March 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: wm_adsp: Split firmware load into smaller chunks
by Charles Keepax 12 Mar '14

12 Mar '14

The firmware files can be quite large and allocating the whole firmware a single DMA safe buffer can be problematic if the system is under a high memory load. Ease the requirements slightly by writing the firmware out in page sized chunks. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 50 ++++++++++++++++++++++++++++--------------- 1 files changed, 32 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f9fd564..937af6f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -684,24 +684,38 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - buf = wm_adsp_buf_alloc(region->data, - le32_to_cpu(region->len), - &buf_list); - if (!buf) { - adsp_err(dsp, "Out of memory\n"); - ret = -ENOMEM; - goto out_fw; - } + size_t to_write = PAGE_SIZE; + size_t remain = le32_to_cpu(region->len); + const u8 *data = region->data; + + while (remain > 0) { + if (remain < PAGE_SIZE) + to_write = remain; + + buf = wm_adsp_buf_alloc(data, + to_write, + &buf_list); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + ret = -ENOMEM; + goto out_fw; + } - ret = regmap_raw_write_async(regmap, reg, buf->buf, - le32_to_cpu(region->len)); - if (ret != 0) { - adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", - file, regions, - le32_to_cpu(region->len), offset, - region_name, ret); - goto out_fw; + ret = regmap_raw_write_async(regmap, reg, + buf->buf, + to_write); + if (ret != 0) { + adsp_err(dsp, + "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + file, regions, + to_write, offset, + region_name, ret); + goto out_fw; + } + + data += to_write; + reg += to_write / 2; + remain -= to_write; } } -- 1.7.2.5

2 3

[alsa-devel] snd_pcm_drain causes repeated playback of last buffer contents
by Sanjeev Chugh 12 Mar '14

12 Mar '14

Hello, I'm not an alsa expert at all but trying to learn alsa. I'm experiencing an issue where on snd_pcm_drain() the slave PCM driven via plugin DSHARE is not filled with silence. Result is endless playback of buffer content until pcm is closed. I'm trying to analyze what could be the reason for this. In my understanding in Alsa, audio data is played as a unit of period. Once playing one period is finished, then next period will be played. If period has lesser frames than the period size, audio data previously played will be re-played until pcm is closed out. Could it be a reason for this continuous playback when snd_pcm_drain is called ? I'm not even sure if it is desired behavior that silence be padded after driver drains the pcm :( I guess it is too big as a bug to be ignored. Thanks Pthread

1 0

[alsa-devel] [PATCH] ASoC: Fix use after free
by Lars-Peter Clausen 12 Mar '14

12 Mar '14

Freeing the current list element while iterating over the list will cause a use after free since the iterator function will still use the current element to look up the next. Use list_for_each_safe() and remove the element from the list before freeing it to avoid this. Fixes: 1438c2f60b ("ASoC: Add a per component dai list") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f67cef4..dac616d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3928,11 +3928,12 @@ static inline char *fmt_multiple_name(struct device *dev, */ static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct snd_soc_dai *dai; + struct snd_soc_dai *dai, *_dai; - list_for_each_entry(dai, &component->dai_list, list) { + list_for_each_entry_safe(dai, _dai, &component->dai_list, list) { dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n", dai->name); + list_del(&dai->list); kfree(dai->name); kfree(dai); } -- 1.8.0

2 1

[alsa-devel] [PATCH v5-resend 0/7] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 12 Mar '14

