June 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] wm8994 trivial merge fix
by Takashi Iwai 25 Jun '12

25 Jun '12

FYI, the patch below is what I applied from Jaroslav's tree. I simply picked it up directly into my tree (fix/asoc and for-linus branches) because of triviality. Takashi --- From 6cab3e1e65bfe070f09bb091eeda182b171d5929 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela <perex(a)perex.cz> Date: Mon, 28 May 2012 11:14:33 +0200 Subject: [PATCH] ASoC: wm8994: remove duplicate code It seems that the code duplication was added at a merge operation. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/wm8994.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index aa8c98b..1436b6c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -727,9 +727,6 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (!wm8994->jackdet || !wm8994->jack_cb) return; - if (!wm8994->jackdet || !wm8994->jack_cb) - return; - if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: dwc: Bodge around continuing absence of clock API stubs
by Mark Brown 25 Jun '12

25 Jun '12

The patches for stubbing out the generic clock API still haven't been applied so we need to either add ifdefs here or add a dependency until someone decides to actually apply them. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/dwc/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index 93e9fc3..e334900 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -1,5 +1,6 @@ config SND_DESIGNWARE_I2S tristate "Synopsys I2S Device Driver" + depends on CLKDEV_LOOKUP help Say Y or M if you want to add support for I2S driver for Synopsys desigwnware I2S device. The device supports upto -- 1.7.10

1 0

[alsa-devel] [PATCH] ASoC: dmaengine_pcm: fix typo in comment
by Uwe Kleine-König 25 Jun '12

25 Jun '12

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/soc-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 4756952..5647449 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -243,7 +243,7 @@ static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd * Note that this function will use private_data field of the substream's * runtime. So it is not availabe to your pcm driver implementation. If you need * to keep additional data attached to a substream use - * snd_dmaeinge_pcm_{set,get}_data. + * snd_dmaengine_pcm_{set,get}_data. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) -- 1.7.10

2 1

[alsa-devel] [RFC PATCH] Inverted internal mic
by David Henningsson 25 Jun '12

25 Jun '12

Hi, One of the more common problems on laptop machines, is that the internal mic provides a stereo signal but with one channel phase inverted, or differential output. This means problems for applications recording two channels but later merging them into one, leaving them with zero or near-zero output. There are various ways we can work around this in both the kernel, alsa-lib and PulseAudio layers. It's a matter of picking the right one. I'm leaning towards trying to fix it in the kernel's codec drivers, because 1) we already have quirking infrastructure there 2) we already have some working quirks already in that layer 3) it would benefit other sound applications that use ALSA directly. The downside to that is really that we're silencing out one channel for everyone, leading to no application being able to use both channels, even if they would implement some kind of "auto-detect-and-reverse-one-channel" functionality. For the most part, this has been some Acer Aspire machines running Realtek ALC268 / ALC269 / ALC271X / ALC272X, and for two of these we have proc coefs we can set, but for the other two these proc coefs, if they exist, are unknown to us. Recently I came across a Conexant as well, and I decided to write a patch for it, that would take the approach that the internal mic is forced mono on the kcontrol, and make sure the right channel is always muted. The patch is verified by the reporter to fix this problem. It could use some perfection though - it would make sense to to the same to the internal mic boost as well, and the strcmp('Internal Mic') call could maybe be turned into something more elegant. But before going ahead with that, I'd like to hear your opinion on the matter, if you agree that this is a good approach to the problem? References: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/903853/+attachme… https://bugs.launchpad.net/bugs/903853 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] Exclusive access to capture device
by Guy Rutenberg 25 Jun '12

25 Jun '12

Hi, Is there a way to open a capture device exclusively? E.g. I want to capture audio and be sure no other application can capture from that device at the same time. I've went through the API and searched the mailing list and haven't found a way to do so. Thanks a lot, Guy ---- http://www.guyrutenberg.com/

2 3

[alsa-devel] [PATCH 1/2] ASoC: Add shared code for Wolfson Arizona class devices
by Mark Brown 23 Jun '12

