_dai->dev); pm_runtime_get_sync(platform->dev); @@ -317,6 +318,7 @@ out: pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + pinctrl_pm_select_sleep_state(cpu_dai->dev); return ret; } @@ -426,6 +428,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + pinctrl_pm_select_sleep_state(cpu_dai->dev); return 0; } -- 1.8.4

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[alsa-devel] [PATCH] asoc-tools: Require bash shell when interpreting vizdapm.sh
by Jarkko Nikula 30 Oct '13

30 Oct '13

pushd and popd are built-in functions of bash shell so require it explicitly instead of relying /bin/sh. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- For git://opensource.wolfsonmicro.com/asoc-tools.git --- bin/vizdapm.sh | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/bin/vizdapm.sh b/bin/vizdapm.sh index 29d6b3c..11d1361 100755 --- a/bin/vizdapm.sh +++ b/bin/vizdapm.sh @@ -1,4 +1,4 @@ -#!/bin/sh +#!/bin/bash # # Copyright 2011 Wolfson Microelectronics plc # Author: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> -- 1.8.4.rc3

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[alsa-devel] [PATCH] ALSA: hda - Add a fixup for ASUS N76VZ
by Takashi Iwai 30 Oct '13

30 Oct '13

ASUS N76VZ needs the same fixup as N56VZ for supporting the boost speaker. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=846529 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf313bea7085..8ad554312b69 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4623,6 +4623,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), + SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_ASUS_MODE4), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- 1.8.4.1

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[alsa-devel] [b42378@freescale.com: Re: [PATCH 1/3] dma: imx-sdma: Add ssi dual fifo script support]
by Nicolin Chen 30 Oct '13

30 Oct '13

Just found that I missed Sascha's mail address in my TO list of last reply. So resend it. And sorry for the duplicated mails. ----- Forwarded message from Nicolin Chen <b42378(a)freescale.com> ----- Date: Wed, 30 Oct 2013 12:48:48 +0800 From: Nicolin Chen <b42378(a)freescale.com> Subject: Re: [PATCH 1/3] dma: imx-sdma: Add ssi dual fifo script support User-Agent: Mutt/1.5.21 (2010-09-15) Hi Sascha, On Tue, Oct 29, 2013 at 02:51:43PM +0100, Sascha Hauer wrote: > Look at drivers/dma/imx-sdma.c: > > > /** > > * struct sdma_firmware_header - Layout of the firmware image > > * > > * @magic "SDMA" > > * @version_major increased whenever layout of struct > > * sdma_script_start_addrs > > * changes. > > Can you image why this firmware has a version field? Right, it's because > it encodes the layout of struct sdma_script_start_addrs. > > As the comment clearly states you have to *increase this field* when you > add scripts. > > Obviously you missed that, as the firmware on lkml posted recently > shows: > > > 00000000: 414d4453 00000001 00000001 0000001c SDMA............ > ^^^^^^^^ > Still '1' > > > 00000010: 00000026 000000b4 0000067a 00000282 &.......z....... > > 00000020: ffffffff 00000000 ffffffff ffffffff ................ > > 00000030: ffffffff ffffffff ffffffff ffffffff ................ > > 00000040: ffffffff ffffffff 00001a6a ffffffff ........j....... > > 00000050: 000002eb 000018bb ffffffff 00000408 ................ > > 00000060: ffffffff 000003c0 ffffffff ffffffff ................ > > 00000070: ffffffff 000002ab ffffffff 0000037b ............{... > > 00000080: ffffffff ffffffff 0000044c 0000046e ........L...n... > > 00000090: ffffffff 00001800 ffffffff ffffffff ................ > > 000000a0: 00000000 00001800 00001862 00001a16 ........b....... > ^^^^^^^^^^^^^^^^^ > new script addresses introduced > > > > -#define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V1 34 > > +#define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V1 37 > > And no, this is not a bug. It's your firmware header that is buggy. > I wasn't aware that the problem is far more complicated than I thought. And thank you for telling me all this. > What you need is: > > #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V2 37 > > You (you as a company, not you as a person) knew that it was me who > created this firmware format. So it was absolutely unnecessary to create > an incompatible firmware instead of dropping me a short note. > > Please add a version check to the driver as necessary and provide a proper > firmware. > Just currently it's not easy for me to create a new proper firmware, and I's been told that besides this version number, it also lacks a decent license info. So may I just refine this patch as you suggested to add a version check and add those new scripts first? Thank you, Nicolin Chen > Sascha > > -- > Pengutronix e.K. | | > Industrial Linux Solutions | http://www.pengutronix.de/ | > Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | > Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 | > ----- End forwarded message -----

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[alsa-devel] [PATCH] ALSA: hda - Disable AA-loopback on ALC283 Chromebook
by Takashi Iwai 30 Oct '13

30 Oct '13

ALC283-based Chromebook suffers from occasional white noise, and it turned out that this comes from AA-loopback. Disable this output path by just clearing mixer_nid, then the generic parser will skip the creation of AA-loopback path. Reported-and-tested-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 894703532ab7..c516fa93ee63 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3445,6 +3445,8 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: alc283_chromebook_caps(codec); + /* Disable AA-loopback as it causes white noise */ + spec->gen.mixer_nid = 0; spec->gen.hp_automute_hook = alc283_hp_automute_hook; /* MIC2-VREF control */ /* Set to manual mode */ -- 1.8.4.1

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[alsa-devel] Getting Error While cross compiling to arm linux
by ranjith vs 30 Oct '13

30 Oct '13

hi, I want to cross compile alsa-lib to arm-linux platform.But I Cross compiled successfully but while giving make it is giving following errors ./opt/freescale/usr/local/gcc-4.6.2-glibc-2.13-linaro-multilib-2011.12/fsl-linaro-toolchain/bin/../lib/gcc/arm-fsl-linux-gnueabi/4.6.2/../../../../arm-fsl-linux-gnueabi/bin/ld: .libs/conf.o: Relocations in generic ELF (EM: 3) .libs/conf.o: could not read symbols: File in wrong format collect2: ld returned 1 exit status make[2]: *** [libasound.la] Error 1 make[2]: Leaving directory `/home/ranjithvs/Downloads/alsa-lib-1.0.27.2/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/ranjithvs/Downloads/alsa-lib-1.0.27.2/src' make: *** [all-recursive] Error 1 Regards, Ranjith

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[alsa-devel] [PATCH 0/3] Add dual-fifo mode support of i.MX ssi
by Nicolin Chen 30 Oct '13

30 Oct '13

Changelog v1: * SSI can reduce hardware overrun/underrun possibility when using dual * fifo mode. To support this mode, we need to first update sdma sciprt * list, and then enable dual fifo BIT in SSI driver, and last update DT * bindings of i.MX series. * * ! This series of patches has a direct dependency between them. When * ! applying them, we need to apply in one single branch. Otherwise, * ! it would break currect branches. Nicolin Chen (3): dma: imx-sdma: Add ssi dual fifo script support ASoC: fsl_ssi: Add dual fifo mode support ARM: dts: imx: use dual-fifo sdma script for ssi .../devicetree/bindings/dma/fsl-imx-sdma.txt | 1 + arch/arm/boot/dts/imx51.dtsi | 4 ++-- arch/arm/boot/dts/imx53.dtsi | 4 ++-- arch/arm/boot/dts/imx6qdl.dtsi | 12 +++++------ arch/arm/boot/dts/imx6sl.dtsi | 12 +++++------ drivers/dma/imx-sdma.c | 6 +++++- include/linux/platform_data/dma-imx-sdma.h | 2 ++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/fsl_ssi.c | 24 +++++++++++++++++++++- 9 files changed, 48 insertions(+), 18 deletions(-) -- 1.8.4

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[alsa-devel] [PATCH] ASoC: wm8962: Fix null pointer pdata access in I2C probe()
by Nicolin Chen 30 Oct '13

30 Oct '13

When using DT binding to pass private data, there would be Kernel panic occuring due to NULL pointer access in wm8962_i2c_probe(). Thus fix it. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2bf9ee7..7dd79c4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3616,28 +3616,28 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, 0); /* Apply static configuration for GPIOs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) - if (pdata->gpio_init[i]) { + for (i = 0; i < ARRAY_SIZE(wm8962->pdata.gpio_init); i++) + if (wm8962->pdata.gpio_init[i]) { wm8962_set_gpio_mode(wm8962, i + 1); regmap_write(wm8962->regmap, 0x200 + i, - pdata->gpio_init[i] & 0xffff); + wm8962->pdata.gpio_init[i] & 0xffff); } /* Put the speakers into mono mode? */ - if (pdata->spk_mono) + if (wm8962->pdata.spk_mono) regmap_update_bits(wm8962->regmap, WM8962_CLASS_D_CONTROL_2, WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); /* Micbias setup, detection enable and detection * threasholds. */ - if (pdata->mic_cfg) + if (wm8962->pdata.mic_cfg) regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, WM8962_MICDET_ENA | WM8962_MICDET_THR_MASK | WM8962_MICSHORT_THR_MASK | WM8962_MICBIAS_LVL, - pdata->mic_cfg); + wm8962->pdata.mic_cfg); /* Latch volume update bits */ regmap_update_bits(wm8962->regmap, WM8962_LEFT_INPUT_VOLUME, @@ -3682,7 +3682,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, } if (wm8962->irq) { - if (pdata->irq_active_low) { + if (wm8962->pdata.irq_active_low) { trigger = IRQF_TRIGGER_LOW; irq_pol = WM8962_IRQ_POL; } else { -- 1.8.4

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[alsa-devel] [PATCH] ASoC: rcar: remove original filter from rsnd_dma_init()
by Kuninori Morimoto 29 Oct '13

29 Oct '13

Remove original filter from rsnd_dma_init(), and use SH-DMA suitable filter. This new style can be used from Device Tree. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 31 ++++++++++++++++++++----------- 1 file changed, 20 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index c3d7544..1f53563 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -94,6 +94,7 @@ * */ #include <linux/pm_runtime.h> +#include <linux/shdma-base.h> #include "rsnd.h" #define RSND_RATES SNDRV_PCM_RATE_8000_96000 @@ -253,13 +254,6 @@ int rsnd_dma_available(struct rsnd_dma *dma) return !!dma->chan; } -static bool rsnd_dma_filter(struct dma_chan *chan, void *param) -{ - chan->private = param; - - return true; -} - int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int is_play, int id, int (*inquiry)(struct rsnd_dma *dma, @@ -267,7 +261,9 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int (*complete)(struct rsnd_dma *dma)) { struct device *dev = rsnd_priv_to_dev(priv); + struct dma_slave_config cfg; dma_cap_mask_t mask; + int ret; if (dma->chan) { dev_err(dev, "it already has dma channel\n"); @@ -277,15 +273,23 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - dma->slave.shdma_slave.slave_id = id; - - dma->chan = dma_request_channel(mask, rsnd_dma_filter, - &dma->slave.shdma_slave); + dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, + (void *)id, dev, + is_play ? "tx" : "rx"); if (!dma->chan) { dev_err(dev, "can't get dma channel\n"); return -EIO; } + cfg.slave_id = id; + cfg.dst_addr = 0; /* use default addr when playback */ + cfg.src_addr = 0; /* use default addr when capture */ + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + + ret = dmaengine_slave_config(dma->chan, &cfg); + if (ret < 0) + goto rsnd_dma_init_err; + dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; dma->priv = priv; dma->inquiry = inquiry; @@ -293,6 +297,11 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, INIT_WORK(&dma->work, rsnd_dma_do_work); return 0; + +rsnd_dma_init_err: + rsnd_dma_quit(priv, dma); + + return ret; } void rsnd_dma_quit(struct rsnd_priv *priv, -- 1.7.9.5

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[alsa-devel] [PATCH - alsa-lib] pcm/ioplug: Add mutex in snd_pcm_ioplug_hw_ptr_update
by Vishal Agrawal 29 Oct '13

29 Oct '13

Function snd_pcm_ioplug_hw_ptr_update can be called from many threads, we need to protect the hw_ptr. I came across this issue with GStreamer and Jackd. Function will be called from the thread doing snd_pcm_write and another would be to get the playback position for the track bar. Change in itself is very minimum, but it took me a lot of time to debug. Thanks, Vishal Agrawal

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[alsa-devel] [PATCH] ASoC: fsl_ssi: separately enable and disable TIE/RIE in trigger()
by Nicolin Chen 29 Oct '13

29 Oct '13

This patch enables Tx-related SIER_FLAGS only when direction is PLAYBACK and does same thing for CAPTURE. Also, after TRIGGER_STOP/PAUSE, it will disable SIER_xFLAGS for symmetric. [ Passed compile-test with mpc85xx_defconfig ] Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 35e2773..3797bf0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -107,11 +107,14 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) #endif /* SIER bitflag of interrupts to enable */ -#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ +#define SIER_TFLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ - CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) + CCSR_SSI_SIER_TUE1_EN) +#define SIER_RFLAGS (CCSR_SSI_SIER_RFRC_EN | CCSR_SSI_SIER_RDMAE | \ + CCSR_SSI_SIER_RIE | CCSR_SSI_SIER_ROE0_EN | \ + CCSR_SSI_SIER_ROE1_EN) + +#define SIER_FLAGS (SIER_TFLAGS | SIER_RFLAGS) /** * fsl_ssi_private: per-SSI private data @@ -560,12 +563,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (ssi_private->use_dma) - sier_bits = SIER_FLAGS; + sier_bits = SIER_TFLAGS; else sier_bits = CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN; } else { if (ssi_private->use_dma) - sier_bits = SIER_FLAGS; + sier_bits = SIER_RFLAGS; else sier_bits = CCSR_SSI_SIER_RIE | CCSR_SSI_SIER_RFF0_EN; } @@ -579,6 +582,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, else write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); + + write_ssi_mask(&ssi->sier, 0, sier_bits); break; case SNDRV_PCM_TRIGGER_STOP: @@ -591,14 +596,14 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + + write_ssi_mask(&ssi->sier, sier_bits, 0); break; default: return -EINVAL; } - write_ssi(sier_bits, &ssi->sier); - return 0; } -- 1.8.4

3 8

[alsa-devel] [PATCH 1/2] ALSA: memalloc: Make snd_{malloc|free}_dev_iram() static
by Takashi Iwai 29 Oct '13

29 Oct '13

These are used only locally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 809fd79ecad6..278248b8f22a 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -167,7 +167,7 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, * * This function requires iram phandle provided via of_node */ -void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) +static void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) { struct device *dev = dmab->dev.dev; struct gen_pool *pool = NULL; @@ -192,7 +192,7 @@ void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) * snd_free_dev_iram - free allocated specific memory from on-chip internal ram * @dmab: buffer allocation record to store the allocated data */ -void snd_free_dev_iram(struct snd_dma_buffer *dmab) +static void snd_free_dev_iram(struct snd_dma_buffer *dmab) { struct gen_pool *pool = dmab->private_data; -- 1.8.4.1

1 1

[alsa-devel] [PATCH 2/2] [trivial]doc:alsa: Fix typo in documentation/alsa
by Masanari Iida 29 Oct '13

