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

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

28 Oct '13

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

4 31

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

28 Oct '13

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

6 20

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

28 Oct '13

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

2 10

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

28 Oct '13

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

4 6

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

28 Oct '13

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

2 1

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

28 Oct '13

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

1 6

[alsa-devel] [PATCH] ALSA: pcm_dmaengine: Remove hardcoded PCM formats
by Takashi Iwai 27 Oct '13

27 Oct '13

Use the standard PCM helper function to figure out the sample bytes instead of hardcodec PCM format checks in snd_hwparams_to_dma_slave_config(). The patch also extends the format check for 8 bytes formats although no one should match so far. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I stumbled upon this, but am not 100% sure whether the current hardcoded formats are the designed limitation... sound/core/pcm_dmaengine.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index aa924d9..94d0873 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -63,23 +63,19 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, struct dma_slave_config *slave_config) { enum dma_slave_buswidth buswidth; + int bits; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + bits = snd_pcm_format_physical_width(params_format(params)); + if (bits < 8 || bits > 64) + return -EINVAL; + else if (bits == 8) buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; - break; - case SNDRV_PCM_FORMAT_S16_LE: + else if (bits == 16) buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; - break; - case SNDRV_PCM_FORMAT_S18_3LE: - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + else if (bits <= 32) buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - break; - default: - return -EINVAL; - } + else + buswidth = DMA_SLAVE_BUSWIDTH_8_BYTES; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config->direction = DMA_MEM_TO_DEV; -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: fsl_spdif: fix return value check in fsl_spdif_probe()
by Wei Yongjun 26 Oct '13

26 Oct '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> In case of error, the function platform_get_resource() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e..e5bfafe 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1107,9 +1107,9 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (IS_ERR(res)) { + if (!res) { dev_err(&pdev->dev, "could not determine device resources\n"); - return PTR_ERR(res); + return -ENXIO; } regs = devm_ioremap_resource(&pdev->dev, res);

2 1

[alsa-devel] [PATCH v2] ASoC: wm5110: Add missing routes for AEC Loopback
by Charles Keepax 26 Oct '13

26 Oct '13

Reported-by: Nariman Poushin <nariman.poushin(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd6438..8c91be5 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -983,24 +983,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, + { "AEC Loopback", "HPOUT2L", "OUT2L" }, + { "AEC Loopback", "HPOUT2R", "OUT2R" }, { "HPOUT2L", NULL, "OUT2L" }, { "HPOUT2R", NULL, "OUT2R" }, + { "AEC Loopback", "HPOUT3L", "OUT3L" }, + { "AEC Loopback", "HPOUT3R", "OUT3R" }, { "HPOUT3L", NULL, "OUT3L" }, { "HPOUT3R", NULL, "OUT3L" }, + { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, + { "AEC Loopback", "SPKOUTR", "OUT4R" }, { "SPKOUTRN", NULL, "OUT4R" }, { "SPKOUTRP", NULL, "OUT4R" }, + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, { "SPKDAT1L", NULL, "OUT5L" }, { "SPKDAT1R", NULL, "OUT5R" }, + { "AEC Loopback", "SPKDAT2L", "OUT6L" }, + { "AEC Loopback", "SPKDAT2R", "OUT6R" }, { "SPKDAT2L", NULL, "OUT6L" }, { "SPKDAT2R", NULL, "OUT6R" }, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm5110: Add missing routes for AEC Loopback
by Charles Keepax 26 Oct '13

26 Oct '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd6438..8c91be5 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -983,24 +983,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, + { "AEC Loopback", "HPOUT2L", "OUT2L" }, + { "AEC Loopback", "HPOUT2R", "OUT2R" }, { "HPOUT2L", NULL, "OUT2L" }, { "HPOUT2R", NULL, "OUT2R" }, + { "AEC Loopback", "HPOUT3L", "OUT3L" }, + { "AEC Loopback", "HPOUT3R", "OUT3R" }, { "HPOUT3L", NULL, "OUT3L" }, { "HPOUT3R", NULL, "OUT3L" }, + { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, + { "AEC Loopback", "SPKOUTR", "OUT4R" }, { "SPKOUTRN", NULL, "OUT4R" }, { "SPKOUTRP", NULL, "OUT4R" }, + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, { "SPKDAT1L", NULL, "OUT5L" }, { "SPKDAT1R", NULL, "OUT5R" }, + { "AEC Loopback", "SPKDAT2L", "OUT6L" }, + { "AEC Loopback", "SPKDAT2R", "OUT6R" }, { "SPKDAT2L", NULL, "OUT6L" }, { "SPKDAT2R", NULL, "OUT6R" }, -- 1.7.2.5

1 0

[alsa-devel] [PATCH] ALSA: hda - Sync EAPD with vmaster on AD1984A Thinkpads
by Takashi Iwai 26 Oct '13

26 Oct '13

As EAPD on NID 0x12 (speaker pin) is used as the master amp on Thinkpads with AD1984A codec, we can hook this to vmaster for saving a bit more power at master mute state. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2aa2f57..87d2e03 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -973,8 +973,11 @@ static void ad1884_fixup_thinkpad(struct hda_codec *codec, { struct ad198x_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.keep_eapd_on = 1; + spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + spec->eapd_nid = 0x12; + } } /* set magic COEFs for dmic */ -- 1.8.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix silent headphone on Thinkpads with AD1984A codec
by Takashi Iwai 26 Oct '13

26 Oct '13

AD1984A codec has a couple of pins with EAPD controls, and the generic codec driver tries to turn each of them on/off depending on the pin active state. However, Thinkpads seem to use EAPD of the speaker pin as a master EAPD for controlling the mute of all outputs, including the headphone. This results in the dead headphone output via the headphone plugging because it mutes the speaker and turns off EAPD. The fix is to simply add spec->gen.keep_on_eapd flag. [This is a regression fix on 3.12 where we moved the AD codec parser to the generic parser. 3.11 and earlier didn't show this problem because still static quirks have been used.] Reported-and-tested-by: Vito Caputo <vcaputo(a)gnugeneration.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0cbdd87..2aa2f57 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -968,6 +968,15 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, } } +static void ad1884_fixup_thinkpad(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.keep_eapd_on = 1; +} + /* set magic COEFs for dmic */ static const struct hda_verb ad1884_dmic_init_verbs[] = { {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, @@ -979,6 +988,7 @@ enum { AD1884_FIXUP_AMP_OVERRIDE, AD1884_FIXUP_HP_EAPD, AD1884_FIXUP_DMIC_COEF, + AD1884_FIXUP_THINKPAD, AD1884_FIXUP_HP_TOUCHSMART, }; @@ -997,6 +1007,12 @@ static const struct hda_fixup ad1884_fixups[] = { .type = HDA_FIXUP_VERBS, .v.verbs = ad1884_dmic_init_verbs, }, + [AD1884_FIXUP_THINKPAD] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1884_fixup_thinkpad, + .chained = true, + .chain_id = AD1884_FIXUP_DMIC_COEF, + }, [AD1884_FIXUP_HP_TOUCHSMART] = { .type = HDA_FIXUP_VERBS, .v.verbs = ad1884_dmic_init_verbs, @@ -1008,7 +1024,7 @@ static const struct hda_fixup ad1884_fixups[] = { static const struct snd_pci_quirk ad1884_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART), SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD), - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_THINKPAD), {} }; -- 1.8.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Add missing initial vmaster hook at build_controls callback
by Takashi Iwai 25 Oct '13

25 Oct '13

The generic parser has a support of vmaster hook, but this is initialized only in the init callback with the check of the presence of the corresponding kctl. However, since kctl is NULL at the very first init callback that is called before build_controls callback, the vmaster hook sync is skipped there. Eventually this leads to the uninitialized state depending on the hook implementation. This patch adds a simple workaround, just calling the sync function explicitly at build_controls callback. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 26ad4f0..b7c89df 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4475,9 +4475,11 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) true, &spec->vmaster_mute.sw_kctl); if (err < 0) return err; - if (spec->vmaster_mute.hook) + if (spec->vmaster_mute.hook) { snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, spec->vmaster_mute_enum); + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + } } free_kctls(spec); /* no longer needed */ -- 1.8.4

1 0

[alsa-devel] [PATCH RESEND v2 0/4] Cleanups/Bugfixes and DT Prep for CS42L52
by Brian Austin 25 Oct '13

25 Oct '13

This series fixes a bug and cleans up the platform data so the driver can nicely move to DT. Convert platform data initialization to regmap_update_bits in i2c_probe. Add GPIO for CODEC Reset. Add print message for Rev ID. Version 2 splits the regmap_patch bug with the MAX_REGISTER increase. Series based off of a clean for-next branch Brian Austin (4): ASoC: cs42l52: Add platform data for reset gpio ASoC: cs42l52: increase MAX_REGISTER for regmap_register_patch ASoC: cs42l52: convert pdata config to regmap_update_bits ASoC: cs42l52: Add chip rev id message include/sound/cs42l52.h | 2 + sound/soc/codecs/cs42l52.c | 93 +++++++++++++++++++++++++++------------------- sound/soc/codecs/cs42l52.h | 2 +- 3 files changed, 57 insertions(+), 40 deletions(-) -- 1.8.4.rc3

2 12

[alsa-devel] [PATCH] ASoC: dmaengine: Use SNDRV_PCM_STREAM_LAST for array size
by Takashi Iwai 25 Oct '13

25 Oct '13

... to make the meaning more obvious. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 0c469cb..ee07903 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -25,7 +25,7 @@ #include <sound/dmaengine_pcm.h> struct dmaengine_pcm { - struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; + struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; unsigned int flags; -- 1.8.4

3 2

[alsa-devel] Enable display power well management in HD-A driver on Baytrail platform (Valleyview SOC)
by Lin, Mengdong 25 Oct '13

25 Oct '13

Hi Takashi, We want to enable display power well management on Batrail platform (Valleyview SOC), similar to what we did on Haswell. Would you please check if our ideas are ok and give some advice? For Valleyview/VLV the display controller has a power well island. When display is off, the display island can be power gated. Unlike Haswell, the VLV HD-A controller is not in the display power well. The controller is in south bridge and shared by multiple codecs, same as the predecessors of Haswell. To assure audio is not disturbed by power well off, the Gfx i915 driver will provide the same API as Haswell: i915_request_power_well & i915_release_power_well So audio driver can indicate whether audio codec needs the power well. In HD-A driver, the display codec can suspend to D3 when there this no active streaming or ioctls which will actually trigger transactions to the display codec on HD-A bus. So our basic idea is to request power before codec resumes to D0 and release power after codec suspends to D3. And there can be two options for this: 1. Overwrite the 'set_power_state' ops for the VLV display codec, and request/release Gfx power well there via the i915 API in this ops. And future VLV specific fixes may also go to this ops. 2. Add two new codec ops like 'pre_resume/post_suspend', and request/release Gfx power well here. Which way do you think is better? Personally I don't want to add new ops and prefer 1st option, and VLV-like processors can use the same ops. And there is an open question: Can we assume that middleware (e.g. pulseaudio or audio flinger) will close the PCM device if there is no need to play audio for power saving? Is this a usual way for middleware to work with ALSA? Now in HD-A driver, once the PCM device is open, the codec will resume to D0. And so the codec will request display power well. If middleware opens PCM device but doesn't play, there will be power waste. We want to avoid this. Thanks Mengdong

2 3

[alsa-devel] ASoC updates for v3.13
by Mark Brown 25 Oct '13

25 Oct '13

The following changes since commit 31d141e3a666269a3b6fcccddb0351caf7454240: Linux 3.12-rc6 (2013-10-19 12:28:15 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13 for you to fetch changes up to 9645083ca5ef365b7b750cf219bb20b61bb925f8: Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next (2013-10-24 11:24:18 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.13 - Further work on the dmaengine helpers, including support for configuring the parameters for DMA by reading the capabilities of the DMA controller which removes some guesswork and magic numbers fromm drivers. - A refresh of the documentation. - Conversions of many drivers to direct regmap API usage in order to allow the ASoC level register I/O code to be removed, this will hopefully be completed by v3.14. - Support for using async register I/O in DAPM, reducing the time taken to implement power transitions on systems that support it. ---------------------------------------------------------------- Andrew Morton (1): ASoC: atmel-pcm: fix warning Brian Austin (3): ASoC: cs42l73: Add platform data support for cs42l73 codec ASoC: cs42l73: Namespace defines for cs42l73 codec ASoC: cs42l73: Add Device Tree support for CS42L73 Charles Keepax (3): ASoC: samsung: Allow mono in i2s driver ASoC: bells: Add missing route to power up DSP clock ASoC: arizona: Improve handling of setting REFCLK to 0 Chen Gang (1): ASoC: fsl_ssi: let check zero instead of check NO_IRQ Daniel Mack (5): ASoC: ak4104: provide a module device table ASoC: tas5086: move two variables into private struct ASoC: tas5086: move initialization code to own functions ASoC: tas5086: add suspend callback ASoC: davinci-mcasp: add support for suspend and resume Dmitry Eremin-Solenikov (3): ALSA: pxa: slightly refactor reset handling ALSA: ASoC: pxa: fix pxa2xx-ac97 DAI initialization order ALSA: ASoC: pxa: add asoc pm callbacks to pxa audio drivers Fabio Estevam (3): ASoC: mxs: mxs-sgtl5000: Simplify probe function ASoC: fsl: fsl_ssi: Fix simultaneous capture and playback ASoC: fsl: imx-sgtl5000: Add .remove back Felipe Pena (1): ASoC: fsl: Fix memory leak in imx-audmux.c Hebbar, Gururaja (2): ASoC: davinci: Add support for AM33xx SoC Audio ASoC: davinci-evm: Add device tree binding Jarkko Nikula (1): ASoC: pcm: Remove extra spaces from dev_ prints Jean-Francois Moine (4): ASoC: kirkwood: fix loss of external clock at probe time ASoc: kirkwood: Extend the min and max number of bytes per period ASoC: kirkwood: prefer external clock over internal clock ASoC: kirkwood: add S/PDIF support Jyri Sarha (8): ASoC: davinci-evm: Move sysclk logic away from evm_hw_params ASoC: tlv320aic3x: Add regulators to DT bindings document ASoC: tlv320aic3x: Add codec pins to DT bindings document ASoC: davinci-mcasp: Add location for data port registers to DT ASoC: davinci-mcasp: Extract DMA channels directly from DT ASoC: davinci-mcasp: Change compatible property model to more accurate ASoC: davinci-mcasp: Improve DT bindings document ASoC: davinci-mcasp: Remove last reference to num-serializer in DT doc Kuninori Morimoto (14): regmap: add regmap_field_update_bits() ASoC: snd_soc_codec includes snd_soc_component ASoC: add .of_xlate_dai_name on snd_soc_component_driver regmap: Add regmap_fields APIs ASoC: rsnd: gen: rsnd_gen_ops cares .probe and .remove ASoC: rcar: remove unnecessary mach/clock.h ASoC: rsnd: remove rsnd_priv_read/write/bset() ASoC: rsnd: use regmap instead of original register mapping method ASoC: rcar: fixup rsnd_platform_call() return value ASoC: rcar: add ID check on rsnd_dai_get() ASoC: add snd_soc_of_get_dai_name() default of_xlate ASoC: simple-card: un-implemented set_fmt is not error ASoC: rcar: add rsnd_scu_hpbif_is_enable() ASoC: rcar: remove RSND_SSI_CLK_FROM_ADG Lars-Peter Clausen (15): ASoC: Remove support for reg_access_defaults ASoC: Remove snd_soc_bulk_write_raw() ASoC: Remove reg_def_copy ASoC: Remove 'reg_size' field from snd_soc_codec struct ASoC: Remove infrastructure for supporting multiple cache types ASoC: adau1373: Convert to direct regmap usage ASoC: adau1373: Remove ADAU1373_PLL_CTRL7 register definition ASoC: adau1373: Issue soft reset on probe ASoC: adav80x: Convert to direct regmap usage ASoC: dapm: Ignore VMID widgets for target bias ASoC: dapm: Add support for virtual mixer controls ASoC: twl6040: Use virtual DAPM mixer controls ASoC: twl4030: Use virtual DAPM mixer controls ASoC: dmaengine-pcm: Add support for querying DMA capabilities ASoC: dmaengine-pcm: Provide default config Liam Girdwood (10): ASoC: trace: Make sure trace header doesnt depend on any headers ASoC: core utils: Dont set DMA params for BE substreams ASoC: rt5640: Provide more useful hw_params error reasons. ASoC: rt5640: Add ACPI probing support. ASoC: core: Add API for configuration of DAI BCLK ratio ASoC: Docs: Update codec documentation ASoC: Docs: Platform update ASoC: Docs: update DAPM ASoC: Docs: Machine update ASoC: Docs: Add documentation for Dynamic PCM Mark Brown (122): ASoC: cq93vc: Don't use control data for core driver data ASoC: wm8400: Use supplies to manage input power ASoC: wm8400: Use regmap for I/O ASoC: core: Implement devm_snd_soc_register_component() ASoC: core: Add devm_snd_soc_register_card() Merge remote-tracking branch 'asoc/topic/devm' into asoc-samsung ASoC: smdk_wm8994: Use devm_snd_soc_unregister_card() ASoC: samsung-i2s: Use devm_snd_soc_register_component() Merge remote-tracking branch 'asoc/topic/component' into asoc-core ASoC: 88pm60x: Don't use control data for i2c ASoC: 88pm860x: Use regmap for I/O ASoC: ab8500: Downgrade noisy log message ASoC: ab8500: Use ASoC I/O functions ASoC: ab8500: Convert register I/O to regmap ASoC: max98088: Fix indentation ASoC: max98088: Use table based control init ASoC: max98088: Convert to direct regmap API usage ASoC: ab8500: Explicitly set I/O up ASoC: max98095: Remove custom hw_write() implementation ASoC: max90895: Convert to table based control init ASoC: max98095: Convert to direct regmap API usage ASoC: tlv320aic23: Remove #defines for I2C ASoC: tlv320aic23: Convert to table based control init ASoC: tlv320aic3x: Remove nonsense comment for register cache ASoC: tlv320aic3x: Move resource acquisition to I2C probe ASoC: tlv320aic3x: Move to table based control init ASoC: tlv320aic3x: Move to table based DAPM init ASoC: tlv320aic3x: Don't reference cache datastructure directly ASoC: tlv320aic3x: Convert to direct regmap API usage ASoC: max98088: Set max_register ASoC: max9850: Convert to direct regmap API usage ASoC: cq93vc: Use table based control registration ASoC: cq93vc: Use core I/O functions ASoC: cq93vc: Use regmap for I/O Merge tag 'regmap-fields' of git://git.kernel.org/.../broonie/regmap into asoc-rcar ASoC: tlv320aic23: Convert to direct regmap API usage ASoC: tlv320aic26: Convert to table based control init ASoC: tlv320aic26: Use snd_soc_update_bits() ASoC: tlv320aic26: Convert to direct regmap API usage mfd: mc13xxx: Don't require lock for simple register I/O mfd: mc13xxx: Move SPI erratum workaround into SPI I/O function ASoC: mc13783: Use regmap directly from ASoC ASoC: sn95031: Convert to regmap ASoC: tlv320aic32x4: Move GPIO acquisition to I2C probe ASoC: tlv320aic32x4: Convert to direct regmap API usage Merge remote-tracking branch 'asoc/fix/fsl' into asoc-devm ASoC: si476x: Remove custom register I/O implementation Merge remote-tracking branch 'asoc/fix/fsl' into asoc-fsl Merge remote-tracking branch 'asoc/topic/dapm' into asoc-twl6040 regmap: Cache async work structures regmap: Don't generate gather writes for single register raw writes regmap: Simplify the initiation of async I/O regmap: Provide asynchronous write and update bits operations Merge tag 'regmap-async-reg' of git://git.kernel.org/.../broonie/regmap into asoc-dapm ASoC: dapm: Use async I/O for DAPM sequences ASoC: dapm: Run clock and regulator events separately to other supplies ASoC: wm8962: Move interrupt initalisation to probe() ASoC: wm8962: Move register initialisation to I2C probe() ASoC: cirrus: Enable compile test builds ASoC: rt5640: Power down LDO while suspended ASoC: rt5640: Don't go to standby on resume ASoC: dai: Provide interface for setting DMA data at probe time ASoC: samsung: Initialise DMA data at device probe time ASoC: si476x: Fix locking of core ASoC: ep93xx: Open code dma channel request ASoC: tegra: Remove redundant initialisation of compat_filter_fn ASoC: dmaengine: Support custom channel names ASoC: dmaengine: Use filter_data rather than dma_data for compat requests Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/omap' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm1681' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm1792a' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-linus Merge remote-tracking branch 'asoc/topic/88pm860x' into asoc-next Merge remote-tracking branch 'asoc/topic/ab8500' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1373' into asoc-next Merge remote-tracking branch 'asoc/topic/adav80x' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4104' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4642' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/bclk' into asoc-next Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cq93vc' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/devm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/doc' into asoc-next Merge remote-tracking branch 'asoc/topic/ep93xx' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98095' into asoc-next Merge remote-tracking branch 'asoc/topic/max9850' into asoc-next Merge remote-tracking branch 'asoc/topic/mc13783' into asoc-next Merge remote-tracking branch 'asoc/topic/mxs' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1792a' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5640' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/simple' into asoc-next Merge remote-tracking branch 'asoc/topic/sn95031' into asoc-next Merge remote-tracking branch 'asoc/topic/spear' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic23' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic26' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic32x4' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/tpa6130a2' into asoc-next Merge remote-tracking branch 'asoc/topic/twl4030' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8400' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next Markus Pargmann (3): ASoC: snd_soc_dai_ops trigger function description ASoC: mxs-saif: Store saif state ASoC: mxs-saif: Handle errors in trigger function Nicolin Chen (1): ASoC: fsl: Add missing pm to current machine drivers Peter Ujfalusi (1): ASoC: davinci-mcasp: Remove redundant num-serializer DT parameter Russell King (1): ASoC: kirkwood-dma: remove IEC958_SUBFRAME formats Sachin Kamat (25): ASoC: fsl_ssi: Staticize local symbols ASoC: rt5640: Staticize hp_amp_power_on ASoC: twl6040: Remove redundant semicolon ASoC: ak4642: Remove redundant break ASoC: fsl_spdif: Remove redundant semicolon ASoC: fsl_ssi: Remove redundant dev_set_drvdata ASoC: SPEAr spdif_in: Use devm_snd_soc_register_component ASoC: SPEAr spdif_out: Use devm_snd_soc_register_component ASoC: SPEAr spdif_out: Remove redundant variable ASoC: imx-sgtl5000: Use devm_snd_soc_register_card ASoC: imx-spdif: Use devm_snd_soc_register_card ASoC: imx-wm8962: Use devm_snd_soc_register_card ASoC: mfld: Use devm_snd_soc_register_card ASoC: omap-twl4030: Use devm_snd_soc_register_card ASoC: fsl_spdif: Use devm_snd_soc_register_component ASoC: mxs-saif: Use devm_snd_soc_register_component ASoC: omap-mcpdm: Use devm_snd_soc_register_component ASoC: mmp-sspa: Use devm_snd_soc_register_component ASoC: smdk_wm8994: Add .pm to struct smdk_audio_driver ASoC: cs4271: Include linux/of.h header ASoC: pcm1681: Include linux/of.h header ASoC: pcm1792a: Include linux/of.h header ASoC: tas5086: Include linux/of.h header ASoC: tlv320aic3x: Include linux/of.h header ASoC: atmel: Include linux/of.h header Sebastian Reichel (1): ASoC: tpa6130a2: Add device tree support Steffen Trumtrar (2): ASoC: mc13783: add mixer controls ASoC: mc13783: add more DAPM routes Stephen Warren (1): ASoC: tegra: support new register layouts in Tegra124 Thierry Reding (1): ASoC: rt5640: Omit ACPI match table only if !ACPI Vinod Koul (1): ASoC: dont call dapm_sync while reporting jack always Vladimir Murzin (1): ASoC: kirkwood: fix compilation warning in kirkwood_dma_open Wolfram Sang (2): ASoC: atmel: don't use devm_pinctrl_get_select_default() in probe ASoC: fsl: remove leftover release_mem_region .../devicetree/bindings/sound/cs42l73.txt | 22 + .../bindings/sound/davinci-evm-audio.txt | 42 ++ .../bindings/sound/davinci-mcasp-audio.txt | 41 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 26 + .../devicetree/bindings/sound/tpa6130a2.txt | 27 + Documentation/sound/alsa/soc/DPCM.txt | 380 +++++++++++++ Documentation/sound/alsa/soc/codec.txt | 46 +- Documentation/sound/alsa/soc/dapm.txt | 71 ++- Documentation/sound/alsa/soc/machine.txt | 6 +- Documentation/sound/alsa/soc/platform.txt | 19 +- drivers/base/regmap/internal.h | 8 +- drivers/base/regmap/regcache.c | 3 +- drivers/base/regmap/regmap.c | 294 ++++++++-- drivers/mfd/mc13xxx-core.c | 5 - drivers/mfd/mc13xxx-spi.c | 5 + include/linux/mfd/mc13xxx.h | 7 + include/linux/platform_data/davinci_asp.h | 2 + include/linux/regmap.h | 44 ++ include/sound/cs42l73.h | 22 + include/sound/dmaengine_pcm.h | 8 + include/sound/rcar_snd.h | 2 +- include/sound/soc-dai.h | 17 + include/sound/soc-dapm.h | 4 + include/sound/soc.h | 100 ++-- include/trace/events/asoc.h | 1 + sound/arm/pxa2xx-ac97-lib.c | 27 +- sound/soc/Makefile | 2 +- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 8 - sound/soc/atmel/sam9g20_wm8731.c | 1 + sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-pcm.c | 13 + sound/soc/codecs/88pm860x-codec.c | 75 +-- sound/soc/codecs/88pm860x-codec.h | 117 ++-- sound/soc/codecs/ab8500-codec.c | 78 ++- sound/soc/codecs/adau1373.c | 298 ++++++++-- sound/soc/codecs/adav80x.c | 147 +++-- sound/soc/codecs/ak4104.c | 11 +- sound/soc/codecs/ak4642.c | 2 - sound/soc/codecs/arizona.c | 23 +- sound/soc/codecs/cq93vc.c | 46 +- sound/soc/codecs/cs4271.c | 1 + sound/soc/codecs/cs42l73.c | 114 ++-- sound/soc/codecs/cs42l73.h | 105 ++-- sound/soc/codecs/max98088.c | 624 +++++++++------------ sound/soc/codecs/max98095.c | 466 ++++++--------- sound/soc/codecs/max9850.c | 39 +- sound/soc/codecs/mc13783.c | 135 +++-- sound/soc/codecs/pcm1681.c | 3 +- sound/soc/codecs/pcm1792a.c | 3 +- sound/soc/codecs/rt5640.c | 24 +- sound/soc/codecs/si476x.c | 64 +-- sound/soc/codecs/sn95031.c | 35 +- sound/soc/codecs/tas5086.c | 171 ++++-- sound/soc/codecs/tlv320aic23.c | 84 ++- sound/soc/codecs/tlv320aic26.c | 139 +---- sound/soc/codecs/tlv320aic26.h | 5 +- sound/soc/codecs/tlv320aic32x4.c | 101 +--- sound/soc/codecs/tlv320aic3x.c | 238 ++++---- sound/soc/codecs/tpa6130a2.c | 32 +- sound/soc/codecs/twl4030.c | 80 ++- sound/soc/codecs/twl6040.c | 26 +- sound/soc/codecs/wm8400.c | 95 +--- sound/soc/codecs/wm8962.c | 218 ++++--- sound/soc/davinci/Kconfig | 18 +- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 188 ++++++- sound/soc/davinci/davinci-mcasp.c | 169 ++++-- sound/soc/davinci/davinci-mcasp.h | 12 + sound/soc/fsl/fsl_spdif.c | 18 +- sound/soc/fsl/fsl_ssi.c | 22 +- sound/soc/fsl/imx-audmux.c | 9 +- sound/soc/fsl/imx-mc13783.c | 3 +- sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-sgtl5000.c | 4 +- sound/soc/fsl/imx-spdif.c | 4 +- sound/soc/fsl/imx-ssi.c | 26 +- sound/soc/fsl/imx-ssi.h | 2 + sound/soc/fsl/imx-wm8962.c | 4 +- sound/soc/generic/simple-card.c | 5 + sound/soc/kirkwood/kirkwood-dma.c | 6 +- sound/soc/kirkwood/kirkwood-i2s.c | 106 +++- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- sound/soc/kirkwood/kirkwood.h | 4 +- sound/soc/mid-x86/mfld_machine.c | 10 +- sound/soc/mxs/mxs-saif.c | 42 +- sound/soc/mxs/mxs-saif.h | 5 + sound/soc/mxs/mxs-sgtl5000.c | 20 +- sound/soc/omap/Kconfig | 4 +- sound/soc/omap/omap-mcpdm.c | 12 +- sound/soc/omap/omap-twl4030.c | 5 +- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/e740_wm9705.c | 1 + sound/soc/pxa/e750_wm9705.c | 1 + sound/soc/pxa/e800_wm9712.c | 1 + sound/soc/pxa/imote2.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/mmp-sspa.c | 5 +- sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/poodle.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 56 +- sound/soc/pxa/tosa.c | 1 + sound/soc/pxa/ttc-dkb.c | 1 + sound/soc/samsung/bells.c | 1 + sound/soc/samsung/i2s.c | 23 +- sound/soc/samsung/smdk_wm8994.c | 14 +- sound/soc/sh/rcar/adg.c | 11 +- sound/soc/sh/rcar/core.c | 50 +- sound/soc/sh/rcar/gen.c | 261 +++++---- sound/soc/sh/rcar/rsnd.h | 9 +- sound/soc/sh/rcar/scu.c | 12 +- sound/soc/sh/rcar/ssi.c | 52 +- sound/soc/soc-cache.c | 263 ++------- sound/soc/soc-core.c | 338 ++++++----- sound/soc/soc-dapm.c | 120 ++-- sound/soc/soc-devres.c | 86 +++ sound/soc/soc-generic-dmaengine-pcm.c | 101 +++- sound/soc/soc-io.c | 26 - sound/soc/soc-jack.c | 7 +- sound/soc/soc-pcm.c | 8 +- sound/soc/soc-utils.c | 6 +- sound/soc/spear/spdif_in.c | 12 +- sound/soc/spear/spdif_out.c | 14 +- sound/soc/tegra/tegra30_ahub.c | 115 +++- sound/soc/tegra/tegra30_ahub.h | 38 +- sound/soc/tegra/tegra30_i2s.c | 51 +- sound/soc/tegra/tegra30_i2s.h | 7 + sound/soc/tegra/tegra_asoc_utils.c | 2 + sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_pcm.c | 1 - 132 files changed, 4222 insertions(+), 3042 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs42l73.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt create mode 100644 Documentation/sound/alsa/soc/DPCM.txt create mode 100644 include/sound/cs42l73.h create mode 100644 sound/soc/soc-devres.c