23 Jun '12

The Wolfson Arizona series of audio hub CODECs can share a large amount of their driver code as the result of a common register map. This patch adds some of this core support, providing a basis for the initial WM5102 audio driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 781 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 149 +++++++++ 4 files changed, 937 insertions(+) create mode 100644 sound/soc/codecs/arizona.c create mode 100644 sound/soc/codecs/arizona.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 43f5240..2ae8082 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -129,6 +129,11 @@ config SND_SOC_ALL_CODECS config SND_SOC_88PM860X tristate +config SND_SOC_ARIZONA + tristate + default y if SND_SOC_WM5102=y + default m if SND_SOC_WM5102=m + config SND_SOC_WM_HUBS tristate default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d30654..3005ea6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l52-objs := cs42l52.o @@ -128,6 +129,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o +obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c new file mode 100644 index 0000000..3b5730b --- /dev/null +++ b/sound/soc/codecs/arizona.c @@ -0,0 +1,781 @@ +/* + * arizona.c - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gcd.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" + +#define ARIZONA_AIF_BCLK_CTRL 0x00 +#define ARIZONA_AIF_TX_PIN_CTRL 0x01 +#define ARIZONA_AIF_RX_PIN_CTRL 0x02 +#define ARIZONA_AIF_RATE_CTRL 0x03 +#define ARIZONA_AIF_FORMAT 0x04 +#define ARIZONA_AIF_TX_BCLK_RATE 0x05 +#define ARIZONA_AIF_RX_BCLK_RATE 0x06 +#define ARIZONA_AIF_FRAME_CTRL_1 0x07 +#define ARIZONA_AIF_FRAME_CTRL_2 0x08 +#define ARIZONA_AIF_FRAME_CTRL_3 0x09 +#define ARIZONA_AIF_FRAME_CTRL_4 0x0A +#define ARIZONA_AIF_FRAME_CTRL_5 0x0B +#define ARIZONA_AIF_FRAME_CTRL_6 0x0C +#define ARIZONA_AIF_FRAME_CTRL_7 0x0D +#define ARIZONA_AIF_FRAME_CTRL_8 0x0E +#define ARIZONA_AIF_FRAME_CTRL_9 0x0F +#define ARIZONA_AIF_FRAME_CTRL_10 0x10 +#define ARIZONA_AIF_FRAME_CTRL_11 0x11 +#define ARIZONA_AIF_FRAME_CTRL_12 0x12 +#define ARIZONA_AIF_FRAME_CTRL_13 0x13 +#define ARIZONA_AIF_FRAME_CTRL_14 0x14 +#define ARIZONA_AIF_FRAME_CTRL_15 0x15 +#define ARIZONA_AIF_FRAME_CTRL_16 0x16 +#define ARIZONA_AIF_FRAME_CTRL_17 0x17 +#define ARIZONA_AIF_FRAME_CTRL_18 0x18 +#define ARIZONA_AIF_TX_ENABLES 0x19 +#define ARIZONA_AIF_RX_ENABLES 0x1A +#define ARIZONA_AIF_FORCE_WRITE 0x1B + +#define arizona_fll_err(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_warn(_fll, fmt, ...) \ + dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_dbg(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + +#define arizona_aif_err(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_warn(_dai, fmt, ...) \ + dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_dbg(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + +const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { + "None", + "Tone Generator 1", + "Tone Generator 2", + "Haptics", + "AEC", + "Mic Mute Mixer", + "Noise Generator", + "IN1L", + "IN1R", + "IN2L", + "IN2R", + "IN3L", + "IN3R", + "AIF1RX1", + "AIF1RX2", + "AIF1RX3", + "AIF1RX4", + "AIF1RX5", + "AIF1RX6", + "AIF1RX7", + "AIF1RX8", + "AIF2RX1", + "AIF2RX2", + "AIF3RX1", + "AIF3RX2", + "SLIMRX1", + "SLIMRX2", + "SLIMRX3", + "SLIMRX4", + "SLIMRX5", + "SLIMRX6", + "SLIMRX7", + "SLIMRX8", + "EQ1", + "EQ2", + "EQ3", + "EQ4", + "DRC1L", + "DRC1R", + "DRC2L", + "DRC2R", + "LHPF1", + "LHPF2", + "LHPF3", + "LHPF4", + "DSP1.1", + "DSP1.2", + "DSP1.3", + "DSP1.4", + "DSP1.5", + "DSP1.6", + "ASRC1L", + "ASRC1R", + "ASRC2L", + "ASRC2R", +}; +EXPORT_SYMBOL_GPL(arizona_mixer_texts); + +int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { + 0x00, /* None */ + 0x04, /* Tone */ + 0x05, + 0x06, /* Haptics */ + 0x08, /* AEC */ + 0x0c, /* Noise mixer */ + 0x0d, /* Comfort noise */ + 0x10, /* IN1L */ + 0x11, + 0x12, + 0x13, + 0x14, + 0x15, + 0x20, /* AIF1RX1 */ + 0x21, + 0x22, + 0x23, + 0x24, + 0x25, + 0x26, + 0x27, + 0x28, /* AIF2RX1 */ + 0x29, + 0x30, /* AIF3RX1 */ + 0x31, + 0x38, /* SLIMRX1 */ + 0x39, + 0x3a, + 0x3b, + 0x3c, + 0x3d, + 0x3e, + 0x3f, + 0x50, /* EQ1 */ + 0x51, + 0x52, + 0x53, + 0x58, /* DRC1L */ + 0x59, + 0x5a, + 0x5b, + 0x60, /* LHPF1 */ + 0x61, + 0x62, + 0x63, + 0x68, /* DSP1.1 */ + 0x69, + 0x6a, + 0x6b, + 0x6c, + 0x6d, + 0x90, /* ASRC1L */ + 0x91, + 0x92, + 0x93, +}; +EXPORT_SYMBOL_GPL(arizona_mixer_values); + +const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); +EXPORT_SYMBOL_GPL(arizona_mixer_tlv); + +static const char *arizona_lhpf_mode_text[] = { + "Low-pass", "High-pass" +}; + +const struct soc_enum arizona_lhpf1_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf1_mode); + +const struct soc_enum arizona_lhpf2_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf2_mode); + +const struct soc_enum arizona_lhpf3_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf3_mode); + +const struct soc_enum arizona_lhpf4_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); + +int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_in_ev); + +int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_out_ev); + +int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + char *name; + unsigned int reg; + unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK; + unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT; + unsigned int *clk; + + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + name = "SYSCLK"; + reg = ARIZONA_SYSTEM_CLOCK_1; + clk = &priv->sysclk; + mask |= ARIZONA_SYSCLK_FRAC; + break; + case ARIZONA_CLK_ASYNCCLK: + name = "ASYNCCLK"; + reg = ARIZONA_ASYNC_CLOCK_1; + clk = &priv->asyncclk; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 6144000: + break; + case 11289600: + case 12288000: + val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 22579200: + case 24576000: + val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 45158400: + case 49152000: + val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + default: + return -EINVAL; + } + + *clk = freq; + + if (freq % 6144000) + val |= ARIZONA_SYSCLK_FRAC; + + dev_dbg(arizona->dev, "%s set to %uHz", name, freq); + + return regmap_update_bits(arizona->regmap, reg, mask, val); +} +EXPORT_SYMBOL_GPL(arizona_set_sysclk); + +static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int