29 Oct '13

Correct spelling typo in documentation/alsa Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- Documentation/sound/alsa/Audiophile-Usb.txt | 2 +- Documentation/sound/alsa/CMIPCI.txt | 2 +- Documentation/sound/alsa/compress_offload.txt | 6 +++--- Documentation/sound/alsa/soc/DPCM.txt | 4 ++-- Documentation/sound/alsa/soc/dapm.txt | 4 ++-- 6 files changed, 10 insertions(+), 10 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 95731a0..b8dd0df 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -616,7 +616,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. As default, snd-dummy drivers doesn't allocate the real buffers but either ignores read/write or mmap a single dummy page to all - buffer pages, in order to save the resouces. If your apps need + buffer pages, in order to save the resources. If your apps need the read/ written buffer data to be consistent, pass fake_buffer=0 option. diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index 654dd3b..e7a5ed4 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -232,7 +232,7 @@ The parameter can be given: # modprobe snd-usb-audio index=1 device_setup=0x09 * Or while configuring the modules options in your modules configuration file - (tipically a .conf file in /etc/modprobe.d/ directory: + (typically a .conf file in /etc/modprobe.d/ directory: alias snd-card-1 snd-usb-audio options snd-usb-audio index=1 device_setup=0x09 diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt index 16935c8..4e36e6e 100644 --- a/Documentation/sound/alsa/CMIPCI.txt +++ b/Documentation/sound/alsa/CMIPCI.txt @@ -87,7 +87,7 @@ with 4 channels, and use the interleaved 4 channel data. -There are some control switchs affecting to the speaker connections: +There are some control switches affecting to the speaker connections: "Line-In Mode" - an enum control to change the behavior of line-in jack. Either "Line-In", "Rear Output" or "Bass Output" can diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index fd74ff2..630c492 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -217,12 +217,12 @@ Not supported: would be enabled with ALSA kcontrols. - Audio policy/resource management. This API does not provide any - hooks to query the utilization of the audio DSP, nor any premption + hooks to query the utilization of the audio DSP, nor any preemption mechanisms. -- No notion of underun/overrun. Since the bytes written are compressed +- No notion of underrun/overrun. Since the bytes written are compressed in nature and data written/read doesn't translate directly to - rendered output in time, this does not deal with underrun/overun and + rendered output in time, this does not deal with underrun/overrun and maybe dealt in user-library Credits: diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt index aa8546f..0110180 100644 --- a/Documentation/sound/alsa/soc/DPCM.txt +++ b/Documentation/sound/alsa/soc/DPCM.txt @@ -192,7 +192,7 @@ This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets the "no_pcm" flag to mark it has a BE and sets flags for supported stream directions using "dpcm_playback" and "dpcm_capture" above. -The BE has also flags set for ignoreing suspend and PM down time. This allows +The BE has also flags set for ignoring suspend and PM down time. This allows the BE to work in a hostless mode where the host CPU is not transferring data like a BT phone call :- @@ -328,7 +328,7 @@ The host can control the hostless link either by :- between both DAIs. 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM - graph. Control is then carried out by the FE as regualar PCM operations. + graph. Control is then carried out by the FE as regular PCM operations. This method gives more control over the DAI links, but requires much more userspace code to control the link. Its recommended to use CODEC<->CODEC unless your HW needs more fine grained sequencing of the PCM ops. diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 7dfd88c..6faab48 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -30,7 +30,7 @@ There are 4 power domains within DAPM machine driver and responds to asynchronous events e.g when HP are inserted - 3. Path domain - audio susbsystem signal paths + 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. @@ -64,7 +64,7 @@ Audio DAPM widgets fall into a number of types:- o Speaker - Speaker o Supply - Power or clock supply widget used by other widgets. o Regulator - External regulator that supplies power to audio components. - o Clock - External clock that supplies clock to audio componnents. + o Clock - External clock that supplies clock to audio components. o AIF IN - Audio Interface Input (with TDM slot mask). o AIF OUT - Audio Interface Output (with TDM slot mask). o Siggen - Signal Generator. -- 1.8.4.1.600.g3d092bf

2 1

[alsa-devel] [PATCH v2 4/6] ASoC: sirf-soc-inner: add drivers for both CPU and Codec DAIs
by Barry Song 29 Oct '13

29 Oct '13

From: Rongjun Ying <Rongjun.Ying(a)csr.com> there is an internal codec embedded in the SiRF SoC. this is not a typical user scenerios of ASoC. but we can still get benefit by sharing platform DMA codes instead of implementing a pure ALSA driver. This driver adds DAI drivers for this internal codec. The features of Internal Codec Controller include: Support two channel 16-bit resolution playback with fix 48KHz sample rate Support two channel 16-bit resolution record with fix 48KHz sample rate Use dedicated Internal Codec TXFIFO and Internal Codec RXFIFO Supports two DMA channels for Internal Codec TXFIFO and Internal Codec RXFIFO Signed-off-by: Rongjun Ying <Rongjun.Ying(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- -v2: use DAPM and runtime PM to replace legacy kcontrol sound/soc/sirf/Kconfig | 3 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-soc-inner.c | 674 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 679 insertions(+) create mode 100644 sound/soc/sirf/sirf-soc-inner.c diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig index 5e395d3..afa3952 100644 --- a/sound/soc/sirf/Kconfig +++ b/sound/soc/sirf/Kconfig @@ -6,5 +6,8 @@ config SND_SIRF_SOC config SND_SOC_SIRF_I2S tristate +config SND_SIRF_SOC_INNER + tristate + config SND_SOC_SIRF_USP tristate diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile index 630c9be..8517c67 100644 --- a/sound/soc/sirf/Makefile +++ b/sound/soc/sirf/Makefile @@ -1,7 +1,9 @@ snd-soc-sirf-objs := sirf-pcm.o +snd-soc-sirf-soc-inner-objs := sirf-soc-inner.o snd-soc-sirf-i2s-objs := sirf-i2s.o snd-soc-sirf-usp-objs := sirf-usp.o obj-$(CONFIG_SND_SIRF_SOC) += snd-soc-sirf.o +obj-$(CONFIG_SND_SIRF_SOC_INNER) += snd-soc-sirf-soc-inner.o obj-$(CONFIG_SND_SOC_SIRF_I2S) += snd-soc-sirf-i2s.o obj-$(CONFIG_SND_SOC_SIRF_USP) += snd-soc-sirf-usp.o diff --git a/sound/soc/sirf/sirf-soc-inner.c b/sound/soc/sirf/sirf-soc-inner.c new file mode 100644 index 0000000..fc40105 --- /dev/null +++ b/sound/soc/sirf/sirf-soc-inner.c @@ -0,0 +1,674 @@ +/* + * SiRF inner audio codec driver + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later.* + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/rtc/sirfsoc_rtciobrg.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "sirf-audio.h" + +struct sirf_soc_inner_audio_reg_bits { + u32 dig_mic_en_bits; + u32 dig_mic_freq_bits; + u32 adc14b_12_bits; + u32 firdac_hsl_en_bits; + u32 firdac_hsr_en_bits; + u32 firdac_lout_en_bits; + u32 por_bits; + u32 codec_clk_en_bits; +}; + +struct sirf_soc_inner_audio { + void __iomem *base; + struct clk *clk; + spinlock_t lock; + u32 sys_pwrc_reg_base; + struct sirf_soc_inner_audio_reg_bits *reg_bits; + u32 reg_ctrl0, reg_ctrl1; + struct platform_device *sirf_pcm_pdev; +}; + +static struct sirf_soc_inner_audio_reg_bits sirf_soc_inner_audio_reg_bits_prima2 = { + 20, 21, 22, 23, 24, 25, 26, 27, +}; + +static struct sirf_soc_inner_audio_reg_bits sirf_soc_inner_audio_reg_bits_atlas6 = { + 22, 23, 24, 25, 26, 27, 28, 29, +}; +static const char * const input_mode_mux[] = {"Single-ended", + "Differential"}; + +static const struct soc_enum sirf_inner_enum[] = { + SOC_ENUM_SINGLE(AUDIO_IC_CODEC_CTRL1, 4, 2, input_mode_mux), +}; + +static const struct snd_kcontrol_new sirf_inner_input_mode_control = + SOC_DAPM_ENUM("Route", sirf_inner_enum[0]); + +static struct snd_kcontrol_new volume_controls_atlas6[] = { + SOC_DOUBLE("Playback Volume", AUDIO_IC_CODEC_CTRL0, 21, 14, + 0x7F, 0), + SOC_DOUBLE("Capture Volume", AUDIO_IC_CODEC_CTRL1, 16, 10, + 0x3F, 0), +}; + +static struct snd_kcontrol_new volume_controls_prima2[] = { + SOC_DOUBLE("Speaker Volume", AUDIO_IC_CODEC_CTRL0, 21, 14, + 0x7F, 0), + SOC_DOUBLE("Capture Volume", AUDIO_IC_CODEC_CTRL1, 15, 10, + 0x1F, 0), +}; + +static struct snd_kcontrol_new left_input_path_controls[] = { + SOC_DAPM_SINGLE("Line left Switch", AUDIO_IC_CODEC_CTRL1, 6, 1, 0), + SOC_DAPM_SINGLE("Mic left Switch", AUDIO_IC_CODEC_CTRL1, 3, 1, 0), +}; + +static struct snd_kcontrol_new right_input_path_controls[] = { + SOC_DAPM_SINGLE("Line right Switch", AUDIO_IC_CODEC_CTRL1, 5, 1, 0), + SOC_DAPM_SINGLE("Mic right Switch", AUDIO_IC_CODEC_CTRL1, 2, 1, 0), +}; + +static struct snd_kcontrol_new left_dac_to_hp_left_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 9, 1, 0); + +static struct snd_kcontrol_new left_dac_to_hp_right_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 8, 1, 0); + +static struct snd_kcontrol_new right_dac_to_hp_left_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 7, 1, 0); + +static struct snd_kcontrol_new right_dac_to_hp_right_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 6, 1, 0); + +static struct snd_kcontrol_new left_dac_to_speaker_lineout_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 11, 1, 0); + +static struct snd_kcontrol_new right_dac_to_speaker_lineout_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 10, 1, 0); + +static int adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(codec->dev); + u32 val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: +#define PWRC_PDN_CTRL_OFFSET 0 +#define AUDIO_POWER_EN_BIT 14 + /* Enable capture power of codec*/ + val = sirfsoc_rtc_iobrg_readl(sinner_audio->sys_pwrc_reg_base + + PWRC_PDN_CTRL_OFFSET); + val |= (1 << AUDIO_POWER_EN_BIT); + sirfsoc_rtc_iobrg_writel(val, + sinner_audio->sys_pwrc_reg_base + PWRC_PDN_CTRL_OFFSET); + break; + case SND_SOC_DAPM_POST_PMU: + /*After enable adc, Delay 200ms to avoid pop noise*/ + msleep(200); + default: + break; + } + + return 0; +} + +static int hp_amp_left_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(codec->dev); + u32 val; + + val = snd_soc_read(codec, AUDIO_IC_CODEC_CTRL1); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val |= (1 << sinner_audio->reg_bits->firdac_hsl_en_bits); + break; + case SND_SOC_DAPM_POST_PMD: + val &= ~(1 << sinner_audio->reg_bits->firdac_hsl_en_bits); + default: + break; + } + snd_soc_write(codec, AUDIO_IC_CODEC_CTRL1, val); + return 0; +} + +static int hp_amp_right_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(codec->dev); + u32 val; + + val = snd_soc_read(codec, AUDIO_IC_CODEC_CTRL1); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val |= (1 << sinner_audio->reg_bits->firdac_hsr_en_bits); + break; + case SND_SOC_DAPM_POST_PMD: + val &= ~(1 << sinner_audio->reg_bits->firdac_hsr_en_bits); + default: + break; + } + snd_soc_write(codec, AUDIO_IC_CODEC_CTRL1, val); + return 0; +} + +static int speaker_output_enable_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(codec->dev); + u32 val; + + val = snd_soc_read(codec, AUDIO_IC_CODEC_CTRL1); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val |= (1 << sinner_audio->reg_bits->firdac_lout_en_bits); + break; + case SND_SOC_DAPM_POST_PMD: + val &= ~(1 << sinner_audio->reg_bits->firdac_lout_en_bits); + default: + break; + } + snd_soc_write(codec, AUDIO_IC_CODEC_CTRL1, val); + return 0; +} + +static const struct snd_soc_dapm_widget sirf_inner_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC left", NULL, AUDIO_IC_CODEC_CTRL0, 1, 0), + SND_SOC_DAPM_DAC("DAC right", NULL, AUDIO_IC_CODEC_CTRL0, 0, 0), + SND_SOC_DAPM_SWITCH("Left dac to hp left amp", SND_SOC_NOPM, 0, 0, + &left_dac_to_hp_left_amp_switch_control), + SND_SOC_DAPM_SWITCH("Left dac to hp right amp", SND_SOC_NOPM, 0, 0, + &left_dac_to_hp_right_amp_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to hp left amp", SND_SOC_NOPM, 0, 0, + &right_dac_to_hp_left_amp_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to hp right amp", SND_SOC_NOPM, 0, 0, + &right_dac_to_hp_right_amp_switch_control), + SND_SOC_DAPM_OUT_DRV_E("HP amp left driver", AUDIO_IC_CODEC_CTRL0, 3, 0, + NULL, 0, hp_amp_left_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("HP amp right driver", AUDIO_IC_CODEC_CTRL0, 2, 0, + NULL, 0, hp_amp_right_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SWITCH("Left dac to speaker lineout", SND_SOC_NOPM, 0, 0, + &left_dac_to_speaker_lineout_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to speaker lineout", SND_SOC_NOPM, 0, 0, + &right_dac_to_speaker_lineout_switch_control), + SND_SOC_DAPM_OUT_DRV_E("Speaker output driver", AUDIO_IC_CODEC_CTRL0, 4, 0, + NULL, 0, speaker_output_enable_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + + SND_SOC_DAPM_ADC_E("ADC left", NULL, AUDIO_IC_CODEC_CTRL1, 8, 0, + adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADC right", NULL, AUDIO_IC_CODEC_CTRL1, 7, 0, + adc_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MIXER("Left PGA mixer", AUDIO_IC_CODEC_CTRL1, 1, 0, + &left_input_path_controls[0], + ARRAY_SIZE(left_input_path_controls)), + SND_SOC_DAPM_MIXER("Right PGA mixer", AUDIO_IC_CODEC_CTRL1, 0, 0, + &right_input_path_controls[0], + ARRAY_SIZE(right_input_path_controls)), + + SND_SOC_DAPM_MUX("Mic input mode mux", SND_SOC_NOPM, 0, 0, + &sirf_inner_input_mode_control), + SND_SOC_DAPM_MICBIAS("Mic Bias", AUDIO_IC_CODEC_PWR, 3, 0), + SND_SOC_DAPM_INPUT("MICIN1"), + SND_SOC_DAPM_INPUT("MICIN2"), + SND_SOC_DAPM_INPUT("LINEIN1"), + SND_SOC_DAPM_INPUT("LINEIN2"), + + SND_SOC_DAPM_SUPPLY("HSL Phase Opposite", AUDIO_IC_CODEC_CTRL0, + 30, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route sirf_inner_audio_map[] = { + {"SPKOUT", NULL, "Speaker output driver"}, + {"Speaker output driver", NULL, "Left dac to speaker lineout"}, + {"Speaker output driver", NULL, "Right dac to speaker lineout"}, + {"Left dac to speaker lineout", "Switch", "DAC left"}, + {"Right dac to speaker lineout", "Switch", "DAC right"}, + {"HPOUTL", NULL, "HP amp left driver"}, + {"HPOUTR", NULL, "HP amp right driver"}, + {"HP amp left driver", NULL, "Right dac to hp left amp"}, + {"HP amp right driver", NULL , "Right dac to hp right amp"}, + {"HP amp left driver", NULL, "Left dac to hp left amp"}, + {"HP amp right driver", NULL , "Right dac to hp right amp"}, + {"Right dac to hp left amp", "Switch", "DAC left"}, + {"Right dac to hp right amp", "Switch", "DAC right"}, + {"Left dac to hp left amp", "Switch", "DAC left"}, + {"Left dac to hp right amp", "Switch", "DAC right"}, + {"DAC left", NULL, "Playback"}, + {"DAC right", NULL, "Playback"}, + {"DAC left", NULL, "HSL Phase Opposite"}, + {"DAC right", NULL, "HSL Phase Opposite"}, + + {"Capture", NULL, "ADC left"}, + {"Capture", NULL, "ADC right"}, + {"ADC left", NULL, "Left PGA mixer"}, + {"ADC right", NULL, "Right PGA mixer"}, + {"Left PGA mixer", "Line left Switch", "LINEIN2"}, + {"Right PGA mixer", "Line right Switch", "LINEIN1"}, + {"Left PGA mixer", "Mic left Switch", "MICIN2"}, + {"Right PGA mixer", "Mic right Switch", "Mic input mode mux"}, + {"Mic input mode mux", "Single-ended", "MICIN1"}, + {"Mic input mode mux", "Differential", "MICIN1"}, +}; + +static int sirf_inner_codec_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + pm_runtime_get_sync(codec->dev); + return 0; +} + +static void sirf_inner_codec_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + pm_runtime_put(codec->dev); +} + +static int sirf_inner_codec_trigger(struct snd_pcm_substream *substream, + int cmd, + struct snd_soc_dai *dai) +{ + struct sirf_soc_inner_audio *sinner_audio = snd_soc_dai_get_drvdata(dai); + int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + u32 val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock(&sinner_audio->lock); + if (playback) { + /*Disconnect HP amp connect to avoid pop noise*/ + val = snd_soc_read(codec, AUDIO_IC_CODEC_CTRL0); + val &= ~(IC_HSLEN | IC_HSREN); + snd_soc_write(codec, AUDIO_IC_CODEC_CTRL0, val); + + snd_soc_write(codec, AUDIO_CTRL_IC_TXFIFO_OP, 0x0); + + val = snd_soc_read(codec, AUDIO_CTRL_IC_CODEC_TX_CTRL); + val &= ~IC_TX_ENABLE; + snd_soc_write(codec, AUDIO_CTRL_IC_CODEC_TX_CTRL, val); + } else { + val = snd_soc_read(codec, AUDIO_CTRL_IC_CODEC_RX_CTRL); + val &= ~IC_RX_ENABLE; + snd_soc_write(codec, AUDIO_CTRL_IC_CODEC_RX_CTRL, val); + } + spin_unlock(&sinner_audio->lock); + break; + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock(&sinner_audio->lock); + if (playback) { + snd_soc_write(codec, AUDIO_CTRL_IC_TXFIFO_OP, AUDIO_FIFO_RESET); + snd_soc_write(codec, AUDIO_CTRL_IC_TXFIFO_INT_MSK, 0); + snd_soc_write(codec, AUDIO_CTRL_IC_TXFIFO_OP, 0x0); + snd_soc_write(codec, AUDIO_CTRL_IC_TXFIFO_OP, + AUDIO_FIFO_START); + snd_soc_write(codec, AUDIO_CTRL_IC_CODEC_TX_CTRL, + IC_TX_ENABLE); + val = snd_soc_read(codec, AUDIO_IC_CODEC_CTRL0); + val |= (IC_HSLEN | IC_HSREN); + snd_soc_write(codec, AUDIO_IC_CODEC_CTRL0, val); + } else { + snd_soc_write(codec, AUDIO_CTRL_IC_RXFIFO_OP, AUDIO_FIFO_RESET); + /* unmask rx fifo interrupt */ + snd_soc_write(codec, AUDIO_CTRL_IC_RXFIFO_INT_MSK, 0); + + snd_soc_write(codec, AUDIO_CTRL_IC_RXFIFO_OP, 0x0); + /* First start the FIFO, then enable the tx/rx */ + snd_soc_write(codec, AUDIO_CTRL_IC_RXFIFO_OP, + AUDIO_FIFO_START); + /* mono capture from dacr*/ + if (substream->runtime->channels == 1) + snd_soc_write(codec, + AUDIO_CTRL_IC_CODEC_RX_CTRL, 0x1); + else + snd_soc_write(codec, + AUDIO_CTRL_IC_CODEC_RX_CTRL, + IC_RX_ENABLE); + } + spin_unlock(&sinner_audio->lock); + break; + default: + return -EINVAL; + } + return 0; +} + +struct snd_soc_dai_ops sirf_inner_codec_dai_ops = { + .startup = sirf_inner_codec_startup, + .shutdown = sirf_inner_codec_shutdown, + .trigger = sirf_inner_codec_trigger, +}; + +struct snd_soc_dai_driver sirf_inner_codec_dai = { + .name = "sirf-soc-inner", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &sirf_inner_codec_dai_ops, +}; +EXPORT_SYMBOL_GPL(sirf_inner_codec_dai); + +static int sirf_inner_codec_probe(struct snd_soc_codec *codec) +{ + pm_runtime_enable(codec->dev); + if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio")) + return snd_soc_add_codec_controls(codec, + volume_controls_prima2, + ARRAY_SIZE(volume_controls_prima2)); + if (of_device_is_compatible(codec->dev->of_node, "sirf,atlas6-audio")) + return snd_soc_add_codec_controls(codec, + volume_controls_atlas6, + ARRAY_SIZE(volume_controls_atlas6)); + + return -EINVAL; +} + +static int sirf_inner_codec_remove(struct snd_soc_codec *codec) +{ + pm_runtime_disable(codec->dev); + return 0; +} + +static unsigned int sirf_inner_codec_reg_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(codec->dev); + return readl(sinner_audio->base + reg); +} + +static int sirf_inner_codec_reg_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val) +{ + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(codec->dev); + writel(val, sinner_audio->base + reg); + return 0; +} + + +static struct snd_soc_codec_driver soc_codec_device_sirf_inner_codec = { + .probe = sirf_inner_codec_probe, + .remove = sirf_inner_codec_remove, + .read = sirf_inner_codec_reg_read, + .write = sirf_inner_codec_reg_write, + .dapm_widgets = sirf_inner_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sirf_inner_dapm_widgets), + .dapm_routes = sirf_inner_audio_map, + .num_dapm_routes = ARRAY_SIZE(sirf_inner_audio_map), + .idle_bias_off = true, +}; + +static struct snd_dmaengine_dai_dma_data dma_data[2]; + +static int sirf_soc_inner_dai_probe(struct snd_soc_dai *dai) +{ + dai->playback_dma_data = &dma_data[0]; + dai->capture_dma_data = &dma_data[1]; + return 0; +} + +static struct snd_soc_dai_driver sirf_soc_inner_dai = { + .probe = sirf_soc_inner_dai_probe, + .name = "sirf-soc-inner", + .id = 0, + .playback = { + .stream_name = "inner Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "inner Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static const struct snd_soc_component_driver sirf_soc_inner_component = { + .name = "sirf-soc-inner", +}; + +static const struct of_device_id sirf_soc_inner_of_match[] = { + { .compatible = "sirf,prima2-audio", .data = &sirf_soc_inner_audio_reg_bits_prima2 }, + { .compatible = "sirf,atlas6-audio", .data = &sirf_soc_inner_audio_reg_bits_atlas6 }, + {} +}; +MODULE_DEVICE_TABLE(of, sirf_soc_inner_of_match); + +static int sirf_soc_inner_probe(struct platform_device *pdev) +{ + int ret; + u32 rx_dma_ch, tx_dma_ch; + struct sirf_soc_inner_audio *sinner_audio; + struct resource *mem_res; + struct device_node *dn = NULL; + const struct of_device_id *match; + + match = of_match_node(sirf_soc_inner_of_match, pdev->dev.of_node); + + sinner_audio = devm_kzalloc(&pdev->dev, + sizeof(struct sirf_soc_inner_audio), GFP_KERNEL); + if (!sinner_audio) + return -ENOMEM; + + sinner_audio->sirf_pcm_pdev = platform_device_register_simple("sirf-pcm-audio", + 1, NULL, 0); + if (IS_ERR(sinner_audio->sirf_pcm_pdev)) + return PTR_ERR(sinner_audio->sirf_pcm_pdev); + + platform_set_drvdata(pdev, sinner_audio); + + ret = of_property_read_u32(pdev->dev.of_node, + "sirf,inner-audio-dma-rx-channel", &rx_dma_ch); + if (ret < 0) { + dev_err(&pdev->dev, "Unable to audio capture dma channel\n"); + return ret; + } + ret = of_property_read_u32(pdev->dev.of_node, + "sirf,inner-audio-dma-tx-channel", &tx_dma_ch); + if (ret < 0) { + dev_err(&pdev->dev, "Unable to audio playback dma channel\n"); + return ret; + } + + dma_data[0].filter_data = (void *)tx_dma_ch; + dma_data[1].filter_data = (void *)rx_dma_ch; + + dn = of_find_compatible_node(dn, NULL, "sirf,prima2-pwrc"); + if (!dn) { + dev_err(&pdev->dev, "Failed to get sirf,prima2-pwrc node!\n"); + return -ENODEV; + } + + ret = of_property_read_u32(dn, "reg", &sinner_audio->sys_pwrc_reg_base); + if (ret < 0) { + dev_err(&pdev->dev, "Failed tp get pwrc register base address\n"); + return -EINVAL; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + sinner_audio->base = devm_ioremap_resource(&pdev->dev, mem_res); + if (sinner_audio->base == NULL) + return -ENOMEM; + + sinner_audio->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(sinner_audio->clk)) { + dev_err(&pdev->dev, "Get clock failed.\n"); + return PTR_ERR(sinner_audio->clk); + } + clk_prepare_enable(sinner_audio->clk); + + ret = snd_soc_register_component(&pdev->dev, &sirf_soc_inner_component, + &sirf_soc_inner_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register Audio SoC dai failed.\n"); + goto err_clk_put; + } + + ret = snd_soc_register_codec(&(pdev->dev), + &soc_codec_device_sirf_inner_codec, + &sirf_inner_codec_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register Audio Codec dai failed.\n"); + goto err_com_unreg; + } + + sinner_audio->reg_bits = (struct sirf_soc_inner_audio_reg_bits *)match->data; + /* Always open charge pump, if not, when the charge pump closed the + * adc will not stable*/ + writel(readl(sinner_audio->base + AUDIO_IC_CODEC_CTRL0) | IC_CPFREQ, + sinner_audio->base + AUDIO_IC_CODEC_CTRL0); + + spin_lock_init(&sinner_audio->lock); + return 0; + +err_com_unreg: + snd_soc_unregister_component(&pdev->dev); +err_clk_put: + clk_disable_unprepare(sinner_audio->clk); + return ret; +} + +static int sirf_soc_inner_remove(struct platform_device *pdev) +{ + struct sirf_soc_inner_audio *sinner_audio = platform_get_drvdata(pdev); + + clk_disable_unprepare(sinner_audio->clk); + snd_soc_unregister_codec(&(pdev->dev)); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int sirf_inner_runtime_suspend(struct device *dev) +{ + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(dev); + u32 val; + val = readl(sinner_audio->base + AUDIO_IC_CODEC_CTRL1); + val &= ~(1 << sinner_audio->reg_bits->codec_clk_en_bits); + writel(val, sinner_audio->base + AUDIO_IC_CODEC_CTRL1); + return 0; +} + +static int sirf_inner_runtime_resume(struct device *dev) +{ + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(dev); + u32 val; + val = readl(sinner_audio->base + AUDIO_IC_CODEC_CTRL1); + val |= (1 << sinner_audio->reg_bits->codec_clk_en_bits); + val &= ~(1 << sinner_audio->reg_bits->por_bits); + writel(val, sinner_audio->base + AUDIO_IC_CODEC_CTRL1); + + writel(readl(sinner_audio->base + AUDIO_IC_CODEC_CTRL0) | IC_CPFREQ, + sinner_audio->base + AUDIO_IC_CODEC_CTRL0); + + msleep(20); + val |= (1 << sinner_audio->reg_bits->por_bits); + writel(val, sinner_audio->base + AUDIO_IC_CODEC_CTRL1); + + return 0; +} +#endif + +#ifdef CONFIG_PM_SLEEP +static int sirf_soc_inner_suspend(struct device *dev) +{ + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(dev); + + sinner_audio->reg_ctrl0 = readl(sinner_audio->base + AUDIO_IC_CODEC_CTRL0); + sinner_audio->reg_ctrl1 = readl(sinner_audio->base + AUDIO_IC_CODEC_CTRL1); + sirf_inner_runtime_suspend(dev); + clk_disable_unprepare(sinner_audio->clk); + + return 0; +} + +static int sirf_soc_inner_resume(struct device *dev) +{ + struct sirf_soc_inner_audio *sinner_audio = dev_get_drvdata(dev); + + clk_prepare_enable(sinner_audio->clk); + + writel(sinner_audio->reg_ctrl0, + sinner_audio->base + AUDIO_IC_CODEC_CTRL0); + writel(sinner_audio->reg_ctrl1, sinner_audio->base + AUDIO_IC_CODEC_CTRL1); + + if (!pm_runtime_status_suspended(dev)) + sirf_inner_runtime_resume(dev); + + return 0; +} +#endif + +static const struct dev_pm_ops sirf_inner_pm_ops = { + SET_RUNTIME_PM_OPS(sirf_inner_runtime_suspend, sirf_inner_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(sirf_soc_inner_suspend, sirf_soc_inner_resume) +}; + +static struct platform_driver sirf_soc_inner_driver = { + .driver = { + .name = "sirf-soc-inner", + .owner = THIS_MODULE, + .of_match_table = sirf_soc_inner_of_match, + .pm = &sirf_inner_pm_ops, + }, + .probe = sirf_soc_inner_probe, + .remove = sirf_soc_inner_remove, +}; + +module_platform_driver(sirf_soc_inner_driver); + +MODULE_DESCRIPTION("SiRF SoC inner bus and codec driver"); +MODULE_AUTHOR("RongJun Ying <Rongjun.Ying(a)csr.com>"); +MODULE_LICENSE("GPL v2"); -- 1.8.2.3

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[alsa-devel] [PATCH] ASoC: Add pinctrl PM to components of active DAIs
by Nicolin Chen 29 Oct '13