2 1

[alsa-devel] Is that reasonable to support pinctrl PM in ASoC core?
by Nicolin Chen 25 Oct '13

25 Oct '13

Hi all, It's quite popular that more drivers are using pinctrl PM, for example: (Documentation/devicetree/bindings/arm/primecell.txt). Just like what runtime PM does, it would de-active and en-active pin group depending on whether it's being used or not. And I think this pinctrl PM might be also beneficial to cpu dai drivers since they might have actual pins, and they can hypnotize/wake them up along with runtime PM: @@ -183,6 +183,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; + pinctrl_pm_select_default_state(cpu_dai->dev); pm_runtime_get_sync(cpu_dai->dev); pm_runtime_get_sync(codec_dai->dev); pm_runtime_get_sync(platform->dev); As pinctrl PM is also reference counted and would return 0 if there is no pinctrl settings, this would not break current ASoC subsystem and any cpu dai driver. So I just want to ask if it's reasonable to add it. Or another one: is there anything similar to sleep pins? Although I've searched the whole ./sound directory, it seems no drivers under it is using this pinctrl PM. Best regards, Nicolin Chen

3 5

[alsa-devel] [PATCH] ASoC: Samsung: I2S: fix return value check in i2s_alloc_dai()
by Wei Yongjun 25 Oct '13

25 Oct '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> In case of error, the function platform_device_alloc() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index b302f3b..b721443 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1073,7 +1073,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) dev_set_drvdata(&i2s->pdev->dev, i2s); } else { /* Create a new platform_device for Secondary */ i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1); - if (IS_ERR(i2s->pdev)) + if (!i2s->pdev) return NULL; i2s->pdev->dev.parent = &pdev->dev;

2 1

[alsa-devel] [PATCH RESEND 1/2] ASoC: cs42l52: Add support for platform data reset GPIO
by Brian Austin 25 Oct '13

25 Oct '13

This patch adds support for grabbing a reset GPIO from platform data and resetting the codec Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs42l52.h | 3 +++ sound/soc/codecs/cs42l52.c | 25 ++++++++++++++++++++----- sound/soc/codecs/cs42l52.h | 2 +- 3 files changed, 24 insertions(+), 6 deletions(-) diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index a902e51..853df80 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -33,6 +33,9 @@ struct cs42l52_platform_data { /* MCLK Freq */ unsigned int mclk_freq; + + /* !RST GPIO */ + unsigned int reset_gpio; }; #endif /* __CS42L52_H */ diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b..6782d4875 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/input.h> @@ -1205,6 +1206,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l52_private *cs42l52; + struct cs42l52_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1222,12 +1224,23 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } + if (pdata) + cs42l52->pdata = *pdata; + + if (cs42l52->pdata.reset_gpio) { + ret = gpio_request_one(cs42l52->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L52 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l52->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 1); + } + i2c_set_clientdata(i2c_client, cs42l52); - - if (dev_get_platdata(&i2c_client->dev)) - memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), - sizeof(cs42l52->pdata)); - + ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, ARRAY_SIZE(cs42l52_threshold_patch)); if (ret != 0) @@ -1243,6 +1256,8 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, devid, CS42L52_CHIP_ID); return ret; } + dev_info(&i2c_client->dev, + "Cirrus Logic CS42L52, Revision: %02X\n", reg & 0xFF); regcache_cache_only(cs42l52->regmap, true); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 4277012..1a9412d 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -269,6 +269,6 @@ #define CS42L52_FIX_BITS1 0x3E #define CS42L52_FIX_BITS2 0x47 -#define CS42L52_MAX_REGISTER 0x34 +#define CS42L52_MAX_REGISTER 0x47 #endif -- 1.8.4.rc3

2 3

[alsa-devel] [PATCH] ALSA: hda - fix DigEn for Valleyview2 display codec
by mengdong.lin＠intel.com 25 Oct '13

25 Oct '13

From: Mengdong Lin <mengdong.lin(a)intel.com> On some Baytrail/Vallleyview platforms, DigEn is disabled on convertors. Thus audio data is blocked from passing through the node and no sound can be heard over HDMI/DP. This patch fixes the covertor's DigEn when setting up a stream. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e332db8..f619732 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -45,6 +45,7 @@ module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->vendor_id == 0x80862807) +#define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1061,6 +1062,20 @@ static void haswell_verify_D0(struct hda_codec *codec, } } +static void vlv_verify_digen(struct hda_codec *codec, hda_nid_t cvt_nid) +{ + unsigned int digi1; + + digi1 = snd_hda_codec_read(codec, cvt_nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + + if (!(digi1 & AC_DIG1_ENABLE)) + snd_hda_codec_write_cache(codec, cvt_nid, 0, + AC_VERB_SET_DIGI_CONVERT_1, + (digi1 | AC_DIG1_ENABLE) & 0xff); + +} + /* * Callbacks */ @@ -1078,6 +1093,9 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, if (is_haswell(codec)) haswell_verify_D0(codec, cvt_nid, pin_nid); + if (is_valleyview(codec)) + vlv_verify_digen(codec, cvt_nid); + if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: Fix compile error due to patch 'add S/PDIF support'
by Jean-Francois Moine 25 Oct '13

25 Oct '13

This patch fixes the compilation error of kirkwood-i2s.c introduced by the commit 75b9b65ee5a80e99e 'ASoC: kirkwood: add S/PDIF support'. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- Mark, sorry for this error. This patch applies on your repository broonie/sound.git/ topic/kirkwood. I may give you an other patch to apply with the commit removed. --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 9ec38d1..d34d917 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -568,7 +568,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } else { dev_info(&pdev->dev, "found external clock\n"); clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; + soc_dai = kirkwood_i2s_dai_extclk; } } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] Variable Delay through ALSA driver
by Brandon Yates 25 Oct '13

25 Oct '13

I have an application using the ALSA API that writes to playback device and simultaneously reads from capture device. My processing loop is : 1)Read from microphone (blocking) 2)Process Data 3)Write to speaker I am measuring the delay from the time I play a series of frames to the speaker until I detect the echo of this coming back in through the microphone. This delay seems to be changing every time I initialize the devices with ALSA. I have measured in the range of 60-160ms delay. I know things like period size effect delay, but using the exact same software and hardware configuration I would expect this to be constant across runs. What are some parameters that can help me make the delay deterministic? Thanks, Brandon

3 2

[alsa-devel] [PATCH] ALSA: hda - hdmi: Re-setup pin and infoframe on plug-in on all codecs
by Anssi Hannula 25 Oct '13

25 Oct '13

hdmi_setup_audio_infoframe() does not set up pin and infoframe if there is no connected sink. If a sink is connected while audio playback is already in progress, the pin and infoframe will not be properly set up, causing no audio or wrongly mapped audio. On Intel Haswell codecs the hdmi_setup_audio_infoframe() is already called again from hdmi_present_sense() when an ELD appears because transcoder:port mapping may have changed. Make the call non-Haswell-specific so that audio will be properly set up if the playback was started before a sink was connected. Tested on non-Haswell Intel HDMI codec by plugging sink in during playback. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/pci/hda/patch_hdmi.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 804adb8..a1fe4bb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1462,11 +1462,12 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; - /* Haswell-specific workaround: re-setup when the transcoder is - * changed during the stream playback + /* + * Re-setup pin and infoframe. This is needed e.g. when + * - sink is first plugged-in (infoframe is not set up if !monitor_present) + * - transcoder can change during stream playback on Haswell */ - if (is_haswell(codec) && - eld->eld_valid && !old_eld_valid && per_pin->setup) + if (eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } -- 1.8.1.5

2 1

[alsa-devel] Bug report Realtek ALC887-VD HDA Intel tone://15000
by Mihai Moise 24 Oct '13

24 Oct '13

I am writing to report a bug with ALSA that affects XMMS and Audacious. I have a Realtek ALC887-VD HDA Intel. I tested with kernel 3.12-rc4 and alsa-libs 1.0.27.2.10.gc1fbd and alsa-oss-1.0.25.2.g39df1 library. I also compiled and installed alsa-utils-1.0.27.2.6.gf1e99 which made me realize I had to move the libraries from /usr/lib to /usr/lib64. Playing tone://15000 in Audacious or XMMS with the ALSA driver without resampling sounds bad (in XMMS there is no resampling option). How to reproduce the bug: Play in XMMS tone://10 In Audacious you would have to lower the volume by 6 dB to get the same sound volume as in XMMS because it pre-amplifies sounds. Lower the PC-volume until you stop hearing clicks. To hear the clicks you need headphones and to raise the real-life amplifier volume all the way up. For me it is, Master: 91% PCM: 100% Front: 60% Select the ALSA driver and no resampling in Audacious (I think Audacious comes with no resampling as the default setting which is why this is so important). Play tone://15000 It is supposed to sound like this: perl -e '$|++;binmode(STDOUT);$_ = 0;while(1){print pack("v",32767.0*sin(($_&(~1))*3.14159265*15000.0/48000.0));++$_;}' | aplay -f dat but it sounds like this: perl -e '$|++;binmode(STDOUT);$_ = 0;while(1){print pack("v",32767.0*sin(($_&(~1))*3.14159265*15000.0/44100.0));++$_;}' | aplay -f cd This is how it sounds like if the OSS driver is invoked: perl -e '$|++;binmode(STDOUT);$_ = 0;while(1){print pack("v",32767.0*sin(($_&(~1))*3.14159265*15000.0/44100.0));++$_;}' | sox -V4 -b 16 -c 2 -r 44100 -e signed-integer -t raw - -r 44100 -t oss -d On my computer, the first and third command lines sound good, the second command line sounds bad. When using the OSS driver in XMMS it sounds like the first and third command lines. To go into more details: when using the ALSA driver in XMMS, for frequencies higher than tone://10000 the main frequency gets swamped by lower harmonics, so the frequency heard seems to drop the higher it increases. At tone://15000 what is heard is a louder lower harmonic. My suggestion is: could there be a way to force ALSA to resample the sound when using an ALC887-VD and/or a Realtek chipset and/or HDA Intel...

1 3

[alsa-devel] [PATCH] ALSA: hda - hdmi: Fix channel map switch not taking effect
by Anssi Hannula 24 Oct '13

24 Oct '13

Currently hdmi_setup_audio_infoframe() reprograms the HDA channel mapping only when the infoframe is not up-to-date or the non-PCM flag has changed. However, when just the channel map has been changed, the infoframe may still be up-to-date and non-PCM flag may not have changed, so the new channel map is not actually programmed into the HDA codec. Notably, this failing case is also always triggered when the device is already in a prepared state and a new channel map is configured while changing only the channel positions (for example, plain "speaker-test -c2 -m FR,FL"). Fix that by keeping track of the last programmed channel map and making hdmi_setup_audio_infoframe() always update the hardware mapping if the channel map has changed. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- BTW, is there some mechanism preventing hdmi_chmap_ctl_put() to be called at the same time with generic_hdmi_playback_pcm_prepare() that I didn't see with a quick look? Both of them call hdmi_setup_audio_infoframe() which is not thread-safe. sound/pci/hda/patch_hdmi.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7ea0245..09f38a1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -74,6 +74,8 @@ struct hdmi_spec_per_pin { bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ + bool last_chmap_set; /* channel-map override programmed? */ + unsigned char last_chmap[8]; /* last programmed channel-map */ char pcm_name[8]; /* filled in build_pcm callbacks */ }; @@ -955,15 +957,21 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, ai.bytes, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); } else { - /* For non-pcm audio switch, setup new channel mapping - * accordingly */ - if (per_pin->non_pcm != non_pcm) + /* For non-pcm audio switch or channel-map switch, + * setup new channel mapping accordingly */ + if (per_pin->non_pcm != non_pcm + || per_pin->chmap_set != per_pin->last_chmap_set + || (per_pin->chmap_set && memcmp(per_pin->chmap, per_pin->last_chmap, + ARRAY_SIZE(per_pin->chmap)) != 0)) hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, channels, per_pin->chmap, per_pin->chmap_set); } per_pin->non_pcm = non_pcm; + per_pin->last_chmap_set = per_pin->chmap_set; + if (per_pin->chmap_set) + memcpy(per_pin->last_chmap, per_pin->chmap, ARRAY_SIZE(per_pin->chmap)); } -- 1.8.1.5

2 12

[alsa-devel] [PATCH 1/2] ASoC: cs42l52: Add support for platform data reset GPIO
by Brian Austin 24 Oct '13

24 Oct '13

This patch adds support for grabbing a reset GPIO from platform data and resetting the codec Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs42l52.h | 3 +++ sound/soc/codecs/cs42l52.c | 25 ++++++++++++++++++++----- sound/soc/codecs/cs42l52.h | 2 +- 3 files changed, 24 insertions(+), 6 deletions(-) diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index a902e51..853df80 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -33,6 +33,9 @@ struct cs42l52_platform_data { /* MCLK Freq */ unsigned int mclk_freq; + + /* !RST GPIO */ + unsigned int reset_gpio; }; #endif /* __CS42L52_H */ diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b..6782d4875 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/input.h> @@ -1205,6 +1206,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l52_private *cs42l52; + struct cs42l52_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1222,12 +1224,23 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } + if (pdata) + cs42l52->pdata = *pdata; + + if ( cs42l52->pdata.reset_gpio) { + ret = gpio_request_one(cs42l52->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L52 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l52->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 1); + } + i2c_set_clientdata(i2c_client, cs42l52); - - if (dev_get_platdata(&i2c_client->dev)) - memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), - sizeof(cs42l52->pdata)); - + ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, ARRAY_SIZE(cs42l52_threshold_patch)); if (ret != 0) @@ -1243,6 +1256,8 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, devid, CS42L52_CHIP_ID); return ret; } + dev_info(&i2c_client->dev, + "Cirrus Logic CS42L52, Revision: %02X\n", reg & 0xFF); regcache_cache_only(cs42l52->regmap, true); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 4277012..1a9412d 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -269,6 +269,6 @@ #define CS42L52_FIX_BITS1 0x3E #define CS42L52_FIX_BITS2 0x47 -#define CS42L52_MAX_REGISTER 0x34 +#define CS42L52_MAX_REGISTER 0x47 #endif -- 1.8.4.rc3

1 2

[alsa-devel] [PATCH v7] ALSA: Add SoC on-chip internal ram support for DMA buffer allocation
by Nicolin Chen 24 Oct '13

24 Oct '13

Now it's quite common that an SoC contains its on-chip internal RAM. By using this RAM space for DMA buffer during audio playback/record, we can shutdown the voltage for external RAM to save power. So add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v6->v7: * Assigned pool to private_data field and accordingly refined helper functions. v5->v6: * Dropped remaining OF dependency bacause no need to add them: of_node doesn't care about OF; of_get_named_gen_pool() returns NULL if !OF. v4->v5: * Minimized the OF dependency. v3->v4: * Appropriately placed '#ifdef CONFIG_OF' * Tested by adding '#undef CONFIG_OF' to modified files, and passed compiling. v2->v3: * Moved iram specific mmap procedure out of ifdef ARCH_HAS_DMA_MMAP_COHERENT v1->v2: * Added of_node check * Dropped noops and unused physical addr in snd_free_dev_iram() --- include/sound/memalloc.h | 1 + sound/core/memalloc.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 6 ++++++ 3 files changed, 59 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b82..510aec4 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f..18c1d47 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> +#include <linux/genalloc.h> #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> @@ -157,6 +158,46 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal ram + * @dmab: buffer allocation record to store the allocated data + * @size: number of bytes to allocate from the iram + * + * This function requires iram phandle provided via of_node + */ +void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) +{ + struct device *dev = dmab->dev.dev; + struct gen_pool *pool = NULL; + + if (dev->of_node) + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + + if (!pool) + return; + + /* Assign the pool into private_data field */ + dmab->private_data = pool; + + dmab->area = (void *)gen_pool_alloc(pool, size); + if (!dmab->area) + return; + + dmab->addr = gen_pool_virt_to_phys(pool, (unsigned long)dmab->area); +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal ram + * @dmab: buffer allocation record to store the allocated data + */ +void snd_free_dev_iram(struct snd_dma_buffer *dmab) +{ + struct gen_pool *pool = dmab->private_data; + + if (pool && dmab->area) + gen_pool_free(pool, (unsigned long)dmab->area, dmab->bytes); +} #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +238,14 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_malloc_dev_iram(dmab, size); + if (dmab->area) + break; + /* Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +318,9 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_free_dev_iram(dmab); + break; case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..513f095 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3199,6 +3199,12 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) -- 1.8.4

3 6

[alsa-devel] [PATCH v4] ASoC: kirkwood: add S/PDIF support
by Jean-Francois Moine 24 Oct '13

24 Oct '13

This patch adds S/PDIF input/output for mvebu DT boards. Three output DAI's are created: I2S only, S/PDIF only, both I2S and S/PDIF. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v4: declare 3 DAI's v3: (Mark Brown) - giving a name to the DAI's impacts the Openrd and HP t5325 clients - test i2s or spdif from the DAI's ID instead of its name v2: declare 2 DAI's (Mark Brown) --- sound/soc/kirkwood/kirkwood-i2s.c | 119 ++++++++++++--------- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- 3 file changed, 72 insertions(+), 51 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d0504a2..c3a5a3e 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -240,6 +244,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s only */ + else if (dai->id == 1) + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif only */ + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ @@ -258,7 +267,8 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +282,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +313,13 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + if (dai->id == 0) + ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN; /* i2s only */ + else if (dai->id == 1) + ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif only */ + + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -361,9 +379,8 @@ static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int kirkwood_i2s_probe(struct snd_soc_dai *dai) +static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) { - struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); unsigned long value; unsigned int reg_data; @@ -404,45 +421,46 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; +#define DAI_DEF(name_p, id_p, rates_p) {\ + .name = name_p,\ + .id = id_p,\ + .playback = {\ + .channels_min = 1,\ + .channels_max = 2,\ + .rates = rates_p,\ + .formats = KIRKWOOD_I2S_FORMATS,\ + },\ + .capture = {\ + .channels_min = 1,\ + .channels_max = 2,\ + .rates = rates_p,\ + .formats = KIRKWOOD_I2S_FORMATS,\ + },\ + .ops = &kirkwood_i2s_dai_ops,\ +} -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .ops = &kirkwood_i2s_dai_ops, +static struct snd_soc_dai_driver kirkwood_i2s_dai[3] = { + DAI_DEF("i2s", 0, + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000), + DAI_DEF("spdif", 1, + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000), + DAI_DEF("i2s/spdif", 2, + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000), }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .ops = &kirkwood_i2s_dai_ops, +static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[3] = { + DAI_DEF("i2s", 0, + SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT), + DAI_DEF("spdif", 1, + SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT), + DAI_DEF("i2s/spdif", 2, + SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT), }; static const struct snd_soc_component_driver kirkwood_i2s_component = { @@ -452,7 +470,7 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; + struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -506,7 +524,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } else { dev_info(&pdev->dev, "found external clock\n"); clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; + soc_dai = kirkwood_i2s_dai_extclk; } } @@ -524,7 +542,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); + soc_dai, 3); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto err_component; @@ -535,6 +553,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); goto err_platform; } + + kirkwood_i2s_init(priv); + return 0; err_platform: snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 025be0e..65f2a5b 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 27545b0..d213832 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -68,7 +68,7 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2] ASoC: kirkwood: add S/PDIF support
by Jean-Francois Moine 24 Oct '13

24 Oct '13

This patch adds S/PDIF input/output for mvebu DT boards. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v3: (Mark Brown) - giving a name to the DAI's impacts the Openrd and HP t5325 clients - test i2s or spdif from the DAI's ID instead of its name v2: declare 2 DAI's (Mark Brown) --- sound/soc/kirkwood/kirkwood-i2s.c | 99 ++++++++++++++++++---- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- 3 file changed, 84 insertions(+), 19 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index c2433ea..f3e1547 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -240,6 +244,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ @@ -258,7 +267,8 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +282,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +313,13 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + if (dai->id == 0) + ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ + + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -361,9 +379,8 @@ static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int kirkwood_i2s_probe(struct snd_soc_dai *dai) +static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) { - struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); unsigned long value; unsigned int reg_data; @@ -404,9 +421,29 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; - -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { + { + .name = "i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, .playback = { .channels_min = 1, .channels_max = 2, @@ -422,10 +459,34 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { + { + .name = "i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, .playback = { .channels_min = 1, .channels_max = 2, @@ -443,6 +504,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; static const struct snd_soc_component_driver kirkwood_i2s_component = { @@ -452,7 +514,7 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; + struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -521,7 +583,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); + soc_dai, 2); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto err_component; @@ -532,6 +594,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); goto err_platform; } + + kirkwood_i2s_init(priv); + return 0; err_platform: snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 025be0e..65f2a5b 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 27545b0..d213832 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -68,7 +68,7 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH] ASoC: fsl: Add missing pm to current machine drivers
by Nicolin Chen 24 Oct '13

24 Oct '13

Add missing pm to current machine drivers so that all of them would correctly do suspend/resume. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/imx-mc13783.c | 1 + sound/soc/fsl/imx-sgtl5000.c | 1 + sound/soc/fsl/imx-wm8962.c | 1 + 3 files changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index a3d60d4..8ded5ad 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,6 +160,7 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 6f4bdc8..4b65c04 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -201,6 +201,7 @@ static struct platform_driver imx_sgtl5000_driver = { .driver = { .name = "imx-sgtl5000", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 6c60666..72064e9 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -310,6 +310,7 @@ static struct platform_driver imx_wm8962_driver = { .driver = { .name = "imx-wm8962", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_wm8962_dt_ids, }, .probe = imx_wm8962_probe, -- 1.8.4

2 1

[alsa-devel] [PATCHv6][ 1/4] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 24 Oct '13

24 Oct '13

Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Sascha Hauer <kernel(a)pengutronix.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: Eric Bénard <eric(a)eukrea.com> Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 23 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 135 ++++++++++++++++---- 3 files changed, 140 insertions(+), 23 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..8791037 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,23 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,eukrea-tlv320" +- model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- audio-codec : The phandle of the tlv320aic23 audio codec. +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,eukrea-tlv320"; + model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,audio-codec = <&tlv320aic23>; + mux-int-port = <2>; + mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f..9c3cd64 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -161,12 +161,15 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || OF depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9a4a0ca..92710cf 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -17,6 +17,8 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,49 +28,64 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 +struct eukrea_tlv320 { + char of; +}; + static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct eukrea_tlv320 *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; + /* fsl_ssi lacks the set_fmt ops. */ + if (!priv->of) { + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + dev_err(cpu_dai->dev, + "Failed to set the cpu dai format\n"); + return ret; + } } ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec dai format.\n"); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret) { - pr_err("%s: failed setting codec sysclk\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec dai sysclk.\n"); return ret; } - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); - ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, - SND_SOC_CLOCK_IN); - if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); - return ret; - } + /* fsl_ssi lacks the set_sysclk ops */ + if (!priv->of) { + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(cpu_dai->dev, "Can't set CPU system clock IMX_SSP_SYS_CLK.\n"); + return ret; + } + } return 0; } @@ -80,7 +97,6 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, @@ -97,8 +113,56 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct platform_device *ssi_pdev; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np; + struct eukrea_tlv320 *priv; + + priv = devm_kzalloc(&pdev->dev, + sizeof(struct eukrea_tlv320), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; - if (machine_is_eukrea_cpuimx27()) { + if (np) { + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, + "ssi-controller missing or invalid.\n"); + ret = -ENODEV; + goto err; + } + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + eukrea_tlv320_dai.platform_of_node = ssi_np; + priv->of = 1; + } else { + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + } + + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -115,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -130,14 +198,29 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,eukrea-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } eukrea_tlv320.dev = &pdev->dev; + snd_soc_card_set_drvdata(&eukrea_tlv320, priv); ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -149,13 +232,21 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,eukrea-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, - .remove = eukrea_tlv320_remove,}; + .remove = eukrea_tlv320_remove, +}; module_platform_driver(eukrea_tlv320_driver); -- 1.7.9.5

2 4

[alsa-devel] [GIT PULL] [00/29] playback-only DICE driver
by Clemens Ladisch 24 Oct '13