12 Mar '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. The SiRF internal audio codec driver 2. The SiRF audio port driver 3. The SiRF internal audio card driver 4. The SiRF I2S controller driver 5. The SiRF hdmi card driver 6. The SiRF USP controller(DSP mode) driver 7. The SiRF bt-sco card driver -v5: 1. Drop all inlines. 2. Reordering the Kconfig and Makefile 3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it. 4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver 5. Add audio codec clock supply instead of adc event callback 6. Fixed playback and capture can't concurrent work bug. Rongjun Ying (7): ASoC: sirf: Add SiRF internal audio codec driver SoC: sirf: Add SiRF audio port driver is used by SiRF internal audio codec ASoC: sirf: Add SiRF audio card ASoC: sirf: Add SiRF I2S driver ASoC: sirf: Add hdmi card ASoC: sirf: Add usp driver which is used by dsp mode SoC: sirf: Add bt-sco card .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 ++ .../devicetree/bindings/sound/sirf-bt-sco.txt | 18 + .../devicetree/bindings/sound/sirf-hdmi.txt | 15 + .../devicetree/bindings/sound/sirf-i2s.txt | 27 + .../devicetree/bindings/sound/sirf-usp.txt | 27 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sirf-audio-codec.c | 533 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ sound/soc/sirf/Kconfig | 34 ++ sound/soc/sirf/Makefile | 13 + sound/soc/sirf/sirf-audio-port.c | 194 +++++++ sound/soc/sirf/sirf-audio-port.h | 62 +++ sound/soc/sirf/sirf-audio.c | 156 ++++++ sound/soc/sirf/sirf-bt-sco.c | 99 ++++ sound/soc/sirf/sirf-hdmi.c | 98 ++++ sound/soc/sirf/sirf-i2s.c | 434 ++++++++++++++++ sound/soc/sirf/sirf-i2s.h | 88 ++++ sound/soc/sirf/sirf-usp.c | 448 ++++++++++++++++ sound/soc/sirf/sirf-usp.h | 282 +++++++++++ 24 files changed, 2689 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c create mode 100644 sound/soc/sirf/sirf-bt-sco.c create mode 100644 sound/soc/sirf/sirf-hdmi.c create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-i2s.h create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

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[alsa-devel] HDA jack retask for Debian 7.4 NOT working
by Subodh Chiwate 12 Mar '14

12 Mar '14

Hi, I am trying to get surround 7.1 on debian 7.4 Wheezy. As guided by hda-jack-retask readme I built it from source on my system. But the changes are not reflecting and no errors by utility as well. I am getting Alsa 1.0.25 from the repository. Do I need to build the latest version from source to get hdajackretask as part of alsa-tools ? Any other suggestions to get the 7.1 working are welcome. Thanks in advance. -- Regards Subodh

2 2

[alsa-devel] [PATCH] ASoC: simple-card: overwrite DAIFMT_MASTER of cpu_dai->fmt
by Nicolin Chen 12 Mar '14

12 Mar '14

The current simple-card driver separates the daimft for cpu_dai and codec_dai. So we might get different values for them (0x4003 and 0x1003 for example): asoc-simple-card sound-cs42888.12: cpu : 2024000.esai / 4003 / 132000000 asoc-simple-card sound-cs42888.12: codec : cs42888 / 1003 / 24576000 asoc-simple-card sound-cs42888.12: cs42888 <-> 2024000.esai mapping ok It's pretty fair to do it for DAIFMT_INV since two dais might have differnt definitions in their drivers even though it'd be better to keep it same if we could. But for DAIFMT_MASTER bit, we should keep them identical. Otherwise, it'll make one of dai work in an incorrect mode. Thus this patch fixes it by overwriting the DAIFMT_MASTER bit of cpu_dai->fmt with the one of codec_dai->fmt since we defined DAIFMT_MASTER basing on CODEC at the first place. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/generic/simple-card.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5dd4769..f9a3c8a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -157,6 +157,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct device *dev) { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; + struct asoc_simple_dai *codec_dai = &priv->codec_dai; + struct asoc_simple_dai *cpu_dai = &priv->cpu_dai; struct device_node *np; char *name; int ret; @@ -189,7 +191,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->cpu_dai, + cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); if (ret < 0) @@ -200,12 +202,16 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, - &priv->codec_dai, + codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); if (ret < 0) return ret; + /* overwrite DAIFMT_MASTER of cpu_dai since it's based on CODEC */ + cpu_dai->fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + cpu_dai->fmt |= codec_dai->fmt & SND_SOC_DAIFMT_MASTER_MASK; + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) return -EINVAL; @@ -227,12 +233,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, - priv->cpu_dai.fmt, - priv->cpu_dai.sysclk); + cpu_dai->fmt, + cpu_dai->sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", dai_link->codec_dai_name, - priv->codec_dai.fmt, - priv->codec_dai.sysclk); + codec_dai->fmt, + codec_dai->sysclk); /* * soc_bind_dai_link() will check cpu name -- 1.8.4