29 Oct '13

It's quite popular that more drivers are using pinctrl PM, for example: (Documentation/devicetree/bindings/arm/primecell.txt). Just like what runtime PM does, it would de-active and en-active pin group depending on whether it's being used or not. And this pinctrl PM might be also beneficial to cpu dai drivers because they might have actual pinctrl so as to sleep their pins and wake them up as needed. To achieve this goal, this patch sets pins to the default state during resume or startup; While during suspend and shutdown, it would set pins to the sleep state. As pinctrl PM would return zero if there is no such pinctrl sleep state settings, this patch would not break current ASoC subsystem directly. [ However, there is still an exception that the patch can not handle, that is, when cpu dai driver does not have pinctrl property but another device has it. (The AUDMUX <-> SSI on Freescale i.MX6 series for example. SSI as a cpu dai doesn't contain pinctrl property while AUDMUX, an Audio Multiplexer, has it). In this case, this kind of cpu dai driver needs to find a way to obtain the pinctrl property as its own, by moving property from AUDMUX to SSI, or creating a pins link/dependency between these two devices, or using a more decent way after we figure it out. ] Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/soc-core.c | 28 ++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 3 +++ 2 files changed, 31 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4280c70..e8511767 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -684,6 +684,13 @@ int snd_soc_suspend(struct device *dev) if (card->suspend_post) card->suspend_post(card); + /* De-active pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_suspend); @@ -807,6 +814,13 @@ int snd_soc_resume(struct device *dev) if (list_empty(&card->codec_dev_list)) return 0; + /* En-active pins from sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + pinctrl_pm_select_default_state(cpu_dai->dev); + } + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume @@ -1929,6 +1943,13 @@ int snd_soc_poweroff(struct device *dev) snd_soc_dapm_shutdown(card); + /* De-active pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_poweroff); @@ -3766,6 +3787,13 @@ int snd_soc_register_card(struct snd_soc_card *card) if (ret != 0) soc_cleanup_card_debugfs(card); + /* De-active pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6..05bdba0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -183,6 +183,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; + pinctrl_pm_select_default_state(cpu_dai->dev); pm_runtime_get_sync(cpu_dai->dev); pm_runtime_get_sync(codec_dai->dev); pm_runtime_get_sync(platform->dev); @@ -317,6 +318,7 @@ out: pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + pinctrl_pm_select_sleep_state(cpu_dai->dev); return ret; } @@ -426,6 +428,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + pinctrl_pm_select_sleep_state(cpu_dai->dev); return 0; } -- 1.8.4

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[alsa-devel] [PATCH v2 3/6] ASoC: usp-pcm: add CPU DAI driver for PCM simulated from USP
by Barry Song 28 Oct '13