lrclk, bclk, mode, base; + + base = dai->driver->base; + + lrclk = 0; + bclk = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + mode = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + break; + case SND_SOC_DAIFMT_I2S: + mode = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode = 3; + break; + default: + arizona_aif_err(dai, "Unsupported DAI format %d\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + default: + arizona_aif_err(dai, "Unsupported master mode %d\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= ARIZONA_AIF1_BCLK_INV; + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= ARIZONA_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR, + bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL, + ARIZONA_AIF1TX_LRCLK_INV | + ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL, + ARIZONA_AIF1RX_LRCLK_INV | + ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT, + ARIZONA_AIF1_FMT_MASK, mode); + + return 0; +} + +static const int arizona_48k_rates[] = { + -1, + 48000, + 64000, + 96000, + 128000, + 192000, + 256000, + 384000, + 512000, + 768000, + 1024000, + 1536000, + 2048000, + 3072000, + 4096000, + 6144000, + 8192000, + 12288000, + 24576000, +}; + +static const int arizona_44k1_rates[] = { + -1, + 44100, + 58800, + 88200, + 117600, + 177640, + 235200, + 352800, + 470400, + 705600, + 940800, + 1411200, + 1881600, + 2882400, + 3763200, + 5644800, + 7526400, + 11289600, + 22579200, +}; + +static int arizona_sr_vals[] = { + 0, + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 0, + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static int arizona_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + int base = dai->driver->base; + const int *rates; + int i; + int bclk, lrclk, wl, frame, sr_val; + + if (params_rate(params) % 8000) + rates = &arizona_44k1_rates[0]; + else + rates = &arizona_48k_rates[0]; + + for (i = 0; i < ARRAY_SIZE(arizona_44k1_rates); i++) { + if (rates[i] == snd_soc_params_to_bclk(params)) { + bclk = i; + break; + } + } + if (i == ARRAY_SIZE(arizona_44k1_rates)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for the chip. + */ + for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) + if (arizona_sr_vals[i] == params_rate(params)) + break; + if (i == ARRAY_SIZE(arizona_sr_vals)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + sr_val = i; + + lrclk = snd_soc_params_to_bclk(params) / params_rate(params); + + arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", + rates[bclk], rates[bclk] / lrclk); + + wl = snd_pcm_format_width(params_format(params)); + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + + return 0; +} + +const struct snd_soc_dai_ops arizona_dai_ops = { + .set_fmt = arizona_set_fmt, + .hw_params = arizona_hw_params, +}; + +static irqreturn_t arizona_fll_lock(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "Locked\n"); + + complete(&fll->lock); + + return IRQ_HANDLED; +} + +static irqreturn_t arizona_fll_clock_ok(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "clock OK\n"); + + complete(&fll->ok); + + return IRQ_HANDLED; +} + +static struct { + unsigned int min; + unsigned int max; + u16 fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +struct arizona_fll_cfg { + int n; + int theta; + int lambda; + int refdiv; + int outdiv; + int fratio; +}; + +static int arizona_calc_fll(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int Fref, + unsigned int Fout) +{ + unsigned int target, div, gcd_fll; + int i, ratio; + + arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout); + + /* Fref must be <=13.5MHz */ + div = 1; + cfg->refdiv = 0; + while ((Fref / div) > 13500000) { + div *= 2; + cfg->refdiv++; + + if (div > 8) { + arizona_fll_err(fll, + "Can't scale %dMHz in to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 1; + while (Fout * div < 90000000) { + div++; + if (div > 7) { + arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + cfg->outdiv = div; + + arizona_fll_dbg(fll, "Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + cfg->fratio = fll_fratios[i].fratio; + ratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", + Fref); + return -EINVAL; + } + + cfg->n = target / (ratio * Fref); + + if (target % Fref) { + gcd_fll = gcd(target, ratio * Fref); + arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll); + + cfg->theta = (target - (cfg->n * ratio * Fref)) + / gcd_fll; + cfg->lambda = (ratio * Fref) / gcd_fll; + } else { + cfg->theta = 0; + cfg->lambda = 0; + } + + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", + cfg->n, cfg->theta, cfg->lambda); + arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", + cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); + + return 0; + +} + +static void arizona_apply_fll(struct arizona *arizona, unsigned int base, + struct arizona_fll_cfg *cfg, int source) +{ + regmap_update_bits(arizona->regmap, base + 3, + ARIZONA_FLL1_THETA_MASK, cfg->theta); + regmap_update_bits(arizona->regmap, base + 4, + ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); + regmap_update_bits(arizona->regmap, base + 5, + ARIZONA_FLL1_FRATIO_MASK, + cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); + regmap_update_bits(arizona->regmap, base + 6, + ARIZONA_FLL1_CLK_REF_DIV_MASK | + ARIZONA_FLL1_CLK_REF_SRC_MASK, + cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | + source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + + regmap_update_bits(arizona->regmap, base + 2, + ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, + ARIZONA_FLL1_CTRL_UPD | cfg->n); +} + +int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout) +{ + struct arizona *arizona = fll->arizona; + struct arizona_fll_cfg cfg, sync; + unsigned int reg, val; + int syncsrc; + bool ena; + int ret; + + ret = regmap_read(arizona->regmap, fll->base + 1, &reg); + if (ret != 0) { + arizona_fll_err(fll, "Failed to read current state: %d\n", + ret); + return ret; + } + ena = reg & ARIZONA_FLL1_ENA; + + if (Fout) { + /* Do we have a 32kHz reference? */ + regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); + switch (val & ARIZONA_CLK_32K_SRC_MASK) { + case ARIZONA_CLK_SRC_MCLK1: + case ARIZONA_CLK_SRC_MCLK2: + syncsrc = val & ARIZONA_CLK_32K_SRC_MASK; + break; + default: + syncsrc = -1; + } + + if (source == syncsrc) + syncsrc = -1; + + if (syncsrc >= 0) { + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; + + ret = arizona_calc_fll(fll, &cfg, 32768, Fout); + if (ret != 0) + return ret; + } else { + ret = arizona_calc_fll(fll, &cfg, Fref, Fout); + if (ret != 0) + return ret; + } + } else { + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, 0); + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); + + if (ena) + pm_runtime_put_autosuspend(arizona->dev); + + return 0; + } + + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + if (syncsrc >= 0) { + arizona_apply_fll(arizona, fll->base, &cfg, syncsrc); + arizona_apply_fll(arizona, fll->base + 0x10, &sync, source); + } else { + arizona_apply_fll(arizona, fll->base, &cfg, source); + } + + if (!ena) + pm_runtime_get(arizona->dev); + + /* Clear any pending completions */ + try_wait_for_completion(&fll->ok); + + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + if (syncsrc >= 0) + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); + + ret = wait_for_completion_timeout(&fll->ok, + msecs_to_jiffies(25)); + if (ret == 0) + arizona_fll_warn(fll, "Timed out waiting for lock\n"); + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_set_fll); + +int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, + int ok_irq, struct arizona_fll *fll) +{ + int ret; + + init_completion(&fll->lock); + init_completion(&fll->ok); + + fll->id = id; + fll->base = base; + fll->arizona = arizona; + + snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id); + snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), + "FLL%d clock OK", id); + + ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, + arizona_fll_lock, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", + id, ret); + } + + ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, + arizona_fll_clock_ok, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", + id, ret); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_fll); + +MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h new file mode 100644 index 0000000..8c2ca1d --- /dev/null +++ b/sound/soc/codecs/arizona.h @@ -0,0 +1,149 @@ +/* + * arizona.h - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ASOC_ARIZONA_H +#define _ASOC_ARIZONA_H + +#include <linux/completion.h> + +#include <sound/soc.h> + +#define ARIZONA_CLK_SYSCLK 1 +#define ARIZONA_CLK_ASYNCCLK 2 + +#define ARIZONA_CLK_SRC_MCLK1 0x0 +#define ARIZONA_CLK_SRC_MCLK2 0x1 +#define ARIZONA_CLK_SRC_FLL1 0x4 +#define ARIZONA_CLK_SRC_FLL2 0x5 +#define ARIZONA_CLK_SRC_AIF1BCLK 0x8 +#define ARIZONA_CLK_SRC_AIF2BCLK 0x9 +#define ARIZONA_CLK_SRC_AIF3BCLK 0xa + +#define ARIZONA_FLL_SRC_MCLK1 0 +#define ARIZONA_FLL_SRC_MCLK2 1 +#define ARIZONA_FLL_SRC_SLIMCLK 2 +#define ARIZONA_FLL_SRC_FLL1 3 +#define ARIZONA_FLL_SRC_FLL2 4 +#define ARIZONA_FLL_SRC_AIF1BCLK 5 +#define ARIZONA_FLL_SRC_AIF2BCLK 6 +#define ARIZONA_FLL_SRC_AIF3BCLK 7 +#define ARIZONA_FLL_SRC_AIF1LRCLK 8 +#define ARIZONA_FLL_SRC_AIF2LRCLK 9 +#define ARIZONA_FLL_SRC_AIF3LRCLK 10 + +#define ARIZONA_MIXER_VOL_MASK 0x00FE +#define ARIZONA_MIXER_VOL_SHIFT 1 +#define ARIZONA_MIXER_VOL_WIDTH 7 + +struct arizona; + +struct arizona_priv { + struct arizona *arizona; + int sysclk; + int asyncclk; +}; + +#define ARIZONA_NUM_MIXER_INPUTS 55 + +extern const unsigned int arizona_mixer_tlv[]; +extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; +extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; + +#define ARIZONA_MIXER_CONTROLS(name, base) \ + SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 2 Volume", base + 3, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 3 Volume", base + 5, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv) + +#define ARIZONA_MUX_ENUM_DECL(name, reg) \ + SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \ + arizona_mixer_texts, arizona_mixer_values) + +#define ARIZONA_MUX_CTL_DECL(name) \ + const struct snd_kcontrol_new name##_mux = \ + SOC_DAPM_VALUE_ENUM("Route", name##_enum) + +#define ARIZONA_MIXER_ENUMS(name, base_reg) \ + static ARIZONA_MUX_ENUM_DECL(name##_in1_enum, base_reg); \ + static ARIZONA_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \ + static ARIZONA_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \ + static ARIZONA_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \ + static ARIZONA_MUX_CTL_DECL(name##_in1); \ + static ARIZONA_MUX_CTL_DECL(name##_in2); \ + static ARIZONA_MUX_CTL_DECL(name##_in3); \ + static ARIZONA_MUX_CTL_DECL(name##_in4) + +#define ARIZONA_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +#define ARIZONA_MIXER_WIDGETS(name, name_str) \ + ARIZONA_MUX(name_str " Input 1", &name##_in1_mux), \ + ARIZONA_MUX(name_str " Input 2", &name##_in2_mux), \ + ARIZONA_MUX(name_str " Input 3", &name##_in3_mux), \ + ARIZONA_MUX(name_str " Input 4", &name##_in4_mux), \ + SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0) + +#define ARIZONA_MIXER_ROUTES(widget, name) \ + { widget, NULL, name " Mixer" }, \ + { name " Mixer", NULL, name " Input 1" }, \ + { name " Mixer", NULL, name " Input 2" }, \ + { name " Mixer", NULL, name " Input 3" }, \ + { name " Mixer", NULL, name " Input 4" }, \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 1"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 2"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 3"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 4") + +extern const struct soc_enum arizona_lhpf1_mode; +extern const struct soc_enum arizona_lhpf2_mode; +extern const struct soc_enum arizona_lhpf3_mode; +extern const struct soc_enum arizona_lhpf4_mode; + +extern int arizona_in_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); +extern int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); + +extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); + +extern const struct snd_soc_dai_ops arizona_dai_ops; + +#define ARIZONA_FLL_NAME_LEN 20 + +struct arizona_fll { + struct arizona *arizona; + int id; + unsigned int base; + struct completion lock; + struct completion ok; + + char lock_name[ARIZONA_FLL_NAME_LEN]; + char clock_ok_name[ARIZONA_FLL_NAME_LEN]; +}; + +extern int arizona_init_fll(struct arizona *arizona, int id, int base, + int lock_irq, int ok_irq, struct arizona_fll *fll); +extern int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout); + +#endif -- 1.7.10