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[alsa-devel] [PATCHv3 resend 0/3] Simplify the CODEC ASoC probe code.
by Xiubo Li 12 Mar '14

12 Mar '14

3 5

[alsa-devel] Creating single-card driver for multi-codec (TLV320AIC34) board - is it possible at the machine level?
by Matt Kleffner 12 Mar '14

12 Mar '14

Hi ALSA developers, I'm developing an ALSA driver for a new board with two TLV320AIC34 chips, which to a driver is the equivalent of 4 TLV320AIC31/2/3 (TLV320AIC3X driver) codecs. The platform is OMAP4, with the OMAP as I2C master. TDM mode with one McBSP port is used, and it is desired that the four codecs appear as one 8-channel codec (clocks are shared and data buses are connected together). It is OK if there are 4 separate mixers, one for each codec. However, when starting/stopping audio all codecs should act together and audio data presented to the ALSA user interface just as with a single 8-channel sound card. My questions have two parts: one on binding together the codecs into a "single card", and another on alterations to support an external sample-clock generator. BONDING CODECS TOGETHER It is desired to write the driver at the "machine" level (ignore the external-clock modification for now), without modifying the ALSA SoC core or the TLV320AIC3X driver. Is this possible? I've searched the internet and looked through the source code, and I have the following gaps in understanding: The following TI link seems to suggest such TLV320AIC34 operation may already be possible with ALSA: http://e2e.ti.com/support/data_converters/audio_converters/w/design_notes/2… However, one of the threads pointed in that link: http://comments.gmane.org/gmane.linux.alsa.devel/84899 suggests such support is not in the ALSA SoC core or the TLV320AIC3X driver. While it is mentioned that if configured appropriately, "you'll end up with a single card", later it is mentioned that "you're probably trying to break new ground here. You're trying to bind all the CODECs together and treat them as one digital interface, currently the code does a series of point to point links multiplexed on a bus only." What does point-to-point link mean in this context? What does "you'll end up with a single card" mean if all 8 channels are not exposed in one card? If it is new ground, then what does the rx51 driver do (it is often referenced as a starting point for this type of binding effort), which interfaces to an AIC34 codec? The threads referenced are a few years old. Is it now possible to write such a driver at the machine level, or must I additionally either modify the core or write a new codec driver (as the original poster did)? ALTERATIONS TO SUPPORT AN EXTERNAL SAMPLE-CLOCK GENERATOR The sample clock is being generated by a single, I2C-configurable clock chip. So any sample-rate configuration must be sent to the clock chip. Can this be handled purely in the machine layer, or do I need to modify the AIC3X driver? I understand I need to use the regmap API to do such things, and ALSA uses this API, but I do not yet know much about it. Any pointers to documentation on doing this besides the source code are appreciated. A key newbie question on regmap (and Linux I2C) as needed by the driver the following: Does the regmap-i2c or i2c driver "own" the I2C bus so that the each device on the bus can be read from/written to via the regmap-i2c or i2c driver without the device's driver having to request-take-release the bus on each access? Thanks, - Matt

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[alsa-devel] [PATCH v2] ASoC: cs42888: Add codec driver support
by Nicolin Chen 12 Mar '14