28 Oct '13

From: Rongjun Ying <Rongjun.Ying(a)csr.com> Universal Serial Ports (USP) can be used as PCM. this patch adds support for this functionality. The USP provides 128-byte data FIFO which supports DMA and I/O modes. Here we use the common DMA driver. Signed-off-by: Rongjun Ying <Rongjun.Ying(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- -v2: check the validation of div_id and rate for usp_pcm_divider; runtime PM enabled; use devm_clk_get instead of clk_get sound/soc/sirf/Kconfig | 3 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-usp.c | 481 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sirf/sirf-usp.h | 277 ++++++++++++++++++++++++++ 4 files changed, 763 insertions(+) create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig index 5064cfc..5e395d3 100644 --- a/sound/soc/sirf/Kconfig +++ b/sound/soc/sirf/Kconfig @@ -5,3 +5,6 @@ config SND_SIRF_SOC config SND_SOC_SIRF_I2S tristate + +config SND_SOC_SIRF_USP + tristate diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile index 9f754fe..630c9be 100644 --- a/sound/soc/sirf/Makefile +++ b/sound/soc/sirf/Makefile @@ -1,5 +1,7 @@ snd-soc-sirf-objs := sirf-pcm.o snd-soc-sirf-i2s-objs := sirf-i2s.o +snd-soc-sirf-usp-objs := sirf-usp.o obj-$(CONFIG_SND_SIRF_SOC) += snd-soc-sirf.o obj-$(CONFIG_SND_SOC_SIRF_I2S) += snd-soc-sirf-i2s.o +obj-$(CONFIG_SND_SOC_SIRF_USP) += snd-soc-sirf-usp.o diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c new file mode 100644 index 0000000..da377e5 --- /dev/null +++ b/sound/soc/sirf/sirf-usp.c @@ -0,0 +1,481 @@ +/* + * SiRF USP audio transfer interface like I2S + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ +#include <linux/module.h> +#include <linux/io.h> +#include <linux/of.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "sirf-usp.h" + +#define FIFO_RESET 0 +#define FIFO_START 1 +#define FIFO_STOP 2 + +#define AUDIO_WORD_SIZE 16 + +struct sirf_usp { + void __iomem *base; + struct clk *clk; + u32 mode1_reg; + u32 mode2_reg; + struct platform_device *sirf_pcm_pdev; +}; + +static struct snd_dmaengine_dai_dma_data dma_data[2]; + +static void sirf_usp_tx_fifo_op(struct sirf_usp *susp, int cmd) +{ + switch (cmd) { + case FIFO_RESET: + writel(USP_TX_FIFO_RESET, susp->base + USP_TX_FIFO_OP); + writel(0, susp->base + USP_TX_FIFO_OP); + break; + case FIFO_START: + writel(USP_TX_FIFO_START, susp->base + USP_TX_FIFO_OP); + break; + case FIFO_STOP: + writel(0, susp->base + USP_TX_FIFO_OP); + break; + } +} + +static void sirf_usp_rx_fifo_op(struct sirf_usp *susp, int cmd) +{ + switch (cmd) { + case FIFO_RESET: + writel(USP_RX_FIFO_RESET, susp->base + USP_RX_FIFO_OP); + writel(0, susp->base + USP_RX_FIFO_OP); + break; + case FIFO_START: + writel(USP_RX_FIFO_START, susp->base + USP_RX_FIFO_OP); + break; + case FIFO_STOP: + writel(0, susp->base + USP_RX_FIFO_OP); + break; + } +} + +static inline void sirf_usp_tx_enable(struct sirf_usp *susp) +{ + writel(readl(susp->base + USP_TX_RX_ENABLE) | USP_TX_ENA, + susp->base + USP_TX_RX_ENABLE); +} + +static inline void sirf_usp_tx_disable(struct sirf_usp *susp) +{ + writel(readl(susp->base + USP_TX_RX_ENABLE) & ~USP_TX_ENA, + susp->base + USP_TX_RX_ENABLE); +} + +static inline void sirf_usp_rx_enable(struct sirf_usp *susp) +{ + writel(readl(susp->base + USP_TX_RX_ENABLE) | USP_RX_ENA, + susp->base + USP_TX_RX_ENABLE); +} + +static inline void sirf_usp_rx_disable(struct sirf_usp *susp) +{ + writel(readl(susp->base + USP_TX_RX_ENABLE) & ~USP_RX_ENA, + susp->base + USP_TX_RX_ENABLE); +} + +static int sirf_usp_pcm_dai_probe(struct snd_soc_dai *dai) +{ + dai->playback_dma_data = &dma_data[0]; + dai->capture_dma_data = &dma_data[1]; + return 0; +} + +static int sirf_usp_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pm_runtime_get_sync(dai->dev); + return 0; +} + +static void sirf_usp_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pm_runtime_put(dai->dev); +} + +static int sirf_usp_pcm_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct sirf_usp *susp = snd_soc_dai_get_drvdata(dai); + u32 val = readl(susp->base + USP_MODE2); + u32 val1 = readl(susp->base + USP_MODE1); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val1 &= ~USP_CLOCK_MODE_SLAVE; + val &= ~USP_TFS_CLK_SLAVE_MODE; + val &= ~USP_RFS_CLK_SLAVE_MODE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val1 |= USP_CLOCK_MODE_SLAVE; + val |= USP_TFS_CLK_SLAVE_MODE; + val |= USP_RFS_CLK_SLAVE_MODE; + break; + default: + return -EINVAL; + } + writel(val1, susp->base + USP_MODE1); + writel(val, susp->base + USP_MODE2); + + return 0; +} + +static int sirf_usp_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct sirf_usp *susp = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (playback) { + sirf_usp_tx_fifo_op(susp, FIFO_RESET); + sirf_usp_tx_fifo_op(susp, FIFO_START); + sirf_usp_tx_enable(susp); + } else { + sirf_usp_rx_fifo_op(susp, FIFO_RESET); + sirf_usp_rx_fifo_op(susp, FIFO_START); + sirf_usp_rx_enable(susp); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (playback) { + sirf_usp_tx_disable(susp); + sirf_usp_tx_fifo_op(susp, FIFO_STOP); + } else { + sirf_usp_rx_disable(susp); + sirf_usp_rx_fifo_op(susp, FIFO_STOP); + } + break; + } + + return 0; +} + +static int sirf_usp_pcm_divider(struct snd_soc_dai *dai, int div_id, int rate) +{ + struct sirf_usp *susp = snd_soc_dai_get_drvdata(dai); + u32 clk_rate, clk_div, clk_div_hi, clk_div_lo; + + if (div_id != SIRF_USP_DIV_MCLK) + return -EINVAL; + + clk_rate = clk_get_rate(susp->clk); + if (clk_rate < rate * 2) { + dev_err(dai->dev, "Can't get rate(%d) by need.\n", rate); + return -EINVAL; + } + + clk_div = (clk_rate / (2 * rate)) - 1; + clk_div_hi = (clk_div & 0xC00) >> 10; + clk_div_lo = (clk_div & 0x3FF); + + writel((clk_div_lo << 21) | readl(susp->base + USP_MODE2), + susp->base + USP_MODE2); + writel((clk_div_hi << 30) | readl(susp->base + USP_TX_FRAME_CTRL), + susp->base + USP_TX_FRAME_CTRL); + + return 0; +} + +static const struct snd_soc_dai_ops sirf_usp_pcm_dai_ops = { + .startup = sirf_usp_pcm_startup, + .trigger = sirf_usp_pcm_trigger, + .set_fmt = sirf_usp_pcm_set_dai_fmt, + .set_clkdiv = sirf_usp_pcm_divider, + .shutdown = sirf_usp_pcm_shutdown, +}; + +static struct snd_soc_dai_driver sirf_usp_pcm_dai = { + .probe = sirf_usp_pcm_dai_probe, + .name = "sirf-usp-pcm", + .id = 0, + .playback = { + .stream_name = "SiRF USP PCM Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_48000 + | SNDRV_PCM_RATE_44100 + | SNDRV_PCM_RATE_32000 + | SNDRV_PCM_RATE_22050 + | SNDRV_PCM_RATE_16000 + | SNDRV_PCM_RATE_11025 + | SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "SiRF USP PCM Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_48000 + | SNDRV_PCM_RATE_44100 + | SNDRV_PCM_RATE_32000 + | SNDRV_PCM_RATE_22050 + | SNDRV_PCM_RATE_16000 + | SNDRV_PCM_RATE_11025 + | SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &sirf_usp_pcm_dai_ops, +}; + +static void sirf_usp_controller_init(struct sirf_usp *susp) +{ + u32 val; + + /* Configure RISC mode */ + writel(readl(susp->base + USP_RISC_DSP_MODE) & ~USP_RISC_DSP_SEL, + susp->base + USP_RISC_DSP_MODE); + + /* Disable all interrupts status */ + writel(readl(susp->base + USP_INT_STATUS), susp->base + USP_INT_STATUS); + + /* Configure DMA IO Length register */ + writel(0, susp->base + USP_TX_DMA_IO_LEN); + writel(0, susp->base + USP_RX_DMA_IO_LEN); + + /* Configure RX Frame Control */ + val = (AUDIO_WORD_SIZE * 2 - 1) << USP_RXC_DATA_LEN_OFFSET; + val |= (AUDIO_WORD_SIZE * 2 - 1) << USP_RXC_FRAME_LEN_OFFSET; + val |= (AUDIO_WORD_SIZE * 2 - 1) << USP_RXC_SHIFTER_LEN_OFFSET; + val |= USP_SINGLE_SYNC_MODE; + writel(val, susp->base + USP_RX_FRAME_CTRL); + + /* Configure TX Frame Control */ + val = (AUDIO_WORD_SIZE * 2 - 1) << USP_TXC_DATA_LEN_OFFSET; + val |= 0 << USP_TXC_SYNC_LEN_OFFSET; + val |= (AUDIO_WORD_SIZE * 2 - 1) << USP_TXC_FRAME_LEN_OFFSET; + val |= (AUDIO_WORD_SIZE * 2 - 1) << USP_TXC_SHIFTER_LEN_OFFSET; + val |= USP_TXC_SLAVE_CLK_SAMPLE; + writel(val, susp->base + USP_TX_FRAME_CTRL); + + /* Configure Mode2 register */ + val = (1 << USP_RXD_DELAY_LEN_OFFSET) | (0 << USP_TXD_DELAY_LEN_OFFSET); + val &= ~USP_ENA_CTRL_MODE; + val &= ~USP_FRAME_CTRL_MODE; + val &= ~USP_TFS_SOURCE_MODE; + writel(val, susp->base + USP_MODE2); + + /* Configure Mode1 register */ + val = 0; + val |= USP_SYNC_MODE; + val |= USP_ENDIAN_CTRL_LSBF; + val |= USP_EN; + val |= USP_RXD_ACT_EDGE_FALLING; + val &= ~USP_TXD_ACT_EDGE_FALLING; + val |= USP_RFS_ACT_LEVEL_LOGIC1; + val |= USP_TFS_ACT_LEVEL_LOGIC1; + val |= USP_SCLK_IDLE_MODE_TOGGLE; + val |= USP_SCLK_IDLE_LEVEL_LOGIC1; + val &= ~USP_SCLK_PIN_MODE_IO; + val &= ~USP_RFS_PIN_MODE_IO; + val &= ~USP_TFS_PIN_MODE_IO; + val &= ~USP_RXD_PIN_MODE_IO; + val &= ~USP_TXD_PIN_MODE_IO; + val |= USP_TX_UFLOW_REPEAT_ZERO; + writel(val, susp->base + USP_MODE1); + + /* Configure RX DMA IO Control register */ + writel(0x0, susp->base + USP_RX_DMA_IO_CTRL); + + /* Congiure RX FIFO Control register */ + writel((USP_RX_FIFO_THRESHOLD << USP_RX_FIFO_THD_OFFSET) | + (USP_TX_RX_FIFO_WIDTH_DWORD << USP_RX_FIFO_WIDTH_OFFSET), + susp->base + USP_RX_FIFO_CTRL); + + /* Congiure RX FIFO Level Check register */ + writel(RX_FIFO_SC(0x04) | RX_FIFO_LC(0x0E) | RX_FIFO_HC(0x1B), + susp->base + USP_RX_FIFO_LEVEL_CHK); + + /* Configure TX DMA IO Control register*/ + writel(0x0, susp->base + USP_TX_DMA_IO_CTRL); + + /* Configure TX FIFO Control register */ + writel((USP_TX_FIFO_THRESHOLD << USP_TX_FIFO_THD_OFFSET) | + (USP_TX_RX_FIFO_WIDTH_DWORD << USP_TX_FIFO_WIDTH_OFFSET), + susp->base + USP_TX_FIFO_CTRL); + + /* Congiure TX FIFO Level Check register */ + writel(TX_FIFO_SC(0x1B) | TX_FIFO_LC(0x0E) | TX_FIFO_HC(0x04), + susp->base + USP_TX_FIFO_LEVEL_CHK); + + /* Configure RX FIFO */ + writel(USP_RX_FIFO_RESET, susp->base + USP_RX_FIFO_OP); + writel(0, susp->base + USP_RX_FIFO_OP); + + /* Configure TX FIFO */ + writel(USP_TX_FIFO_RESET, susp->base + USP_TX_FIFO_OP); + writel(0, susp->base + USP_TX_FIFO_OP); +} + +static void sirf_usp_controller_uninit(struct sirf_usp *susp) +{ + /* Disable RX/TX */ + writel(0, susp->base + USP_INT_ENABLE); + writel(0, susp->base + USP_TX_RX_ENABLE); +} + +#ifdef CONFIG_PM_RUNTIME +static int sirf_usp_pcm_runtime_suspend(struct device *dev) +{ + struct sirf_usp *susp = dev_get_drvdata(dev); + sirf_usp_controller_uninit(susp); + clk_disable_unprepare(susp->clk); + return 0; +} + +static int sirf_usp_pcm_runtime_resume(struct device *dev) +{ + struct sirf_usp *susp = dev_get_drvdata(dev); + clk_prepare_enable(susp->clk); + sirf_usp_controller_init(susp); + return 0; +} +#endif + +#ifdef CONFIG_PM_SLEEP +static int sirf_usp_pcm_suspend(struct device *dev) +{ + struct sirf_usp *susp = dev_get_drvdata(dev); + + if (!pm_runtime_status_suspended(dev)) { + susp->mode1_reg = readl(susp->base + USP_MODE1); + susp->mode2_reg = readl(susp->base + USP_MODE2); + sirf_usp_pcm_runtime_suspend(dev); + } + return 0; +} + +static int sirf_usp_pcm_resume(struct device *dev) +{ + struct sirf_usp *susp = dev_get_drvdata(dev); + + if (!pm_runtime_status_suspended(dev)) { + sirf_usp_pcm_runtime_resume(dev); + writel(susp->mode1_reg, susp->base + USP_MODE1); + writel(susp->mode2_reg, susp->base + USP_MODE2); + } + return 0; +} +#endif + +static const struct snd_soc_component_driver sirf_usp_component = { + .name = "sirf-usp", +}; + +static int sirf_usp_pcm_probe(struct platform_device *pdev) +{ + struct sirf_usp *susp; + u32 rx_dma_ch, tx_dma_ch; + int ret; + struct resource *mem_res; + + susp = devm_kzalloc(&pdev->dev, sizeof(struct sirf_usp), + GFP_KERNEL); + if (!susp) + return -ENOMEM; + + susp->sirf_pcm_pdev = platform_device_register_simple("sirf-pcm-audio", + 2, NULL, 0); + if (IS_ERR(susp->sirf_pcm_pdev)) + return PTR_ERR(susp->sirf_pcm_pdev); + + platform_set_drvdata(pdev, susp); + + ret = of_property_read_u32(pdev->dev.of_node, + "sirf,usp-dma-rx-channel", &rx_dma_ch); + if (ret < 0) { + dev_err(&pdev->dev, "Unable to USP0 rx dma channel\n"); + return ret; + } + ret = of_property_read_u32(pdev->dev.of_node, + "sirf,usp-dma-tx-channel", &tx_dma_ch); + if (ret < 0) { + dev_err(&pdev->dev, "Unable to USP0 tx dma channel\n"); + return ret; + } + + dma_data[0].filter_data = (void *)tx_dma_ch; + dma_data[1].filter_data = (void *)rx_dma_ch; + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + susp->base = devm_ioremap_resource(&pdev->dev, mem_res); + if (susp->base == NULL) + return -ENOMEM; + + susp->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(susp->clk)) { + dev_err(&pdev->dev, "Get clock failed.\n"); + return PTR_ERR(susp->clk); + } + clk_prepare_enable(susp->clk); + + ret = snd_soc_register_component(&pdev->dev, &sirf_usp_component, + &sirf_usp_pcm_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register Audio SoC dai failed.\n"); + goto err; + } + pm_runtime_enable(&pdev->dev); + return 0; + +err: + return ret; +} + +static int sirf_usp_pcm_remove(struct platform_device *pdev) +{ + struct sirf_usp *susp = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); + pm_runtime_disable(&pdev->dev); + platform_device_unregister(susp->sirf_pcm_pdev); + + return 0; +} + +static const struct of_device_id sirf_usp_pcm_of_match[] = { + { .compatible = "sirf,prima2-usp-pcm", }, + {} +}; +MODULE_DEVICE_TABLE(of, sirf_usp_pcm_of_match); + +static const struct dev_pm_ops sirf_usp_pcm_pm_ops = { + SET_RUNTIME_PM_OPS(sirf_usp_pcm_runtime_suspend, sirf_usp_pcm_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(sirf_usp_pcm_suspend, sirf_usp_pcm_resume) +}; + +static struct platform_driver sirf_usp_pcm_driver = { + .driver = { + .name = "sirf-usp-pcm", + .owner = THIS_MODULE, + .of_match_table = sirf_usp_pcm_of_match, + .pm = &sirf_usp_pcm_pm_ops, + }, + .probe = sirf_usp_pcm_probe, + .remove = sirf_usp_pcm_remove, +}; + +module_platform_driver(sirf_usp_pcm_driver); + +MODULE_DESCRIPTION("SiRF SoC USP PCM bus driver"); +MODULE_AUTHOR("RongJun Ying <Rongjun.Ying(a)csr.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/sirf/sirf-usp.h b/sound/soc/sirf/sirf-usp.h new file mode 100644 index 0000000..62afc0d --- /dev/null +++ b/sound/soc/sirf/sirf-usp.h @@ -0,0 +1,277 @@ +/* + * arch/arm/mach-prima2/include/mach/sirfsoc_usp.h + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#ifndef __SIRFSOC_USP__ +#define __SIRFSOC_USP__ + +/* USP Registers */ +#define USP_MODE1 0x00 +#define USP_MODE2 0x04 +#define USP_TX_FRAME_CTRL 0x08 +#define USP_RX_FRAME_CTRL 0x0C +#define USP_TX_RX_ENABLE 0x10 +#define USP_INT_ENABLE 0x14 +#define USP_INT_STATUS 0x18 +#define USP_PIN_IO_DATA 0x1C +#define USP_RISC_DSP_MODE 0x20 +#define USP_AYSNC_PARAM_REG 0x24 +#define USP_IRDA_X_MODE_DIV 0x28 +#define USP_SM_CFG 0x2C +#define USP_TX_DMA_IO_CTRL 0x100 +#define USP_TX_DMA_IO_LEN 0x104 +#define USP_TX_FIFO_CTRL 0x108 +#define USP_TX_FIFO_LEVEL_CHK 0x10C +#define USP_TX_FIFO_OP 0x110 +#define USP_TX_FIFO_STATUS 0x114 +#define USP_TX_FIFO_DATA 0x118 +#define USP_RX_DMA_IO_CTRL 0x120 +#define USP_RX_DMA_IO_LEN 0x124 +#define USP_RX_FIFO_CTRL 0x128 +#define USP_RX_FIFO_LEVEL_CHK 0x12C +#define USP_RX_FIFO_OP 0x130 +#define USP_RX_FIFO_STATUS 0x134 +#define USP_RX_FIFO_DATA 0x138 + +/* USP MODE register-1 */ +#define USP_SYNC_MODE 0x00000001 +#define USP_CLOCK_MODE_SLAVE 0x00000002 +#define USP_LOOP_BACK_EN 0x00000004 +#define USP_HPSIR_EN 0x00000008 +#define USP_ENDIAN_CTRL_LSBF 0x00000010 +#define USP_EN 0x00000020 +#define USP_RXD_ACT_EDGE_FALLING 0x00000040 +#define USP_TXD_ACT_EDGE_FALLING 0x00000080 +#define USP_RFS_ACT_LEVEL_LOGIC1 0x00000100 +#define USP_TFS_ACT_LEVEL_LOGIC1 0x00000200 +#define USP_SCLK_IDLE_MODE_TOGGLE 0x00000400 +#define USP_SCLK_IDLE_LEVEL_LOGIC1 0x00000800 +#define USP_SCLK_PIN_MODE_IO 0x00001000 +#define USP_RFS_PIN_MODE_IO 0x00002000 +#define USP_TFS_PIN_MODE_IO 0x00004000 +#define USP_RXD_PIN_MODE_IO 0x00008000 +#define USP_TXD_PIN_MODE_IO 0x00010000 +#define USP_SCLK_IO_MODE_INPUT 0x00020000 +#define USP_RFS_IO_MODE_INPUT 0x00040000 +#define USP_TFS_IO_MODE_INPUT 0x00080000 +#define USP_RXD_IO_MODE_INPUT 0x00100000 +#define USP_TXD_IO_MODE_INPUT 0x00200000 +#define USP_IRDA_WIDTH_DIV_MASK 0x3FC00000 +#define USP_IRDA_WIDTH_DIV_OFFSET 0 +#define USP_IRDA_IDLE_LEVEL_HIGH 0x40000000 +#define USP_TX_UFLOW_REPEAT_ZERO 0x80000000 +#define USP_TX_ENDIAN_MODE 0x00000020 +#define USP_RX_ENDIAN_MODE 0x00000020 + +/* USP Mode Register-2 */ +#define USP_RXD_DELAY_LEN_MASK 0x000000FF +#define USP_RXD_DELAY_LEN_OFFSET 0 + +#define USP_TXD_DELAY_LEN_MASK 0x0000FF00 +#define USP_TXD_DELAY_LEN_OFFSET 8 + +#define USP_ENA_CTRL_MODE 0x00010000 +#define USP_FRAME_CTRL_MODE 0x00020000 +#define USP_TFS_SOURCE_MODE 0x00040000 +#define USP_TFS_MS_MODE 0x00080000 +#define USP_CLK_DIVISOR_MASK 0x7FE00000 +#define USP_CLK_DIVISOR_OFFSET 21 + +#define USP_TFS_CLK_SLAVE_MODE (1<<20) +#define USP_RFS_CLK_SLAVE_MODE (1<<19) + +#define USP_IRDA_DATA_WIDTH 0x80000000 + +/* USP Transmit Frame Control Register */ + +#define USP_TXC_DATA_LEN_MASK 0x000000FF +#define USP_TXC_DATA_LEN_OFFSET 0 + +#define USP_TXC_SYNC_LEN_MASK 0x0000FF00 +#define USP_TXC_SYNC_LEN_OFFSET 8 + +#define USP_TXC_FRAME_LEN_MASK 0x00FF0000 +#define USP_TXC_FRAME_LEN_OFFSET 16 + +#define USP_TXC_SHIFTER_LEN_MASK 0x1F000000 +#define USP_TXC_SHIFTER_LEN_OFFSET 24 + +#define USP_TXC_SLAVE_CLK_SAMPLE 0x20000000 + +#define USP_TXC_CLK_DIVISOR_MASK 0xC0000000 +#define USP_TXC_CLK_DIVISOR_OFFSET 30 + +/* USP Receive Frame Control Register */ + +#define USP_RXC_DATA_LEN_MASK 0x000000FF +#define USP_RXC_DATA_LEN_OFFSET 0 + +#define USP_RXC_FRAME_LEN_MASK 0x0000FF00 +#define USP_RXC_FRAME_LEN_OFFSET 8 + +#define USP_RXC_SHIFTER_LEN_MASK 0x001F0000 +#define USP_RXC_SHIFTER_LEN_OFFSET 16 + +#define USP_I2S_SYNC_CHG 0x00200000 + +#define USP_RXC_CLK_DIVISOR_MASK 0x0F000000 +#define USP_RXC_CLK_DIVISOR_OFFSET 24 +#define USP_SINGLE_SYNC_MODE 0x00400000 + +/* Tx - RX Enable Register */ + +#define USP_RX_ENA 0x00000001 +#define USP_TX_ENA 0x00000002 + +/* USP Interrupt Enable and status Register */ +#define USP_RX_DONE_INT 0x00000001 +#define USP_TX_DONE_INT 0x00000002 +#define USP_RX_OFLOW_INT 0x00000004 +#define USP_TX_UFLOW_INT 0x00000008 +#define USP_RX_IO_DMA_INT 0x00000010 +#define USP_TX_IO_DMA_INT 0x00000020 +#define USP_RXFIFO_FULL_INT 0x00000040 +#define USP_TXFIFO_EMPTY_INT 0x00000080 +#define USP_RXFIFO_THD_INT 0x00000100 +#define USP_TXFIFO_THD_INT 0x00000200 +#define USP_UART_FRM_ERR_INT 0x00000400 +#define USP_RX_TIMEOUT_INT 0x00000800 +#define USP_TX_ALLOUT_INT 0x00001000 +#define USP_RXD_BREAK_INT 0x00008000 + +/* All possible TX interruots */ +#define USP_TX_INTERRUPT (USP_TX_DONE_INT|USP_TX_UFLOW_INT|USP_TX_IO_DMA_INT|\ + USP_TXFIFO_EMPTY_INT|USP_TXFIFO_THD_INT) +/* All possible RX interruots */ +#define USP_RX_INTERRUPT (USP_RX_DONE_INT|USP_RX_OFLOW_INT|USP_RX_IO_DMA_INT|\ + USP_RXFIFO_FULL_INT|USP_RXFIFO_THD_INT|USP_RXFIFO_THD_INT|USP_RX_TIMEOUT_INT) + +#define USP_INT_ALL 0x1FFF + +/* USP Pin I/O Data Register */ + +#define USP_RFS_PIN_VALUE_MASK 0x00000001 +#define USP_TFS_PIN_VALUE_MASK 0x00000002 +#define USP_RXD_PIN_VALUE_MASK 0x00000004 +#define USP_TXD_PIN_VALUE_MASK 0x00000008 +#define USP_SCLK_PIN_VALUE_MASK 0x00000010 + +/* USP RISC/DSP Mode Register */ +#define USP_RISC_DSP_SEL 0x00000001 + +/* USP ASYNC PARAMETER Register*/ + +#define USP_ASYNC_TIMEOUT_MASK 0x0000FFFF +#define USP_ASYNC_TIMEOUT_OFFSET 0 +#define USP_ASYNC_TIMEOUT(x) (((x)&USP_ASYNC_TIMEOUT_MASK)<<USP_ASYNC_TIMEOUT_OFFSET) + +#define USP_ASYNC_DIV2_MASK 0x003F0000 +#define USP_ASYNC_DIV2_OFFSET 16 + +/* USP TX DMA I/O MODE Register */ +#define USP_TX_MODE_IO 0x00000001 + +/* USP TX DMA I/O Length Register */ +#define USP_TX_DATA_LEN_MASK 0xFFFFFFFF +#define USP_TX_DATA_LEN_OFFSET 0 + +/* USP TX FIFO Control Register */ +#define USP_TX_FIFO_WIDTH_MASK 0x00000003 +#define USP_TX_FIFO_WIDTH_OFFSET 0 + +#define USP_TX_FIFO_THD_MASK 0x000001FC +#define USP_TX_FIFO_THD_OFFSET 2 + +/* USP TX FIFO Level Check Register */ +#define USP_TX_FIFO_LEVEL_CHECK_MASK 0x1F +#define USP_TX_FIFO_SC_OFFSET 0 +#define USP_TX_FIFO_LC_OFFSET 10 +#define USP_TX_FIFO_HC_OFFSET 20 + +#define TX_FIFO_SC(x) (((x) & USP_TX_FIFO_LEVEL_CHECK_MASK) << USP_TX_FIFO_SC_OFFSET) +#define TX_FIFO_LC(x) (((x) & USP_TX_FIFO_LEVEL_CHECK_MASK) << USP_TX_FIFO_LC_OFFSET) +#define TX_FIFO_HC(x) (((x) & USP_TX_FIFO_LEVEL_CHECK_MASK) << USP_TX_FIFO_HC_OFFSET) + +/* USP TX FIFO Operation Register */ +#define USP_TX_FIFO_RESET 0x00000001 +#define USP_TX_FIFO_START 0x00000002 + +/* USP TX FIFO Status Register */ +#define USP_TX_FIFO_LEVEL_MASK 0x0000007F +#define USP_TX_FIFO_LEVEL_OFFSET 0 + +#define USP_TX_FIFO_FULL 0x00000080 +#define USP_TX_FIFO_EMPTY 0x00000100 + +/* USP TX FIFO Data Register */ +#define USP_TX_FIFO_DATA_MASK 0xFFFFFFFF +#define USP_TX_FIFO_DATA_OFFSET 0 + +/* USP RX DMA I/O MODE Register */ +#define USP_RX_MODE_IO 0x00000001 +#define USP_RX_DMA_FLUSH 0x00000004 + +/* USP RX DMA I/O Length Register */ +#define USP_RX_DATA_LEN_MASK 0xFFFFFFFF +#define USP_RX_DATA_LEN_OFFSET 0 + +/* USP RX FIFO Control Register */ +#define USP_RX_FIFO_WIDTH_MASK 0x00000003 +#define USP_RX_FIFO_WIDTH_OFFSET 0 + +#define USP_RX_FIFO_THD_MASK 0x000001FC +#define USP_RX_FIFO_THD_OFFSET 2 + +/* USP RX FIFO Level Check Register */ + +#define USP_RX_FIFO_LEVEL_CHECK_MASK 0x1F +#define USP_RX_FIFO_SC_OFFSET 0 +#define USP_RX_FIFO_LC_OFFSET 10 +#define USP_RX_FIFO_HC_OFFSET 20 + +#define RX_FIFO_SC(x) (((x) & USP_RX_FIFO_LEVEL_CHECK_MASK) << USP_RX_FIFO_SC_OFFSET) +#define RX_FIFO_LC(x) (((x) & USP_RX_FIFO_LEVEL_CHECK_MASK) << USP_RX_FIFO_LC_OFFSET) +#define RX_FIFO_HC(x) (((x) & USP_RX_FIFO_LEVEL_CHECK_MASK) << USP_RX_FIFO_HC_OFFSET) + +/* USP RX FIFO Operation Register */ +#define USP_RX_FIFO_RESET 0x00000001 +#define USP_RX_FIFO_START 0x00000002 + +/* USP RX FIFO Status Register */ + +#define USP_RX_FIFO_LEVEL_MASK 0x0000007F +#define USP_RX_FIFO_LEVEL_OFFSET 0 + +#define USP_RX_FIFO_FULL 0x00000080 +#define USP_RX_FIFO_EMPTY 0x00000100 + +/* USP RX FIFO Data Register */ + +#define USP_RX_FIFO_DATA_MASK 0xFFFFFFFF +#define USP_RX_FIFO_DATA_OFFSET 0 + +/* + * When rx thd irq occur, sender just disable tx empty irq, + * Remaining data in tx fifo wil also be sent out. + */ +#define USP_FIFO_SIZE 128 +#define USP_TX_FIFO_THRESHOLD (USP_FIFO_SIZE/2) +#define USP_RX_FIFO_THRESHOLD (USP_FIFO_SIZE/2) + +/* FIFO_WIDTH for the USP_TX_FIFO_CTRL and USP_RX_FIFO_CTRL registers */ +#define USP_FIFO_WIDTH_BYTE 0x00 +#define USP_FIFO_WIDTH_WORD 0x01 +#define USP_FIFO_WIDTH_DWORD 0x02 + +#define USP_ASYNC_DIV2 16 + +#define USP_PLUGOUT_RETRY_CNT 2 + +#define USP_TX_RX_FIFO_WIDTH_DWORD 2 + +#define SIRF_USP_DIV_MCLK 0 +#endif -- 1.8.2.3