1 1

Re: [alsa-devel] [PATCH] sound: fix MODULE_LICENSE typos
by Mark Brown 23 Jun '12

23 Jun '12

On Fri, Jun 22, 2012 at 04:21:49PM -0700, Kees Cook wrote: > include/linux/license.h does not recognize "GPLv2", so adjust these > declarations to add the needed space before the version. This patch isn't against current -next which doesn't have this issue, I've no idea what you generated against as a version with this has never even been in -next. Please do also use subject lines that resemble those used in the rest of the subsystem, having to hand edit commit logs is tedious. It seems more sensible to fix license.h, GPLv2 is a pretty common way of writing the license name in the wider community.

1 0

[alsa-devel] [PATCH] ASoC: core: Add DOUBLE_R variants of the _RANGE controls
by Mark Brown 22 Jun '12

22 Jun '12

The code handles this fine already, we just need new macros in the header for drivers to create the controls. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index e4348d2..e063380 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -42,6 +42,10 @@ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ .max = xmax, .platform_max = xmax, .invert = xinvert}) +#define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ + .min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -96,6 +100,13 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -114,6 +125,16 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE_TLV(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ -- 1.7.10

2 1

[alsa-devel] [PATCH] ASoC: dapm: Try to add all routes even if one fails
by Mark Brown 22 Jun '12

22 Jun '12

We may as well print as many errors as we can in one go rather than requiring developers to iterate through all their typos. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 667a941..3e7baab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2276,15 +2276,15 @@ err: int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret = 0; + int i, r, ret = 0; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(dapm, route); - if (ret < 0) { + r = snd_soc_dapm_add_route(dapm, route); + if (r < 0) { dev_err(dapm->dev, "Failed to add route %s->%s\n", route->source, route->sink); - break; + ret = r; } route++; } -- 1.7.10

2 1

[PATCH] ALSA: hda - Add inverted mic quirks for Asus U41SV, Acer 1810TZ and AOD260
by David Henningsson 22 Jun '12

22 Jun '12

These machines have inverted phase on right channel for their internal mics. BugLink: https://bugs.launchpad.net/bugs/997227 BugLink: https://bugs.launchpad.net/bugs/996611 BugLink: https://bugs.launchpad.net/bugs/1006089 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53cd730..9be7da1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5918,6 +5918,14 @@ static void alc269_fixup_mic2_mute(struct hda_codec *codec, } } +static void alc269_fixup_inv_dmic(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PROBE) + alc_add_inv_dmic_mixer(codec, 0x12); +} + + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5936,6 +5944,7 @@ enum { ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, ALC269_FIXUP_MIC2_MUTE_LED, + ALC269_FIXUP_INV_DMIC, }; static const struct alc_fixup alc269_fixups[] = { @@ -6060,12 +6069,19 @@ static const struct alc_fixup alc269_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, }, + [ALC269_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_inv_dmic, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), + SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), @@ -6534,12 +6550,7 @@ static void alc272_fixup_mario(struct hda_codec *codec, "hda_codec: failed to override amp caps for NID 0x2\n"); } -static void alc662_fixup_inv_dmic(struct hda_codec *codec, - const struct alc_fixup *fix, int action) -{ - if (action == ALC_FIXUP_ACT_PROBE) - alc_add_inv_dmic_mixer(codec, 0x12); -} +#define alc662_fixup_inv_dmic alc269_fixup_inv_dmic enum { ALC662_FIXUP_ASPIRE, -- 1.7.9.5 --------------030106040506020209090705--

1 0

[alsa-devel] [PATCH] ASoC: da732x: Staticise non-exported symbol soc_codec_dev_da732x
by Mark Brown 22 Jun '12

22 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/da732x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 04af369..01be2a3 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1543,7 +1543,7 @@ static int da732x_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver soc_codec_dev_da732x = { +static struct snd_soc_codec_driver soc_codec_dev_da732x = { .probe = da732x_probe, .remove = da732x_remove, .set_bias_level = da732x_set_bias_level, -- 1.7.10

2 1

[alsa-devel] [PATCH] ASoC: isabelle: Staticise non-exported isabelle_dai
by Mark Brown 22 Jun '12

22 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/isabelle.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 0d62f3b..5d8f39e 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1036,7 +1036,7 @@ static struct snd_soc_dai_ops isabelle_ul_dai_ops = { }; /* ISABELLE dai structure */ -struct snd_soc_dai_driver isabelle_dai[] = { +static struct snd_soc_dai_driver isabelle_dai[] = { { .name = "isabelle-dl1", .playback = { -- 1.7.10

1 0

[alsa-devel] [PATCH] ALSA: hda - Don't power up when not powered down.
by Dylan Reid 22 Jun '12

22 Jun '12

After cancel_delayed_work_sync returns, the power down work either never started (power_on == 1) or finished (power_on == 0). In the former case there is no need to power up again. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 604699c..045b5e7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4444,6 +4444,13 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) cancel_delayed_work_sync(&codec->power_work); spin_lock(&codec->power_lock); + /* If the power down delayed work was cancelled above before starting, + * then there is no need to go through power up here. + */ + if (codec->power_on) { + spin_unlock(&codec->power_lock); + return; + } trace_hda_power_up(codec); snd_hda_update_power_acct(codec); codec->power_on = 1; -- 1.7.9.rc0

2 3

[alsa-devel] Need ALSA soc architecture Documents
by raj raja 22 Jun '12

22 Jun '12

Hi, I am very new to ALSA soc core . I want to have some basic knowledge on ALSA soc core architecture Where can i find the related pfd or ppt format documents Thanks Yuvaraj R

1 0

[alsa-devel] [PATCH V3 12/16] ALSA: HDA: Make hda sound card usable for Loongson.
by Huacai Chen 22 Jun '12

22 Jun '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch modify patch_conexant.c to add Lemote specific code. Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Clemens Ladisch <clemens(a)ladisch.de> Cc: alsa-devel(a)alsa-project.org --- include/linux/pci_ids.h | 2 + sound/pci/hda/patch_conexant.c | 52 ++++++++++++++++++++++++++++++++++++++- 2 files changed, 52 insertions(+), 2 deletions(-) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index ab741b0..d8b0a52 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -2906,3 +2906,5 @@ #define PCI_DEVICE_ID_XEN_PLATFORM 0x0001 #define PCI_VENDOR_ID_OCZ 0x1b85 + +#define PCI_VENDOR_ID_LEMOTE 0x1c06 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 172370b..ca3cc237 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int thinkpad:1; unsigned int hp_laptop:1; unsigned int asus:1; + unsigned int lemote:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; @@ -2278,7 +2279,7 @@ static void cxt5066_automic(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); - else if (spec->asus) + else if (spec->asus || spec->lemote) cxt5066_asus_automic(codec); } @@ -2911,6 +2912,32 @@ static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_lemote[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1: Connection index: 0 */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* ADC2, ADC3: unused */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -2948,6 +2975,8 @@ enum { CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ + CXT5066_LEMOTE_A1004, /* Lemote Laptop A1004 */ + CXT5066_LEMOTE_A1205, /* Lemote All-In-One A1205 */ CXT5066_AUTO, /* BIOS auto-parser */ CXT5066_MODELS }; @@ -2961,6 +2990,8 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_THINKPAD] = "thinkpad", [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", + [CXT5066_LEMOTE_A1004] = "lemote-laptop-a1004", + [CXT5066_LEMOTE_A1205] = "lemote-aio-a1205", [CXT5066_AUTO] = "auto", }; @@ -2993,6 +3024,8 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2011, "Lemote A1004", CXT5066_LEMOTE_A1004), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2012, "Lemote A1205", CXT5066_LEMOTE_A1205), SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; @@ -3073,7 +3106,22 @@ static int patch_cxt5066(struct hda_codec *codec) spec->port_d_mode = 0; spec->mic_boost = 3; /* default 30dB gain */ break; - + case CXT5066_LEMOTE_A1004: + case CXT5066_LEMOTE_A1205: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_lemote; + spec->num_init_verbs++; + spec->lemote = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- 1.7.7.3