12 Mar '14

This patch adds support for the Cirrus Logic CS42888 Audio CODEC that has four 24-bit A/D and eight 24-bit D/A converters. [ CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Separated driver into I2C and main part drivers. * Added error out if failed to get clock. * Enabled Auto-mute function as default. * Disabled runtime-pm in remove(). * Added return value checking for regcache_sync(). * Fixed some trival parts like 'const' as Lars mentioned. .../devicetree/bindings/sound/cs42888.txt | 27 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42888-i2c.c | 68 +++ sound/soc/codecs/cs42888.c | 536 +++++++++++++++++++++ sound/soc/codecs/cs42888.h | 213 ++++++++ 6 files changed, 858 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42888.txt create mode 100644 sound/soc/codecs/cs42888-i2c.c create mode 100644 sound/soc/codecs/cs42888.c create mode 100644 sound/soc/codecs/cs42888.h diff --git a/Documentation/devicetree/bindings/sound/cs42888.txt b/Documentation/devicetree/bindings/sound/cs42888.txt new file mode 100644 index 0000000..7736527 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42888.txt @@ -0,0 +1,27 @@ +CS42888 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : phandle to the clock source for MCLK + + - clock-names : must contain "mclk". + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 32d7a6f..1642578 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS42888_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -303,6 +304,15 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42888 + tristate + +config SND_SOC_CS42888_I2C + tristate "Cirrus Logic CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42888 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cb46c4c..f5612fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,8 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42888-objs := cs42888.o +snd-soc-cs42888-i2c-objs := cs42888-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -178,6 +180,8 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42888) += snd-soc-cs42888.o +obj-$(CONFIG_SND_SOC_CS42888_I2C) += snd-soc-cs42888-i2c.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs42888-i2c.c b/sound/soc/codecs/cs42888-i2c.c new file mode 100644 index 0000000..54813bc --- /dev/null +++ b/sound/soc/codecs/cs42888-i2c.c @@ -0,0 +1,68 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +static int cs42888_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42888_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42888_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42888_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42888_i2c_id[] = { + {"cs42888", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42888_i2c_id); + +static struct i2c_driver cs42888_i2c_driver = { + .driver = { + .name = "cs42888", + .owner = THIS_MODULE, + .pm = &cs42888_pm, + }, + .probe = cs42888_i2c_probe, + .remove = cs42888_i2c_remove, + .id_table = cs42888_i2c_id, +}; + +module_i2c_driver(cs42888_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.c b/sound/soc/codecs/cs42888.c new file mode 100644 index 0000000..8cd3d4a --- /dev/null +++ b/sound/soc/codecs/cs42888.c @@ -0,0 +1,536 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +#define CS42888_NUM_SUPPLIES 4 +static const char *const cs42888_supply_names[CS42888_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42888_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42888_priv { + struct regulator_bulk_data supplies[CS42888_NUM_SUPPLIES]; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *const cs42888_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42888_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 4, 2, cs42888_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42888_ADCCTL, 3, 2, cs42888_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 5, 4, cs42888_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42888_TXCTL, 0, 4, cs42888_szc); + +static const struct snd_kcontrol_new cs42888_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42888_VOLAOUT1, + CS42888_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42888_VOLAOUT3, + CS42888_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42888_VOLAOUT5, + CS42888_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42888_VOLAOUT7, + CS42888_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42888_VOLAIN1, + CS42888_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42888_VOLAIN3, + CS42888_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42888_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42888_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42888_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42888_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42888_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42888_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42888_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42888_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum), + SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum), + SOC_SINGLE("DAC Single Volume Control Switch", CS42888_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42888_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42888_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42888_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_soc_dapm_widget cs42888_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42888_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42888_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42888_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42888_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42888_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42888_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42888_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_route cs42888_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +struct cs42888_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42888_ratios cs42888_ratios[] = { + { 64, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_256(4) }, + { 96, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_384(4) }, + { 128, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_512(4) }, + { 192, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_768(4) }, + { 256, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_256(1) }, + { 384, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_384(1) }, + { 512, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_512(1) }, + { 768, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42888_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + cs42888->sysclk = freq; + + return 0; +} + +static int cs42888_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42888_INTF_DAC_DIF_LEFTJ | CS42888_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42888_INTF_DAC_DIF_I2S | CS42888_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42888_INTF_DAC_DIF_RIGHTJ | CS42888_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42888->regmap, CS42888_INTF, + CS42888_INTF_DAC_DIF_MASK | + CS42888_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42888->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42888->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42888_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42888->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42888_ratios); i++) { + if (cs42888_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42888_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42888_FUNCMOD_MFREQ_MASK; + val = cs42888_ratios[i].mclk; + + fm = cs42888->slave_mode ? CS42888_FM_AUTO : cs42888_ratios[i].speed; + + regmap_update_bits(cs42888->regmap, CS42888_FUNCMOD, + CS42888_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42888_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42888_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42888->regmap, CS42888_DACMUTE, + CS42888_DACMUTE_ALL, mute ? CS42888_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42888_dai_ops = { + .set_fmt = cs42888_set_dai_fmt, + .set_sysclk = cs42888_set_dai_sysclk, + .hw_params = cs42888_hw_params, + .digital_mute = cs42888_digital_mute, +}; + +static struct snd_soc_dai_driver cs42888_dai = { + .name = "cs42888", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .ops = &cs42888_dai_ops, +}; + +static const struct reg_default cs42888_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42888_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_STATUS: + return true; + default: + return false; + } +} + +static bool cs42888_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_CHIPID: + case CS42888_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42888_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42888_LASTREG, + .reg_defaults = cs42888_reg, + .num_reg_defaults = ARRAY_SIZE(cs42888_reg), + .volatile_reg = cs42888_volatile_register, + .writeable_reg = cs42888_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42888_regmap_config); + +static int cs42888_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + /* Mute all DAC channels */ + regmap_write(cs42888->regmap, CS42888_DACMUTE, CS42888_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42888_driver = { + .probe = cs42888_codec_probe, + .idle_bias_off = true, + + .controls = cs42888_snd_controls, + .num_controls = ARRAY_SIZE(cs42888_snd_controls), + .dapm_widgets = cs42888_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42888_dapm_widgets), + .dapm_routes = cs42888_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42888_dapm_routes), +}; + +int cs42888_probe(struct device *dev, struct regmap *regmap) +{ + struct cs42888_priv *cs42888; + int ret, val, i; + + cs42888 = devm_kzalloc(dev, sizeof(*cs42888), GFP_KERNEL); + if (cs42888 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42888); + + cs42888->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42888->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42888->clk)); + return -EINVAL; + } + + cs42888->sysclk = clk_get_rate(cs42888->clk); + + for (i = 0; i < ARRAY_SIZE(cs42888->supplies); i++) + cs42888->supplies[i].supply = cs42888_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42888->regmap = regmap; + if (IS_ERR(cs42888->regmap)) { + ret = PTR_ERR(cs42888->regmap); + dev_err(dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42888->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42888->regmap, CS42888_CHIPID, &val); + if (val < 0) { + dev_err(dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42888_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42888_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42888_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42888->regmap, false); + + ret = snd_soc_register_codec(dev, &cs42888_driver, &cs42888_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42888->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42888_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42888_runtime_resume(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42888->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + goto err_clk; + } + + /* Make sure hardware reset done */ + msleep(5); + + regcache_cache_only(cs42888->regmap, false); + + ret = regcache_sync(cs42888->regmap); + if (ret) { + dev_err(dev, "failed to sync regmap: %d\n", ret); + goto err_bulk; + } + + return 0; + +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); +err_clk: + clk_disable_unprepare(cs42888->clk); + + return ret; +} + +static int cs42888_runtime_suspend(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + + regcache_cache_only(cs42888->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + clk_disable_unprepare(cs42888->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42888_pm = { + SET_RUNTIME_PM_OPS(cs42888_runtime_suspend, cs42888_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42888_pm); + +const struct of_device_id cs42888_of_match[] = { + { .compatible = "cirrus,cs42888", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42888_of_match); +EXPORT_SYMBOL_GPL(cs42888_of_match); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.h b/sound/soc/codecs/cs42888.h new file mode 100644 index 0000000..eda55ac --- /dev/null +++ b/sound/soc/codecs/cs42888.h @@ -0,0 +1,213 @@ +/* + * cs42888.h - Cirrus Logic CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _CS42888_H +#define _CS42888_H + +extern const struct dev_pm_ops cs42888_pm; +extern const struct regmap_config cs42888_regmap_config; +int cs42888_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42888_CHIPID 0x01 /* Chip ID */ +#define CS42888_PWRCTL 0x02 /* Power Control */ +#define CS42888_FUNCMOD 0x03 /* Functional Mode */ +#define CS42888_INTF 0x04 /* Interface Formats */ +#define CS42888_ADCCTL 0x05 /* ADC Control */ +#define CS42888_TXCTL 0x06 /* Transition Control */ +#define CS42888_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42888_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42888_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42888_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42888_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42888_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42888_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42888_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42888_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42888_DACINV 0x10 /* DAC Channel Invert */ +#define CS42888_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42888_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42888_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42888_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42888_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42888_STATUSCTL 0x18 /* Status Control */ +#define CS42888_STATUS 0x19 /* Status */ +#define CS42888_STATUSM 0x1A /* Status Mask */ +#define CS42888_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42888_FIRSTREG CS42888_CHIPID +#define