2 1

[alsa-devel] [PATCH v4 00/15] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 28 Oct '13

28 Oct '13

Hi, This patch series attempts to fix a long standing problem with the concurrent usage of playback and capture on implicit feedback devices. Now the following work properly on my system: * Full-duplex with jack :) * All tests I ran with my tool. Will post it in case anyone want to try it, it takes about 15 minutes to run all the tests. * Starting a capture-only jackd instance, e.g.: jackd -n 1 -d alsa -d hw:2,0 -C and then playback-only: jackd -n 2 -d alsa -d hw:2,0 -P and then stop and start the playback. The same with playback running and restarting capture. * Trying to start a second substream with parameter mismatch (rate or period bytes is all I can test with my device) will fail, without ill effects. With this series applied there are now 3 usage tracking variables: - substream usage flag - endpoint param_set (to protect the parameters until the endpoint is started) - the existing endpoint use_count Changes from v3: * Added substream "used" flag, to prevent failure in some combinations of ops. (#13) * Moved start_endpoints for capture to prepare (#14) * Added more constraints to the endpoint_may_set_params function (#10) * Cleanups (#12, #15) v3: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065744.html Applies against for-next, tested on upstream 3.12-rc3 (needs Alan Stern's recent patch "improve buffer size computations for USB PCM audio" to apply cleanly on current mainline). Daniel and Nikolay: if you can test again with this series I'd appreciate it. Cheers, Eldad Eldad Zack (15): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: correct ep use_count semantics (add set_param flag) ALSA: usb-audio: rename alt_idx to altsetting ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: remove altset_idx from snd_usb_substream ALSA: usb-audio: remove unused endpoint flag EP_FLAG_ACTIVATED ALSA: usb-audio: clear sync subs hw_params ALSA: usb-audio: always wait in start_endpoints ALSA: usb-audio: improve logging messages sound/usb/card.h | 9 ++- sound/usb/endpoint.c | 122 ++++++++++++++++++++--------- sound/usb/endpoint.h | 13 +++- sound/usb/pcm.c | 214 ++++++++++++++++++++++++++++++++++++--------------- sound/usb/pcm.h | 2 + sound/usb/proc.c | 4 +- 6 files changed, 258 insertions(+), 106 deletions(-) -- 1.8.1.5

4 31

[alsa-devel] [RFC] ASoC: core: Allow DAI links to be specified by using ACPI names
by Jarkko Nikula 28 Oct '13

28 Oct '13

This patch adds alternative way to specify DAI links by using ACPI names. ACPI name here is considered to be used as an alias for device name during DAI link binding time. Rationale behind this is that device name especially with bus connected codecs may not be constant enough to be used in machines with dynamic and variable amount of bus controllers due changing bus/adapter number. One way to solve this problem on ACPI 5 based systems is to use ACPI names of those devices that are enumerated by the ACPI subsystem. For instance, matchine drivers could specify codec in DAI link by using: .codec_name = "INT33CA:00", instead of .codec_name = "rt5640.0-001c", Note that ACPI name is used just an alias during bind time and core continues to use device name. In fact, machine drivers could use either names. Thanks to Mika Westerberg for initial idea to use ACPI handles for binding. Author extended and simplified the idea so that struct snd_soc_dai_link doesn't need a new field, machine drivers don't need any additional code and soc-core.c can handle matching when executing soc_bind_dai_link(). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Mika Westerberg <mika.westerberg(a)linux.intel.com> --- linux-acpi(a)vger.kernel.org cc'd if ACPI folks would like to check API usage here. --- sound/soc/soc-core.c | 38 ++++++++++++++++++++++++++++++++------ 1 file changed, 32 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4280c70..5cb440e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -36,6 +36,7 @@ #include <linux/of.h> #include <linux/gpio.h> #include <linux/of_gpio.h> +#include <linux/acpi.h> #include <sound/ac97_codec.h> #include <sound/core.h> #include <sound/jack.h> @@ -836,6 +837,27 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; +static bool soc_match_name(struct device *dev, + const char *comp_name, const char *bind_name) +{ + if (!strcmp(comp_name, bind_name)) + return true; + +#if IS_ENABLED(CONFIG_ACPI) + if (ACPI_HANDLE(dev)) { + struct acpi_device *adev; + if (!acpi_bus_get_device(ACPI_HANDLE(dev), &adev) && + !strcmp(dev_name(&adev->dev), bind_name)) { + dev_dbg(dev, "Matching with ACPI name %s\n", + dev_name(&adev->dev)); + return true; + } + } +#endif + + return false; +} + static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; @@ -853,10 +875,12 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) (cpu_dai->dev->of_node != dai_link->cpu_of_node)) continue; if (dai_link->cpu_name && - strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name)) + !soc_match_name(cpu_dai->dev, dev_name(cpu_dai->dev), + dai_link->cpu_name)) continue; if (dai_link->cpu_dai_name && - strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + !soc_match_name(cpu_dai->dev, cpu_dai->name, + dai_link->cpu_dai_name)) continue; rtd->cpu_dai = cpu_dai; @@ -874,7 +898,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) if (codec->dev->of_node != dai_link->codec_of_node) continue; } else { - if (strcmp(codec->name, dai_link->codec_name)) + if (!soc_match_name(codec->dev, codec->name, + dai_link->codec_name)) continue; } @@ -886,8 +911,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) */ list_for_each_entry(codec_dai, &dai_list, list) { if (codec->dev == codec_dai->dev && - !strcmp(codec_dai->name, - dai_link->codec_dai_name)) { + soc_match_name(codec->dev, codec_dai->name, + dai_link->codec_dai_name)) { rtd->codec_dai = codec_dai; } @@ -918,7 +943,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) dai_link->platform_of_node) continue; } else { - if (strcmp(platform->name, platform_name)) + if (!soc_match_name(platform->dev, platform->name, + platform_name)) continue; } -- 1.8.4.rc3

6 20

[alsa-devel] [PATCH 0/5] ASoC: small fixes for bugs reported by Coverity
by Takashi Iwai 28 Oct '13

28 Oct '13

Hi, this is another series of small fixes for bugs reported by Coverity, this time about ASoC core codes. Takashi

2 10

[alsa-devel] [PATCH v2 2/6] ASoC: sirf: add I2S CPU DAI driver
by Barry Song 28 Oct '13