1 0

[alsa-devel] [PATCH] asoc: bfin: use dev_err to print error log instead of dev_dbg
by Scott Jiang 21 Jun '12

21 Jun '12

Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/bf6xx-sport.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index f19a72b..318c5ba 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -256,9 +256,9 @@ static irqreturn_t sport_err_irq(int irq, void *dev_id) struct device *dev = &sport->pdev->dev; if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) - dev_dbg(dev, "sport error: TUVF\n"); + dev_err(dev, "sport error: TUVF\n"); if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) - dev_dbg(dev, "sport error: ROVF\n"); + dev_err(dev, "sport error: ROVF\n"); return IRQ_HANDLED; } -- 1.7.0.4

4 7

[alsa-devel] [PATCH] ALSA: HDA: Add inverted internal mic quirk for Lenovo S205
by David Henningsson 21 Jun '12

21 Jun '12

The Lenovo Ideapad S205 has an internal mic where the right channel is phase inverted. BugLink: https://bugs.launchpad.net/bugs/884652 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 172370b..2af0868 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4466,6 +4466,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), {} }; -- 1.7.9.5

2 1

[alsa-devel] why snd_soc_suspend is not called
by zhangfei gao 21 Jun '12

21 Jun '12

Hi, Mark Excuse me, I am confused about snd_soc_suspend. When realizing cpu_dai->driver->suspend, we found snd_soc_suspend is not called. In fact, even soc_probe is not called since no soc-audio device. We manually alloc soc-audio deivce before in platfrom.c. dev = platform_device_alloc("soc-audio", -1); platform_set_drvdata(device, &snd_soc_card); platform_device_add(dev); When running, system recommend using snd_soc_register_card instead. After doing such modification, the audio could work. But strangely snd_soc_suspend is not called, as well as soc-probe. So cpu_dai->driver->suspend will no be called accordingly. If manually add "soc-audio" in arch/, the soc-probe can be called. But snd_soc_suspend still fail to be called. Could you give me some suggestion? Thanks a lot. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 5

[alsa-devel] Tempo event in a Midi sequencer...
by Ru Vuott 21 Jun '12

21 Jun '12

Hello, I am finally able to make a simple midi sequencer with the Gambas language. It works, but I have some little problem about changing the metronomic Tempo within a midi song. Since I used the method of sending an Echo event to send Midi Messages groups to ALSA, I thought I'd solve the problem of an eventual return of metronomic Time by sending an echo and backing from Alsa custom (Type: 90). At the time of receipt of the echo by the Alsa function: int snd_seq_event_input(snd_seq_t *handle, snd_seq_event_t **ev) the code sends the value of the tempo to a routine, where there are these Alsa functions: snd_seq_queue_tempo_malloc(...) snd_seq_queue_tempo_set_tempo(...) snd_seq_set_queue_tempo(...) snd_seq_queue_tempo_free(...) ...obviously with teir parametres. I noticed, though, unfortunately (incomprehensibly) that often the change is a bit 'earlier than required. I do not think there is a problem of Delta Time, since the Delta Time of metronomic Tempo event is treated the same way as that relating to other MIDI events. Looking for a solution, I had a look at the code of your Aplaymidi, and I saw this statement: case 0x51: /* tempo */ ...... ...etc event->type = SND_SEQ_EVENT_TEMPO; ...etc, so, I thought that the change of Tempo occur simply by sending an event by identifying it with precisely the type number: 35. I performed the test, checking the result is listening to the song, seeing both as reported by Aseqdump. I guess it Aseqdump me signals a correct delivery, because it gives me back this response: queue + a number depending on the value sent with the Tempo event. Instead, listening to the midi track, the change does not happen! I am writing to ask you, if my analysis is correct, and if it is sufficient to send only time that event, or if some function of Alsa is absolutely necessary Thanks Vuotttttttt

2 1

Re: [alsa-devel] [PATCH v2 0/5] mmp audio support
by zhangfei gao 21 Jun '12

21 Jun '12

On Thu, Jun 14, 2012 at 12:37 AM, Chris Ball <cjb(a)laptop.org> wrote: > Hi Zhangfei, > > On Mon, Jun 11 2012, Zhangfei Gao wrote: >> v1->v2: >> modify accroingly with suggestion from Vinod and Mark >> remove deprecated chan->private and dma.h >> remove prep_slave_sg >> remove module_init and move operation to probe >> misc change >> >> v0->v1: >> dmaengine -> soc-dmaengine. >> update according to Mark's suggestion >> >> These patches provide mmp audio support under alsa via dmaengine >> Support platfrom pxa688 and pxa910 >> mmp_tdma.c and mmp-pcm.c is shared for pxa688 and pxa910 >> pxa688 (mmp2) use mmp2-sspa.c >> pxa910 directly use pxa-ssp.c >> mmp_tdma.c is under dmaengine framework >> >> Verified on brownstone (pxa688) and ttc-dkb > > Great, we're looking into adopting this for OLPC XO-1.75 too. > > Could you explain how system suspend/resume should work with this > patchset? You don't seem to be adding any suspend/resume handlers. > > Thanks, > > - Chris. Chris Some confusion about audio suspend. We use android environment, wake lock is hold preventing system entering suspend when audio work. Only entering early-suspend, which close LCD etc. If this is the case, we may not need consider suspend during audio play back. If support suspend during audio work, some modification required. I found SNDRV_PCM_TRIGGER_START/STOP is send in suspend/resume instead of SNDRV_PCM_TRIGGER_RESUME/SUSPEND as before. The result is dmaengine_pcm_prepare_and_submit will be re-called every suspend/resume. Unfortunately we malloc descriptor in prep function, but no function to free at such case. Thanks -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