CS42888_LASTREG CS42888_MUTEC +#define CS42888_NUMREGS (CS42888_LASTREG - CS42888_FIRSTREG + 1) +#define CS42888_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42888_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42888_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42888_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42888_PWRCTL_PDN_ADC2_MASK (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC2 (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42888_PWRCTL_PDN_ADC1_MASK (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1 (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42888_PWRCTL_PDN_DAC4_MASK (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4 (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42888_PWRCTL_PDN_DAC3_MASK (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3 (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42888_PWRCTL_PDN_DAC2_MASK (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2 (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42888_PWRCTL_PDN_DAC1_MASK (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1 (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_SHIFT 0 +#define CS42888_PWRCTL_PDN_MASK (1 << CS42888_PWRCTL_PDN_SHIFT) +#define CS42888_PWRCTL_PDN (1 << CS42888_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42888_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42888_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42888_FUNCMOD_DAC_FM_MASK (((1 << CS42888_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_DAC_FM(v) ((v) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42888_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42888_FUNCMOD_ADC_FM_MASK (((1 << CS42888_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM(v) ((v) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42888_FUNCMOD_DAC_FM_MASK : CS42888_FUNCMOD_ADC_FM_MASK) +#define CS42888_FUNCMOD_xC_FM(x, v) ((x) ? CS42888_FUNCMOD_DAC_FM(v) : CS42888_FUNCMOD_ADC_FM(v)) +#define CS42888_FUNCMOD_MFREQ_SHIFT 1 +#define CS42888_FUNCMOD_MFREQ_WIDTH 3 +#define CS42888_FUNCMOD_MFREQ_MASK (((1 << CS42888_FUNCMOD_MFREQ_WIDTH) - 1) << CS42888_FUNCMOD_MFREQ_SHIFT) +#define CS42888_FUNCMOD_MFREQ_256(s) ((0 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_384(s) ((1 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_512(s) ((2 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_768(s) ((3 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_1024(s) ((4 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42888_FM_SINGLE 0 +#define CS42888_FM_DOUBLE 1 +#define CS42888_FM_QUAD 2 +#define CS42888_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42888_INTF_FREEZE_SHIFT 7 +#define CS42888_INTF_FREEZE_MASK (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_FREEZE (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_AUX_DIF_SHIFT 6 +#define CS42888_INTF_AUX_DIF_MASK (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_AUX_DIF (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_SHIFT 3 +#define CS42888_INTF_DAC_DIF_WIDTH 3 +#define CS42888_INTF_DAC_DIF_MASK (((1 << CS42888_INTF_DAC_DIF_WIDTH) - 1) << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_LEFTJ (0 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_I2S (1 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ (2 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_20 (4 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_24 (6 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_TDM (7 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_SHIFT 0 +#define CS42888_INTF_ADC_DIF_WIDTH 3 +#define CS42888_INTF_ADC_DIF_MASK (((1 << CS42888_INTF_ADC_DIF_WIDTH) - 1) << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_LEFTJ (0 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_I2S (1 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ (2 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_20 (4 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_24 (6 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_TDM (7 << CS42888_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42888_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_ADC_HPF_FREEZE (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42888_ADCCTL_DAC_DEM_MASK (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_DAC_DEM (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42888_ADCCTL_ADC1_SINGLE_MASK (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42888_ADCCTL_ADC2_SINGLE_MASK (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42888_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42888_TXCTL_DAC_SNGVOL_MASK (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SNGVOL (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SHIFT 5 +#define CS42888_TXCTL_DAC_SZC_WIDTH 2 +#define CS42888_TXCTL_DAC_SZC_MASK (((1 << CS42888_TXCTL_DAC_SZC_WIDTH) - 1) << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_IC (0 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_ZC (1 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SR (2 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SRZC (3 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_AMUTE_SHIFT 4 +#define CS42888_TXCTL_AMUTE_MASK (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_AMUTE (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42888_TXCTL_MUTE_ADC_SP_MASK (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42888_TXCTL_ADC_SNGVOL_MASK (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SHIFT 0 +#define CS42888_TXCTL_ADC_SZC_MASK (((1 << CS42888_TXCTL_ADC_SZC_WIDTH) - 1) << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_IC (0 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_ZC (1 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SR (2 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SRZC (3 << CS42888_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42888_DACMUTE_AOUT(n) (0x1 << n) +#define CS42888_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42888_STATUSCTL_INI_SHIFT 2 +#define CS42888_STATUSCTL_INI_WIDTH 2 +#define CS42888_STATUSCTL_INI_MASK (((1 << CS42888_STATUSCTL_INI_WIDTH) - 1) << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_LOW (1 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_OPEN_DRAIN (2 << CS42888_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42888_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_MASK (1 << CS42888_STATUS_ADC2_OVFL_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_MASK (1 << CS42888_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42888_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_M_MASK (1 << CS42888_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_M_MASK (1 << CS42888_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42888_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42888_MUTEC_MCPOLARITY_MASK (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42888_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42888_H */ -- 1.8.4