28 Oct '13

From: Rongjun Ying <Rongjun.Ying(a)csr.com> This patch adds I2S DAI driver for SiRFprima2 and SiRFatlas6. The hardware supports: I2S bus specification compliant (Released by Philips Semiconductors in June, 1996) Supports I2S master and I2S slave mode Provides MCLK to I2S CODEC in I2S master mode Supports 16-bit resolution playback Supports 16-bit resolution record Supports 2 channel record Supports 2 channel and 6 channel mode playback Frame length, left channel length and right channel length are all programmable Supports two DMA channels for TXFIFO and RXFIFO Supports floating mode (x_ac97_dout can be configured as input) headfile sound/soc/sirf/sirf-audio.h will be shard by all I2S and soc-inner DAIs. Signed-off-by: Rongjun Ying <Rongjun.Ying(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- -v2: runtime pm enabled; fix wrong device_reset path in v1; other minor fixes according to Mark's comments sound/soc/sirf/Kconfig | 3 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-audio.h | 268 +++++++++++++++++++++++++ sound/soc/sirf/sirf-i2s.c | 464 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 737 insertions(+) create mode 100644 sound/soc/sirf/sirf-audio.h create mode 100644 sound/soc/sirf/sirf-i2s.c diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig index 1637089..5064cfc 100644 --- a/sound/soc/sirf/Kconfig +++ b/sound/soc/sirf/Kconfig @@ -2,3 +2,6 @@ config SND_SIRF_SOC tristate "Platform DMA driver for the SiRF SoC chips" depends on ARCH_SIRF && SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM + +config SND_SOC_SIRF_I2S + tristate diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile index f268b83..9f754fe 100644 --- a/sound/soc/sirf/Makefile +++ b/sound/soc/sirf/Makefile @@ -1,3 +1,5 @@ snd-soc-sirf-objs := sirf-pcm.o +snd-soc-sirf-i2s-objs := sirf-i2s.o obj-$(CONFIG_SND_SIRF_SOC) += snd-soc-sirf.o +obj-$(CONFIG_SND_SOC_SIRF_I2S) += snd-soc-sirf-i2s.o diff --git a/sound/soc/sirf/sirf-audio.h b/sound/soc/sirf/sirf-audio.h new file mode 100644 index 0000000..b6fdf06 --- /dev/null +++ b/sound/soc/sirf/sirf-audio.h @@ -0,0 +1,268 @@ +/* + * SiRF inner codec controllers define + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#ifndef _SIRF_INNER_AUDIO_CTRL_H +#define _SIRF_INNER_AUDIO_CTRL_H + +#define AUDIO_CTRL_TX_FIFO_LEVEL_CHECK_MASK 0x3F +#define AUDIO_CTRL_TX_FIFO_SC_OFFSET 0 +#define AUDIO_CTRL_TX_FIFO_LC_OFFSET 10 +#define AUDIO_CTRL_TX_FIFO_HC_OFFSET 20 + +#define TX_FIFO_SC(x) (((x) & AUDIO_CTRL_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_CTRL_TX_FIFO_SC_OFFSET) +#define TX_FIFO_LC(x) (((x) & AUDIO_CTRL_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_CTRL_TX_FIFO_LC_OFFSET) +#define TX_FIFO_HC(x) (((x) & AUDIO_CTRL_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_CTRL_TX_FIFO_HC_OFFSET) + +#define AUDIO_CTRL_RX_FIFO_LEVEL_CHECK_MASK 0x0F +#define AUDIO_CTRL_RX_FIFO_SC_OFFSET 0 +#define AUDIO_CTRL_RX_FIFO_LC_OFFSET 10 +#define AUDIO_CTRL_RX_FIFO_HC_OFFSET 20 + +#define RX_FIFO_SC(x) (((x) & AUDIO_CTRL_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_CTRL_RX_FIFO_SC_OFFSET) +#define RX_FIFO_LC(x) (((x) & AUDIO_CTRL_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_CTRL_RX_FIFO_LC_OFFSET) +#define RX_FIFO_HC(x) (((x) & AUDIO_CTRL_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_CTRL_RX_FIFO_HC_OFFSET) + +#define AUDIO_CTRL_MODE_SEL (0x0000) +#define AUDIO_CTRL_AC97_CTRL (0x0004) +#define AUDIO_CTRL_AC97_CMD (0x0008) +#define AUDIO_CTRL_AC97_OP_STATUS (0x000C) +#define AUDIO_CTRL_AC97_RD_CODEC_REG (0x0010) +#define AUDIO_CTRL_TXSLOT_EN (0x0014) +#define AUDIO_CTRL_RXSLOT_EN (0x0018) +#define AUDIO_CTRL_AC97_AUX_SLOT_EN (0x001c) +#define AUDIO_CTRL_I2S_CTRL (0x0020) +#define AUDIO_CTRL_I2S_TX_RX_EN (0x0024) + +#define AUDIO_CTRL_EXT_TXFIFO1_OP (0x0040) +#define AUDIO_CTRL_EXT_TXFIFO1_LEV_CHK (0x0044) +#define AUDIO_CTRL_EXT_TXFIFO1_STS (0x0048) +#define AUDIO_CTRL_EXT_TXFIFO1_INT (0x004C) +#define AUDIO_CTRL_EXT_TXFIFO1_INT_MSK (0x0050) + +#define AUDIO_CTRL_EXT_TXFIFO2_OP (0x0054) +#define AUDIO_CTRL_EXT_TXFIFO2_LEV_CHK (0x0058) +#define AUDIO_CTRL_EXT_TXFIFO2_STS (0x005C) +#define AUDIO_CTRL_EXT_TXFIFO2_INT (0x0060) +#define AUDIO_CTRL_EXT_TXFIFO2_INT_MSK (0x0064) + +#define AUDIO_CTRL_EXT_TXFIFO3_OP (0x0068) +#define AUDIO_CTRL_EXT_TXFIFO3_LEV_CHK (0x006C) +#define AUDIO_CTRL_EXT_TXFIFO3_STS (0x0070) +#define AUDIO_CTRL_EXT_TXFIFO3_INT (0x0074) +#define AUDIO_CTRL_EXT_TXFIFO3_INT_MSK (0x0078) + +#define AUDIO_CTRL_EXT_TXFIFO4_OP (0x007C) +#define AUDIO_CTRL_EXT_TXFIFO4_LEV_CHK (0x0080) +#define AUDIO_CTRL_EXT_TXFIFO4_STS (0x0084) +#define AUDIO_CTRL_EXT_TXFIFO4_INT (0x0088) +#define AUDIO_CTRL_EXT_TXFIFO4_INT_MSK (0x008C) + +#define AUDIO_CTRL_EXT_TXFIFO5_OP (0x0090) +#define AUDIO_CTRL_EXT_TXFIFO5_LEV_CHK (0x0094) +#define AUDIO_CTRL_EXT_TXFIFO5_STS (0x0098) +#define AUDIO_CTRL_EXT_TXFIFO5_INT (0x009C) +#define AUDIO_CTRL_EXT_TXFIFO5_INT_MSK (0x00A0) + +#define AUDIO_CTRL_EXT_TXFIFO6_OP (0x00A4) +#define AUDIO_CTRL_EXT_TXFIFO6_LEV_CHK (0x00A8) +#define AUDIO_CTRL_EXT_TXFIFO6_STS (0x00AC) +#define AUDIO_CTRL_EXT_TXFIFO6_INT (0x00B0) +#define AUDIO_CTRL_EXT_TXFIFO6_INT_MSK (0x00B4) + +#define AUDIO_CTRL_RXFIFO_OP (0x00B8) +#define AUDIO_CTRL_RXFIFO_LEV_CHK (0x00BC) +#define AUDIO_CTRL_RXFIFO_STS (0x00C0) +#define AUDIO_CTRL_RXFIFO_INT (0x00C4) +#define AUDIO_CTRL_RXFIFO_INT_MSK (0x00C8) + +#define AUDIO_CTRL_AUXFIFO_OP (0x00CC) +#define AUDIO_CTRL_AUXFIFO_LEV_CHK (0x00D0) +#define AUDIO_CTRL_AUXFIFO_STS (0x00D4) +#define AUDIO_CTRL_AUXFIFO_INT (0x00D8) +#define AUDIO_CTRL_AUXFIFO_INT_MSK (0x00DC) + +#define AUDIO_IC_CODEC_PWR (0x00E0) +#define AUDIO_IC_CODEC_CTRL0 (0x00E4) +#define AUDIO_IC_CODEC_CTRL1 (0x00E8) +#define AUDIO_IC_CODEC_CTRL2 (0x00EC) +#define AUDIO_IC_CODEC_CTRL3 (0x00F0) + +#define AUDIO_CTRL_IC_CODEC_TX_CTRL (0x00F4) +#define AUDIO_CTRL_IC_CODEC_RX_CTRL (0x00F8) + +#define AUDIO_CTRL_IC_TXFIFO_OP (0x00FC) +#define AUDIO_CTRL_IC_TXFIFO_LEV_CHK (0x0100) +#define AUDIO_CTRL_IC_TXFIFO_STS (0x0104) +#define AUDIO_CTRL_IC_TXFIFO_INT (0x0108) +#define AUDIO_CTRL_IC_TXFIFO_INT_MSK (0x010C) + +#define AUDIO_CTRL_IC_RXFIFO_OP (0x0110) +#define AUDIO_CTRL_IC_RXFIFO_LEV_CHK (0x0114) +#define AUDIO_CTRL_IC_RXFIFO_STS (0x0118) +#define AUDIO_CTRL_IC_RXFIFO_INT (0x011C) +#define AUDIO_CTRL_IC_RXFIFO_INT_MSK (0x0120) + +#define I2S_MODE (1<<0) +#define AC97_TX_SLOT3_WIDTH_MASK (3<<1) +#define AC97_TX_SLOT4_WIDTH_MASK (3<<3) +#define AC97_TX_SLOT6_WIDTH_MASK (3<<5) +#define AC97_TX_SLOT7_WIDTH_MASK (3<<7) +#define AC97_TX_SLOT8_WIDTH_MASK (3<<9) +#define AC97_TX_SLOT9_WIDTH_MASK (3<<11) +#define AC97_FIFO_SYNC_MASK (3<<13) +#define AC97_FIFO_SYNC_ALL (1<<13) + +#define SYNC_START (1<<0) +#define AC97_START (1<<1) +#define WARM_WAKEUP (1<<2) + +#define AC97_CMD_TYPE_MASK (1<<7) +#define AC97_CMD_ADDR_MASK (0x7F) +#define AC97_CMD_TYPE_WRITE (0<<7) +#define AC97_CMD_TYPE_READ (1<<7) + +#define CMD_ISSUE_BIT (1<<0) +#define RD_CMD_FINISH_BIT (1<<1) +#define CODEC_READY_BIT (1<<2) + +#define AC97_RDBACK_ADDR_BITS (0x7F) +#define AC97_RDBACK_DATA_BITS (0xFF<<16) + +#define AC97_TX_SLOT3_EN (1<<0) +#define AC97_TX_SLOT4_EN (1<<1) +#define AC97_TX_SLOT6_EN (1<<2) +#define AC97_TX_SLOT7_EN (1<<3) +#define AC97_TX_SLOT8_EN (1<<4) +#define AC97_TX_SLOT9_EN (1<<5) + +#define AC97_RX_SLOT3_EN (1<<0) +#define AC97_RX_SLOT4_EN (1<<1) +#define AC97_RX_SLOT5_EN (1<<2) +#define AC97_RX_SLOT6_EN (1<<3) +#define AC97_RX_SLOT7_EN (1<<4) +#define AC97_RX_SLOT8_EN (1<<5) +#define AC97_RX_SLOT9_EN (1<<6) +#define AC97_RX_SLOT10_EN (1<<7) +#define AC97_RX_SLOT11_EN (1<<8) +#define AC97_RX_SLOT12_EN (1<<9) + +#define AC97_AUX_SLOT3_EN (1<<0) +#define AC97_AUX_SLOT4_EN (1<<1) +#define AC97_AUX_SLOT5_EN (1<<2) +#define AC97_AUX_SLOT6_EN (1<<3) +#define AC97_AUX_SLOT7_EN (1<<4) +#define AC97_AUX_SLOT8_EN (1<<5) +#define AC97_AUX_SLOT9_EN (1<<6) +#define AC97_AUX_SLOT10_EN (1<<7) +#define AC97_AUX_SLOT11_EN (1<<8) +#define AC97_AUX_SLOT12_EN (1<<9) + +#define I2S_LOOP_BACK (1<<3) +#define I2S_MCLK_DIV_SHIFT 15 +#define I2S_MCLK_DIV_MASK (0x1FF<<I2S_MCLK_DIV_SHIFT) +#define I2S_BITCLK_DIV_SHIFT 24 +#define I2S_BITCLK_DIV_MASK (0xFF<<I2S_BITCLK_DIV_SHIFT) + +#define I2S_MCLK_EN (1<<2) +#define I2S_REF_CLK_SEL_EXT (1<<3) +#define I2S_DOUT_OE (1<<4) +#define i2s_R2X_LP_TO_TX0 (1<<30) +#define i2s_R2X_LP_TO_TX1 (2<<30) +#define i2s_R2X_LP_TO_TX2 (3<<30) + +#define AUDIO_FIFO_START (1 << 0) +#define AUDIO_FIFO_RESET (1 << 1) + +#define AUDIO_FIFO_FULL (1 << 0) +#define AUDIO_FIFO_EMPTY (1 << 1) +#define AUDIO_FIFO_OFLOW (1 << 2) +#define AUDIO_FIFO_UFLOW (1 << 3) + +#define I2S_RX_ENABLE (1 << 0) +#define I2S_TX_ENABLE (1 << 1) + +/* Codec I2S Control Register defines */ +#define I2S_SLAVE_MODE (1 << 0) +#define I2S_SIX_CHANNELS (1 << 1) +#define I2S_L_CHAN_LEN_MASK (0x1f << 4) +#define I2S_FRAME_LEN_MASK (0x3f << 9) + +#define AC97_WRITE_FRAME_VALID 0X10 +#define AC97_WRITE_SLOT1_VALID 0X08 +#define AC97_WRITE_SLOT2_VALID 0X04 +#define AC97_WRITE_SLOT3_VALID 0X02 +#define AC97_WRITE_SLOT4_VALID 0X01 + +#define IC_TX_ENABLE (0x03) +#define IC_RX_ENABLE (0x03) + +#define MICBIASEN (1 << 3) + +#define IC_RDACEN (1 << 0) +#define IC_LDACEN (1 << 1) +#define IC_HSREN (1 << 2) +#define IC_HSLEN (1 << 3) +#define IC_SPEN (1 << 4) +#define IC_CPEN (1 << 5) + +#define IC_HPRSELR (1 << 6) +#define IC_HPLSELR (1 << 7) +#define IC_HPRSELL (1 << 8) +#define IC_HPLSELL (1 << 9) +#define IC_SPSELR (1 << 10) +#define IC_SPSELL (1 << 11) + +#define IC_MONOR (1 << 12) +#define IC_MONOL (1 << 13) + +#define IC_RXOSRSEL (1 << 28) +#define IC_CPFREQ (1 << 29) +#define IC_HSINVEN (1 << 30) + +#define IC_MICINREN (1 << 0) +#define IC_MICINLEN (1 << 1) +#define IC_MICIN1SEL (1 << 2) +#define IC_MICIN2SEL (1 << 3) +#define IC_MICDIFSEL (1 << 4) +#define IC_LINEIN1SEL (1 << 5) +#define IC_LINEIN2SEL (1 << 6) +#define IC_RADCEN (1 << 7) +#define IC_LADCEN (1 << 8) +#define IC_ALM (1 << 9) + +#define IC_DIGMICEN (1 << 22) +#define IC_DIGMICFREQ (1 << 23) +#define IC_ADC14B_12 (1 << 24) +#define IC_FIRDAC_HSL_EN (1 << 25) +#define IC_FIRDAC_HSR_EN (1 << 26) +#define IC_FIRDAC_LOUT_EN (1 << 27) +#define IC_POR (1 << 28) +#define IC_CODEC_CLK_EN (1 << 29) +#define IC_HP_3DB_BOOST (1 << 30) + +#define IC_ADC_LEFT_GAIN_SHIFT 16 +#define IC_ADC_RIGHT_GAIN_SHIFT 10 +#define IC_ADC_GAIN_MASK 0x3F +#define IC_MIC_MAX_GAIN 0x39 + +#define IC_RXPGAR_MASK 0x3F +#define IC_RXPGAR_SHIFT 14 +#define IC_RXPGAL_MASK 0x3F +#define IC_RXPGAL_SHIFT 21 +#define IC_RXPGAR 0x7B +#define IC_RXPGAL 0x7B + +#define SIRF_I2S_EXT_CLK 0x0 +#define SIRF_I2S_PWM_CLK 0x1 +#endif /*__SIRF_INNER_AUDIO_CTRL_H*/ diff --git a/sound/soc/sirf/sirf-i2s.c b/sound/soc/sirf/sirf-i2s.c new file mode 100644 index 0000000..ae4102f --- /dev/null +++ b/sound/soc/sirf/sirf-i2s.c @@ -0,0 +1,464 @@ +/* + * SiRF I2S driver + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ +#include <linux/module.h> +#include <linux/io.h> +#include <linux/of.h> +#include <linux/of_address.h> +#include <linux/pm_runtime.h> +#include <linux/clk.h> +#include <linux/reset.h> + +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> + +#include "sirf-audio.h" + +struct sirf_i2s { + void __iomem *base; + struct clk *clk; + u32 i2s_ctrl; + u32 i2s_ctrl_tx_rx_en; + spinlock_t lock; + struct platform_device *sirf_pcm_pdev; +}; + +static struct snd_dmaengine_dai_dma_data dma_data[2]; + +static int sirf_i2s_dai_probe(struct snd_soc_dai *dai) +{ + dai->playback_dma_data = &dma_data[0]; + dai->capture_dma_data = &dma_data[1]; + return 0; +} + +static int sirf_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pm_runtime_get_sync(dai->dev); + return 0; +} + +static void sirf_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pm_runtime_put(dai->dev); +} + +static int sirf_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct sirf_i2s *si2s = snd_soc_dai_get_drvdata(dai); + int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock(&si2s->lock); + + if (playback) { + /* First start the FIFO, then enable the tx/rx */ + writel(AUDIO_FIFO_RESET, + si2s->base + AUDIO_CTRL_EXT_TXFIFO1_OP); + writel(AUDIO_FIFO_START, + si2s->base + AUDIO_CTRL_EXT_TXFIFO1_OP); + + writel(readl(si2s->base+AUDIO_CTRL_I2S_TX_RX_EN) + | I2S_TX_ENABLE | I2S_DOUT_OE, + si2s->base + AUDIO_CTRL_I2S_TX_RX_EN); + + } else { + /* First start the FIFO, then enable the tx/rx */ + writel(AUDIO_FIFO_RESET, + si2s->base + AUDIO_CTRL_RXFIFO_OP); + writel(AUDIO_FIFO_START, + si2s->base + AUDIO_CTRL_RXFIFO_OP); + + writel(readl(si2s->base+AUDIO_CTRL_I2S_TX_RX_EN) + | I2S_RX_ENABLE, + si2s->base + AUDIO_CTRL_I2S_TX_RX_EN); + } + + spin_unlock(&si2s->lock); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock(&si2s->lock); + + if (playback) { + writel(readl(si2s->base + AUDIO_CTRL_I2S_TX_RX_EN) + & ~(I2S_TX_ENABLE | I2S_MCLK_EN), + si2s->base + AUDIO_CTRL_I2S_TX_RX_EN); + /* First disable the tx/rx, then stop the FIFO */ + writel(0, si2s->base + AUDIO_CTRL_EXT_TXFIFO1_OP); + } else { + writel(readl(si2s->base + AUDIO_CTRL_I2S_TX_RX_EN) + & ~(I2S_RX_ENABLE | I2S_MCLK_EN), + si2s->base+AUDIO_CTRL_I2S_TX_RX_EN); + + /* First disable the tx/rx, then stop the FIFO */ + writel(0, si2s->base + AUDIO_CTRL_RXFIFO_OP); + } + + spin_unlock(&si2s->lock); + break; + default: + return -EINVAL; + } + return 0; +} + +static int sirf_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct sirf_i2s *si2s = snd_soc_dai_get_drvdata(dai); + u32 i2s_ctrl = readl(si2s->base + AUDIO_CTRL_I2S_CTRL); + u32 left_len, frame_len; + int channels = params_channels(params); + + /* + * SiRFSoC I2S controller only support 2 and 6 channells output. + * I2S_SIX_CHANNELS bit clear: select 2 channels mode. + * I2S_SIX_CHANNELS bit set: select 6 channels mode. + */ + switch (channels) { + case 2: + i2s_ctrl &= ~I2S_SIX_CHANNELS; + break; + case 6: + i2s_ctrl |= I2S_SIX_CHANNELS; + break; + default: + dev_err(dai->dev, "%d channels unsupported\n", channels); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + left_len = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + left_len = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + left_len = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + left_len = 32; + break; + default: + dev_err(dai->dev, "Format unsupported\n"); + return -EINVAL; + } + + frame_len = left_len * 2; + i2s_ctrl &= ~(I2S_L_CHAN_LEN_MASK | I2S_FRAME_LEN_MASK); + /* Fill the actual len - 1 */ + i2s_ctrl |= ((frame_len - 1) << 9) | ((left_len - 1) << 4) + | (0 << 15) | (3 << 24); + writel(i2s_ctrl, si2s->base + AUDIO_CTRL_I2S_CTRL); + return 0; +} + +static int sirf_i2s_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct sirf_i2s *si2s = snd_soc_dai_get_drvdata(dai); + u32 i2s_ctrl, i2s_tx_rx_ctrl; + + i2s_ctrl = readl(si2s->base + AUDIO_CTRL_I2S_CTRL); + i2s_tx_rx_ctrl = readl(si2s->base + AUDIO_CTRL_I2S_TX_RX_EN); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s_ctrl |= I2S_SLAVE_MODE; + i2s_tx_rx_ctrl &= ~I2S_MCLK_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s_ctrl &= ~I2S_SLAVE_MODE; + i2s_tx_rx_ctrl |= I2S_MCLK_EN; + break; + default: + return -EINVAL; + } + writel(i2s_ctrl, si2s->base + AUDIO_CTRL_I2S_CTRL); + writel(i2s_tx_rx_ctrl, si2s->base + AUDIO_CTRL_I2S_TX_RX_EN); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + writel(readl(si2s->base + AUDIO_CTRL_MODE_SEL) + | I2S_MODE, + si2s->base + AUDIO_CTRL_MODE_SEL); + break; + default: + dev_err(dai->dev, "Only I2S format supported\n"); + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + dev_err(dai->dev, " Only normal bit clock, normal frame clock supported\n"); + return -EINVAL; + } + + return 0; +} + +static int sirf_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct sirf_i2s *si2s = snd_soc_dai_get_drvdata(dai); + u32 val; + u32 bclk_div_coefficient; + + if (div < 2 || div % 2) { + dev_err(dai->dev, "BITCLK divider must greater than 1," + "And must is a multiple of 2\n"); + return -EINVAL; + } + + /* + * Calculate the divider coefficient of I2S reference + * clock frequency divider. + */ + bclk_div_coefficient = div / 2 - 1; + + if (bclk_div_coefficient >= (1 << 9)) { + dev_err(dai->dev, "The BITCLK divider(%d) must less than " + "%d.\n", div, (1 << 9) * 2); + return -EINVAL; + } + + val = readl(si2s->base + AUDIO_CTRL_I2S_TX_RX_EN); + switch (div_id) { + case SIRF_I2S_EXT_CLK: + val |= I2S_REF_CLK_SEL_EXT; + break; + case SIRF_I2S_PWM_CLK: + val &= ~I2S_REF_CLK_SEL_EXT; + break; + default: + return -EINVAL; + } + writel(val, si2s->base + AUDIO_CTRL_I2S_TX_RX_EN); + + val = readl(si2s->base + AUDIO_CTRL_I2S_CTRL); + val |= (bclk_div_coefficient << 24); + /* + * MCLK coefficient must set to 0, means + * divide-by-two from reference clock. + */ + val &= ~(((1 << 10) - 1) << 15); + writel(val, si2s->base + AUDIO_CTRL_I2S_CTRL); + + return 0; +} + +struct snd_soc_dai_ops sirfsoc_i2s_dai_ops = { + .startup = sirf_i2s_startup, + .shutdown = sirf_i2s_shutdown, + .trigger = sirf_i2s_trigger, + .hw_params = sirf_i2s_hw_params, + .set_fmt = sirf_i2s_set_dai_fmt, + .set_clkdiv = sirf_i2s_set_clkdiv, +}; + +static struct snd_soc_dai_driver sirf_i2s_dai = { + .probe = sirf_i2s_dai_probe, + .name = "sirf-i2s", + .id = 0, + .playback = { + .stream_name = "SiRF I2S Playback", + .channels_min = 2, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "SiRF I2S Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &sirfsoc_i2s_dai_ops, +}; +#ifdef CONFIG_PM_RUNTIME +static int sirf_i2s_runtime_suspend(struct device *dev) +{ + struct sirf_i2s *si2s = dev_get_drvdata(dev); + clk_disable_unprepare(si2s->clk); + + return 0; +} + +static int sirf_i2s_runtime_resume(struct device *dev) +{ + struct sirf_i2s *si2s = dev_get_drvdata(dev); + clk_prepare_enable(si2s->clk); + device_reset(dev); + return 0; +} +#endif + +#ifdef CONFIG_PM_SLEEP +static int sirf_i2s_suspend(struct device *dev) +{ + struct sirf_i2s *si2s = dev_get_drvdata(dev); + + if (!pm_runtime_status_suspended(dev)) { + si2s->i2s_ctrl = readl(si2s->base + AUDIO_CTRL_I2S_CTRL); + si2s->i2s_ctrl_tx_rx_en = + readl(si2s->base + AUDIO_CTRL_I2S_TX_RX_EN); + sirf_i2s_runtime_suspend(dev); + } + return 0; +} + +static int sirf_i2s_resume(struct device *dev) +{ + struct sirf_i2s *si2s = dev_get_drvdata(dev); + if (!pm_runtime_status_suspended(dev)) { + sirf_i2s_runtime_resume(dev); + writel(readl(si2s->base + AUDIO_CTRL_MODE_SEL) + | I2S_MODE, + si2s->base + AUDIO_CTRL_MODE_SEL); + writel(si2s->i2s_ctrl, si2s->base + AUDIO_CTRL_I2S_CTRL); + /*Restore MCLK enable and reference clock select bits.*/ + writel(si2s->i2s_ctrl_tx_rx_en & + (I2S_MCLK_EN | I2S_REF_CLK_SEL_EXT), + si2s->base + AUDIO_CTRL_I2S_TX_RX_EN); + + writel(0, si2s->base + AUDIO_CTRL_EXT_TXFIFO1_INT_MSK); + writel(0, si2s->base + AUDIO_CTRL_RXFIFO_INT_MSK); + } + + return 0; +} +#endif + +static const struct snd_soc_component_driver sirf_i2s_component = { + .name = "sirf-i2s", +}; + +static int sirf_i2s_probe(struct platform_device *pdev) +{ + struct sirf_i2s *si2s; + u32 rx_dma_ch, tx_dma_ch; + int ret; + struct resource mem_res; + + si2s = devm_kzalloc(&pdev->dev, sizeof(struct sirf_i2s), + GFP_KERNEL); + if (!si2s) + return -ENOMEM; + + si2s->sirf_pcm_pdev = platform_device_register_simple("sirf-pcm-audio", + 0, NULL, 0); + if (IS_ERR(si2s->sirf_pcm_pdev)) + return PTR_ERR(si2s->sirf_pcm_pdev); + + platform_set_drvdata(pdev, si2s); + + spin_lock_init(&si2s->lock); + + ret = of_property_read_u32(pdev->dev.of_node, + "sirf,i2s-dma-rx-channel", &rx_dma_ch); + if (ret < 0) { + dev_err(&pdev->dev, "Unable to USP0 rx dma channel\n"); + return ret; + } + ret = of_property_read_u32(pdev->dev.of_node, + "sirf,i2s-dma-tx-channel", &tx_dma_ch); + if (ret < 0) { + dev_err(&pdev->dev, "Unable to USP0 tx dma channel\n"); + return ret; + } + + dma_data[0].filter_data = (void *)tx_dma_ch; + dma_data[1].filter_data = (void *)rx_dma_ch; + + ret = of_address_to_resource(pdev->dev.of_node, 0, &mem_res); + if (ret < 0) { + dev_err(&pdev->dev, "Unable to get i2s memory resource.\n"); + return ret; + } + si2s->base = devm_ioremap(&pdev->dev, mem_res.start, + resource_size(&mem_res)); + if (!si2s->base) + return -ENOMEM; + + si2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(si2s->clk)) { + dev_err(&pdev->dev, "Get clock failed.\n"); + ret = PTR_ERR(si2s->clk); + goto err; + } + + ret = snd_soc_register_component(&pdev->dev, &sirf_i2s_component, + &sirf_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register Audio SoC dai failed.\n"); + goto err; + } + + pm_runtime_enable(&pdev->dev); + return 0; + +err: + return ret; +} + +static int sirf_i2s_remove(struct platform_device *pdev) +{ + struct sirf_i2s *si2s = platform_get_drvdata(pdev); + + snd_soc_unregister_component(&pdev->dev); + pm_runtime_disable(&pdev->dev); + platform_device_unregister(si2s->sirf_pcm_pdev); + return 0; +} + +static const struct of_device_id sirf_i2s_of_match[] = { + { .compatible = "sirf,prima2-i2s", }, + {} +}; +MODULE_DEVICE_TABLE(of, sirf_i2s_of_match); + +static const struct dev_pm_ops sirf_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(sirf_i2s_runtime_suspend, sirf_i2s_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(sirf_i2s_suspend, sirf_i2s_resume) +}; + +static struct platform_driver sirf_i2s_driver = { + .driver = { + .name = "sirf-i2s", + .owner = THIS_MODULE, + .of_match_table = sirf_i2s_of_match, + .pm = &sirf_i2s_pm_ops, + }, + .probe = sirf_i2s_probe, + .remove = sirf_i2s_remove, +}; + +module_platform_driver(sirf_i2s_driver); + +MODULE_DESCRIPTION("SiRF SoC I2S driver"); +MODULE_AUTHOR("RongJun Ying <Rongjun.Ying(a)csr.com>"); +MODULE_LICENSE("GPL v2"); -- 1.8.2.3