24 Oct '13

Hi Takashi, please pull this driver for FireWire DACs based on the DICE chip family. While the driver supports only PCM playback at the moment, it fills a gap left by FFADO, which supports only full- duplex devices. Thanks, Clemens ---------------------------------------------------------------- The following changes since commit 6e4664525b1db28f8c4e1130957f70a94c19213e: Linux 3.11 (2013-09-02 13:46:10 -0700) are available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git dice-driver-playback-only for you to fetch changes up to b20be8de1b3972ccf9af72850b045214faa8d830: ALSA: dice: restrict the driver to playback-only devices (2013-10-20 22:07:57 +0200) ---------------------------------------------------------------- Clemens Ladisch (28): ALSA: add DICE driver ALSA: dice: optimize bus reset handling ALSA: dice: allow all sample rates ALSA: dice: reduce noisy logging ALSA: dice, firewire-lib: add blocking mode ALSA: dice: fix hang when unplugging a running device ALSA: dice: implement hwdep device ALSA: dice: clear device lock when closing hwdep device ALSA: firewire: introduce amdtp_out_stream_running() ALSA: dice: reorganize interface definitions ALSA: dice: fix device detection for other vendors ALSA: dice: support dual-wire stream format at 192 kHz ALSA: dice: optimize reading of consecutive registers ALSA: firewire: extend snd_fw_transaction() ALSA: dice: avoid superflous write at bus reset ALSA: dice: remove 10s period length limit ALSA: dice: remove superfluous field ALSA: dice: make amdtp_rates[] const ALSA: dice: get clock capabilities ALSA: dice: allow notifications during initialization ALSA: dice: get rate-dependent parameters ALSA: dice: dynamic sample rate selection ALSA: dice: check clock change timeout ALSA: dice: add a proc file to show device information ALSA: dice: document quadlet alignment ALSA: dice: dice_proc_read: remove wrong typecast ALSA: dice: fix detection of Weiss devices ALSA: dice: restrict the driver to playback-only devices Stefan Richter (1): ALSA: dice: fix locking Documentation/ioctl/ioctl-number.txt | 1 + include/uapi/sound/Kbuild | 1 + include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 51 ++ sound/firewire/Kconfig | 15 + sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 209 +++-- sound/firewire/amdtp.h | 46 +- sound/firewire/cmp.c | 50 +- sound/firewire/dice-interface.h | 371 +++++++++ sound/firewire/dice.c | 1494 ++++++++++++++++++++++++++++++++++ sound/firewire/fcp.c | 2 +- sound/firewire/isight.c | 43 +- sound/firewire/lib.c | 24 +- sound/firewire/lib.h | 7 +- sound/firewire/scs1x.c | 8 +- sound/firewire/speakers.c | 16 +- 17 files changed, 2192 insertions(+), 151 deletions(-) create mode 100644 include/uapi/sound/firewire.h create mode 100644 sound/firewire/dice-interface.h create mode 100644 sound/firewire/dice.c

2 30

[alsa-devel] [PATCH 1/2] ALSA: hda - Add support of ALC255 codecs
by Takashi Iwai 24 Oct '13

24 Oct '13

From: Kailang Yang <kailang(a)realtek.com> It's just another variant of ALC269 & co. Signed-off-by: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52c26d3..1c4f8c5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2540,6 +2540,7 @@ enum { ALC269_TYPE_ALC283, ALC269_TYPE_ALC284, ALC269_TYPE_ALC286, + ALC269_TYPE_ALC255, }; /* @@ -2565,6 +2566,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC282: case ALC269_TYPE_ALC283: case ALC269_TYPE_ALC286: + case ALC269_TYPE_ALC255: ssids = alc269_ssids; break; default: @@ -4102,6 +4104,9 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0286: spec->codec_variant = ALC269_TYPE_ALC286; break; + case 0x10ec0255: + spec->codec_variant = ALC269_TYPE_ALC255; + break; } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { @@ -4814,6 +4819,7 @@ static int patch_alc680(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, + { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, -- 1.8.4

1 1

[alsa-devel] [PATCH] ALSA: hda - Fix mute LED on HP laptops in runtime suspend
by Takashi Iwai 24 Oct '13

24 Oct '13

When HP laptops with mute and mic-record LEDs go to runtime suspend, these LEDs are turned on forcibly no matter whether GPIO pis are on or off. This strange behavior seems triggered by resetting the HD-audio bus link at azx_rutime_suspend(). So, just add a new hda_bus flag to avoid the link reset at runtime suspend and set it for these HP machines. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 3 ++- sound/pci/hda/patch_sigmatel.c | 4 +++- 3 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 7aa9870..77db694 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -698,6 +698,7 @@ struct hda_bus { unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ + unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */ int primary_dig_out_type; /* primary digital out PCM type */ }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e61a01..a0a06f7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2986,7 +2986,8 @@ static int azx_runtime_suspend(struct device *dev) STATESTS_INT_MASK); azx_stop_chip(chip); - azx_enter_link_reset(chip); + if (!chip->bus->avoid_link_reset) + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index fba0cef..69a549a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2091,8 +2091,10 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, { struct sigmatel_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ + codec->bus->avoid_link_reset = 1; + } } static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, -- 1.8.4

1 0

[alsa-devel] [PATCH 0/8] treewide: Remove OOM message after input_alloc_device
by Joe Perches 24 Oct '13

24 Oct '13

Joe Perches (8): Documentation: Remove OOM message after input_allocate_device cell: Remove OOM message after input_allocate_device hid: Remove OOM message after input_allocate_device input: Remove OOM message after input_allocate_device media: Remove OOM message after input_allocate_device platform:x86: Remove OOM message after input_allocate_device staging: Remove OOM message after input_allocate_device sound: Remove OOM message after input_allocate_device Documentation/input/input-programming.txt | 1 - arch/powerpc/platforms/cell/cbe_powerbutton.c | 1 - drivers/hid/hid-input.c | 1 - drivers/hid/hid-picolcd_core.c | 5 ++--- drivers/input/joystick/as5011.c | 2 -- drivers/input/joystick/db9.c | 1 - drivers/input/joystick/gamecon.c | 4 +--- drivers/input/joystick/turbografx.c | 1 - drivers/input/joystick/walkera0701.c | 1 - drivers/input/keyboard/amikbd.c | 4 +--- drivers/input/keyboard/davinci_keyscan.c | 1 - drivers/input/keyboard/gpio_keys.c | 1 - drivers/input/keyboard/lpc32xx-keys.c | 1 - drivers/input/keyboard/max7359_keypad.c | 1 - drivers/input/keyboard/mcs_touchkey.c | 1 - drivers/input/keyboard/mpr121_touchkey.c | 1 - drivers/input/keyboard/nomadik-ske-keypad.c | 1 - drivers/input/keyboard/opencores-kbd.c | 1 - drivers/input/keyboard/pmic8xxx-keypad.c | 1 - drivers/input/keyboard/pxa27x_keypad.c | 1 - drivers/input/keyboard/pxa930_rotary.c | 1 - drivers/input/keyboard/qt1070.c | 1 - drivers/input/keyboard/qt2160.c | 1 - drivers/input/keyboard/sh_keysc.c | 1 - drivers/input/keyboard/tc3589x-keypad.c | 1 - drivers/input/keyboard/tnetv107x-keypad.c | 1 - drivers/input/keyboard/w90p910_keypad.c | 1 - drivers/input/misc/88pm80x_onkey.c | 1 - drivers/input/misc/88pm860x_onkey.c | 1 - drivers/input/misc/arizona-haptics.c | 4 +--- drivers/input/misc/atlas_btns.c | 4 +--- drivers/input/misc/da9052_onkey.c | 1 - drivers/input/misc/da9055_onkey.c | 4 +--- drivers/input/misc/ideapad_slidebar.c | 1 - drivers/input/misc/ims-pcu.c | 7 +------ drivers/input/misc/kxtj9.c | 4 +--- drivers/input/misc/max8997_haptic.c | 1 - drivers/input/misc/mc13783-pwrbutton.c | 4 +--- drivers/input/misc/mpu3050.c | 1 - drivers/input/misc/pcf8574_keypad.c | 1 - drivers/input/misc/pm8xxx-vibrator.c | 1 - drivers/input/misc/pmic8xxx-pwrkey.c | 1 - drivers/input/misc/pwm-beeper.c | 1 - drivers/input/misc/twl4030-pwrbutton.c | 4 +--- drivers/input/misc/twl6040-vibra.c | 1 - drivers/input/mouse/appletouch.c | 4 +--- drivers/input/mouse/bcm5974.c | 4 +--- drivers/input/mouse/cyapa.c | 4 +--- drivers/input/mouse/inport.c | 1 - drivers/input/mouse/logibm.c | 1 - drivers/input/mouse/pc110pad.c | 1 - drivers/input/mouse/pxa930_trkball.c | 1 - drivers/input/tablet/aiptek.c | 5 +---- drivers/input/tablet/gtco.c | 1 - drivers/input/touchscreen/88pm860x-ts.c | 1 - drivers/input/touchscreen/atmel_mxt_ts.c | 1 - drivers/input/touchscreen/atmel_tsadcc.c | 1 - drivers/input/touchscreen/bu21013_ts.c | 1 - drivers/input/touchscreen/cyttsp4_core.c | 2 -- drivers/input/touchscreen/da9034-ts.c | 1 - drivers/input/touchscreen/edt-ft5x06.c | 1 - drivers/input/touchscreen/eeti_ts.c | 5 +---- drivers/input/touchscreen/htcpen.c | 1 - drivers/input/touchscreen/intel-mid-touch.c | 1 - drivers/input/touchscreen/lpc32xx_ts.c | 1 - drivers/input/touchscreen/mcs5000_ts.c | 1 - drivers/input/touchscreen/migor_ts.c | 1 - drivers/input/touchscreen/mk712.c | 1 - drivers/input/touchscreen/pixcir_i2c_ts.c | 1 - drivers/input/touchscreen/s3c2410_ts.c | 1 - drivers/input/touchscreen/ti_am335x_tsc.c | 1 - drivers/input/touchscreen/tnetv107x-ts.c | 1 - drivers/media/rc/imon.c | 8 ++------ drivers/media/usb/em28xx/em28xx-input.c | 4 +--- drivers/media/usb/pwc/pwc-if.c | 1 - drivers/platform/x86/asus-laptop.c | 5 ++--- drivers/platform/x86/eeepc-laptop.c | 4 +--- drivers/platform/x86/ideapad-laptop.c | 4 +--- drivers/platform/x86/intel_mid_powerbtn.c | 4 +--- drivers/platform/x86/panasonic-laptop.c | 5 +---- drivers/platform/x86/thinkpad_acpi.c | 1 - drivers/platform/x86/topstar-laptop.c | 4 +--- drivers/platform/x86/toshiba_acpi.c | 4 +--- drivers/staging/cptm1217/clearpad_tm1217.c | 2 -- drivers/staging/iio/adc/mxs-lradc.c | 4 +--- drivers/staging/ste_rmi4/synaptics_i2c_rmi4.c | 2 -- sound/pci/hda/hda_beep.c | 4 +--- 87 files changed, 29 insertions(+), 152 deletions(-) -- 1.8.1.2.459.gbcd45b4.dirty

2 2

Re: [alsa-devel] [PATCH] ASoC: tpa6130a2: Add device tree support
by Mark Brown 24 Oct '13

24 Oct '13

On Wed, Oct 23, 2013 at 02:03:28PM +0200, Sebastian Reichel wrote: > Add device tree support to tpa6130a2 driver and document the > bindings. Applied, thanks.

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix unbalanced runtime PM refcount after S3/S4
by Takashi Iwai 24 Oct '13

24 Oct '13

When a machine goes to S3/S4 after power-save is enabled, the runtime PM refcount might be incorrectly decreased because the power-down triggered soon after resume assumes that the controller was already powered up, and issues the pm_notify down. This patch fixes the incorrect pm_notify call simply by checking the current value properly. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5b6c4e3..748c6a9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4864,8 +4864,8 @@ static void hda_power_work(struct work_struct *work) spin_unlock(&codec->power_lock); state = hda_call_codec_suspend(codec, true); - codec->pm_down_notified = 0; - if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { + if (!codec->pm_down_notified && + !bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { codec->pm_down_notified = 1; hda_call_pm_notify(bus, false); } -- 1.8.4

1 0

[alsa-devel] [PATCH v2 00/11] Fix AM335x-evm analog audio support
by jsarha＠ti.com 23 Oct '13

23 Oct '13

From: Jyri Sarha <jsarha(a)ti.com> The first version of patches can been found here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-September/066178.… Changes for v2 - Dropped our "ASoC: davinci-mcasp: Add pinctrl support" since driver core is taking care of this now. - Cleanup am33xx audio build - Add regulators to tlv320aic3x DT binding document - Remove dm365-voice-codec-audio DT support as it has never been tested an probably does not work - Add output pins and Line In connector to davinci-evm-audio DT binding doc - Remove asp_chan_q and ram_chan_q properties from mcasp DT node in DT mode mcasp is hardcoded to event queue 0 (highest priority) - Add pins to tlv320aic3x DT bindings document. If I misunderstood Marks comment and this patch is not needed, then just leave it out Changes based on TI internal discussions - Move system clock rate logic away from from evm_hw_params soc-op - Remove unnecesary #if defined(CONFIG_OF) from davinci-evm.c - Make dma property DT binding document more exact - Add only "dma" reg location instead of separate "dma-tx" and "dma-rx" - Primarily look for "mpu" reg property, but fall back to index 0 if not found - Remove interrupt property from mcasp DT node as it is not used - Remove #address-cells and #size-cells mcasp properties as they are not needed This set of patches fixes the basic audio support for am335x-evm. It should also be relatively simple to add the necessary nodes to relevant dts files to get BeagleBone + AudioCape and am335x-evmsk working too. The patch set depends on following patches: [PATCH v11 4/8] ARM: dts: add AM33XX EDMA support https://lkml.org/lkml/2013/6/18/49 [PATCH v11 5/8] ARM: dts: add AM33XX SPI DMA support https://lkml.org/lkml/2013/6/18/55 [PATCH v2] ARM: EDMA: Fix clearing of unused list for DT DMA resources https://lkml.org/lkml/2013/7/22/441 I have tried my best not to break the existing support for older davinci boards, but since I do not have those boards I can not be sure. Some commit comments refer to a dmaengine based davinci audio implementation which is planned for but nothing has been done yet. Best regards, Jyri Darren Etheridge (1): ARM/dts: am335x-evm: Add audio support for am335x-evm.dts Hebbar, Gururaja (2): ASoC: davinci-evm: Add device tree binding ASoC: davinci: Add support for AM33xx SoC Audio Jyri Sarha (7): ASoC: davinci-evm: Move sysclk logic away from evm_hw_params ASoC: davinci-mcasp: Add DMA register locations to DT ASoC: davinci-mcasp: Extract DMA channels directly from DT ASoC: davinci-mcasp: Remove interrupt property from DT bindings doc ASoC: tlv320aic3x: Add regulators to DT bindings document ASoC: tlv320aic3x: Add codec pins to DT bindings document ARM/dts: am33xx: mcasp: Add new dma register location to reg-property Pantelis Antoniou (1): ARM/dts: am33xx: Add mcasp0 and mcasp1 device tree entries .../bindings/sound/davinci-evm-audio.txt | 58 ++++++ .../bindings/sound/davinci-mcasp-audio.txt | 15 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 26 +++ arch/arm/boot/dts/am335x-evm.dts | 56 ++++++ arch/arm/boot/dts/am33xx.dtsi | 25 +++ include/linux/platform_data/davinci_asp.h | 2 + sound/soc/davinci/Kconfig | 18 +- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 184 +++++++++++++++++--- sound/soc/davinci/davinci-mcasp.c | 106 +++++++---- 10 files changed, 428 insertions(+), 63 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

9 121

[alsa-devel] [PATCH 1/3] ASoC: samsung: Provide helper for DMA init
by Mark Brown 23 Oct '13

23 Oct '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 38 insertions(+), 44 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 2acf987..bd09410 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_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_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_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_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b..ee23194 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315..fb09a1c 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2c4d250..9cd0a2a 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -963,7 +963,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256f..6a5e4bf 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.4.rc3

2 5

[alsa-devel] [PATCH RESEND] ASoC: dont call dapm_sync while reporting jack always
by Vinod Koul 23 Oct '13

23 Oct '13

While reporting the jack status snd_soc_jack_report() invokes snd_soc_dapm_sync() always. This should be required when we have pins associated with jack and reporting enables or disables these. So add a check for this case Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Sorry previous one had some non upstream cruft sound/soc/soc-jack.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index c9f7953..9fd4ed8 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -65,6 +65,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) struct snd_soc_codec *codec; struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; + unsigned int sync = 0; int enable; trace_snd_soc_jack_report(jack, mask, status); @@ -92,12 +93,16 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_soc_dapm_enable_pin(dapm, pin->pin); else snd_soc_dapm_disable_pin(dapm, pin->pin); + + /* we need to sync for this case only */ + sync = 1; } /* Report before the DAPM sync to help users updating micbias status */ blocking_notifier_call_chain(&jack->notifier, jack->status, jack); - snd_soc_dapm_sync(dapm); + if (sync) + snd_soc_dapm_sync(dapm); snd_jack_report(jack->jack, jack->status); -- 1.7.0.4

3 4

[alsa-devel] [PATCH] ASoC: tegra: Remove redundant initialisation of compat_filter_fn
by Mark Brown 22 Oct '13

22 Oct '13

From: Mark Brown <broonie(a)linaro.org> Setting a field in a static struct to NULL has no effect so don't bother (and don't generate false positives for grep). Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/tegra/tegra_pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index f056f63..7b2d23b 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -56,7 +56,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { static const struct snd_dmaengine_pcm_config tegra_dmaengine_pcm_config = { .pcm_hardware = &tegra_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = NULL, .prealloc_buffer_size = PAGE_SIZE * 8, }; -- 1.8.4.rc3

3 4

[alsa-devel] [PATCH v6] ALSA: Add SoC on-chip internal memory support for DMA buffer allocation
by Nicolin Chen 22 Oct '13

22 Oct '13

Now it's quite common that an SoC contains its on-chip internal memory. By using this memory space for DMA buffer during audio playback/record, we can shutdown the voltage for external memory to save power. Thus add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v5->v6: * Dropped remaining OF dependency bacause no need to add them: of_node doesn't care about OF; of_get_named_gen_pool() returns NULL if !OF. v4->v5: * Minimized the OF dependency. v3->v4: * Appropriately placed '#ifdef CONFIG_OF' * Tested by adding '#undef CONFIG_OF' to modified files, and passed compiling. v2->v3: * Moved iram specific mmap procedure out of ifdef ARCH_HAS_DMA_MMAP_COHERENT v1->v2: * Added of_node check * Dropped noops and unused physical addr in snd_free_dev_iram() --- include/sound/memalloc.h | 1 + sound/core/memalloc.c | 65 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 6 +++++ 3 files changed, 72 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b82..510aec4 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f..ee505e9 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> +#include <linux/genalloc.h> #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> @@ -157,6 +158,59 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal memory + * @dev: DMA device pointer + * @size: number of bytes to allocate from the iram + * @dma: dma-view physical address + * + * Return cpu-view address or NULL indicating allocating failure + * + * This function requires iram phandle provided via of_node + */ +void *snd_malloc_dev_iram(struct device *dev, size_t size, dma_addr_t *dma) +{ + struct gen_pool *pool = NULL; + unsigned long vaddr; + + if (!dev->of_node) + return NULL; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + if (!pool) + return NULL; + + vaddr = gen_pool_alloc(pool, size); + if (!vaddr) + return NULL; + + *dma = gen_pool_virt_to_phys(pool, vaddr); + + return (void *)vaddr; +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal memory + * @dev: DMA device pointer + * @size: size in bytes of memory to free + * @ptr: cpu-view address returned from snd_malloc_dev_iram + * + * This function requires iram phandle provided via of_node + */ +void snd_free_dev_iram(struct device *dev, size_t size, void *ptr) +{ + struct gen_pool *pool = NULL; + + if (!dev->of_node) + return; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + if (!pool) + return; + + gen_pool_free(pool, (unsigned long)ptr, size); +} #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +251,14 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + dmab->area = snd_malloc_dev_iram(device, size, &dmab->addr); + if (dmab->area) + break; + /* Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +331,9 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_free_dev_iram(dmab->dev.dev, dmab->bytes, dmab->area); + break; case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..513f095 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3199,6 +3199,12 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) -- 1.8.4

4 4

[alsa-devel] [PATCH 1/2] ASoC: cirrus: Enable compile test builds
by Mark Brown 22 Oct '13

22 Oct '13

From: Mark Brown <broonie(a)linaro.org> The core support for ep93xx (currently only the DMA driver) does not depend on the architecture at all and everything else has more strict dependencies so enable compile test builds for improved build coverage. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/cirrus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 2c20f01..06f938d 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,6 +1,6 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" - depends on ARCH_EP93XX && SND_SOC + depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 1.8.4.rc3

2 4

[alsa-devel] [PATCH] ASoC: dmaengine: Use filter_data rather than dma_data for compat requests
by Mark Brown 22 Oct '13

22 Oct '13

From: Mark Brown <broonie(a)linaro.org> When using the legacy filter function channel requests we currently pass the audio specific struct snd_dmaengine_dai_dma_data which isn't likely to be helpful for actual filtering. Since there's already a field in the structure called filter_data clearly intended for use here convert the driver to use that. All existing users of plain filter functions have been converted to use an explict compat function to override this behaviour except i.MX which is working around this issue in its filter function and is updated to just use filter_data directly here. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/imx-pcm-dma.c | 4 +--- sound/soc/soc-generic-dmaengine-pcm.c | 5 ++++- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 4dc1296..aee2307 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -25,12 +25,10 @@ static bool filter(struct dma_chan *chan, void *param) { - struct snd_dmaengine_dai_dma_data *dma_data = param; - if (!imx_dma_is_general_purpose(chan)) return false; - chan->private = dma_data->filter_data; + chan->private = param; return true; } diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index e29ec3c..17dd379 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -126,6 +126,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( struct snd_pcm_substream *substream) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; @@ -134,7 +137,7 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( return pcm->config->compat_request_channel(rtd, substream); return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); + dma_data->filter_data); } static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) -- 1.8.4.rc3

3 2

[alsa-devel] no sound on Haswell Lenovo T440s with 3.12-rc5
by John Twideldum 22 Oct '13

22 Oct '13

Hi all... There is a shiny new Lenovo T440s on my desk, which is a Haswell generation Ultrabook. Configuration dump running 3.12-rc5: http://www.alsa-project.org/db/?f=2eec0890ae63625c1b47a7e7cbcbb1722c8c5080 Symptoms: I have console beep, but no sound. Specifically, a) it appears "Intel Haswell HDMI" on displayport appears to be picked as default output, b) alternative output 2 "HDA Intel PCH/Realtek ALC292" has no PCM control. So my wild guess would be a) PCH needs to be default and b) the PCM controls are missing due to a bug Is there a patch already queued for 3.13? If not - willing to test, but may take some time as I'm travelling... Thank you!

1 0

[alsa-devel] [PATCHv5][ 1/4] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 21 Oct '13

21 Oct '13

Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Sascha Hauer <kernel(a)pengutronix.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: Eric Bénard <eric(a)eukrea.com> Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- ChangeLog v4->v5: - The pr_err are now replaced with (non line wrapped) dev_err. - The place of the "};" in eukrea_tlv320_driver is now fixed. --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 23 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 135 ++++++++++++++++---- 3 files changed, 140 insertions(+), 23 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..8791037 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,23 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,eukrea-tlv320" +- model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- audio-codec : The phandle of the tlv320aic23 audio codec. +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,eukrea-tlv320"; + model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,audio-codec = <&tlv320aic23>; + mux-int-port = <2>; + mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f..9c3cd64 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -161,12 +161,15 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || OF depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9a4a0ca..92710cf 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -17,6 +17,8 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,49 +28,64 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 +struct eukrea_tlv320 { + char of; +}; + static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct eukrea_tlv320 *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; + /* fsl_ssi lacks the set_fmt ops. */ + if (!priv->of) { + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + dev_err(cpu_dai->dev, + "Failed to set the cpu dai format\n"); + return ret; + } } ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec dai format.\n"); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret) { - pr_err("%s: failed setting codec sysclk\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec dai sysclk.\n"); return ret; } - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); - ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, - SND_SOC_CLOCK_IN); - if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); - return ret; - } + /* fsl_ssi lacks the set_sysclk ops */ + if (!priv->of) { + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(cpu_dai->dev, "Can't set CPU system clock IMX_SSP_SYS_CLK.\n"); + return ret; + } + } return 0; } @@ -80,7 +97,6 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, @@ -97,8 +113,56 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct platform_device *ssi_pdev; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np; + struct eukrea_tlv320 *priv; + + priv = devm_kzalloc(&pdev->dev, + sizeof(struct eukrea_tlv320), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; - if (machine_is_eukrea_cpuimx27()) { + if (np) { + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, + "ssi-controller missing or invalid.\n"); + ret = -ENODEV; + goto err; + } + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + eukrea_tlv320_dai.platform_of_node = ssi_np; + priv->of = 1; + } else { + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + } + + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -115,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -130,14 +198,29 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,eukrea-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } eukrea_tlv320.dev = &pdev->dev; + snd_soc_card_set_drvdata(&eukrea_tlv320, priv); ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -149,13 +232,21 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,eukrea-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, - .remove = eukrea_tlv320_remove,}; + .remove = eukrea_tlv320_remove, +}; module_platform_driver(eukrea_tlv320_driver); -- 1.7.9.5

1 3

[alsa-devel] Questions about virtual ALSA driver (dummy), PortAudio and full-duplex drops
by Smilen Dimitrov 21 Oct '13