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[alsa-devel] About snd_dmaengine_pcm_trigger()
by Kuninori Morimoto 12 Mar '14

12 Mar '14

Hi Mark, Lars and all I need your opinion Now, I'm working for ${LINUX}/sound/soc/sh/fsi and ${LINUX}/sound/soc/sh/rcar drirvers. These drivers are supporting DMAEngine transfer, but it is using original DMAEngine method, not using snd_dmaengine_pcm_trigger() or snd_dmaengine_pcm_register(). I was requested to use it from Mark. The reason why I couldn't use it was our DMAEngine didn't have "cyclic" tranfer support which is used on dmaengine_pcm_prepare_and_submit() But now, I created cyclic support on our DMA driver (on my local PC at this point) Then, I noticed our drivers still can't use snd_dmaengine_pcm_xxx() method. 1st, our device needs PIO tranfer support too. Current PIO/DMA transfer are sharing many methods/functions. Code will become difficult to read if it is separated forcibly. (to using snd_dmaengine_pcm_xxx()) 2nd, our device DMA ON/OFF timing has relation ship to other register settings. Unfortunately, our device is picky, it needs like this HW init setting -> DMA ON -> HW start setting It can be solved if snd_dmaengine_pcm_xxx() has callbacks, but many callbacks are needed... 3rd, our device needs special method on snd_pcm_ops int the future. From my point of view, our driver can replace to use "cyclic" DMA transfer instead of current original DMAEngine method. I guess, it is not difficult. But, using snd_dmaengine_pcm_xxx() is difficult. But, what do you think ?

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