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[alsa-devel] [PATCH v2 1/6] ASoC: sirf: add sirf platform driver which provides DMA
by Barry Song 28 Oct '13

28 Oct '13

From: Rongjun Ying <Rongjun.Ying(a)csr.com> this driver uses dmaengine APIs and provides DMA to the CPU DAIs of I2S, USP and SiRF-soc-inner. SiRFSoC has 3 audio DAIs: I2S, USP(Universal Serial Ports) and DAI connected to soc-inner-codec, all of them will use the same DMA driver here. Signed-off-by: Rongjun Ying <Rongjun.Ying(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- -v2: use the generic dmaengine PCM driver as Lars-Peter Clausen's comments; sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sirf/Kconfig | 4 ++ sound/soc/sirf/Makefile | 3 ++ sound/soc/sirf/sirf-pcm.c | 99 +++++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 108 insertions(+) create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-pcm.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b84..d5dce2b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -48,6 +48,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/sirf/Kconfig" source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 61a64d2..1a47ff9 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -25,6 +25,7 @@ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += sirf/ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig new file mode 100644 index 0000000..1637089 --- /dev/null +++ b/sound/soc/sirf/Kconfig @@ -0,0 +1,4 @@ +config SND_SIRF_SOC + tristate "Platform DMA driver for the SiRF SoC chips" + depends on ARCH_SIRF && SND_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile new file mode 100644 index 0000000..f268b83 --- /dev/null +++ b/sound/soc/sirf/Makefile @@ -0,0 +1,3 @@ +snd-soc-sirf-objs := sirf-pcm.o + +obj-$(CONFIG_SND_SIRF_SOC) += snd-soc-sirf.o diff --git a/sound/soc/sirf/sirf-pcm.c b/sound/soc/sirf/sirf-pcm.c new file mode 100644 index 0000000..1d53786 --- /dev/null +++ b/sound/soc/sirf/sirf-pcm.c @@ -0,0 +1,99 @@ +/* + * ALSA PCM interface for the SiRF SoC + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#include <linux/module.h> +#include <linux/sirfsoc_dma.h> +#include <sound/dmaengine_pcm.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +static struct snd_pcm_hardware sirf_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_BLOCK_TRANSFER + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_48000), + .rate_min = 512, + .rate_max = 115200, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 128, + .period_bytes_max = 32 * 1024, + .periods_min = 2, + .periods_max = 2, +}; + +static bool filter(struct dma_chan *chan, void *param) +{ + struct snd_dmaengine_dai_dma_data *dma_data = param; + + if (!sirfsoc_dma_filter_id(chan, dma_data->filter_data)) + return false; + + chan->private = dma_data->filter_data; + + return true; +} + +static const struct snd_dmaengine_pcm_config sirf_dmaengine_pcm_config = { + .pcm_hardware = &sirf_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .compat_filter_fn = filter, + .prealloc_buffer_size = 64 * 1024, +}; + +static int sirf_pcm_probe(struct platform_device *pdev) +{ + return snd_dmaengine_pcm_register(&pdev->dev, + &sirf_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT); +} + +static int sirf_pcm_remove(struct platform_device *pdev) +{ + snd_dmaengine_pcm_unregister(&pdev->dev); + return 0; +} + +static struct platform_driver sirf_pcm_driver = { + .driver = { + .name = "sirf-pcm-audio", + .owner = THIS_MODULE, + }, + .probe = sirf_pcm_probe, + .remove = sirf_pcm_remove, +}; + +static int __init sirf_pcm_init(void) +{ + int ret = 0; + + ret = platform_driver_register(&sirf_pcm_driver); + if (ret) + pr_err("failed to register platform driver\n"); + return ret; +} + +static void __exit sirf_pcm_exit(void) +{ + platform_driver_unregister(&sirf_pcm_driver); +} + +module_init(sirf_pcm_init); +module_exit(sirf_pcm_exit); + +MODULE_DESCRIPTION("SiRF PCM audio interface driver"); +MODULE_AUTHOR("RongJun Ying <Rongjun.Ying(a)csr.com>"); +MODULE_LICENSE("GPL v2"); -- 1.8.2.3

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[alsa-devel] [PATCH 0/6] Small fixes for bugs Coverity reported
by Takashi Iwai 28 Oct '13

28 Oct '13

Hi, here is a series of small fixes for bugs reported by Coverity. The first one for 6fire is a real bad one, but I guess it hits little users just because of the number of users for such cases. Others are just minor fixes and code improvements in HD-audio and core codes. Takashi

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