21 Oct '13

Hi list(s), Apologies in advance for the longish exposition - however, I have bumped into a problem with ALSA driver development, which raised some questions that I cannot really understand at the moment. I find it especially frustrating that I cannot formulate a simple question, but instead I have to resort to test code, captures and plots, to discuss my (mis)conceptions through those - so I hope at least someone will bear with this wall of text (some 23KB plaintext), and I'll be able to get some help with this. I originally started working on a 16-bit, 44.1 kHz stereo ALSA driver for a device; and came to a point where I could achieve (what I thought was) full-duplex by running `arecord` and `aplay`, in separate shells, without any problems. However, if I tried to do the same from `audacity` - that is, have "record" running, while another track plays on the same card, with "Audacity Preferences/Recording/Overdub: Play other tracks while recording new one" enabled - then I'd experience some strange drops in the capture, which were not reported by any debug messages, even after I rebuilt and used PortAudio with debug messages on (my eventual goal is to have this experiment running in Audacity without drops). (Therefore CC to audacity-devel, since I first started looking for "audacity full duplex drops" and couldn't find much; and CC to portaudio since some of my questions depend on correct understanding of PortAudio). Anyways, I think I managed to reconstruct the drop problem I get (with my device driver and `audacity`), by using a modified command line program from the tests in the PortAudio distribution, and a modified `dummy` driver from the ALSA distribution. The complete code and test scripts are available in this directory: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/ I ran this on Ubuntu 11.04 Natty, Linux 2.6.38 and corresponding ALSA 1.0.24.2, audacity-1.3.13, PortAudio V19 - more about my machine and environment setup is in the Readme: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/Readme In that directory there is a script, `run-alsa-pa-tests.sh`; which compiles: the ALSA driver from `dummy-2.6.32-patest.c`; and the PortAudio program from `patest_duplex_wire.c`; with different compile-time defines - and then runs the program, obtaining verbose logs (some timestamped through `ftrace`), which end up as .csv files. The .csv files are finally plotted in the time domain using a gnuplot script, `traceLogGraph.gp`, through a batch script `batch_traceLogFile.sh`. Note you may have to change some hardcoded values in the source codes, if you want to run this collection on your machine - see the Readme for more. The file `captures01-04.tar.gz` contains complete logs from four runs of `run-alsa-pa-tests.sh` (NB: +70MB expanded), with respective `traceLogGraph.gp` gnuplot scripts that allow for reconstruction of all .png images; while the `captures_0*/` directories contain only some .pngs (those that will be referenced here). Back to the problem: once I started working with `patest_duplex_wire.c`, I realised there are actually two ways to implement "full-duplex", understood simply as "playback and record at the same time", in PortAudio: * By using two separate `PaStream*`s, with two `Pa_OpenStream`s and `Pa_StartStream`s, started one right after the other - and using their separate "record" (capture) and "play" callbacks * By using a single `PaStream*`, which correspondingly has only one callback ("wire" callback), which handles memory copying in both playback and capture directions In Audacity, as far as I could see, there is only one PortAudio callback defined (audacity-1.3.13/src/AudioIO.cpp has audacityAudioCallback;), so Audacity basically has a single "wire" callback for full-duplex. Because of that, both Audacity and `patest_duplex_wire.c` (when using a single stream/"wire" callback), can enter this section of portaudio-v19/src/hostapi/alsa/pa_linux_alsa.c: if( !xrun ) { /* Get the number of available frames for the pcms that are marked ready. * @concern FullDuplex If only one direction is marked ready (from poll), the number of frames available for * the other direction is returned. Output is normally preferred over capture however, so capture frames may be * discarded to avoid overrun unless paNeverDropInput is specified. */ int captureReady = self->capture.pcm ? self->capture.ready : 0, playbackReady = self->playback.pcm ? self->playback.ready : 0; PA_ENSURE( PaAlsaStream_GetAvailableFrames( self, captureReady, playbackReady, framesAvail, &xrun ) ); if( self->capture.pcm && self->playback.pcm ) { if( !self->playback.ready && !self->neverDropInput ) { /* Drop input, a period's worth */ PA_MDEBUG(( "%s: full-duplex (not xrun): Drop input, a period's worth - fra:%lu \n", __FUNCTION__, *framesAvail )); //added assert( self->capture.ready ); PaAlsaStreamComponent_EndProcessing( &self->capture, PA_MIN( self->capture.framesPerBuffer, *framesAvail ), &xrun ); *framesAvail = 0; self->capture.ready = 0; } } else if( self->capture.pcm ) assert( self->capture.ready ); else assert( self->playback.ready ); } ... ... which is precisely the section that seems to be causing the full-duplex capture drops I experience. I think I tried `neverDropInput` with `patest_duplex_wire.c`, and it kept segfaulting - but in any case, I don't think Audacity (at least this version) has any UI to manipulate this parameter; so I'd rather code the ALSA driver properly, so PortAudio thinks it's getting a proper full-duplex stream from it. However, this is where I get stuck - because all I can see in the code snippet are boolean conditions. Thus I cannot see what the condition is to enter this segment - in terms of variables visible from the ALSA driver (like dpcm->pcm_buf_pos, or snd_pcm_capture_avail/_hw_avail(substream->runtime)). In other words, I'd like to know what sort of rule should the ALSA driver variables observe, so the above "drop input" section is never entered when using a single PortAudio stream/"wire" callback. This is the reason why I tried adding some extra debugging statements to PortAudio (see `audacity-portaudio.patch` in the online folder), to obtain logs through `ftrace` - in hope that I'd get accurate timestamps from both user- and kernel- space, against which variable values from logs could be plotted. In brief, some of the variables on the plots are: * frabCC/frabPC = framesAvail (Capture/Playback) in bytes from CallbackThreadFunc (pa_linux_alsa.c) * frabCW/frabPW = framesAvail (Capture/Playback) in bytes from PaAlsaStream_WaitForFrames (pa_linux_alsa.c) * frgbtc/frgbtp = framesGot (Capture/Playback) in bytes total (cumulative); CallbackThreadFunc (pa_linux_alsa.c) * pav/cav/phwav/chwav = snd_pcm_playback/capture_avail/_hw_avail(substream->runtime) in bytes (dummy-2.6.32-patest.c) * aptbC/aptbP/hptbC/hptbP = (Capture/Playback) substream->runtime->control->appl_ptr/hw_ptr in bytes (dummy-2.6.32-patest.c) [ When a single PortAudio "wire" callback is used, only frabCC, frabCW and frgbtc appear (there are no separate playback variants then) - however, the driver still has separate timer functions/tasklets for each direction (playback or capture). ] The original `dummy` driver has code for both Linux kernel system timers, and for high-resolution timers; the driver builds on my system automatically with hrtimers, which is what I'm interested in. Initially, I started by working with hrtimer functions set to a period of one jiffy: since HZ in my OS is defined as 250, one jiffy is 1/HZ = 1/250 = 4 ms - and then I calculated bytes per period as total bytes per second 44100*2*2 = 176400 divided by HZ, which gives 176400/250 = 705.6 ; and then I tried to increase dpcm->pcm_buf_pos with 705 each time the hrtimer tasklet was called (those captures have folder names ending with `_jif`). That results in a behavior like this: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_02_jif/tr… That is a capture with driver debug tracing (D), fixed bytes per period in driver (F), playback+record callbacks in `patest_duplex_wire.c` (pr), with request of 512 FRAMES_PER_BUFFER (512). What I find surprising here, is that driver's `chwav` (and `cav`) stay mostly constant, "resetting" themselves in just three "hits" of the timer/tasklet after 0.12 sec; while `pav` slowly raises (and `phwav` slowly falls) - until a point after 0.12 sec, where they "reset" themselves in just one "hit" of the timer function. Furthermore, `frabPC`, `frabCC` and `frabPW` also follow the `pav` line (indicating that some frames were left 'avail', after the requested 512 frames were 'got' in PA's `CallbackThreadFunc`). For a while I thought that this slow "raise" was the cause for the drop - but it isn't, as that particular capture doesn't display a drop (and it cannot either, because it doesn't use a single "wire" callback) Then, I thought that the reason for the `pav` rising slope could be, that I'm taking too much time processing in the driver callbacks; but that can't be it, because the capture above is obtained with the "snd-dummy" driver - which does pretty much nothing in its hrtimer/takslet function, but increasing the pcm_buf_pos counters (and memsetting a couple of bytes in the capture direction); I cannot really see that as a CPU hog?! Then I thought, maybe it's a rounding problem, since I should be increasing driver pointers by 705.6 bytes per timer tick, and I'm increasing for 705 bytes instead - but then, that should result with _less_ data available for playback; while `pav` rising implies that _more_ data than needed is delivered to the PortAudio callback. Then I did another test, trying to check the delta between the kernel timestamps for capture and playback driver tasklets separately; it turns out, even if the period is specified as 4000000e-9 nsecs (4 ms) - the deltas still gr avitate around 3.99x ms; I tried also calculating cumulative bytes per second (based on deltas and bytes per period) - and for different captures, they typically tended to stabilize to some +80 Bps to some +100 Bps above the required (176400 Bps). So, if the hrtimer indeed runs a bit faster, then indeed we may feed more data than needed from the driver to userspace (PA) - but then again, the slope of `pav` is at least 668 bytes for 12.6 ms, which is >> 80 Bps; so I'm still not sure where exactly does that slope come from. Then, I noticed that the original `dummy.c` driver doesn't increase its counters by a fixed amount of bytes - but instead, by what I'd call "adaptive" amount of bytes: delta to a base time is taken; and it can be used to calculate the expected (wrapped) buffer position depending on the requested rate (although `dummy.c` does this in frames, and does it in the `_pointer` callback). So I did the same in the hrtimer tasklet - and, this seemingly does very little in respect to the `pav` slope: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_02_jif/tr… ... which is still present, with nearly the same slope as before. Unfortunately, passing "adaptive" amount of bytes/frames around is not the solution to the drops either - because here is a capture of a "drop input" (`frabDI`), with single "wire" callback requesting 512 FRAMES_PER_BUFFER - happening at approx 60 ms, even before the first correction happens (after 0.12 sec): http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_02_jif/tr… The only thing I can notice here, is that when the drop happens, `phwav` briefly "crosses" below the value of `pav` (which otherwise never happens; the same goes for cav/chwav) - but I'm not sure if this is the cause, or the effect, of the drop. Also, it can be noted that for a PA single "wire" callback, the driver's playback and capture hrtimer tasklets hit very close to each other, and the corresponding variables are also very close - at the above resolution, the trace points for both nearly overlap. Another thing I suspected was FRAMES_PER_BUFFER in the PortAudio program, so the test script switches the setting in `patest_duplex_wire.c` between 512 and 0 (which is `paFramesPerBufferUnspecified`). Note that when using 0 frames per buffer: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_02_jif/tr… ... and playback+record callbacks, the is no more long slope of `pav`/`phwav`, and it self-corrects more quickly; while `cav`/`chwav` stay nearly constant - however, also note the condition where `phwav` briefly "crosses" below the value of `pav`, appears here again (but no "drop input" is detected, because we're not using a single "wire" callback). However, not even FRAMES_PER_BUFFER=0 seems to help; as it doesn't prevent the "drop input" for a single "wire" callback, in the capture below: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_02_jif/tr… Since I was also suspecting the debug traces from the driver for having an influence on the drop, the debug traces are disabled for the driver (x) in that capture - and obviously, this is not the sole cause of the drop either. Note that the original `dummy.c` kernel module calculates the period in nanoseconds, so it matches the pcm period size; in that case, one does not have to calculate if the current buffer position went over a period size - but can, instead, issue `snd_pcm_period_elapsed` directly at each call of the hrtimer tasklet. In this case, say for a pcm_period_size: 512 bytes, we might get a hrtimer period of 2902495 ns = 2.9 ms. This is controlled by the USE_JIFFY_PERIOD variable in the `dummy-2.6.32-patest.c` driver code, though not controlled by the `run-alsa-pa-tests.sh` - thus it has to be enabled manually; captures which USE_JIFFY_PERIOD have `_jif` as extension to the folder name, those that match to period size have `_psz` as extension. Again, even this doesn't prevent occurrence of the full-duplex "drop input"s - in all, these of the provided captures in the .tar.gz file feature drops: captures_01_psz/trace_patest__07_DA_w_0_drop.csv captures_03_psz/trace_patest__16_DF_w_512_drop.csv captures_01_psz/trace_patest__15_DF_w_0_drop.csv captures_04_jif/trace_patest__03_xA_w_0_drop.csv captures_02_jif/trace_patest__03_xA_w_0_drop.csv captures_04_jif/trace_patest__07_DA_w_0_drop.csv captures_02_jif/trace_patest__07_DA_w_0_drop.csv captures_04_jif/trace_patest__11_xF_w_0_drop.csv captures_02_jif/trace_patest__08_DA_w_512_drop.csv captures_04_jif/trace_patest__15_DF_w_0_drop.csv captures_02_jif/trace_patest__11_xF_w_0_drop.csv captures_04_jif/trace_patest__16_DF_w_512_drop.csv captures_03_psz/trace_patest__07_DA_w_0_drop.csv As the filenames indicate: * Regardless of whether the driver tunes its timer period to a jiffy (_jif) or to the period size (_psz); * Regardless of whether I use trace debug statement in the driver (D), or not (x); * Regardless of whether the driver "returns" adaptive (A) or fixed (F) amount of bytes; * Regardless of whether we use paFramesPerBufferUnspecified=0 or 512 frames per buffer in PortAudio; ... as long as there is a single stream/"wire callback" used from PortAudio, a "drop input" is liable to happen when running `patest_duplex_wire.c` - and this, even with a virtual `dummy` driver, which has no communication with real hardware at all, and pretty much does nothing: its role of "reading"/"writing" or "returning" bytes is merely simulated by increasing the stream buffer position counters! The mind boggles - especially since, the way I see it, one should do nothing else but properly increase these buffer counters from the driver, in order to have PortAudio "believe" a "proper" full-duplex transfer is occurring?!... and yet, what this "proper" rule (of increasing driver buffer counters) should be, completely evades me. Here I'd also like to note a few other observations - first, note that when scaling the driver hrtimer period to the period size (_psz), the `pav`/`phwav` became nearly constant, in comparison to the slopes still present upon FRAMES_PER_BUFFER=0 (which confuses me, since both: 705 bytes @ 4ms; and 2.9 bytes @ 2.9ms; should both be nearly as close to a rate of 44100 frames = 176400 bytes per second): http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_03_psz/tr… ... but then, as the filenames above indicate, drops happen even then - although somewhat less often, and only when debug tracing in the driver is enabled. Could it be, that debug tracing directly from the driver can have such an effect on performance, so as to influence appearance of full-duplex "drop inputs" - even if precautions were taken, such as using `ftrace`s `trace_printk` (instead of the usual syslog `printk`), and piping the debug output directly to RAM (by using `/dev/shm`)? On a scale of a complete capture, I'd usually expect all cumulative pointers to essentially follow a single line; and usually when a capture goes well, that is how it appears - like on the image below: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_04_jif/tr… While it may be a bit difficult to follow all variables (recreate the plot in the interactive `wxt` terminal in `gnuplot`, to be able to turn off individual function plots), they all generally follow each other in the same general slope; also note that the capture ends at about 2 seconds (as programmed in `patest_duplex_wire.c`) - but playback lingers on for half a second more. However, a full-duplex drop can seemingly happen even on a line that otherwise appears straight - like on the plot below (although the reason for that could be that the drop happens early in that capture): http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_02_jif/tr… ... but usually, a "drop input" resets (at least) the driver's playback `hw_ptr` (and others), like on the capture below: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_04_jif/tr… Note that when using playback/record PA callbacks (which cannot detect a "drop input"), may have similar resets of playback `hw_ptr`: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_03_psz/tr… ... but also the capture variables may reset: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_03_psz/tr… The above capture also shows `frgbtc` starting to deviate from `frgbtp`, when these "resets" occur - but piecewise, `frgbtc` still seems to keep slope parallel to `frgbtp` (and the rest of the cumulative position variables). However, this seems to be a case of proper XRUN, as the respective log for that capture includes "PaAlsaStream_HandleXrun: restarting Alsa to recover from XRUN" (while the full-duplex "drop input" is specifically _not_ an xrun). Note also, that sometimes I get some variables to be completely non-linear: http://sdaaubckp.sourceforge.net/post/alsa-patest-duplex/captures_01_psz/tr… ... however, note also that `captures_01_psz` was taken with an older version of the scripts, where I may have possibly failed to increase `ftrace` buffering - so I think those interruptions in the capture data are actually an artifact of a tracing problem. Otherwise, is it possible that ALSA could "pause" the playback/capture hrtimer tasklets for a while? Interestingly, when I tried using `patest_duplex_wire.c` with my onboard HDA Intel soundcard (naturally, without any kernel driver variable tracing, only the PA part) - a lot of the times it ended up producing similar non-linear (and diverging) cumulative variable traces - however, it never triggered a full-duplex "drop input". Finally, I suspect that these settings in the `dummy-2.6.32-patest.c` module, may also play a role in the behavior that results with full-duplex drops (they are otherwise kept the same as in the original `dummy` module): static struct snd_pcm_hardware dummy_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID), ... .buffer_bytes_max = MAX_BUFFER_SIZE, .period_bytes_min = 64, .period_bytes_max = MAX_PERIOD_SIZE, //(64*1024) .periods_min = USE_PERIODS_MIN, //1 .periods_max = USE_PERIODS_MAX, //1024 .fifo_size = 0, ... ... and my questions in respect to them, would be this (considering the capture data should be available through address substream->runtime->dma_area in the driver): * Given that this is a virtual driver, there is no hardware with buffers whose status (how filled they are) could generate (hardware) interrupts; instead we "raise" hrtimer functions (softirq priority) directly from the kernel module - which further schedule a tasklet (with even lower priority), and where we have the opportunity to control the repetition period of the timer functions (and thus the expected transferred bytes per period, for a given audio rate). In this context, what is the meaning of `period_bytes_min` and `periods_min`? Can they be arbitrarily set - or should one set them in relation to the expected bytes per period achieved through the timer functions? * Given that `dummy` is a virtual driver, and doesn't communicate with hardware - there is no actual DMA operation performed through it, right? * If there is no actual DMA - is there actual MMAP? In other words, does the MMAP in SNDRV_PCM_INFO_MMAP refer to: memory mapping of a hardware card DMA bus address, to substream->runtime->dma_area; or to: memory mapping of kernel-space substream->runtime->dma_area, to whatever address user-space (e.g. snd_pcm_read()) uses to access it? * If there is no actual DMA - should SNDRV_PCM_INFO_MMAP_VALID be then kept? (original `dummy` driver keeps it) What would be the meaning of SNDRV_PCM_INFO_MMAP_VALID in this context (of a virtual driver with hrtimer functions/tasklets?) * What would be the meaning of SNDRV_PCM_INFO_JOINT_DUPLEX in this context? (I've seen [http://mailman.alsa-project.org/pipermail/alsa-devel/2007-December/004801.h… "[alsa-devel] How do I use SNDRV_PCM_INFO_JOINT_DUPLEX?"], but I can't really relate it to this context) Could usage of SNDRV_PCM_INFO_JOINT_DUPLEX help avoid the PortAudio full-duplex drops? And to summarize my general questions: * Given the 'telecom' definition of full-duplex as "a point-to-point system composed of two connected parties or devices that can communicate with one another in both directions, simultaneously" (wiki), is there a stricter/more specific definition of "full-duplex": ** in terms of digital audio generally? ** specifically in PortAudio (I guess yes: "when a single callback is used for both playback and capture", but there's probably more in terms of buffer positions), and in ALSA (I guess no)? * Given that none of the approaches used in the test (debug driver traces on vs. off; driver returning fixed vs. adaptive number of frames/bytes per period; driver tuning the period to pcm_period_size vs. to a jiffy; PA using fixed vs. using unspecified amount of frames per period) have explicitly prevented the appearance of single "wire" callback (full-duplex) drops; and given that the driver is a virtual one (merely updating buffer positions, which shouldn't be a CPU hog) - what could possibly be the reason for the full-duplex drops? * Is it possible at all, to use the architecture of the `dummy` driver (hrtimer + tasklet just updating buffer positions) to code an ALSA driver, that will not generate full-duplex drops when used from PortAudio with single "wire" callback? ** If so, is there a metric/conditions in terms of ALSA driver variables, that should be followed to ensure proper full-duplex operation (under PortAudio with single callback)? * Is it possible that the driver is OK - and instead I have a problem with coding of the single "wire" callback in PortAudio? (even though most of it is copied from examples) Well, that was certainly a mouthful - but I hope at least I managed to provide a reproducible example, if not a coherent explanation of the problem... Thanks in advance for any answers/pointers, Cheers!

6 15

[alsa-devel] [PATCHv4][ 1/4] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 21 Oct '13

21 Oct '13

Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Sascha Hauer <kernel(a)pengutronix.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: Eric Bénard <eric(a)eukrea.com> Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- ChangeLog v2->v4: - a multiline string was avoided. --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 23 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 125 +++++++++++++++++--- 3 files changed, 133 insertions(+), 20 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..8791037 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,23 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,eukrea-tlv320" +- model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- audio-codec : The phandle of the tlv320aic23 audio codec. +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,eukrea-tlv320"; + model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,audio-codec = <&tlv320aic23>; + mux-int-port = <2>; + mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f..9c3cd64 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -161,12 +161,15 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || OF depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9a4a0ca..773863d 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -17,6 +17,8 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,24 +28,33 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 +struct eukrea_tlv320 { + char of; +}; + static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct eukrea_tlv320 *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; + /* fsl_ssi lacks the set_fmt ops. */ + if (!priv->of) { + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } } ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | @@ -60,15 +71,18 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, pr_err("%s: failed setting codec sysclk\n", __func__); return ret; } - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); - ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, - SND_SOC_CLOCK_IN); - if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); - return ret; - } + /* fsl_ssi lacks the set_sysclk ops */ + if (!priv->of) { + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + } return 0; } @@ -80,7 +94,6 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, @@ -97,8 +110,56 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct platform_device *ssi_pdev; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np; + struct eukrea_tlv320 *priv; + + priv = devm_kzalloc(&pdev->dev, + sizeof(struct eukrea_tlv320), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (np) { + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, + "ssi-controller missing or invalid.\n"); + ret = -ENODEV; + goto err; + } + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + eukrea_tlv320_dai.platform_of_node = ssi_np; + priv->of = 1; + } else { + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + } - if (machine_is_eukrea_cpuimx27()) { + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -115,8 +176,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -130,14 +195,29 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,eukrea-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } eukrea_tlv320.dev = &pdev->dev; + snd_soc_card_set_drvdata(&eukrea_tlv320, priv); ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -149,10 +229,17 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,eukrea-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, .remove = eukrea_tlv320_remove,}; -- 1.7.9.5

1 4

[alsa-devel] [PATCH] ASoC: dont call dapm_sync while reporting jack always
by Vinod Koul 21 Oct '13

21 Oct '13

While reporting the jack status snd_soc_jack_report() invokes snd_soc_dapm_sync() always. This should be required when we have pins associated with jack and reporting enables or disables these. So add a check for this case Change-Id: I671cd444d1b6610dcb83b28f7339477556373e5c Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-jack.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index c9f7953..9fd4ed8 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -65,6 +65,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) struct snd_soc_codec *codec; struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; + unsigned int sync = 0; int enable; trace_snd_soc_jack_report(jack, mask, status); @@ -92,12 +93,16 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_soc_dapm_enable_pin(dapm, pin->pin); else snd_soc_dapm_disable_pin(dapm, pin->pin); + + /* we need to sync for this case only */ + sync = 1; } /* Report before the DAPM sync to help users updating micbias status */ blocking_notifier_call_chain(&jack->notifier, jack->status, jack); - snd_soc_dapm_sync(dapm); + if (sync) + snd_soc_dapm_sync(dapm); snd_jack_report(jack->jack, jack->status); -- 1.7.0.4

1 0

[alsa-devel] mxc-spdif delay: miss 1-2s at the beginning of the song
by stéphane cerveau 21 Oct '13

21 Oct '13

Dear all, I'm facing a delay during startup missing 1-2s of the song when i'm using mxc spdif with aplay. I guess its a problem with spdif synchronization. If i put a msleep(1000) at the end of the function mxc-spdif.c:mxc_spdif_playback_start, the playback is ok. Is there anyone who already faced this issue? BR. Stéphane

2 3

[alsa-devel] alsa project installation not working
by Ranjith 21 Oct '13

21 Oct '13

Hi, I am using Ubuntu 12.10. I am not able to install alsa project. When I'm giving ./configure it is throwing error as configure command not found. Thanks & Regards, Ranjith

2 1

[alsa-devel] [PATCH] ALSA: hda - add codec ID for Valleyview2 display codec
by mengdong.lin＠intel.com 21 Oct '13

21 Oct '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds codec ID (0x80862882) and module alias for Valleyview2 display codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7ea0245..e332db8 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2671,6 +2671,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -2725,6 +2726,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:80862807"); MODULE_ALIAS("snd-hda-codec-id:80862880"); +MODULE_ALIAS("snd-hda-codec-id:80862882"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_LICENSE("GPL"); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: cleanup the external clock probe
by Jean-Francois Moine 21 Oct '13

21 Oct '13

This patch checks the return value of the clk_prepare_enable of the external clock and removes the test about a same internal and external clock which would quite never occur and won't work in most cases it would occur. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- This patch applies against broonie/sound for-next. --- sound/soc/kirkwood/kirkwood-i2s.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c.old b/sound/soc/kirkwood/kirkwood-i2s.c.pat2 index d0504a2..c2433ea 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c.old +++ b/sound/soc/kirkwood/kirkwood-i2s.c.pat2 @@ -500,14 +500,11 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) return -EPROBE_DEFER; } else { - if (priv->extclk == priv->clk) { - devm_clk_put(&pdev->dev, priv->extclk); - priv->extclk = ERR_PTR(-EINVAL); - } else { - dev_info(&pdev->dev, "found external clock\n"); - clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; - } + dev_info(&pdev->dev, "found external clock\n"); + ret = clk_prepare_enable(priv->extclk); + if (err < 0) + return err; + soc_dai = kirkwood_i2s_dai_extclk; } /* Some sensible defaults - this reflects the powerup values */ -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 4

[alsa-devel] Fw: alsa project installation not working
by Ranjith 21 Oct '13

21 Oct '13

-------------------------------------------------- From: "Ranjith" <ranjithvs(a)iwavesystems.com> Sent: Monday, October 21, 2013 2:32 PM To: <alsa-devel(a)alsa-project.org> Subject: [alsa-devel] alsa project installation not working > Hi, > > I am using Ubuntu 12.10. I am not able to install alsa project. When I'm > giving ./configure it is throwing error as configure command not found. > > > Thanks & Regards, > > Ranjith > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] Audio Miniconference 2013 October
by Mark Brown 21 Oct '13

21 Oct '13

The room is Harris 2 upstairs on the 1st floor.

1 0

[alsa-devel] [PATCH v3 2/2] ASoC: kirkwood: fix loss of external clock at probe time
by Jean-Francois Moine 21 Oct '13

21 Oct '13

At probe time, when the clock driver is not yet initialized, the external clock of the kirkwood sound device will not be usable. This patch fixes this problem defering the device probe. It also removes the test about same internal and external clocks which can never occur. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v3: remove the test of same internal and external clocks The associated patch: [PATCH v3 1/2] ASoC: kirkwood: clk: probe defer when clock not yet ready is unchanged and not resent --- sound/soc/kirkwood/kirkwood-i2s.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 8ac89f5..2bbbab5 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -496,15 +496,13 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return err; priv->extclk = devm_clk_get(&pdev->dev, "extclk"); - if (!IS_ERR(priv->extclk)) { - if (priv->extclk == priv->clk) { - devm_clk_put(&pdev->dev, priv->extclk); - priv->extclk = ERR_PTR(-EINVAL); - } else { - dev_info(&pdev->dev, "found external clock\n"); - clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; - } + if (IS_ERR(priv->extclk)) { + if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { + dev_info(&pdev->dev, "found external clock\n"); + clk_prepare_enable(priv->extclk); + soc_dai = kirkwood_i2s_dai_extclk; } /* Some sensible defaults - this reflects the powerup values */ -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 6

[alsa-devel] [PATCH v2 3/3] ASoC: cs42l73: Add Device Tree support for CS42L73
by Brian Austin 21 Oct '13

21 Oct '13

This patch adds support for device tree for the CS42L73 CODEC Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- .../devicetree/bindings/sound/cs42l73.txt | 22 ++++++++++++++++ sound/soc/codecs/cs42l73.c | 29 ++++++++++++++++++++-- 2 files changed, 49 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs42l73.txt diff --git a/Documentation/devicetree/bindings/sound/cs42l73.txt b/Documentation/devicetree/bindings/sound/cs42l73.txt new file mode 100644 index 0000000..80ae910 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l73.txt @@ -0,0 +1,22 @@ +CS42L73 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l73" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset_gpio : a GPIO spec for the reset pin. + - chgfreq : Charge Pump Frequency values 0x00-0x0F + + +Example: + +codec: cs42l73@4a { + compatible = "cirrus,cs42l73"; + reg = <0x4a>; + reset_gpio = <&gpio 10 0>; + chgfreq = <0x05>; +}; \ No newline at end of file diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 89efc3c..549d5d6 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -17,7 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -#include <linux/gpio.h> +#include <linux/of_gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -1416,6 +1416,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, int ret; unsigned int devid = 0; unsigned int reg; + u32 val32; cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private), GFP_KERNEL); @@ -1431,8 +1432,25 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, return ret; } - if (pdata) + if (pdata) { cs42l73->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l73_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + if (of_property_read_u32(i2c_client->dev.of_node, + "chgfreq", &val32) >= 0) + pdata->chgfreq = val32; + } + pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node, + "reset-gpio", 0); + cs42l73->pdata = *pdata; + } i2c_set_clientdata(i2c_client, cs42l73); @@ -1493,6 +1511,12 @@ static int cs42l73_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id cs42l73_of_match[] = { + { .compatible = "cirrus,cs42l73", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs42l73_of_match); + static const struct i2c_device_id cs42l73_id[] = { {"cs42l73", 0}, {} @@ -1504,6 +1528,7 @@ static struct i2c_driver cs42l73_i2c_driver = { .driver = { .name = "cs42l73", .owner = THIS_MODULE, + .of_match_table = cs42l73_of_match, }, .id_table = cs42l73_id, .probe = cs42l73_i2c_probe, -- 1.8.4.rc3

3 3

[alsa-devel] Audio Miniconference 2013 schedule II - Edinburgh 21st October
by Mark Brown 20 Oct '13

20 Oct '13

Another go at the schedule for the mini-summit based on the feedback last time. Same three major topics: - Compressed audio review: tinycompress, integration into userspace, driver upstreaming. - UCM and other use case management status & review (Tanu). - Effects offload to DSPs with dynamically instantiatable algorithms, and general dynamic DSP firmware handling including DSP resource management (Pierre). Plus a few things that only one or two people brought up: - Process stuff (the list, bug tracking and so on). - Some relatively specific hardware support topics: - HDMI hotplug handling throughout the stack - Mic status LED handling throughout the stack. - Code cleanups: - Moving away from the custom ALSA device registration infrastructure. - General code modernisation. - Performance, especially DAPM (Patrick) - General routing control in userspace - UIs, automation (Daniel). Hopefully this caught everything that was missed last time. It'd be good to have volunteers to lead the discussions in each area, I've marked those we got already. As previously mentioned it'd be good if anyone planning to attend could fill in the form at: https://docs.google.com/forms/d/105uav9tkwwkDU4P1T3IOWjfsa3kgtDbFAJOlHgBQR5… to help with planning & coordination.