Re: [alsa-devel] [PATCH 1/5] dmaengine: mmp_tdma: add mmp tdma support
by Vinod Koul 21 Jun '12

21 Jun '12

On Fri, 2012-06-15 at 11:04 +0800, Zhangfei Gao wrote: > Add support for two-channel dma under dmaengine > support: mmp-adma and pxa910-squ Applied, thanks -- ~Vinod -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH 1/2 v2] asoc: add sport driver for bf6xx soc
by Scott Jiang 21 Jun '12

21 Jun '12

The SPORT(Serial Port) module on bf6xx soc has a totally different ip comparing to bf5xx soc. An individual SPORT module consists of two independently configurable SPORT halves with identical functionality. Each SPORT half can be configured for either transmitter or receiver. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/Kconfig | 10 +- sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf6xx-sport.c | 422 ++++++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf6xx-sport.h | 82 ++++++++ 4 files changed, 513 insertions(+), 3 deletions(-) create mode 100644 sound/soc/blackfin/bf6xx-sport.c create mode 100644 sound/soc/blackfin/bf6xx-sport.h diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 9f6bc55..0374a39 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,7 +1,8 @@ config SND_BF5XX_I2S - tristate "SoC I2S Audio for the ADI BF5xx chip" + tristate "SoC I2S Audio for the ADI Blackfin chip" depends on BLACKFIN - select SND_BF5XX_SOC_SPORT + select SND_BF5XX_SOC_SPORT if !BF60x + select SND_BF6XX_SOC_SPORT if BF60x help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -162,6 +163,9 @@ config SND_BF5XX_SOC_AD1980 config SND_BF5XX_SOC_SPORT tristate +config SND_BF6XX_SOC_SPORT + tristate + config SND_BF5XX_SOC_I2S tristate @@ -173,7 +177,7 @@ config SND_BF5XX_SOC_AC97 config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) + depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT) range 0 3 if BF54x range 0 1 if !BF54x default 0 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 1bf86cc..13b0922 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -3,6 +3,7 @@ snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o +snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o @@ -11,6 +12,7 @@ obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o +obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c new file mode 100644 index 0000000..f19a72b --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -0,0 +1,422 @@ +/* + * bf6xx_sport.c Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <asm/blackfin.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "bf6xx-sport.h" + +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN; + sport->tx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_tx_params); + +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN; + sport->rx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_rx_params); + +static int compute_wdsize(size_t wdsize) +{ + switch (wdsize) { + case 1: + return WDSIZE_8 | PSIZE_8; + case 2: + return WDSIZE_16 | PSIZE_16; + default: + return WDSIZE_32 | PSIZE_32; + } +} + +void sport_tx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc); + set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->tx_dma_chan); + sport->tx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_tx_start); + +void sport_rx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc); + set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->rx_dma_chan); + sport->rx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_rx_start); + +void sport_tx_stop(struct sport_device *sport) +{ + sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->tx_dma_chan); +} +EXPORT_SYMBOL(sport_tx_stop); + +void sport_rx_stop(struct sport_device *sport) +{ + sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->rx_dma_chan); +} +EXPORT_SYMBOL(sport_rx_stop); + +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data) +{ + sport->tx_callback = tx_callback; + sport->tx_data = tx_data; +} +EXPORT_SYMBOL(sport_set_tx_callback); + +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data) +{ + sport->rx_callback = rx_callback; + sport->rx_data = rx_data; +} +EXPORT_SYMBOL(sport_set_rx_callback); + +static void setup_desc(struct dmasg *desc, void *buf, int fragcount, + size_t fragsize, unsigned int cfg, + unsigned int count, size_t wdsize) +{ + + int i; + + for (i = 0; i < fragcount; ++i) { + desc[i].next_desc_addr = &(desc[i + 1]); + desc[i].start_addr = (unsigned long)buf + i*fragsize; + desc[i].cfg = cfg; + desc[i].x_count = count; + desc[i].x_modify = wdsize; + desc[i].y_count = 0; + desc[i].y_modify = 0; + } + + /* make circular */ + desc[fragcount-1].next_desc_addr = desc; +} + +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + + sport->tx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->tx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->tx_desc) + return -ENOMEM; + + sport->tx_buf = buf; + sport->tx_fragsize = fragsize; + sport->tx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6; + + setup_desc(sport->tx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_tx_dma); + +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); + + sport->rx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->rx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->rx_desc) + return -ENOMEM; + + sport->rx_buf = buf; + sport->rx_fragsize = fragsize; + sport->rx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) + | WNR | NDSIZE_6; + + setup_desc(sport->rx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_rx_dma); + +unsigned long sport_curr_offset_tx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan); + + return (unsigned char *)curr - sport->tx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_tx); + +unsigned long sport_curr_offset_rx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan); + + return (unsigned char *)curr - sport->rx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_rx); + +static irqreturn_t sport_tx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + static unsigned long status; + + status = get_dma_curr_irqstat(sport->tx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->tx_dma_chan); + SSYNC(); + } + if (sport->tx_callback) + sport->tx_callback(sport->tx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_rx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + unsigned long status; + + status = get_dma_curr_irqstat(sport->rx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->rx_dma_chan); + SSYNC(); + } + if (sport->rx_callback) + sport->rx_callback(sport->rx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_err_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + struct device *dev = &sport->pdev->dev; + + if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) + dev_dbg(dev, "sport error: TUVF\n"); + if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) + dev_dbg(dev, "sport error: ROVF\n"); + + return IRQ_HANDLED; +} + +static int sport_get_resource(struct sport_device *sport) +{ + struct platform_device *pdev = sport->pdev; + struct device *dev = &pdev->dev; + struct bfin_snd_platform_data *pdata = dev->platform_data; + struct resource *res; + + if (!pdata) { + dev_err(dev, "No platform data\n"); + return -ENODEV; + } + sport->pin_req = pdata->pin_req; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "No tx MEM resource\n"); + return -ENODEV; + } + sport->tx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res) { + dev_err(dev, "No rx MEM resource\n"); + return -ENODEV; + } + sport->rx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(dev, "No tx DMA resource\n"); + return -ENODEV; + } + sport->tx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(dev, "No rx DMA resource\n"); + return -ENODEV; + } + sport->rx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(dev, "No tx error irq resource\n"); + return -ENODEV; + } + sport->tx_err_irq = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 1); + if (!res) { + dev_err(dev, "No rx error irq resource\n"); + return -ENODEV; + } + sport->rx_err_irq = res->start; + + return 0; +} + +static int sport_request_resource(struct sport_device *sport) +{ + struct device *dev = &sport->pdev->dev; + int ret; + + ret = peripheral_request_list(sport->pin_req, "soc-audio"); + if (ret) { + dev_err(dev, "Unable to request sport pin\n"); + return ret; + } + + ret = request_dma(sport->tx_dma_chan, "SPORT TX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport tx\n"); + goto err_tx_dma; + } + set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport); + + ret = request_dma(sport->rx_dma_chan, "SPORT RX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport rx\n"); + goto err_rx_dma; + } + set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport); + + ret = request_irq(sport->tx_err_irq, sport_err_irq, + 0, "SPORT TX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate tx error IRQ for sport\n"); + goto err_tx_irq; + } + + ret = request_irq(sport->rx_err_irq, sport_err_irq, + 0, "SPORT RX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate rx error IRQ for sport\n"); + goto err_rx_irq; + } + + return 0; +err_rx_irq: + free_irq(sport->tx_err_irq, sport); +err_tx_irq: + free_dma(sport->rx_dma_chan); +err_rx_dma: + free_dma(sport->tx_dma_chan); +err_tx_dma: + peripheral_free_list(sport->pin_req); + return ret; +} + +static void sport_free_resource(struct sport_device *sport) +{ + free_irq(sport->rx_err_irq, sport); + free_irq(sport->tx_err_irq, sport); + free_dma(sport->rx_dma_chan); + free_dma(sport->tx_dma_chan); + peripheral_free_list(sport->pin_req); +} + +struct sport_device *sport_create(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct sport_device *sport; + int ret; + + sport = kzalloc(sizeof(*sport), GFP_KERNEL); + if (!sport) { + dev_err(dev, "Unable to allocate memory for sport device\n"); + return NULL; + } + sport->pdev = pdev; + + ret = sport_get_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + ret = sport_request_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + dev_dbg(dev, "SPORT create success\n"); + return sport; +} +EXPORT_SYMBOL(sport_create); + +void sport_delete(struct sport_device *sport) +{ + sport_free_resource(sport); +} +EXPORT_SYMBOL(sport_delete); + +MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h new file mode 100644 index 0000000..307d193 --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.h @@ -0,0 +1,82 @@ +/* + * bf6xx_sport - Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef _BF6XX_SPORT_H_ +#define _BF6XX_SPORT_H_ + +#include <linux/platform_device.h> +#include <asm/bfin_sport3.h> + +struct sport_device { + struct platform_device *pdev; + const unsigned short *pin_req; + struct sport_register *tx_regs; + struct sport_register *rx_regs; + int tx_dma_chan; + int rx_dma_chan; + int tx_err_irq; + int rx_err_irq; + + void (*tx_callback)(void *data); + void *tx_data; + void (*rx_callback)(void *data); + void *rx_data; + + struct dmasg *tx_desc; + struct dmasg *rx_desc; + unsigned int tx_desc_size; + unsigned int rx_desc_size; + unsigned char *tx_buf; + unsigned char *rx_buf; + unsigned int tx_fragsize; + unsigned int rx_fragsize; + unsigned int tx_frags; + unsigned int rx_frags; + unsigned int wdsize; +}; + +struct sport_params { + u32 spctl; + u32 div; +}; + +struct sport_device *sport_create(struct platform_device *pdev); +void sport_delete(struct sport_device *sport); +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params); +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params); +void sport_tx_start(struct sport_device *sport); +void sport_rx_start(struct sport_device *sport); +void sport_tx_stop(struct sport_device *sport); +void sport_rx_stop(struct sport_device *sport); +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data); +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data); +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +unsigned long sport_curr_offset_tx(struct sport_device *sport); +unsigned long sport_curr_offset_rx(struct sport_device *sport); + + + +#endif -- 1.7.0.4