6 10

[alsa-devel] [PATCH] ASoC: si476x: Fix locking of core
by Mark Brown 20 Oct '13

20 Oct '13

From: Mark Brown <broonie(a)linaro.org> The conversion of the si476x to regmap removed locking of the core during register updates, allowing things like power state changes for the MFD to happen during a register update. Avoid this by taking the core lock in the DAI operations (which are the only things that do register updates) as we used to do in the open coded register I/O functions. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/si476x.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 03645ce..52e7cb0 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -73,6 +73,7 @@ static const struct snd_soc_dapm_route si476x_dapm_routes[] = { static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { + struct si476x_core *core = i2c_mfd_cell_to_core(codec_dai->dev); int err; u16 format = 0; @@ -136,9 +137,14 @@ static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } + si476x_core_lock(core); + err = snd_soc_update_bits(codec_dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK, format); + + si476x_core_unlock(core); + if (err < 0) { dev_err(codec_dai->codec->dev, "Failed to set output format\n"); return err; @@ -151,6 +157,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct si476x_core *core = i2c_mfd_cell_to_core(dai->dev); int rate, width, err; rate = params_rate(params); @@ -176,11 +183,13 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + si476x_core_lock(core); + err = snd_soc_write(dai->codec, SI476X_DIGITAL_IO_OUTPUT_SAMPLE_RATE, rate); if (err < 0) { dev_err(dai->codec->dev, "Failed to set sample rate\n"); - return err; + goto out; } err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, @@ -189,10 +198,13 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, (width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)); if (err < 0) { dev_err(dai->codec->dev, "Failed to set output width\n"); - return err; + goto out; } - return 0; +out: + si476x_core_unlock(core); + + return err; } static int si476x_codec_probe(struct snd_soc_codec *codec) -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH v2] ASoC: kirkwood: add S/PDIF support
by Jean-Francois Moine 20 Oct '13

20 Oct '13

This patch adds S/PDIF input/output for mvebu DT boards. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2: declare 2 DAI's (Mark Brown) --- sound/soc/kirkwood/kirkwood-i2s.c | 95 ++++++++++++++++++---- 1 file changed, 78 insertions(+), 17 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 2bbbab5..20c56c8 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -240,6 +244,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } + if (dai->name[0] == 'i') + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ @@ -258,7 +267,8 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +282,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +313,13 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + if (dai->name[0] == 'i') + ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ + + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -361,9 +379,8 @@ static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int kirkwood_i2s_probe(struct snd_soc_dai *dai) +static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) { - struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); unsigned long value; unsigned int reg_data; @@ -404,9 +421,27 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; - -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { + { + .name = "i2s", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", .playback = { .channels_min = 1, .channels_max = 2, @@ -422,10 +457,32 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { + { + .name = "i2s", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", .playback = { .channels_min = 1, .channels_max = 2, @@ -443,6 +500,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; static const struct snd_soc_component_driver kirkwood_i2s_component = { @@ -452,7 +510,7 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; + struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -519,7 +577,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); + soc_dai, 2); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto err_component; @@ -530,6 +588,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); goto err_platform; } + + kirkwood_i2s_init(priv); + return 0; err_platform: snd_soc_unregister_component(&pdev->dev); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 2

[alsa-devel] [PATCH RESEND] ASoC: kirkwood: prefer external clock over internal clock
by Jean-Francois Moine 20 Oct '13

20 Oct '13

When there is an external clock, always use this one. This prevents the two Dove audio devices to use the same DCO clock at different rates. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- This patch was initialy proposed by Russell, but with an erroneous 'else if'. --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d..8ac89f5 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -103,7 +103,7 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, { uint32_t clks_ctrl; - if (rate == 44100 || rate == 48000 || rate == 96000) { + if (IS_ERR(priv->extclk)) { /* use internal dco for the supported rates * defined in kirkwood_i2s_dai */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH 0/2] ASoC: rcar: remove un-needed flags
by Kuninori Morimoto 20 Oct '13

20 Oct '13

Hi Mark These 2 patches remove un-needed flags. These are prepare patches for DT support Kuninori Morimoto (2): ASoC: rcar: add rsnd_scu_hpbif_is_enable() ASoC: rcar: remove RSND_SSI_CLK_FROM_ADG include/sound/rcar_snd.h | 1 - sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/scu.c | 12 ++++++++--- sound/soc/sh/rcar/ssi.c | 52 +++++++++++++++++++--------------------------- 4 files changed, 31 insertions(+), 35 deletions(-) Best regards --- Kuninori Morimoto

2 3

[alsa-devel] [PATCH] ALSA: Add MAINTAINERS entry for dmaengine helpers
by Mark Brown 19 Oct '13

19 Oct '13

From: Mark Brown <broonie(a)linaro.org> Help ensure that Lars-Peter gets CCed on dmaengine related patches by adding a MAINTAINERS entry for the helpers. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 34ca01a..3a7eaaa 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7877,6 +7877,13 @@ F: Documentation/sound/alsa/soc/ F: sound/soc/ F: include/sound/soc* +SOUND - DMAENGINE HELPERS +M: Lars-Peter Clausen <lars(a)metafoo.de> +S: Supported +F: include/sound/dmaengine_pcm.h +F: sound/core/pcm_dmaengine.c +F: sound/soc/soc-generic-dmaengine-pcm.c + SPARC + UltraSPARC (sparc/sparc64) M: "David S. Miller" <davem(a)davemloft.net> L: sparclinux(a)vger.kernel.org -- 1.8.4.rc3

3 2

[alsa-devel] ASoC updates for v3.12
by Mark Brown 19 Oct '13

19 Oct '13

The following changes since commit 61e6cfa80de5760bbe406f4e815b7739205754d2: Linux 3.12-rc5 (2013-10-13 15:41:28 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-rc5 for you to fetch changes up to 40f8989695660dee984338861ca9900b8e0b5183: Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-linus (2013-10-18 14:17:14 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.12 A number of small, fairly unexciting, fixes for drivers - nothing stand out. ---------------------------------------------------------------- Axel Lin (3): ASoC: pcm1681: Fix max_register setting ASoC: pcm1681: Fix max_register setting ASoC: pcm1792a: Fix max_register setting Guenter Roeck (1): ASoC: fsl_ssi: Fix irq_of_parse_and_map() return value check Jarkko Nikula (1): ASoC: omap: Fix incorrect ARM dependency Kuninori Morimoto (1): ASoC: rcar: fixup generation checker Mark Brown (6): Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/omap' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm1681' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm1792a' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-linus Peter Ujfalusi (1): ASoC: tlv320aic3x: Connect 'Left Line1R Mux' and 'Right Line1L Mux' Philippe Rétornaz (2): ASoC: fsl: Fix sound on mx31moboard ASoC: fsl: imx-ssi: fix probe on imx31 include/sound/rcar_snd.h | 1 + sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/pcm1792a.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 4 ++++ sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-ssi.c | 23 ++++++++++++----------- sound/soc/fsl/imx-ssi.h | 2 ++ sound/soc/omap/Kconfig | 4 ++-- sound/soc/sh/rcar/rsnd.h | 4 ++-- 10 files changed, 27 insertions(+), 19 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: imx: Open code dma channel request
by Mark Brown 19 Oct '13

19 Oct '13

From: Mark Brown <broonie(a)linaro.org> Currently the imx DMA code is one of the few users relying on the fact that the compat code uses the dma_data as the filter data for non-DT channel requests. Since the rest of the core expects this to be a struct snd_dmaengine_dai_data this isn't terribly helpful this will be changed to use the already existing filter data so avoid breaking imx by open coding the current behaviour. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/imx-pcm-dma.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 4dc1296..318f1fc 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -35,6 +35,17 @@ static bool filter(struct dma_chan *chan, void *param) return true; } +static struct dma_chan *imx_compat_request_channel( + struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) +{ + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + return snd_dmaengine_pcm_request_channel(filter, dma_data); +} + static const struct snd_pcm_hardware imx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -58,6 +69,7 @@ static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { .pcm_hardware = &imx_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, .compat_filter_fn = filter, + .compat_request_channel = imx_compat_request_channel, .prealloc_buffer_size = IMX_SSI_DMABUF_SIZE, }; -- 1.8.4.rc3

2 2

[alsa-devel] [PATCH] ASoC: dmaengine: Use filter_data rather than dma_data for compat requests
by Mark Brown 19 Oct '13

19 Oct '13

From: Mark Brown <broonie(a)linaro.org> When using the legacy filter function channel requests we currently pass the audio specific struct snd_dmaengine_dai_dma_data which isn't likely to be helpful for actual filtering. Since there's already a field in the structure called filter_data clearly intended for use here convert the driver to use that. All existing users of plain filter functions have been converted to use an explict compat function to override this behaviour. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 99f9495..2eea2e0 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -168,6 +168,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( struct snd_pcm_substream *substream) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; @@ -176,7 +179,7 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( return pcm->config->compat_request_channel(rtd, substream); return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); + dma_data->filter_data); } static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) -- 1.8.4.rc3

2 2

Re: [alsa-devel] [Alsa-user] hdmi sound rerouted?
by Dagg Stompler 19 Oct '13

19 Oct '13

> ----- Original Message ----- > From: Takashi Iwai > Sent: 10/08/13 10:49 AM > To: Deucher, Alexander > Subject: Re: [alsa-devel] [Alsa-user] hdmi sound rerouted? > > At Mon, 7 Oct 2013 13:36:24 +0000, > Deucher, Alexander wrote: > > > > > -----Original Message----- > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > Sent: Monday, September 30, 2013 5:18 AM > > > To: Deucher, Alexander > > > Cc: Clemens Ladisch; Dagg Stompler; alsa-user(a)lists.sourceforge.net; alsa- > > > devel(a)alsa-project.org > > > Subject: Re: [alsa-devel] [Alsa-user] hdmi sound rerouted? > > > > > > At Mon, 30 Sep 2013 02:23:22 +0000, > > > Deucher, Alexander wrote: > > > > > > > > > -----Original Message----- > > > > > From: Clemens Ladisch [mailto:cladisch@googlemail.com] > > > > > Sent: Saturday, September 28, 2013 6:33 PM > > > > > To: Dagg Stompler > > > > > Cc: alsa-user(a)lists.sourceforge.net; Deucher, Alexander; alsa- > > > devel@alsa- > > > > > project.org > > > > > Subject: Re: [Alsa-user] [alsa-devel] hdmi sound rerouted? > > > > > > > > > > Dagg Stompler wrote: > > > > > >> daggs(a)gmx.com wrote: > > > > > >>> card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > > > > > >>> card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] > > > > > >>> card 2: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] > > > > > >> > > > > > >> The second one is called "Generic" because your kernel does not (yet) > > > > > >> know the hardware. > > > > > >> > > > > > >>> 01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Cedar > > > > > HDMI Audio [Radeon HD 5400/6300 Series] > > > > > >>> 06:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] RV635 > > > > > HDMI Audio [Radeon HD 3600 Series] > > > > > >> > > > > > >> These are the HDMI outputs of *two* graphics cards. The first one's > > > > > >> model is so new that the driver doesn't know it. > > > > > > > > > > > > the 3600 doesn't have hdmi socket so I think it is irrelevant for my issue. > > > > > > > > > > The RV635 chip does support HDMI outputs (even if none are actually > > > > > connected), so the corresponding PCI device shows up here. > > > > > > > > > > >>> If you update the kernel, you will get two devices named "HDA ATI > > > > > HDMI". > > > > > > > > > > > > upgrade to where? > > > > > > > > > > Oops. Even the latest driver lacks the PCI IDs of many HDMI HDA devices > > > > > (aa50/aa58/aa60/aa68/aa80/aa88/aa90/aa98: > > > > > Cypress/Juniper/Redwood/Cedar/ > > > > > Cayman+Antilles/Barts/Turks/Caicos). > > > > > > > > > > Alex, are these devices that require disabling snooping? > > > > > > > > Sorry what does snooping mean in this context? > > > > > > Most of HD-audio controller have a snoop control, but the recent AMD > > > HDMI audio controller don't have the feature, thus the driver must use > > > the non-cached pages. At least, PCI ID 1002:aaa0, aaa8 and aab0 are > > > known to require such. > > > > > > Now the question is whether all new chips above (aa50,...) require the > > > same workaround, too. Could you check this or forward the inquiry to > > > someone in AMD? > > > > According to our internal teams, disabling snooping is only required on the audio parts included in Evergreen and Northern Islands GPUs. Older chips like R7xx (aa48) and newer chips like Southern Islands (aaa0) are able to use snooping. I'm not real familiar with the audio portions of the GPUs so if you need further clarification, let me know. > > Thanks for the information. > > Though, it's strange that the information doesn't match with what I've > heard (and also with what I've tested with a laptop). I've heard that > it's Southern Island HDMI that needs the snoop-disablement. > > Could you ask again whether it's really Evergreen / Nothern Islands? > > > thanks, > > Takashi > > > > > Alex > > > > > > > > FWIW, such a chip needs to set AZX_DRIVER_ATIHDMI_NS bits in > > > driver_data of pci entries in sound/pci/hda/hda_intel.c. > > > > > > > > > thanks, > > > > > > Takashi > > > > any updates on the matter guys?

2 1

[alsa-devel] make randconfig issue with v3.12-rc5-126-g9219cec and config #4:
by Toralf Förster 19 Oct '13