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[alsa-devel] [PATCH] ALSA: Add missing include of pcm.h to pcm_params.h
by Mark Brown 20 Jun '12

20 Jun '12

There's a dependency but no #include. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/pcm_params.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index f494f1e..37ae12e 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -22,6 +22,8 @@ * */ +#include <sound/pcm.h> + int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir); -- 1.7.10

2 3

[alsa-devel] [PATCH v2 1/3] dmaengine: Add wrapper for device_tx_status callback
by Lars-Peter Clausen 20 Jun '12

20 Jun '12

This patch adds a small inline wrapper for the devivce_tx_status callback of a dma device. This makes the source code of users of this function a bit more compact and a bit more legible. E.g.: -status = chan->device->device_tx_status(chan, cookie, &state) +status = dmaengine_tx_status(chan, cookie, &state) Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes since v1 --- include/linux/dmaengine.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 56377df..cc0756a 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -670,6 +670,12 @@ static inline int dmaengine_resume(struct dma_chan *chan) return dmaengine_device_control(chan, DMA_RESUME, 0); } +static inline enum dma_status dmaengine_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *state) +{ + return chan->device->device_tx_status(chan, cookie, state); +} + static inline dma_cookie_t dmaengine_submit(struct dma_async_tx_descriptor *desc) { return desc->tx_submit(desc); -- 1.7.10

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[alsa-devel] [PATCH] ASoC: wm2200: Add missing BCLK rate
by Mark Brown 20 Jun '12

20 Jun '12

Without this very high BCLKs will be configured incorrectly. Reported-by: Axel Lin <axel.lin(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index acbdc5f..32682c1 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1491,6 +1491,7 @@ static int wm2200_bclk_rates_dat[WM2200_NUM_BCLK_RATES] = { static int wm2200_bclk_rates_cd[WM2200_NUM_BCLK_RATES] = { 5644800, + 3763200, 2882400, 1881600, 1411200, -- 1.7.10

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