19 Oct '13

Hello, the build (.config attached) failed, make gives : ... CC drivers/extcon/extcon-adc-jack.o CC drivers/extcon/extcon-max77693.o CC drivers/firewire/core-cdev.o CC sound/soc/fsl/fsl_ssi.o CC sound/pci/asihpi/hpidebug.o CC drivers/extcon/extcon-max8997.o sound/soc/fsl/fsl_ssi.c: In function ‘fsl_ssi_probe’: sound/soc/fsl/fsl_ssi.c:939:26: error: ‘NO_IRQ’ undeclared (first use in this function) sound/soc/fsl/fsl_ssi.c:939:26: note: each undeclared identifier is reported only once for each function it appears in make[3]: *** [sound/soc/fsl/fsl_ssi.o] Error 1 make[2]: *** [sound/soc/fsl] Error 2 make[1]: *** [sound/soc] Error 2 -- CC drivers/hid/hid-wiimote-debug.o LD drivers/gpu/drm/r128/built-in.o CC drivers/hid/hid-debug.o CC drivers/gpu/drm/tdfx/tdfx_drv.o LD drivers/hsi/built-in.o CC drivers/hwmon/hwmon.o LD drivers/gpu/drm/tdfx/tdfx.o LD drivers/gpu/drm/tdfx/built-in.o CC drivers/hid/hid-core.o CC drivers/gpu/drm/ttm/ttm_agp_backend.o make: *** [sound] Error 2 drivers/char/hw_random/via-rng.c:224:26: warning: ‘via_rng_cpu_id’ defined but not used [-Wunused-variable] Here's the config: # # Automatically generated file; DO NOT EDIT. # Linux/x86 3.12.0-rc5 Kernel Configuration # # CONFIG_64BIT is not set CONFIG_X86_32=y CONFIG_X86=y CONFIG_INSTRUCTION_DECODER=y CONFIG_OUTPUT_FORMAT="elf32-i386" CONFIG_ARCH_DEFCONFIG="arch/x86/configs/i386_defconfig" CONFIG_LOCKDEP_SUPPORT=y CONFIG_STACKTRACE_SUPPORT=y CONFIG_HAVE_LATENCYTOP_SUPPORT=y CONFIG_MMU=y CONFIG_NEED_DMA_MAP_STATE=y CONFIG_NEED_SG_DMA_LENGTH=y CONFIG_GENERIC_ISA_DMA=y CONFIG_GENERIC_BUG=y CONFIG_GENERIC_HWEIGHT=y CONFIG_ARCH_MAY_HAVE_PC_FDC=y CONFIG_RWSEM_XCHGADD_ALGORITHM=y CONFIG_GENERIC_CALIBRATE_DELAY=y CONFIG_ARCH_HAS_CPU_RELAX=y CONFIG_ARCH_HAS_CACHE_LINE_SIZE=y CONFIG_ARCH_HAS_CPU_AUTOPROBE=y CONFIG_HAVE_SETUP_PER_CPU_AREA=y CONFIG_NEED_PER_CPU_EMBED_FIRST_CHUNK=y CONFIG_NEED_PER_CPU_PAGE_FIRST_CHUNK=y CONFIG_ARCH_HIBERNATION_POSSIBLE=y CONFIG_ARCH_SUSPEND_POSSIBLE=y CONFIG_ARCH_WANT_HUGE_PMD_SHARE=y CONFIG_ARCH_WANT_GENERAL_HUGETLB=y # CONFIG_ZONE_DMA32 is not set # CONFIG_AUDIT_ARCH is not set CONFIG_ARCH_SUPPORTS_OPTIMIZED_INLINING=y CONFIG_ARCH_SUPPORTS_DEBUG_PAGEALLOC=y CONFIG_X86_32_LAZY_GS=y CONFIG_ARCH_HWEIGHT_CFLAGS="-fcall-saved-ecx -fcall-saved-edx" CONFIG_ARCH_SUPPORTS_UPROBES=y CONFIG_DEFCONFIG_LIST="/lib/modules/$UNAME_RELEASE/.config" CONFIG_CONSTRUCTORS=y CONFIG_IRQ_WORK=y CONFIG_BUILDTIME_EXTABLE_SORT=y # # General setup # CONFIG_BROKEN_ON_SMP=y CONFIG_INIT_ENV_ARG_LIMIT=32 CONFIG_CROSS_COMPILE="" CONFIG_COMPILE_TEST=y CONFIG_LOCALVERSION="" # CONFIG_LOCALVERSION_AUTO is not set CONFIG_HAVE_KERNEL_GZIP=y CONFIG_HAVE_KERNEL_BZIP2=y CONFIG_HAVE_KERNEL_LZMA=y CONFIG_HAVE_KERNEL_XZ=y CONFIG_HAVE_KERNEL_LZO=y CONFIG_HAVE_KERNEL_LZ4=y CONFIG_KERNEL_GZIP=y # CONFIG_KERNEL_BZIP2 is not set # CONFIG_KERNEL_LZMA is not set # CONFIG_KERNEL_XZ is not set # CONFIG_KERNEL_LZO is not set # CONFIG_KERNEL_LZ4 is not set CONFIG_DEFAULT_HOSTNAME="(none)" CONFIG_SYSVIPC=y # CONFIG_FHANDLE is not set # # IRQ subsystem # CONFIG_GENERIC_IRQ_PROBE=y CONFIG_GENERIC_IRQ_SHOW=y CONFIG_GENERIC_IRQ_CHIP=y CONFIG_IRQ_DOMAIN=y CONFIG_IRQ_DOMAIN_DEBUG=y CONFIG_IRQ_FORCED_THREADING=y CONFIG_SPARSE_IRQ=y CONFIG_CLOCKSOURCE_WATCHDOG=y CONFIG_KTIME_SCALAR=y CONFIG_GENERIC_CLOCKEVENTS=y CONFIG_GENERIC_CLOCKEVENTS_BUILD=y CONFIG_GENERIC_CLOCKEVENTS_BROADCAST=y CONFIG_GENERIC_CLOCKEVENTS_MIN_ADJUST=y CONFIG_GENERIC_CMOS_UPDATE=y # # Timers subsystem # CONFIG_TICK_ONESHOT=y CONFIG_NO_HZ_COMMON=y # CONFIG_HZ_PERIODIC is not set CONFIG_NO_HZ_IDLE=y CONFIG_NO_HZ=y # CONFIG_HIGH_RES_TIMERS is not set # # CPU/Task time and stats accounting # CONFIG_TICK_CPU_ACCOUNTING=y # CONFIG_IRQ_TIME_ACCOUNTING is not set # CONFIG_BSD_PROCESS_ACCT is not set # # RCU Subsystem # CONFIG_TINY_RCU=y # CONFIG_PREEMPT_RCU is not set CONFIG_RCU_STALL_COMMON=y # CONFIG_TREE_RCU_TRACE is not set CONFIG_IKCONFIG=y CONFIG_IKCONFIG_PROC=y CONFIG_LOG_BUF_SHIFT=17 CONFIG_HAVE_UNSTABLE_SCHED_CLOCK=y CONFIG_ARCH_SUPPORTS_NUMA_BALANCING=y CONFIG_ARCH_WANTS_PROT_NUMA_PROT_NONE=y # CONFIG_CHECKPOINT_RESTORE is not set CONFIG_NAMESPACES=y # CONFIG_UTS_NS is not set # CONFIG_IPC_NS is not set # CONFIG_USER_NS is not set CONFIG_PID_NS=y CONFIG_UIDGID_STRICT_TYPE_CHECKS=y # CONFIG_SCHED_AUTOGROUP is not set CONFIG_SYSFS_DEPRECATED=y # CONFIG_SYSFS_DEPRECATED_V2 is not set CONFIG_RELAY=y # CONFIG_BLK_DEV_INITRD is not set CONFIG_CC_OPTIMIZE_FOR_SIZE=y CONFIG_ANON_INODES=y CONFIG_HAVE_UID16=y CONFIG_SYSCTL_EXCEPTION_TRACE=y CONFIG_HAVE_PCSPKR_PLATFORM=y CONFIG_EXPERT=y CONFIG_UID16=y CONFIG_KALLSYMS=y CONFIG_KALLSYMS_ALL=y # CONFIG_PRINTK is not set CONFIG_BUG=y # CONFIG_PCSPKR_PLATFORM is not set CONFIG_BASE_FULL=y CONFIG_FUTEX=y # CONFIG_EPOLL is not set # CONFIG_SIGNALFD is not set CONFIG_TIMERFD=y # CONFIG_EVENTFD is not set CONFIG_SHMEM=y CONFIG_AIO=y # CONFIG_PCI_QUIRKS is not set # CONFIG_EMBEDDED is not set CONFIG_HAVE_PERF_EVENTS=y # # Kernel Performance Events And Counters # CONFIG_PERF_EVENTS=y # CONFIG_DEBUG_PERF_USE_VMALLOC is not set # CONFIG_VM_EVENT_COUNTERS is not set CONFIG_SLUB_DEBUG=y # CONFIG_COMPAT_BRK is not set # CONFIG_SLAB is not set CONFIG_SLUB=y # CONFIG_SLOB is not set # CONFIG_PROFILING is not set CONFIG_TRACEPOINTS=y CONFIG_HAVE_OPROFILE=y CONFIG_OPROFILE_NMI_TIMER=y # CONFIG_JUMP_LABEL is not set CONFIG_UPROBES=y # CONFIG_HAVE_64BIT_ALIGNED_ACCESS is not set CONFIG_HAVE_EFFICIENT_UNALIGNED_ACCESS=y CONFIG_ARCH_USE_BUILTIN_BSWAP=y CONFIG_HAVE_IOREMAP_PROT=y CONFIG_HAVE_KPROBES=y CONFIG_HAVE_KRETPROBES=y CONFIG_HAVE_OPTPROBES=y CONFIG_HAVE_KPROBES_ON_FTRACE=y CONFIG_HAVE_ARCH_TRACEHOOK=y CONFIG_HAVE_DMA_ATTRS=y CONFIG_HAVE_DMA_CONTIGUOUS=y CONFIG_GENERIC_SMP_IDLE_THREAD=y CONFIG_HAVE_REGS_AND_STACK_ACCESS_API=y CONFIG_HAVE_DMA_API_DEBUG=y CONFIG_HAVE_HW_BREAKPOINT=y CONFIG_HAVE_MIXED_BREAKPOINTS_REGS=y CONFIG_HAVE_USER_RETURN_NOTIFIER=y CONFIG_HAVE_PERF_EVENTS_NMI=y CONFIG_HAVE_PERF_REGS=y CONFIG_HAVE_PERF_USER_STACK_DUMP=y CONFIG_HAVE_ARCH_JUMP_LABEL=y CONFIG_ARCH_HAVE_NMI_SAFE_CMPXCHG=y CONFIG_HAVE_ALIGNED_STRUCT_PAGE=y CONFIG_HAVE_CMPXCHG_LOCAL=y CONFIG_HAVE_CMPXCHG_DOUBLE=y CONFIG_ARCH_WANT_IPC_PARSE_VERSION=y CONFIG_HAVE_ARCH_SECCOMP_FILTER=y CONFIG_HAVE_IRQ_TIME_ACCOUNTING=y CONFIG_HAVE_ARCH_TRANSPARENT_HUGEPAGE=y CONFIG_HAVE_ARCH_SOFT_DIRTY=y CONFIG_MODULES_USE_ELF_REL=y CONFIG_CLONE_BACKWARDS=y CONFIG_OLD_SIGSUSPEND3=y CONFIG_OLD_SIGACTION=y # # GCOV-based kernel profiling # CONFIG_GCOV_KERNEL=y CONFIG_GCOV_PROFILE_ALL=y CONFIG_HAVE_GENERIC_DMA_COHERENT=y CONFIG_SLABINFO=y CONFIG_RT_MUTEXES=y CONFIG_BASE_SMALL=0 # CONFIG_MODULES is not set # CONFIG_BLOCK is not set CONFIG_UNINLINE_SPIN_UNLOCK=y CONFIG_FREEZER=y # # Processor type and features # CONFIG_ZONE_DMA=y # CONFIG_SMP is not set CONFIG_X86_MPPARSE=y CONFIG_GOLDFISH=y CONFIG_X86_EXTENDED_PLATFORM=y CONFIG_X86_GOLDFISH=y CONFIG_X86_WANT_INTEL_MID=y CONFIG_X86_INTEL_MID=y CONFIG_X86_MDFLD=y CONFIG_X86_RDC321X=y CONFIG_X86_32_IRIS=y CONFIG_SCHED_OMIT_FRAME_POINTER=y CONFIG_HYPERVISOR_GUEST=y # CONFIG_PARAVIRT is not set # CONFIG_XEN_PRIVILEGED_GUEST is not set CONFIG_NO_BOOTMEM=y CONFIG_MEMTEST=y # CONFIG_M486 is not set # CONFIG_M586 is not set # CONFIG_M586TSC is not set # CONFIG_M586MMX is not set CONFIG_M686=y # CONFIG_MPENTIUMII is not set # CONFIG_MPENTIUMIII is not set # CONFIG_MPENTIUMM is not set # CONFIG_MPENTIUM4 is not set # CONFIG_MK6 is not set # CONFIG_MK7 is not set # CONFIG_MK8 is not set # CONFIG_MCRUSOE is not set # CONFIG_MEFFICEON is not set # CONFIG_MWINCHIPC6 is not set # CONFIG_MWINCHIP3D is not set # CONFIG_MELAN is not set # CONFIG_MGEODEGX1 is not set # CONFIG_MGEODE_LX is not set # CONFIG_MCYRIXIII is not set # CONFIG_MVIAC3_2 is not set # CONFIG_MVIAC7 is not set # CONFIG_MCORE2 is not set # CONFIG_MATOM is not set CONFIG_X86_GENERIC=y CONFIG_X86_INTERNODE_CACHE_SHIFT=6 CONFIG_X86_L1_CACHE_SHIFT=6 # CONFIG_X86_PPRO_FENCE is not set CONFIG_X86_INTEL_USERCOPY=y CONFIG_X86_USE_PPRO_CHECKSUM=y CONFIG_X86_TSC=y CONFIG_X86_CMPXCHG64=y CONFIG_X86_CMOV=y CONFIG_X86_MINIMUM_CPU_FAMILY=5 CONFIG_X86_DEBUGCTLMSR=y # CONFIG_PROCESSOR_SELECT is not set CONFIG_CPU_SUP_INTEL=y CONFIG_CPU_SUP_CYRIX_32=y CONFIG_CPU_SUP_AMD=y CONFIG_CPU_SUP_CENTAUR=y CONFIG_CPU_SUP_TRANSMETA_32=y CONFIG_CPU_SUP_UMC_32=y CONFIG_HPET_TIMER=y CONFIG_HPET_EMULATE_RTC=y CONFIG_APB_TIMER=y CONFIG_DMI=y CONFIG_NR_CPUS=1 CONFIG_PREEMPT_NONE=y # CONFIG_PREEMPT_VOLUNTARY is not set # CONFIG_PREEMPT is not set CONFIG_PREEMPT_COUNT=y CONFIG_X86_LOCAL_APIC=y CONFIG_X86_IO_APIC=y CONFIG_X86_REROUTE_FOR_BROKEN_BOOT_IRQS=y # CONFIG_X86_MCE is not set CONFIG_VM86=y CONFIG_TOSHIBA=y CONFIG_I8K=y CONFIG_X86_REBOOTFIXUPS=y # CONFIG_MICROCODE is not set # CONFIG_MICROCODE_INTEL_EARLY is not set # CONFIG_MICROCODE_AMD_EARLY is not set CONFIG_X86_MSR=y # CONFIG_X86_CPUID is not set # CONFIG_NOHIGHMEM is not set CONFIG_HIGHMEM4G=y # CONFIG_HIGHMEM64G is not set CONFIG_VMSPLIT_3G=y # CONFIG_VMSPLIT_3G_OPT is not set # CONFIG_VMSPLIT_2G is not set # CONFIG_VMSPLIT_2G_OPT is not set # CONFIG_VMSPLIT_1G is not set CONFIG_PAGE_OFFSET=0xC0000000 CONFIG_HIGHMEM=y CONFIG_ARCH_FLATMEM_ENABLE=y CONFIG_ARCH_SPARSEMEM_ENABLE=y CONFIG_ARCH_SELECT_MEMORY_MODEL=y CONFIG_ILLEGAL_POINTER_VALUE=0 CONFIG_SELECT_MEMORY_MODEL=y CONFIG_FLATMEM_MANUAL=y # CONFIG_SPARSEMEM_MANUAL is not set CONFIG_FLATMEM=y CONFIG_FLAT_NODE_MEM_MAP=y CONFIG_SPARSEMEM_STATIC=y CONFIG_HAVE_MEMBLOCK=y CONFIG_HAVE_MEMBLOCK_NODE_MAP=y CONFIG_ARCH_DISCARD_MEMBLOCK=y # CONFIG_HAVE_BOOTMEM_INFO_NODE is not set CONFIG_PAGEFLAGS_EXTENDED=y CONFIG_SPLIT_PTLOCK_CPUS=999999 CONFIG_BALLOON_COMPACTION=y CONFIG_COMPACTION=y CONFIG_MIGRATION=y # CONFIG_PHYS_ADDR_T_64BIT is not set CONFIG_ZONE_DMA_FLAG=1 CONFIG_VIRT_TO_BUS=y CONFIG_KSM=y CONFIG_DEFAULT_MMAP_MIN_ADDR=4096 CONFIG_TRANSPARENT_HUGEPAGE=y CONFIG_TRANSPARENT_HUGEPAGE_ALWAYS=y # CONFIG_TRANSPARENT_HUGEPAGE_MADVISE is not set CONFIG_CROSS_MEMORY_ATTACH=y CONFIG_NEED_PER_CPU_KM=y # CONFIG_CLEANCACHE is not set # CONFIG_CMA is not set # CONFIG_ZBUD is not set CONFIG_HIGHPTE=y CONFIG_X86_CHECK_BIOS_CORRUPTION=y # CONFIG_X86_BOOTPARAM_MEMORY_CORRUPTION_CHECK is not set CONFIG_X86_RESERVE_LOW=64 CONFIG_MATH_EMULATION=y # CONFIG_MTRR is not set # CONFIG_ARCH_RANDOM is not set CONFIG_X86_SMAP=y # CONFIG_SECCOMP is not set # CONFIG_CC_STACKPROTECTOR is not set # CONFIG_HZ_100 is not set CONFIG_HZ_250=y # CONFIG_HZ_300 is not set # CONFIG_HZ_1000 is not set CONFIG_HZ=250 # CONFIG_SCHED_HRTICK is not set CONFIG_KEXEC=y # CONFIG_CRASH_DUMP is not set CONFIG_PHYSICAL_START=0x1000000 CONFIG_RELOCATABLE=y CONFIG_X86_NEED_RELOCS=y CONFIG_PHYSICAL_ALIGN=0x1000000 CONFIG_COMPAT_VDSO=y # CONFIG_CMDLINE_BOOL is not set CONFIG_ARCH_ENABLE_MEMORY_HOTPLUG=y # # Power management and ACPI options # CONFIG_SUSPEND=y CONFIG_SUSPEND_FREEZER=y CONFIG_PM_SLEEP=y # CONFIG_PM_AUTOSLEEP is not set # CONFIG_PM_WAKELOCKS is not set CONFIG_PM_RUNTIME=y CONFIG_PM=y # CONFIG_PM_DEBUG is not set # CONFIG_WQ_POWER_EFFICIENT_DEFAULT is not set # CONFIG_ACPI is not set CONFIG_SFI=y CONFIG_X86_APM_BOOT=y CONFIG_APM=y CONFIG_APM_IGNORE_USER_SUSPEND=y # CONFIG_APM_DO_ENABLE is not set CONFIG_APM_CPU_IDLE=y # CONFIG_APM_DISPLAY_BLANK is not set # CONFIG_APM_ALLOW_INTS is not set # # CPU Frequency scaling # CONFIG_CPU_FREQ=y CONFIG_CPU_FREQ_TABLE=y CONFIG_CPU_FREQ_GOV_COMMON=y CONFIG_CPU_FREQ_STAT=y CONFIG_CPU_FREQ_STAT_DETAILS=y CONFIG_CPU_FREQ_DEFAULT_GOV_PERFORMANCE=y # CONFIG_CPU_FREQ_DEFAULT_GOV_POWERSAVE is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_USERSPACE is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_ONDEMAND is not set # CONFIG_CPU_FREQ_DEFAULT_GOV_CONSERVATIVE is not set CONFIG_CPU_FREQ_GOV_PERFORMANCE=y CONFIG_CPU_FREQ_GOV_POWERSAVE=y CONFIG_CPU_FREQ_GOV_USERSPACE=y # CONFIG_CPU_FREQ_GOV_ONDEMAND is not set CONFIG_CPU_FREQ_GOV_CONSERVATIVE=y # # x86 CPU frequency scaling drivers # CONFIG_X86_INTEL_PSTATE=y CONFIG_X86_POWERNOW_K6=y CONFIG_X86_POWERNOW_K7=y # CONFIG_X86_GX_SUSPMOD is not set # CONFIG_X86_SPEEDSTEP_CENTRINO is not set # CONFIG_X86_SPEEDSTEP_ICH is not set CONFIG_X86_SPEEDSTEP_SMI=y # CONFIG_X86_P4_CLOCKMOD is not set CONFIG_X86_CPUFREQ_NFORCE2=y CONFIG_X86_LONGRUN=y # # shared options # CONFIG_X86_SPEEDSTEP_LIB=y CONFIG_X86_SPEEDSTEP_RELAXED_CAP_CHECK=y # # CPU Idle # CONFIG_CPU_IDLE=y CONFIG_CPU_IDLE_MULTIPLE_DRIVERS=y # CONFIG_CPU_IDLE_GOV_LADDER is not set CONFIG_CPU_IDLE_GOV_MENU=y # CONFIG_ARCH_NEEDS_CPU_IDLE_COUPLED is not set CONFIG_INTEL_IDLE=y # # Bus options (PCI etc.) # CONFIG_PCI=y # CONFIG_PCI_GOBIOS is not set # CONFIG_PCI_GOMMCONFIG is not set # CONFIG_PCI_GODIRECT is not set # CONFIG_PCI_GOOLPC is not set CONFIG_PCI_GOANY=y CONFIG_PCI_BIOS=y CONFIG_PCI_DIRECT=y CONFIG_PCI_MMCONFIG=y CONFIG_PCI_OLPC=y CONFIG_PCI_DOMAINS=y # CONFIG_PCI_CNB20LE_QUIRK is not set # CONFIG_PCIEPORTBUS is not set CONFIG_PCI_MSI=y # CONFIG_PCI_DEBUG is not set # CONFIG_PCI_REALLOC_ENABLE_AUTO is not set CONFIG_PCI_STUB=y # CONFIG_HT_IRQ is not set CONFIG_PCI_ATS=y CONFIG_PCI_IOV=y CONFIG_PCI_PRI=y # CONFIG_PCI_PASID is not set CONFIG_PCI_LABEL=y # # PCI host controller drivers # CONFIG_ISA_DMA_API=y # CONFIG_ISA is not set CONFIG_SCx200=y CONFIG_SCx200HR_TIMER=y CONFIG_OLPC=y # CONFIG_OLPC_XO1_PM is not set CONFIG_ALIX=y CONFIG_NET5501=y # CONFIG_GEOS is not set CONFIG_AMD_NB=y CONFIG_PCCARD=y # CONFIG_PCMCIA is not set # CONFIG_CARDBUS is not set # # PC-card bridges # CONFIG_YENTA=y CONFIG_YENTA_O2=y # CONFIG_YENTA_RICOH is not set CONFIG_YENTA_TI=y # CONFIG_YENTA_TOSHIBA is not set CONFIG_HOTPLUG_PCI=y CONFIG_HOTPLUG_PCI_COMPAQ=y # CONFIG_HOTPLUG_PCI_COMPAQ_NVRAM is not set # CONFIG_HOTPLUG_PCI_IBM is not set CONFIG_HOTPLUG_PCI_CPCI=y CONFIG_HOTPLUG_PCI_CPCI_ZT5550=y CONFIG_HOTPLUG_PCI_CPCI_GENERIC=y CONFIG_HOTPLUG_PCI_SHPC=y # CONFIG_RAPIDIO is not set CONFIG_X86_SYSFB=y # # Executable file formats / Emulations # CONFIG_BINFMT_ELF=y CONFIG_ARCH_BINFMT_ELF_RANDOMIZE_PIE=y # CONFIG_BINFMT_SCRIPT is not set CONFIG_HAVE_AOUT=y # CONFIG_BINFMT_AOUT is not set # CONFIG_BINFMT_MISC is not set # CONFIG_COREDUMP is not set CONFIG_HAVE_ATOMIC_IOMAP=y # CONFIG_NET is not set # # Device Drivers # # # Generic Driver Options # CONFIG_UEVENT_HELPER_PATH="" # CONFIG_DEVTMPFS is not set CONFIG_STANDALONE=y CONFIG_PREVENT_FIRMWARE_BUILD=y CONFIG_FW_LOADER=y # CONFIG_FIRMWARE_IN_KERNEL is not set CONFIG_EXTRA_FIRMWARE="" CONFIG_FW_LOADER_USER_HELPER=y CONFIG_DEBUG_DRIVER=y CONFIG_DEBUG_DEVRES=y # CONFIG_SYS_HYPERVISOR is not set # CONFIG_GENERIC_CPU_DEVICES is not set CONFIG_REGMAP=y CONFIG_REGMAP_I2C=y CONFIG_REGMAP_SPI=y CONFIG_REGMAP_MMIO=y CONFIG_REGMAP_IRQ=y CONFIG_DMA_SHARED_BUFFER=y # # Bus devices # CONFIG_MTD=y CONFIG_MTD_REDBOOT_PARTS=y CONFIG_MTD_REDBOOT_DIRECTORY_BLOCK=-1 # CONFIG_MTD_REDBOOT_PARTS_UNALLOCATED is not set CONFIG_MTD_REDBOOT_PARTS_READONLY=y CONFIG_MTD_CMDLINE_PARTS=y CONFIG_MTD_OF_PARTS=y CONFIG_MTD_AR7_PARTS=y # # User Modules And Translation Layers # CONFIG_MTD_OOPS=y # # RAM/ROM/Flash chip drivers # # CONFIG_MTD_CFI is not set CONFIG_MTD_JEDECPROBE=y CONFIG_MTD_GEN_PROBE=y CONFIG_MTD_CFI_ADV_OPTIONS=y CONFIG_MTD_CFI_NOSWAP=y # CONFIG_MTD_CFI_BE_BYTE_SWAP is not set # CONFIG_MTD_CFI_LE_BYTE_SWAP is not set # CONFIG_MTD_CFI_GEOMETRY is not set CONFIG_MTD_MAP_BANK_WIDTH_1=y CONFIG_MTD_MAP_BANK_WIDTH_2=y CONFIG_MTD_MAP_BANK_WIDTH_4=y # CONFIG_MTD_MAP_BANK_WIDTH_8 is not set # CONFIG_MTD_MAP_BANK_WIDTH_16 is not set # CONFIG_MTD_MAP_BANK_WIDTH_32 is not set CONFIG_MTD_CFI_I1=y CONFIG_MTD_CFI_I2=y # CONFIG_MTD_CFI_I4 is not set # CONFIG_MTD_CFI_I8 is not set CONFIG_MTD_OTP=y CONFIG_MTD_CFI_INTELEXT=y CONFIG_MTD_CFI_AMDSTD=y CONFIG_MTD_CFI_STAA=y CONFIG_MTD_CFI_UTIL=y CONFIG_MTD_RAM=y CONFIG_MTD_ROM=y CONFIG_MTD_ABSENT=y # # Mapping drivers for chip access # CONFIG_MTD_COMPLEX_MAPPINGS=y CONFIG_MTD_PHYSMAP=y CONFIG_MTD_PHYSMAP_COMPAT=y CONFIG_MTD_PHYSMAP_START=0x8000000 CONFIG_MTD_PHYSMAP_LEN=0 CONFIG_MTD_PHYSMAP_BANKWIDTH=2 CONFIG_MTD_PHYSMAP_OF=y CONFIG_MTD_TS5500=y CONFIG_MTD_SBC_GXX=y # CONFIG_MTD_AMD76XROM is not set # CONFIG_MTD_ICHXROM is not set # CONFIG_MTD_ESB2ROM is not set # CONFIG_MTD_CK804XROM is not set CONFIG_MTD_SCB2_FLASH=y # CONFIG_MTD_NETtel is not set # CONFIG_MTD_L440GX is not set CONFIG_MTD_PCI=y CONFIG_MTD_GPIO_ADDR=y # CONFIG_MTD_INTEL_VR_NOR is not set CONFIG_MTD_PLATRAM=y # CONFIG_MTD_LATCH_ADDR is not set # # Self-contained MTD device drivers # # CONFIG_MTD_PMC551 is not set CONFIG_MTD_DATAFLASH=y # CONFIG_MTD_DATAFLASH_WRITE_VERIFY is not set CONFIG_MTD_DATAFLASH_OTP=y CONFIG_MTD_M25P80=y CONFIG_M25PXX_USE_FAST_READ=y CONFIG_MTD_SST25L=y CONFIG_MTD_SLRAM=y CONFIG_MTD_PHRAM=y CONFIG_MTD_MTDRAM=y CONFIG_MTDRAM_TOTAL_SIZE=4096 CONFIG_MTDRAM_ERASE_SIZE=128 CONFIG_MTDRAM_ABS_POS=0 # # Disk-On-Chip Device Drivers # CONFIG_MTD_DOCG3=y CONFIG_BCH_CONST_M=14 CONFIG_BCH_CONST_T=4 CONFIG_MTD_NAND_ECC=y CONFIG_MTD_NAND_ECC_SMC=y CONFIG_MTD_NAND=y CONFIG_MTD_NAND_BCH=y CONFIG_MTD_NAND_ECC_BCH=y # CONFIG_MTD_SM_COMMON is not set CONFIG_MTD_NAND_DENALI=y CONFIG_MTD_NAND_DENALI_PCI=y CONFIG_MTD_NAND_DENALI_SCRATCH_REG_ADDR=0xFF108018 CONFIG_MTD_NAND_GPIO=y CONFIG_MTD_NAND_IDS=y # CONFIG_MTD_NAND_RICOH is not set CONFIG_MTD_NAND_DISKONCHIP=y # CONFIG_MTD_NAND_DISKONCHIP_PROBE_ADVANCED is not set CONFIG_MTD_NAND_DISKONCHIP_PROBE_ADDRESS=0 CONFIG_MTD_NAND_DISKONCHIP_BBTWRITE=y CONFIG_MTD_NAND_DOCG4=y CONFIG_MTD_NAND_CAFE=y # CONFIG_MTD_NAND_CS553X is not set CONFIG_MTD_NAND_NANDSIM=y CONFIG_MTD_NAND_PLATFORM=y CONFIG_MTD_ONENAND=y # CONFIG_MTD_ONENAND_VERIFY_WRITE is not set # CONFIG_MTD_ONENAND_GENERIC is not set CONFIG_MTD_ONENAND_OTP=y CONFIG_MTD_ONENAND_2X_PROGRAM=y # # LPDDR flash memory drivers # # CONFIG_MTD_LPDDR is not set CONFIG_MTD_UBI=y CONFIG_MTD_UBI_WL_THRESHOLD=4096 CONFIG_MTD_UBI_BEB_LIMIT=20 CONFIG_MTD_UBI_FASTMAP=y CONFIG_MTD_UBI_GLUEBI=y CONFIG_OF=y # # Device Tree and Open Firmware support # CONFIG_PROC_DEVICETREE=y CONFIG_OF_SELFTEST=y CONFIG_OF_PROMTREE=y CONFIG_OF_ADDRESS=y CONFIG_OF_IRQ=y CONFIG_OF_PCI=y CONFIG_OF_PCI_IRQ=y CONFIG_OF_MTD=y CONFIG_PARPORT=y CONFIG_PARPORT_PC=y # CONFIG_PARPORT_PC_FIFO is not set CONFIG_PARPORT_PC_SUPERIO=y # CONFIG_PARPORT_GSC is not set CONFIG_PARPORT_AX88796=y # CONFIG_PARPORT_1284 is not set CONFIG_PARPORT_NOT_PC=y # # Misc devices # CONFIG_SENSORS_LIS3LV02D=y # CONFIG_AD525X_DPOT is not set CONFIG_DUMMY_IRQ=y # CONFIG_IBM_ASM is not set # CONFIG_PHANTOM is not set # CONFIG_INTEL_MID_PTI is not set CONFIG_SGI_IOC4=y CONFIG_TIFM_CORE=y CONFIG_TIFM_7XX1=y # CONFIG_ICS932S401 is not set CONFIG_ATMEL_SSC=y CONFIG_ENCLOSURE_SERVICES=y CONFIG_CS5535_MFGPT=y CONFIG_CS5535_MFGPT_DEFAULT_IRQ=7 CONFIG_CS5535_CLOCK_EVENT_SRC=y CONFIG_HP_ILO=y CONFIG_APDS9802ALS=y # CONFIG_ISL29003 is not set CONFIG_ISL29020=y # CONFIG_SENSORS_TSL2550 is not set CONFIG_SENSORS_BH1780=y CONFIG_SENSORS_BH1770=y CONFIG_SENSORS_APDS990X=y CONFIG_HMC6352=y # CONFIG_DS1682 is not set # CONFIG_TI_DAC7512 is not set # CONFIG_VMWARE_BALLOON is not set CONFIG_BMP085=y # CONFIG_BMP085_I2C is not set CONFIG_BMP085_SPI=y CONFIG_PCH_PHUB=y CONFIG_USB_SWITCH_FSA9480=y CONFIG_LATTICE_ECP3_CONFIG=y # CONFIG_SRAM is not set CONFIG_C2PORT=y CONFIG_C2PORT_DURAMAR_2150=y # # EEPROM support # CONFIG_EEPROM_AT24=y CONFIG_EEPROM_AT25=y # CONFIG_EEPROM_LEGACY is not set # CONFIG_EEPROM_MAX6875 is not set CONFIG_EEPROM_93CX6=y # CONFIG_EEPROM_93XX46 is not set # CONFIG_CB710_CORE is not set # # Texas Instruments shared transport line discipline # # CONFIG_SENSORS_LIS3_SPI is not set CONFIG_SENSORS_LIS3_I2C=y # # Altera FPGA firmware download module # # CONFIG_ALTERA_STAPL is not set CONFIG_VMWARE_VMCI=y CONFIG_HAVE_IDE=y # # SCSI device support # CONFIG_SCSI_MOD=y # CONFIG_SCSI_DMA is not set # CONFIG_SCSI_NETLINK is not set # CONFIG_FUSION is not set # # IEEE 1394 (FireWire) support # CONFIG_FIREWIRE=y # CONFIG_FIREWIRE_OHCI is not set CONFIG_FIREWIRE_NOSY=y # CONFIG_I2O is not set CONFIG_MACINTOSH_DRIVERS=y # # Input device support # CONFIG_INPUT=y CONFIG_INPUT_FF_MEMLESS=y CONFIG_INPUT_POLLDEV=y CONFIG_INPUT_SPARSEKMAP=y CONFIG_INPUT_MATRIXKMAP=y # # Userland interfaces # CONFIG_INPUT_MOUSEDEV=y # CONFIG_INPUT_MOUSEDEV_PSAUX is not set CONFIG_INPUT_MOUSEDEV_SCREEN_X=1024 CONFIG_INPUT_MOUSEDEV_SCREEN_Y=768 CONFIG_INPUT_JOYDEV=y # CONFIG_INPUT_EVDEV is not set CONFIG_INPUT_EVBUG=y # # Input Device Drivers # # CONFIG_INPUT_KEYBOARD is not set # CONFIG_INPUT_MOUSE is not set # CONFIG_INPUT_JOYSTICK is not set CONFIG_INPUT_TABLET=y CONFIG_INPUT_TOUCHSCREEN=y CONFIG_TOUCHSCREEN_88PM860X=y # CONFIG_TOUCHSCREEN_ADS7846 is not set # CONFIG_TOUCHSCREEN_AD7877 is not set CONFIG_TOUCHSCREEN_AD7879=y # CONFIG_TOUCHSCREEN_AD7879_I2C is not set # CONFIG_TOUCHSCREEN_AD7879_SPI is not set # CONFIG_TOUCHSCREEN_ATMEL_MXT is not set # CONFIG_TOUCHSCREEN_AUO_PIXCIR is not set CONFIG_TOUCHSCREEN_BU21013=y # CONFIG_TOUCHSCREEN_CY8CTMG110 is not set # CONFIG_TOUCHSCREEN_CYTTSP_CORE is not set CONFIG_TOUCHSCREEN_CYTTSP4_CORE=y # CONFIG_TOUCHSCREEN_CYTTSP4_I2C is not set CONFIG_TOUCHSCREEN_CYTTSP4_SPI=y # CONFIG_TOUCHSCREEN_DA9034 is not set # CONFIG_TOUCHSCREEN_DA9052 is not set CONFIG_TOUCHSCREEN_DYNAPRO=y CONFIG_TOUCHSCREEN_HAMPSHIRE=y CONFIG_TOUCHSCREEN_EETI=y CONFIG_TOUCHSCREEN_EGALAX=y CONFIG_TOUCHSCREEN_FUJITSU=y # CONFIG_TOUCHSCREEN_ILI210X is not set CONFIG_TOUCHSCREEN_GUNZE=y # CONFIG_TOUCHSCREEN_ELO is not set CONFIG_TOUCHSCREEN_WACOM_W8001=y CONFIG_TOUCHSCREEN_WACOM_I2C=y # CONFIG_TOUCHSCREEN_MAX11801 is not set CONFIG_TOUCHSCREEN_MCS5000=y CONFIG_TOUCHSCREEN_MMS114=y CONFIG_TOUCHSCREEN_MTOUCH=y CONFIG_TOUCHSCREEN_INEXIO=y # CONFIG_TOUCHSCREEN_INTEL_MID is not set CONFIG_TOUCHSCREEN_MK712=y CONFIG_TOUCHSCREEN_PENMOUNT=y CONFIG_TOUCHSCREEN_EDT_FT5X06=y CONFIG_TOUCHSCREEN_TOUCHRIGHT=y # CONFIG_TOUCHSCREEN_TOUCHWIN is not set # CONFIG_TOUCHSCREEN_PIXCIR is not set # CONFIG_TOUCHSCREEN_WM831X is not set CONFIG_TOUCHSCREEN_WM97XX=y CONFIG_TOUCHSCREEN_WM9705=y CONFIG_TOUCHSCREEN_WM9712=y # CONFIG_TOUCHSCREEN_WM9713 is not set CONFIG_TOUCHSCREEN_MC13783=y CONFIG_TOUCHSCREEN_TOUCHIT213=y # CONFIG_TOUCHSCREEN_TSC_SERIO is not set CONFIG_TOUCHSCREEN_TSC2005=y CONFIG_TOUCHSCREEN_TSC2007=y CONFIG_TOUCHSCREEN_PCAP=y # CONFIG_TOUCHSCREEN_ST1232 is not set CONFIG_TOUCHSCREEN_STMPE=y # CONFIG_TOUCHSCREEN_TPS6507X is not set # CONFIG_INPUT_MISC is not set # # Hardware I/O ports # CONFIG_SERIO=y CONFIG_SERIO_I8042=y CONFIG_SERIO_SERPORT=y CONFIG_SERIO_CT82C710=y CONFIG_SERIO_PARKBD=y CONFIG_SERIO_PCIPS2=y # CONFIG_SERIO_LIBPS2 is not set CONFIG_SERIO_RAW=y CONFIG_SERIO_ALTERA_PS2=y # CONFIG_SERIO_PS2MULT is not set # CONFIG_SERIO_ARC_PS2 is not set CONFIG_SERIO_APBPS2=y CONFIG_SERIO_OLPC_APSP=y CONFIG_GAMEPORT=y # CONFIG_GAMEPORT_NS558 is not set CONFIG_GAMEPORT_L4=y CONFIG_GAMEPORT_EMU10K1=y CONFIG_GAMEPORT_FM801=y # # Character devices # CONFIG_TTY=y CONFIG_VT=y # CONFIG_CONSOLE_TRANSLATIONS is not set # CONFIG_VT_CONSOLE is not set CONFIG_HW_CONSOLE=y CONFIG_VT_HW_CONSOLE_BINDING=y # CONFIG_UNIX98_PTYS is not set CONFIG_LEGACY_PTYS=y CONFIG_LEGACY_PTY_COUNT=256 # CONFIG_SERIAL_NONSTANDARD is not set # CONFIG_NOZOMI is not set CONFIG_TRACE_ROUTER=y CONFIG_TRACE_SINK=y CONFIG_GOLDFISH_TTY=y CONFIG_DEVKMEM=y # # Serial drivers # CONFIG_SERIAL_8250=y # CONFIG_SERIAL_8250_DEPRECATED_OPTIONS is not set # CONFIG_SERIAL_8250_CONSOLE is not set CONFIG_FIX_EARLYCON_MEM=y # CONFIG_SERIAL_8250_PCI is not set CONFIG_SERIAL_8250_NR_UARTS=4 CONFIG_SERIAL_8250_RUNTIME_UARTS=4 CONFIG_SERIAL_8250_EXTENDED=y CONFIG_SERIAL_8250_MANY_PORTS=y # CONFIG_SERIAL_8250_SHARE_IRQ is not set CONFIG_SERIAL_8250_DETECT_IRQ=y # CONFIG_SERIAL_8250_RSA is not set CONFIG_SERIAL_8250_DW=y # # Non-8250 serial port support # CONFIG_SERIAL_MAX3100=y CONFIG_SERIAL_MAX310X=y CONFIG_SERIAL_MRST_MAX3110=y CONFIG_SERIAL_MFD_HSU=y CONFIG_SERIAL_MFD_HSU_CONSOLE=y CONFIG_SERIAL_CORE=y CONFIG_SERIAL_CORE_CONSOLE=y CONFIG_SERIAL_JSM=y # CONFIG_SERIAL_OF_PLATFORM is not set # CONFIG_SERIAL_SCCNXP is not set CONFIG_SERIAL_TIMBERDALE=y # CONFIG_SERIAL_ALTERA_JTAGUART is not set # CONFIG_SERIAL_ALTERA_UART is not set # CONFIG_SERIAL_IFX6X60 is not set CONFIG_SERIAL_PCH_UART=y # CONFIG_SERIAL_PCH_UART_CONSOLE is not set CONFIG_SERIAL_XILINX_PS_UART=y CONFIG_SERIAL_XILINX_PS_UART_CONSOLE=y CONFIG_SERIAL_ARC=y # CONFIG_SERIAL_ARC_CONSOLE is not set CONFIG_SERIAL_ARC_NR_PORTS=1 # CONFIG_SERIAL_RP2 is not set # CONFIG_SERIAL_FSL_LPUART is not set CONFIG_SERIAL_ST_ASC=y CONFIG_SERIAL_ST_ASC_CONSOLE=y CONFIG_TTY_PRINTK=y CONFIG_PRINTER=y CONFIG_LP_CONSOLE=y CONFIG_PPDEV=y # CONFIG_VIRTIO_CONSOLE is not set CONFIG_IPMI_HANDLER=y CONFIG_IPMI_PANIC_EVENT=y CONFIG_IPMI_PANIC_STRING=y CONFIG_IPMI_DEVICE_INTERFACE=y CONFIG_IPMI_SI=y CONFIG_IPMI_WATCHDOG=y CONFIG_IPMI_POWEROFF=y CONFIG_HW_RANDOM=y # CONFIG_HW_RANDOM_TIMERIOMEM is not set CONFIG_HW_RANDOM_INTEL=y # CONFIG_HW_RANDOM_AMD is not set CONFIG_HW_RANDOM_GEODE=y CONFIG_HW_RANDOM_VIA=y # CONFIG_HW_RANDOM_VIRTIO is not set CONFIG_HW_RANDOM_TPM=y # CONFIG_NVRAM is not set CONFIG_R3964=y # CONFIG_APPLICOM is not set CONFIG_SONYPI=y CONFIG_MWAVE=y CONFIG_SCx200_GPIO=y CONFIG_PC8736x_GPIO=y CONFIG_NSC_GPIO=y CONFIG_HANGCHECK_TIMER=y CONFIG_TCG_TPM=y CONFIG_TCG_TIS=y CONFIG_TCG_TIS_I2C_INFINEON=y CONFIG_TCG_NSC=y CONFIG_TCG_ATMEL=y CONFIG_TCG_ST33_I2C=y CONFIG_TELCLOCK=y CONFIG_DEVPORT=y CONFIG_I2C=y CONFIG_I2C_BOARDINFO=y # CONFIG_I2C_COMPAT is not set # CONFIG_I2C_CHARDEV is not set CONFIG_I2C_MUX=y # # Multiplexer I2C Chip support # CONFIG_I2C_ARB_GPIO_CHALLENGE=y CONFIG_I2C_MUX_GPIO=y CONFIG_I2C_MUX_PCA9541=y CONFIG_I2C_MUX_PCA954x=y CONFIG_I2C_HELPER_AUTO=y CONFIG_I2C_SMBUS=y CONFIG_I2C_ALGOBIT=y # # I2C Hardware Bus support # # # PC SMBus host controller drivers # CONFIG_I2C_ALI1535=y # CONFIG_I2C_ALI1563 is not set CONFIG_I2C_ALI15X3=y # CONFIG_I2C_AMD756 is not set # CONFIG_I2C_AMD8111 is not set CONFIG_I2C_I801=y CONFIG_I2C_ISCH=y # CONFIG_I2C_ISMT is not set # CONFIG_I2C_PIIX4 is not set CONFIG_I2C_NFORCE2=y CONFIG_I2C_NFORCE2_S4985=y # CONFIG_I2C_SIS5595 is not set CONFIG_I2C_SIS630=y CONFIG_I2C_SIS96X=y CONFIG_I2C_VIA=y CONFIG_I2C_VIAPRO=y # # I2C system bus drivers (mostly embedded / system-on-chip) # CONFIG_I2C_CBUS_GPIO=y CONFIG_I2C_DESIGNWARE_CORE=y CONFIG_I2C_DESIGNWARE_PCI=y # CONFIG_I2C_EG20T is not set CONFIG_I2C_GPIO=y # CONFIG_I2C_KEMPLD is not set # CONFIG_I2C_OCORES is not set # CONFIG_I2C_PCA_PLATFORM is not set CONFIG_I2C_PXA=y CONFIG_I2C_PXA_PCI=y # CONFIG_I2C_SIMTEC is not set # CONFIG_I2C_XILINX is not set # # External I2C/SMBus adapter drivers # # CONFIG_I2C_PARPORT is not set CONFIG_I2C_PARPORT_LIGHT=y # CONFIG_I2C_TAOS_EVM is not set # # Other I2C/SMBus bus drivers # CONFIG_SCx200_I2C=y CONFIG_SCx200_I2C_SCL=12 CONFIG_SCx200_I2C_SDA=13 CONFIG_SCx200_ACB=y CONFIG_I2C_DEBUG_CORE=y # CONFIG_I2C_DEBUG_ALGO is not set CONFIG_I2C_DEBUG_BUS=y CONFIG_SPI=y CONFIG_SPI_DEBUG=y CONFIG_SPI_MASTER=y # # SPI Master Controller Drivers # CONFIG_SPI_ALTERA=y CONFIG_SPI_ATMEL=y CONFIG_SPI_BCM2835=y CONFIG_SPI_BITBANG=y # CONFIG_SPI_BUTTERFLY is not set CONFIG_SPI_EP93XX=y # CONFIG_SPI_GPIO is not set CONFIG_SPI_IMX=y CONFIG_SPI_LM70_LLP=y # CONFIG_SPI_FSL_SPI is not set # CONFIG_SPI_FSL_DSPI is not set # CONFIG_SPI_OC_TINY is not set CONFIG_SPI_OMAP24XX=y # CONFIG_SPI_TI_QSPI is not set # CONFIG_SPI_OMAP_100K is not set # CONFIG_SPI_ORION is not set CONFIG_SPI_PXA2XX_DMA=y CONFIG_SPI_PXA2XX=y CONFIG_SPI_PXA2XX_PCI=y # CONFIG_SPI_SC18IS602 is not set CONFIG_SPI_SH=y CONFIG_SPI_SH_HSPI=y CONFIG_SPI_TEGRA114=y # CONFIG_SPI_TEGRA20_SFLASH is not set # CONFIG_SPI_TEGRA20_SLINK is not set # CONFIG_SPI_TOPCLIFF_PCH is not set CONFIG_SPI_TXX9=y CONFIG_SPI_XCOMM=y # CONFIG_SPI_XILINX is not set CONFIG_SPI_DESIGNWARE=y CONFIG_SPI_DW_PCI=y # # SPI Protocol Masters # CONFIG_SPI_SPIDEV=y CONFIG_SPI_TLE62X0=y # CONFIG_HSI is not set # # PPS support # CONFIG_PPS=y # CONFIG_PPS_DEBUG is not set # # PPS clients support # CONFIG_PPS_CLIENT_KTIMER=y # CONFIG_PPS_CLIENT_LDISC is not set CONFIG_PPS_CLIENT_PARPORT=y CONFIG_PPS_CLIENT_GPIO=y # # PPS generators support # # # PTP clock support # CONFIG_PTP_1588_CLOCK=y # # Enable PHYLIB and NETWORK_PHY_TIMESTAMPING to see the additional clocks. # CONFIG_PTP_1588_CLOCK_PCH=y CONFIG_ARCH_WANT_OPTIONAL_GPIOLIB=y CONFIG_GPIO_DEVRES=y CONFIG_GPIOLIB=y CONFIG_OF_GPIO=y # CONFIG_DEBUG_GPIO is not set CONFIG_GPIO_SYSFS=y CONFIG_GPIO_GENERIC=y # CONFIG_GPIO_DA9052 is not set # # Memory mapped GPIO drivers: # CONFIG_GPIO_GENERIC_PLATFORM=y CONFIG_GPIO_IT8761E=y # CONFIG_GPIO_F7188X is not set # CONFIG_GPIO_TS5500 is not set CONFIG_GPIO_SCH=y CONFIG_GPIO_ICH=y CONFIG_GPIO_VX855=y # CONFIG_GPIO_GRGPIO is not set # # I2C GPIO expanders: # CONFIG_GPIO_ARIZONA=y # CONFIG_GPIO_MAX7300 is not set # CONFIG_GPIO_MAX732X is not set CONFIG_GPIO_PCA953X=y CONFIG_GPIO_PCA953X_IRQ=y # CONFIG_GPIO_PCF857X is not set CONFIG_GPIO_SX150X=y CONFIG_GPIO_STMPE=y # CONFIG_GPIO_TC3589X is not set CONFIG_GPIO_TPS65912=y # CONFIG_GPIO_TWL6040 is not set CONFIG_GPIO_WM831X=y CONFIG_GPIO_WM8350=y # CONFIG_GPIO_WM8994 is not set CONFIG_GPIO_ADP5520=y CONFIG_GPIO_ADP5588=y # CONFIG_GPIO_ADP5588_IRQ is not set CONFIG_GPIO_ADNP=y # # PCI GPIO expanders: # CONFIG_GPIO_CS5535=y CONFIG_GPIO_BT8XX=y CONFIG_GPIO_AMD8111=y CONFIG_GPIO_LANGWELL=y # CONFIG_GPIO_PCH is not set # CONFIG_GPIO_ML_IOH is not set CONFIG_GPIO_SODAVILLE=y CONFIG_GPIO_RDC321X=y # # SPI GPIO expanders: # # CONFIG_GPIO_MAX7301 is not set CONFIG_GPIO_MCP23S08=y # CONFIG_GPIO_MC33880 is not set # CONFIG_GPIO_74X164 is not set # # AC97 GPIO expanders: # # # LPC GPIO expanders: # CONFIG_GPIO_KEMPLD=y # # MODULbus GPIO expanders: # CONFIG_GPIO_PALMAS=y CONFIG_GPIO_TPS65910=y # CONFIG_GPIO_MSIC is not set # # USB GPIO expanders: # CONFIG_W1=y # # 1-wire Bus Masters # CONFIG_W1_MASTER_MATROX=y # CONFIG_W1_MASTER_DS2482 is not set CONFIG_W1_MASTER_DS1WM=y CONFIG_W1_MASTER_GPIO=y # # 1-wire Slaves # # CONFIG_W1_SLAVE_THERM is not set # CONFIG_W1_SLAVE_SMEM is not set CONFIG_W1_SLAVE_DS2408=y # CONFIG_W1_SLAVE_DS2408_READBACK is not set CONFIG_W1_SLAVE_DS2413=y CONFIG_W1_SLAVE_DS2423=y CONFIG_W1_SLAVE_DS2431=y CONFIG_W1_SLAVE_DS2433=y CONFIG_W1_SLAVE_DS2433_CRC=y CONFIG_W1_SLAVE_DS2760=y CONFIG_W1_SLAVE_DS2780=y CONFIG_W1_SLAVE_DS2781=y CONFIG_W1_SLAVE_DS28E04=y # CONFIG_W1_SLAVE_BQ27000 is not set CONFIG_POWER_SUPPLY=y CONFIG_POWER_SUPPLY_DEBUG=y CONFIG_PDA_POWER=y CONFIG_GENERIC_ADC_BATTERY=y # CONFIG_WM831X_BACKUP is not set CONFIG_WM831X_POWER=y CONFIG_WM8350_POWER=y CONFIG_TEST_POWER=y # CONFIG_BATTERY_88PM860X is not set CONFIG_BATTERY_DS2760=y CONFIG_BATTERY_DS2780=y # CONFIG_BATTERY_DS2781 is not set CONFIG_BATTERY_DS2782=y # CONFIG_BATTERY_OLPC is not set CONFIG_BATTERY_WM97XX=y CONFIG_BATTERY_SBS=y # CONFIG_BATTERY_BQ27x00 is not set CONFIG_BATTERY_DA9030=y # CONFIG_BATTERY_DA9052 is not set CONFIG_BATTERY_MAX17040=y CONFIG_BATTERY_MAX17042=y CONFIG_CHARGER_PCF50633=y # CONFIG_BATTERY_INTEL_MID is not set # CONFIG_CHARGER_MAX8903 is not set CONFIG_CHARGER_LP8727=y CONFIG_CHARGER_LP8788=y CONFIG_CHARGER_GPIO=y # CONFIG_CHARGER_MANAGER is not set # CONFIG_CHARGER_MAX8997 is not set # CONFIG_CHARGER_MAX8998 is not set # CONFIG_CHARGER_BQ2415X is not set # CONFIG_CHARGER_BQ24190 is not set CONFIG_CHARGER_SMB347=y # CONFIG_CHARGER_TPS65090 is not set CONFIG_BATTERY_GOLDFISH=y # CONFIG_POWER_RESET is not set # CONFIG_POWER_AVS is not set CONFIG_HWMON=y CONFIG_HWMON_VID=y CONFIG_HWMON_DEBUG_CHIP=y # # Native drivers # # CONFIG_SENSORS_ABITUGURU is not set CONFIG_SENSORS_ABITUGURU3=y CONFIG_SENSORS_AD7314=y CONFIG_SENSORS_AD7414=y # CONFIG_SENSORS_AD7418 is not set # CONFIG_SENSORS_ADCXX is not set CONFIG_SENSORS_ADM1021=y # CONFIG_SENSORS_ADM1025 is not set CONFIG_SENSORS_ADM1026=y CONFIG_SENSORS_ADM1029=y # CONFIG_SENSORS_ADM1031 is not set CONFIG_SENSORS_ADM9240=y # CONFIG_SENSORS_ADT7310 is not set # CONFIG_SENSORS_ADT7410 is not set CONFIG_SENSORS_ADT7411=y CONFIG_SENSORS_ADT7462=y CONFIG_SENSORS_ADT7470=y CONFIG_SENSORS_ADT7475=y CONFIG_SENSORS_ASC7621=y CONFIG_SENSORS_K8TEMP=y CONFIG_SENSORS_K10TEMP=y # CONFIG_SENSORS_FAM15H_POWER is not set # CONFIG_SENSORS_ASB100 is not set # CONFIG_SENSORS_ATXP1 is not set CONFIG_SENSORS_DS620=y CONFIG_SENSORS_DS1621=y CONFIG_SENSORS_DA9052_ADC=y CONFIG_SENSORS_I5K_AMB=y CONFIG_SENSORS_F71805F=y CONFIG_SENSORS_F71882FG=y CONFIG_SENSORS_F75375S=y CONFIG_SENSORS_FSCHMD=y # CONFIG_SENSORS_G760A is not set CONFIG_SENSORS_G762=y CONFIG_SENSORS_GL518SM=y CONFIG_SENSORS_GL520SM=y CONFIG_SENSORS_GPIO_FAN=y CONFIG_SENSORS_HIH6130=y # CONFIG_SENSORS_HTU21 is not set CONFIG_SENSORS_CORETEMP=y CONFIG_SENSORS_IBMAEM=y CONFIG_SENSORS_IBMPEX=y CONFIG_SENSORS_IIO_HWMON=y CONFIG_SENSORS_IT87=y CONFIG_SENSORS_JC42=y CONFIG_SENSORS_LINEAGE=y CONFIG_SENSORS_LM63=y # CONFIG_SENSORS_LM70 is not set CONFIG_SENSORS_LM73=y CONFIG_SENSORS_LM75=y CONFIG_SENSORS_LM77=y CONFIG_SENSORS_LM78=y CONFIG_SENSORS_LM80=y CONFIG_SENSORS_LM83=y # CONFIG_SENSORS_LM85 is not set # CONFIG_SENSORS_LM87 is not set CONFIG_SENSORS_LM90=y CONFIG_SENSORS_LM92=y # CONFIG_SENSORS_LM93 is not set # CONFIG_SENSORS_LTC4151 is not set # CONFIG_SENSORS_LTC4215 is not set CONFIG_SENSORS_LTC4245=y CONFIG_SENSORS_LTC4261=y CONFIG_SENSORS_LM95234=y # CONFIG_SENSORS_LM95241 is not set # CONFIG_SENSORS_LM95245 is not set CONFIG_SENSORS_MAX1111=y # CONFIG_SENSORS_MAX16065 is not set CONFIG_SENSORS_MAX1619=y CONFIG_SENSORS_MAX1668=y CONFIG_SENSORS_MAX197=y CONFIG_SENSORS_MAX6639=y CONFIG_SENSORS_MAX6642=y # CONFIG_SENSORS_MAX6650 is not set CONFIG_SENSORS_MAX6697=y CONFIG_SENSORS_MCP3021=y # CONFIG_SENSORS_NCT6775 is not set CONFIG_SENSORS_NTC_THERMISTOR=y # CONFIG_SENSORS_PC87360 is not set CONFIG_SENSORS_PC87427=y CONFIG_SENSORS_PCF8591=y CONFIG_PMBUS=y # CONFIG_SENSORS_PMBUS is not set # CONFIG_SENSORS_ADM1275 is not set # CONFIG_SENSORS_LM25066 is not set # CONFIG_SENSORS_LTC2978 is not set CONFIG_SENSORS_MAX16064=y # CONFIG_SENSORS_MAX34440 is not set CONFIG_SENSORS_MAX8688=y # CONFIG_SENSORS_UCD9000 is not set # CONFIG_SENSORS_UCD9200 is not set CONFIG_SENSORS_ZL6100=y # CONFIG_SENSORS_SHT15 is not set CONFIG_SENSORS_SHT21=y CONFIG_SENSORS_SIS5595=y # CONFIG_SENSORS_SMM665 is not set CONFIG_SENSORS_DME1737=y CONFIG_SENSORS_EMC1403=y CONFIG_SENSORS_EMC2103=y CONFIG_SENSORS_EMC6W201=y CONFIG_SENSORS_SMSC47M1=y CONFIG_SENSORS_SMSC47M192=y CONFIG_SENSORS_SMSC47B397=y # CONFIG_SENSORS_SCH56XX_COMMON is not set CONFIG_SENSORS_ADS1015=y # CONFIG_SENSORS_ADS7828 is not set # CONFIG_SENSORS_ADS7871 is not set # CONFIG_SENSORS_AMC6821 is not set # CONFIG_SENSORS_INA209 is not set CONFIG_SENSORS_INA2XX=y CONFIG_SENSORS_THMC50=y # CONFIG_SENSORS_TMP102 is not set # CONFIG_SENSORS_TMP401 is not set CONFIG_SENSORS_TMP421=y CONFIG_SENSORS_VIA_CPUTEMP=y CONFIG_SENSORS_VIA686A=y CONFIG_SENSORS_VT1211=y CONFIG_SENSORS_VT8231=y # CONFIG_SENSORS_W83781D is not set CONFIG_SENSORS_W83791D=y CONFIG_SENSORS_W83792D=y CONFIG_SENSORS_W83793=y # CONFIG_SENSORS_W83795 is not set CONFIG_SENSORS_W83L785TS=y CONFIG_SENSORS_W83L786NG=y # CONFIG_SENSORS_W83627HF is not set # CONFIG_SENSORS_W83627EHF is not set CONFIG_SENSORS_WM831X=y CONFIG_SENSORS_WM8350=y CONFIG_SENSORS_APPLESMC=y # CONFIG_SENSORS_MC13783_ADC is not set CONFIG_THERMAL=y CONFIG_THERMAL_HWMON=y CONFIG_THERMAL_DEFAULT_GOV_STEP_WISE=y # CONFIG_THERMAL_DEFAULT_GOV_FAIR_SHARE is not set # CONFIG_THERMAL_DEFAULT_GOV_USER_SPACE is not set CONFIG_THERMAL_GOV_FAIR_SHARE=y CONFIG_THERMAL_GOV_STEP_WISE=y CONFIG_THERMAL_GOV_USER_SPACE=y CONFIG_CPU_THERMAL=y CONFIG_THERMAL_EMULATION=y CONFIG_INTEL_POWERCLAMP=y # # Texas Instruments thermal drivers # # CONFIG_WATCHDOG is not set CONFIG_SSB_POSSIBLE=y # # Sonics Silicon Backplane # # CONFIG_SSB is not set CONFIG_BCMA_POSSIBLE=y # # Broadcom specific AMBA # CONFIG_BCMA=y CONFIG_BCMA_HOST_PCI_POSSIBLE=y # CONFIG_BCMA_HOST_PCI is not set CONFIG_BCMA_HOST_SOC=y # CONFIG_BCMA_DRIVER_GMAC_CMN is not set CONFIG_BCMA_DRIVER_GPIO=y CONFIG_BCMA_DEBUG=y # # Multifunction device drivers # CONFIG_MFD_CORE=y CONFIG_MFD_CS5535=y # CONFIG_MFD_AS3711 is not set CONFIG_PMIC_ADP5520=y # CONFIG_MFD_AAT2870_CORE is not set CONFIG_MFD_CROS_EC=y CONFIG_MFD_CROS_EC_I2C=y # CONFIG_MFD_CROS_EC_SPI is not set CONFIG_PMIC_DA903X=y CONFIG_PMIC_DA9052=y CONFIG_MFD_DA9052_SPI=y CONFIG_MFD_DA9052_I2C=y # CONFIG_MFD_DA9055 is not set # CONFIG_MFD_DA9063 is not set CONFIG_MFD_MC13783=y CONFIG_MFD_MC13XXX=y CONFIG_MFD_MC13XXX_SPI=y CONFIG_MFD_MC13XXX_I2C=y CONFIG_HTC_PASIC3=y CONFIG_HTC_I2CPLD=y CONFIG_LPC_ICH=y CONFIG_LPC_SCH=y CONFIG_MFD_INTEL_MSIC=y # CONFIG_MFD_JANZ_CMODIO is not set CONFIG_MFD_KEMPLD=y CONFIG_MFD_88PM800=y CONFIG_MFD_88PM805=y CONFIG_MFD_88PM860X=y # CONFIG_MFD_MAX77686 is not set CONFIG_MFD_MAX77693=y # CONFIG_MFD_MAX8907 is not set # CONFIG_MFD_MAX8925 is not set CONFIG_MFD_MAX8997=y CONFIG_MFD_MAX8998=y CONFIG_EZX_PCAP=y CONFIG_MFD_RETU=y CONFIG_MFD_PCF50633=y # CONFIG_PCF50633_ADC is not set CONFIG_PCF50633_GPIO=y # CONFIG_UCB1400_CORE is not set CONFIG_MFD_RDC321X=y # CONFIG_MFD_RTSX_PCI is not set # CONFIG_MFD_RC5T583 is not set CONFIG_MFD_SEC_CORE=y # CONFIG_MFD_SI476X_CORE is not set # CONFIG_MFD_SM501 is not set CONFIG_MFD_SMSC=y CONFIG_ABX500_CORE=y CONFIG_AB3100_CORE=y CONFIG_AB3100_OTP=y CONFIG_MFD_STMPE=y # # STMicroelectronics STMPE Interface Drivers # # CONFIG_STMPE_I2C is not set # CONFIG_STMPE_SPI is not set # CONFIG_MFD_SYSCON is not set # CONFIG_MFD_TI_AM335X_TSCADC is not set CONFIG_MFD_LP8788=y CONFIG_MFD_PALMAS=y # CONFIG_TPS6105X is not set CONFIG_TPS65010=y # CONFIG_TPS6507X is not set CONFIG_MFD_TPS65090=y CONFIG_MFD_TPS65217=y # CONFIG_MFD_TPS6586X is not set CONFIG_MFD_TPS65910=y CONFIG_MFD_TPS65912=y CONFIG_MFD_TPS65912_I2C=y # CONFIG_MFD_TPS65912_SPI is not set # CONFIG_MFD_TPS80031 is not set # CONFIG_TWL4030_CORE is not set CONFIG_TWL6040_CORE=y CONFIG_MFD_WL1273_CORE=y CONFIG_MFD_LM3533=y # CONFIG_MFD_TIMBERDALE is not set CONFIG_MFD_TC3589X=y # CONFIG_MFD_TMIO is not set CONFIG_MFD_VX855=y CONFIG_MFD_ARIZONA=y # CONFIG_MFD_ARIZONA_I2C is not set CONFIG_MFD_ARIZONA_SPI=y CONFIG_MFD_WM5102=y # CONFIG_MFD_WM5110 is not set # CONFIG_MFD_WM8997 is not set CONFIG_MFD_WM8400=y CONFIG_MFD_WM831X=y CONFIG_MFD_WM831X_I2C=y CONFIG_MFD_WM831X_SPI=y CONFIG_MFD_WM8350=y CONFIG_MFD_WM8350_I2C=y CONFIG_MFD_WM8994=y CONFIG_REGULATOR=y CONFIG_REGULATOR_DEBUG=y CONFIG_REGULATOR_DUMMY=y CONFIG_REGULATOR_FIXED_VOLTAGE=y # CONFIG_REGULATOR_VIRTUAL_CONSUMER is not set CONFIG_REGULATOR_USERSPACE_CONSUMER=y # CONFIG_REGULATOR_88PM800 is not set # CONFIG_REGULATOR_88PM8607 is not set CONFIG_REGULATOR_AD5398=y CONFIG_REGULATOR_AB3100=y CONFIG_REGULATOR_ARIZONA=y CONFIG_REGULATOR_DA903X=y # CONFIG_REGULATOR_DA9052 is not set # CONFIG_REGULATOR_DA9210 is not set # CONFIG_REGULATOR_FAN53555 is not set CONFIG_REGULATOR_GPIO=y # CONFIG_REGULATOR_ISL6271A is not set # CONFIG_REGULATOR_LP3971 is not set # CONFIG_REGULATOR_LP3972 is not set CONFIG_REGULATOR_LP872X=y # CONFIG_REGULATOR_LP8755 is not set CONFIG_REGULATOR_LP8788=y CONFIG_REGULATOR_MAX1586=y # CONFIG_REGULATOR_MAX8649 is not set CONFIG_REGULATOR_MAX8660=y # CONFIG_REGULATOR_MAX8952 is not set # CONFIG_REGULATOR_MAX8973 is not set CONFIG_REGULATOR_MAX8997=y CONFIG_REGULATOR_MAX8998=y # CONFIG_REGULATOR_MAX77693 is not set CONFIG_REGULATOR_MC13XXX_CORE=y CONFIG_REGULATOR_MC13783=y CONFIG_REGULATOR_MC13892=y CONFIG_REGULATOR_PALMAS=y # CONFIG_REGULATOR_PCAP is not set CONFIG_REGULATOR_PCF50633=y CONFIG_REGULATOR_PFUZE100=y # CONFIG_REGULATOR_S2MPS11 is not set # CONFIG_REGULATOR_S5M8767 is not set CONFIG_REGULATOR_TPS51632=y # CONFIG_REGULATOR_TPS62360 is not set CONFIG_REGULATOR_TPS65023=y CONFIG_REGULATOR_TPS6507X=y CONFIG_REGULATOR_TPS65090=y CONFIG_REGULATOR_TPS65217=y CONFIG_REGULATOR_TPS6524X=y CONFIG_REGULATOR_TPS65910=y CONFIG_REGULATOR_TPS65912=y CONFIG_REGULATOR_WM831X=y CONFIG_REGULATOR_WM8350=y # CONFIG_REGULATOR_WM8400 is not set CONFIG_REGULATOR_WM8994=y CONFIG_MEDIA_SUPPORT=y # # Multimedia core support # # CONFIG_MEDIA_CAMERA_SUPPORT is not set # CONFIG_MEDIA_ANALOG_TV_SUPPORT is not set CONFIG_MEDIA_DIGITAL_TV_SUPPORT=y # CONFIG_MEDIA_RADIO_SUPPORT is not set # CONFIG_MEDIA_RC_SUPPORT is not set # CONFIG_VIDEO_ADV_DEBUG is not set # CONFIG_VIDEO_FIXED_MINOR_RANGES is not set CONFIG_DVB_CORE=y CONFIG_TTPCI_EEPROM=y CONFIG_DVB_MAX_ADAPTERS=8 CONFIG_DVB_DYNAMIC_MINORS=y # # Media drivers # CONFIG_MEDIA_PCI_SUPPORT=y # # Media capture/analog/hybrid TV support # # # Media digital TV PCI Adapters # CONFIG_DVB_BUDGET_CORE=y CONFIG_DVB_BUDGET=y # CONFIG_DVB_B2C2_FLEXCOP_PCI is not set # CONFIG_DVB_PLUTO2 is not set CONFIG_DVB_PT1=y CONFIG_DVB_NGENE=y CONFIG_DVB_DDBRIDGE=y # # Supported MMC/SDIO adapters # # CONFIG_SMS_SDIO_DRV is not set # # Supported FireWire (IEEE 1394) Adapters # CONFIG_DVB_FIREDTV=y CONFIG_DVB_FIREDTV_INPUT=y CONFIG_VIDEO_SAA7146=y # # Media ancillary drivers (tuners, sensors, i2c, frontends) # # CONFIG_MEDIA_SUBDRV_AUTOSELECT is not set # # Customize TV tuners # # CONFIG_MEDIA_TUNER_SIMPLE is not set CONFIG_MEDIA_TUNER_TDA8290=y CONFIG_MEDIA_TUNER_TDA827X=y CONFIG_MEDIA_TUNER_TDA18271=y CONFIG_MEDIA_TUNER_TDA9887=y CONFIG_MEDIA_TUNER_TEA5761=y CONFIG_MEDIA_TUNER_TEA5767=y CONFIG_MEDIA_TUNER_MT20XX=y CONFIG_MEDIA_TUNER_MT2060=y CONFIG_MEDIA_TUNER_MT2063=y CONFIG_MEDIA_TUNER_MT2266=y CONFIG_MEDIA_TUNER_MT2131=y CONFIG_MEDIA_TUNER_QT1010=y # CONFIG_MEDIA_TUNER_XC2028 is not set # CONFIG_MEDIA_TUNER_XC5000 is not set CONFIG_MEDIA_TUNER_XC4000=y # CONFIG_MEDIA_TUNER_MXL5005S is not set CONFIG_MEDIA_TUNER_MXL5007T=y CONFIG_MEDIA_TUNER_MC44S803=y CONFIG_MEDIA_TUNER_MAX2165=y # CONFIG_MEDIA_TUNER_TDA18218 is not set CONFIG_MEDIA_TUNER_FC0011=y CONFIG_MEDIA_TUNER_FC0012=y # CONFIG_MEDIA_TUNER_FC0013 is not set CONFIG_MEDIA_TUNER_TDA18212=y CONFIG_MEDIA_TUNER_E4000=y # CONFIG_MEDIA_TUNER_FC2580 is not set CONFIG_MEDIA_TUNER_TUA9001=y CONFIG_MEDIA_TUNER_IT913X=y CONFIG_MEDIA_TUNER_R820T=y # # Customise DVB Frontends # # # Multistandard (satellite) frontends # # CONFIG_DVB_STB0899 is not set CONFIG_DVB_STB6100=y # CONFIG_DVB_STV090x is not set CONFIG_DVB_STV6110x=y # # Multistandard (cable + terrestrial) frontends # CONFIG_DVB_DRXK=y CONFIG_DVB_TDA18271C2DD=y # # DVB-S (satellite) frontends # # CONFIG_DVB_CX24110 is not set CONFIG_DVB_CX24123=y CONFIG_DVB_MT312=y CONFIG_DVB_ZL10036=y CONFIG_DVB_ZL10039=y CONFIG_DVB_S5H1420=y CONFIG_DVB_STV0288=y CONFIG_DVB_STB6000=y # CONFIG_DVB_STV0299 is not set CONFIG_DVB_STV6110=y # CONFIG_DVB_STV0900 is not set CONFIG_DVB_TDA8083=y # CONFIG_DVB_TDA10086 is not set CONFIG_DVB_TDA8261=y # CONFIG_DVB_VES1X93 is not set CONFIG_DVB_TUNER_ITD1000=y # CONFIG_DVB_TUNER_CX24113 is not set CONFIG_DVB_TDA826X=y CONFIG_DVB_TUA6100=y CONFIG_DVB_CX24116=y CONFIG_DVB_SI21XX=y CONFIG_DVB_TS2020=y CONFIG_DVB_DS3000=y CONFIG_DVB_MB86A16=y # CONFIG_DVB_TDA10071 is not set # # DVB-T (terrestrial) frontends # # CONFIG_DVB_SP8870 is not set CONFIG_DVB_SP887X=y # CONFIG_DVB_CX22700 is not set CONFIG_DVB_CX22702=y # CONFIG_DVB_S5H1432 is not set CONFIG_DVB_DRXD=y CONFIG_DVB_L64781=y CONFIG_DVB_TDA1004X=y CONFIG_DVB_NXT6000=y CONFIG_DVB_MT352=y # CONFIG_DVB_ZL10353 is not set CONFIG_DVB_DIB3000MB=y CONFIG_DVB_DIB3000MC=y # CONFIG_DVB_DIB7000M is not set CONFIG_DVB_DIB7000P=y CONFIG_DVB_DIB9000=y CONFIG_DVB_TDA10048=y # CONFIG_DVB_AF9013 is not set # CONFIG_DVB_EC100 is not set CONFIG_DVB_HD29L2=y CONFIG_DVB_STV0367=y CONFIG_DVB_CXD2820R=y CONFIG_DVB_RTL2830=y CONFIG_DVB_RTL2832=y # # DVB-C (cable) frontends # # CONFIG_DVB_VES1820 is not set CONFIG_DVB_TDA10021=y # CONFIG_DVB_TDA10023 is not set # CONFIG_DVB_STV0297 is not set # # ATSC (North American/Korean Terrestrial/Cable DTV) frontends # # CONFIG_DVB_NXT200X is not set # CONFIG_DVB_OR51211 is not set CONFIG_DVB_OR51132=y CONFIG_DVB_BCM3510=y # CONFIG_DVB_LGDT330X is not set # CONFIG_DVB_LGDT3305 is not set CONFIG_DVB_LG2160=y CONFIG_DVB_S5H1409=y # CONFIG_DVB_AU8522_DTV is not set # CONFIG_DVB_S5H1411 is not set # # ISDB-T (terrestrial) frontends # CONFIG_DVB_S921=y CONFIG_DVB_DIB8000=y # CONFIG_DVB_MB86A20S is not set # # Digital terrestrial only tuners/PLL # # CONFIG_DVB_PLL is not set # CONFIG_DVB_TUNER_DIB0070 is not set # CONFIG_DVB_TUNER_DIB0090 is not set # # SEC control devices for DVB-S # # CONFIG_DVB_LNBP21 is not set # CONFIG_DVB_LNBP22 is not set CONFIG_DVB_ISL6405=y # CONFIG_DVB_ISL6421 is not set CONFIG_DVB_ISL6423=y # CONFIG_DVB_A8293 is not set # CONFIG_DVB_LGS8GL5 is not set # CONFIG_DVB_LGS8GXX is not set CONFIG_DVB_ATBM8830=y # CONFIG_DVB_TDA665x is not set CONFIG_DVB_IX2505V=y CONFIG_DVB_IT913X_FE=y CONFIG_DVB_M88RS2000=y CONFIG_DVB_AF9033=y # # Tools to develop new frontends # CONFIG_DVB_DUMMY_FE=y # # Graphics support # # CONFIG_AGP is not set CONFIG_VGA_ARB=y CONFIG_VGA_ARB_MAX_GPUS=16 CONFIG_DRM=y CONFIG_DRM_KMS_HELPER=y # CONFIG_DRM_LOAD_EDID_FIRMWARE is not set CONFIG_DRM_TTM=y # # I2C encoder or helper chips # CONFIG_DRM_I2C_CH7006=y CONFIG_DRM_I2C_SIL164=y # CONFIG_DRM_I2C_NXP_TDA998X is not set CONFIG_DRM_TDFX=y CONFIG_DRM_R128=y # CONFIG_DRM_RADEON is not set # CONFIG_DRM_NOUVEAU is not set # CONFIG_DRM_MGA is not set CONFIG_DRM_VIA=y # CONFIG_DRM_SAVAGE is not set # CONFIG_DRM_VMWGFX is not set CONFIG_DRM_GMA500=y # CONFIG_DRM_GMA600 is not set # CONFIG_DRM_GMA3600 is not set CONFIG_DRM_MEDFIELD=y CONFIG_DRM_AST=y # CONFIG_DRM_MGAG200 is not set # CONFIG_DRM_CIRRUS_QEMU is not set CONFIG_DRM_QXL=y CONFIG_VGASTATE=y # CONFIG_VIDEO_OUTPUT_CONTROL is not set CONFIG_HDMI=y CONFIG_FB=y # CONFIG_FIRMWARE_EDID is not set CONFIG_FB_DDC=y CONFIG_FB_BOOT_VESA_SUPPORT=y CONFIG_FB_CFB_FILLRECT=y CONFIG_FB_CFB_COPYAREA=y CONFIG_FB_CFB_IMAGEBLIT=y # CONFIG_FB_CFB_REV_PIXELS_IN_BYTE is not set CONFIG_FB_SYS_FILLRECT=y CONFIG_FB_SYS_COPYAREA=y CONFIG_FB_SYS_IMAGEBLIT=y # CONFIG_FB_FOREIGN_ENDIAN is not set CONFIG_FB_SYS_FOPS=y CONFIG_FB_DEFERRED_IO=y CONFIG_FB_HECUBA=y CONFIG_FB_SVGALIB=y # CONFIG_FB_MACMODES is not set CONFIG_FB_BACKLIGHT=y CONFIG_FB_MODE_HELPERS=y CONFIG_FB_TILEBLITTING=y # # Frame buffer hardware drivers # CONFIG_FB_CIRRUS=y CONFIG_FB_PM2=y CONFIG_FB_PM2_FIFO_DISCONNECT=y # CONFIG_FB_CYBER2000 is not set CONFIG_FB_ARC=y # CONFIG_FB_ASILIANT is not set CONFIG_FB_IMSTT=y # CONFIG_FB_VGA16 is not set CONFIG_FB_VESA=y CONFIG_FB_N411=y CONFIG_FB_HGA=y CONFIG_FB_S1D13XXX=y # CONFIG_FB_NVIDIA is not set CONFIG_FB_RIVA=y # CONFIG_FB_RIVA_I2C is not set CONFIG_FB_RIVA_DEBUG=y CONFIG_FB_RIVA_BACKLIGHT=y CONFIG_FB_I740=y CONFIG_FB_LE80578=y CONFIG_FB_CARILLO_RANCH=y # CONFIG_FB_MATROX is not set CONFIG_FB_RADEON=y # CONFIG_FB_RADEON_I2C is not set # CONFIG_FB_RADEON_BACKLIGHT is not set CONFIG_FB_RADEON_DEBUG=y CONFIG_FB_ATY128=y # CONFIG_FB_ATY128_BACKLIGHT is not set # CONFIG_FB_ATY is not set CONFIG_FB_S3=y # CONFIG_FB_S3_DDC is not set CONFIG_FB_SAVAGE=y # CONFIG_FB_SAVAGE_I2C is not set # CONFIG_FB_SAVAGE_ACCEL is not set # CONFIG_FB_SIS is not set CONFIG_FB_VIA=y # CONFIG_FB_VIA_DIRECT_PROCFS is not set # CONFIG_FB_VIA_X_COMPATIBILITY is not set CONFIG_FB_NEOMAGIC=y # CONFIG_FB_KYRO is not set # CONFIG_FB_3DFX is not set CONFIG_FB_VOODOO1=y CONFIG_FB_VT8623=y # CONFIG_FB_TRIDENT is not set CONFIG_FB_ARK=y CONFIG_FB_PM3=y CONFIG_FB_CARMINE=y CONFIG_FB_CARMINE_DRAM_EVAL=y # CONFIG_CARMINE_DRAM_CUSTOM is not set # CONFIG_FB_GEODE is not set CONFIG_FB_TMIO=y CONFIG_FB_TMIO_ACCELL=y CONFIG_FB_GOLDFISH=y CONFIG_FB_VIRTUAL=y CONFIG_FB_METRONOME=y CONFIG_FB_MB862XX=y CONFIG_FB_MB862XX_PCI_GDC=y # CONFIG_FB_MB862XX_I2C is not set # CONFIG_FB_BROADSHEET is not set # CONFIG_FB_AUO_K190X is not set # CONFIG_FB_SIMPLE is not set # CONFIG_EXYNOS_VIDEO is not set CONFIG_BACKLIGHT_LCD_SUPPORT=y # CONFIG_LCD_CLASS_DEVICE is not set CONFIG_BACKLIGHT_CLASS_DEVICE=y CONFIG_BACKLIGHT_GENERIC=y # CONFIG_BACKLIGHT_LM3533 is not set CONFIG_BACKLIGHT_PWM=y CONFIG_BACKLIGHT_DA903X=y # CONFIG_BACKLIGHT_DA9052 is not set CONFIG_BACKLIGHT_SAHARA=y CONFIG_BACKLIGHT_WM831X=y # CONFIG_BACKLIGHT_ADP5520 is not set CONFIG_BACKLIGHT_ADP8860=y CONFIG_BACKLIGHT_ADP8870=y # CONFIG_BACKLIGHT_88PM860X is not set # CONFIG_BACKLIGHT_PCF50633 is not set CONFIG_BACKLIGHT_LM3630=y CONFIG_BACKLIGHT_LM3639=y CONFIG_BACKLIGHT_LP855X=y CONFIG_BACKLIGHT_LP8788=y # CONFIG_BACKLIGHT_OT200 is not set CONFIG_BACKLIGHT_TPS65217=y CONFIG_BACKLIGHT_GPIO=y # CONFIG_BACKLIGHT_LV5207LP is not set # CONFIG_BACKLIGHT_BD6107 is not set # # Console display driver support # CONFIG_VGA_CONSOLE=y # CONFIG_VGACON_SOFT_SCROLLBACK is not set CONFIG_DUMMY_CONSOLE=y CONFIG_FRAMEBUFFER_CONSOLE=y CONFIG_FRAMEBUFFER_CONSOLE_DETECT_PRIMARY=y # CONFIG_FRAMEBUFFER_CONSOLE_ROTATION is not set CONFIG_LOGO=y CONFIG_LOGO_LINUX_MONO=y # CONFIG_LOGO_LINUX_VGA16 is not set CONFIG_LOGO_LINUX_CLUT224=y CONFIG_FB_SSD1307=y CONFIG_SOUND=y CONFIG_SOUND_OSS_CORE=y CONFIG_SOUND_OSS_CORE_PRECLAIM=y CONFIG_SND=y CONFIG_SND_TIMER=y CONFIG_SND_PCM=y CONFIG_SND_DMAENGINE_PCM=y CONFIG_SND_HWDEP=y CONFIG_SND_RAWMIDI=y CONFIG_SND_COMPRESS_OFFLOAD=y CONFIG_SND_JACK=y CONFIG_SND_SEQUENCER=y CONFIG_SND_SEQ_DUMMY=y CONFIG_SND_OSSEMUL=y # CONFIG_SND_MIXER_OSS is not set CONFIG_SND_PCM_OSS=y CONFIG_SND_PCM_OSS_PLUGINS=y # CONFIG_SND_SEQUENCER_OSS is not set # CONFIG_SND_DYNAMIC_MINORS is not set CONFIG_SND_SUPPORT_OLD_API=y # CONFIG_SND_VERBOSE_PROCFS is not set CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y CONFIG_SND_VMASTER=y CONFIG_SND_DMA_SGBUF=y CONFIG_SND_RAWMIDI_SEQ=y CONFIG_SND_OPL3_LIB_SEQ=y # CONFIG_SND_OPL4_LIB_SEQ is not set # CONFIG_SND_SBAWE_SEQ is not set CONFIG_SND_EMU10K1_SEQ=y CONFIG_SND_MPU401_UART=y CONFIG_SND_OPL3_LIB=y CONFIG_SND_VX_LIB=y CONFIG_SND_AC97_CODEC=y # CONFIG_SND_DRIVERS is not set CONFIG_SND_SB_COMMON=y CONFIG_SND_SB16_DSP=y CONFIG_SND_PCI=y # CONFIG_SND_AD1889 is not set CONFIG_SND_ALS300=y CONFIG_SND_ALS4000=y # CONFIG_SND_ALI5451 is not set CONFIG_SND_ASIHPI=y # CONFIG_SND_ATIIXP is not set CONFIG_SND_ATIIXP_MODEM=y # CONFIG_SND_AU8810 is not set # CONFIG_SND_AU8820 is not set # CONFIG_SND_AU8830 is not set # CONFIG_SND_AW2 is not set CONFIG_SND_AZT3328=y CONFIG_SND_BT87X=y CONFIG_SND_BT87X_OVERCLOCK=y # CONFIG_SND_CA0106 is not set CONFIG_SND_CMIPCI=y CONFIG_SND_OXYGEN_LIB=y CONFIG_SND_OXYGEN=y CONFIG_SND_CS4281=y CONFIG_SND_CS46XX=y CONFIG_SND_CS46XX_NEW_DSP=y CONFIG_SND_CS5530=y CONFIG_SND_CS5535AUDIO=y # CONFIG_SND_CTXFI is not set CONFIG_SND_DARLA20=y # CONFIG_SND_GINA20 is not set # CONFIG_SND_LAYLA20 is not set # CONFIG_SND_DARLA24 is not set # CONFIG_SND_GINA24 is not set # CONFIG_SND_LAYLA24 is not set # CONFIG_SND_MONA is not set CONFIG_SND_MIA=y # CONFIG_SND_ECHO3G is not set CONFIG_SND_INDIGO=y CONFIG_SND_INDIGOIO=y # CONFIG_SND_INDIGODJ is not set CONFIG_SND_INDIGOIOX=y # CONFIG_SND_INDIGODJX is not set CONFIG_SND_EMU10K1=y CONFIG_SND_EMU10K1X=y CONFIG_SND_ENS1370=y CONFIG_SND_ENS1371=y CONFIG_SND_ES1938=y CONFIG_SND_ES1968=y CONFIG_SND_ES1968_INPUT=y CONFIG_SND_FM801=y # CONFIG_SND_HDA_INTEL is not set CONFIG_SND_HDSP=y # # Don't forget to add built-in firmwares for HDSP driver # # CONFIG_SND_HDSPM is not set # CONFIG_SND_ICE1712 is not set CONFIG_SND_ICE1724=y # CONFIG_SND_INTEL8X0 is not set # CONFIG_SND_INTEL8X0M is not set # CONFIG_SND_KORG1212 is not set CONFIG_SND_LOLA=y CONFIG_SND_LX6464ES=y # CONFIG_SND_MAESTRO3 is not set # CONFIG_SND_MIXART is not set CONFIG_SND_NM256=y CONFIG_SND_PCXHR=y CONFIG_SND_RIPTIDE=y CONFIG_SND_RME32=y CONFIG_SND_RME96=y # CONFIG_SND_RME9652 is not set CONFIG_SND_SIS7019=y CONFIG_SND_SONICVIBES=y # CONFIG_SND_TRIDENT is not set CONFIG_SND_VIA82XX=y CONFIG_SND_VIA82XX_MODEM=y CONFIG_SND_VIRTUOSO=y CONFIG_SND_VX222=y # CONFIG_SND_YMFPCI is not set CONFIG_SND_SPI=y CONFIG_SND_AT73C213=y CONFIG_SND_AT73C213_TARGET_BITRATE=48000 CONFIG_SND_FIREWIRE=y CONFIG_SND_FIREWIRE_LIB=y # CONFIG_SND_FIREWIRE_SPEAKERS is not set # CONFIG_SND_ISIGHT is not set CONFIG_SND_SCS1X=y CONFIG_SND_SOC=y CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM=y CONFIG_SND_ATMEL_SOC=y CONFIG_SND_SOC_FSL_SSI=y CONFIG_SND_SOC_FSL_SPDIF=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_IMX_PCM_DMA=y CONFIG_SND_SOC_IMX_AUDMUX=y CONFIG_SND_SOC_IMX_WM8962=y # CONFIG_SND_SOC_IMX_SGTL5000 is not set CONFIG_SND_SOC_IMX_SPDIF=y CONFIG_SND_KIRKWOOD_SOC=y CONFIG_SND_KIRKWOOD_SOC_OPENRD=y CONFIG_SND_KIRKWOOD_SOC_T5325=y CONFIG_SND_MFLD_MACHINE=y CONFIG_SND_SST_PLATFORM=y CONFIG_SND_SOC_TEGRA=y # CONFIG_SND_SOC_TEGRA_RT5640 is not set CONFIG_SND_SOC_TEGRA_WM8753=y CONFIG_SND_SOC_TEGRA_WM8903=y CONFIG_SND_SOC_TEGRA_TRIMSLICE=y CONFIG_SND_SOC_TEGRA_ALC5632=y CONFIG_SND_SOC_I2C_AND_SPI=y # CONFIG_SND_SOC_ALL_CODECS is not set CONFIG_SND_SOC_ALC5623=y CONFIG_SND_SOC_ALC5632=y CONFIG_SND_SOC_CS42L51=y CONFIG_SND_SOC_SN95031=y CONFIG_SND_SOC_SPDIF=y CONFIG_SND_SOC_TLV320AIC23=y CONFIG_SND_SOC_WM8753=y CONFIG_SND_SOC_WM8903=y CONFIG_SND_SOC_WM8962=y # CONFIG_SND_SIMPLE_CARD is not set # CONFIG_SOUND_PRIME is not set CONFIG_AC97_BUS=y # # HID support # CONFIG_HID=y # CONFIG_HID_BATTERY_STRENGTH is not set # CONFIG_HIDRAW is not set CONFIG_UHID=y CONFIG_HID_GENERIC=y # # Special HID drivers # CONFIG_HID_A4TECH=y CONFIG_HID_ACRUX=y CONFIG_HID_ACRUX_FF=y CONFIG_HID_APPLE=y CONFIG_HID_AUREAL=y # CONFIG_HID_BELKIN is not set CONFIG_HID_CHERRY=y CONFIG_HID_CHICONY=y CONFIG_HID_PRODIKEYS=y # CONFIG_HID_CYPRESS is not set CONFIG_HID_DRAGONRISE=y # CONFIG_DRAGONRISE_FF is not set # CONFIG_HID_EMS_FF is not set # CONFIG_HID_ELECOM is not set # CONFIG_HID_EZKEY is not set CONFIG_HID_KEYTOUCH=y CONFIG_HID_KYE=y CONFIG_HID_UCLOGIC=y CONFIG_HID_WALTOP=y CONFIG_HID_GYRATION=y # CONFIG_HID_ICADE is not set CONFIG_HID_TWINHAN=y CONFIG_HID_KENSINGTON=y # CONFIG_HID_LCPOWER is not set CONFIG_HID_LOGITECH=y CONFIG_HID_LOGITECH_DJ=y CONFIG_LOGITECH_FF=y CONFIG_LOGIRUMBLEPAD2_FF=y # CONFIG_LOGIG940_FF is not set CONFIG_LOGIWHEELS_FF=y CONFIG_HID_MAGICMOUSE=y # CONFIG_HID_MICROSOFT is not set # CONFIG_HID_MONTEREY is not set # CONFIG_HID_MULTITOUCH is not set CONFIG_HID_ORTEK=y # CONFIG_HID_PANTHERLORD is not set CONFIG_HID_PETALYNX=y # CONFIG_HID_PICOLCD is not set CONFIG_HID_PRIMAX=y CONFIG_HID_SAITEK=y CONFIG_HID_SAMSUNG=y CONFIG_HID_SPEEDLINK=y # CONFIG_HID_STEELSERIES is not set CONFIG_HID_SUNPLUS=y CONFIG_HID_GREENASIA=y CONFIG_GREENASIA_FF=y CONFIG_HID_SMARTJOYPLUS=y # CONFIG_SMARTJOYPLUS_FF is not set CONFIG_HID_TIVO=y CONFIG_HID_TOPSEED=y CONFIG_HID_THINGM=y # CONFIG_HID_THRUSTMASTER is not set CONFIG_HID_WACOM=y CONFIG_HID_WIIMOTE=y CONFIG_HID_XINMO=y # CONFIG_HID_ZEROPLUS is not set # CONFIG_HID_ZYDACRON is not set CONFIG_HID_SENSOR_HUB=y # # I2C HID support # CONFIG_I2C_HID=y CONFIG_USB_OHCI_LITTLE_ENDIAN=y # CONFIG_USB_SUPPORT is not set # CONFIG_UWB is not set CONFIG_MMC=y # CONFIG_MMC_DEBUG is not set CONFIG_MMC_UNSAFE_RESUME=y CONFIG_MMC_CLKGATE=y # # MMC/SD/SDIO Card Drivers # CONFIG_SDIO_UART=y CONFIG_MMC_TEST=y # # MMC/SD/SDIO Host Controller Drivers # CONFIG_MMC_SDHCI=y CONFIG_MMC_SDHCI_PCI=y # CONFIG_MMC_RICOH_MMC is not set CONFIG_MMC_SDHCI_PLTFM=y # CONFIG_MMC_OMAP_HS is not set CONFIG_MMC_WBSD=y CONFIG_MMC_TIFM_SD=y CONFIG_MMC_GOLDFISH=y # CONFIG_MMC_CB710 is not set # CONFIG_MMC_VIA_SDMMC is not set CONFIG_MEMSTICK=y # CONFIG_MEMSTICK_DEBUG is not set # # MemoryStick drivers # # CONFIG_MEMSTICK_UNSAFE_RESUME is not set # # MemoryStick Host Controller Drivers # CONFIG_MEMSTICK_TIFM_MS=y CONFIG_MEMSTICK_JMICRON_38X=y CONFIG_MEMSTICK_R592=y CONFIG_NEW_LEDS=y CONFIG_LEDS_CLASS=y # # LED drivers # # CONFIG_LEDS_88PM860X is not set CONFIG_LEDS_LM3530=y CONFIG_LEDS_LM3533=y CONFIG_LEDS_LM3642=y CONFIG_LEDS_NET48XX=y # CONFIG_LEDS_WRAP is not set CONFIG_LEDS_PCA9532=y CONFIG_LEDS_PCA9532_GPIO=y # CONFIG_LEDS_GPIO is not set CONFIG_LEDS_LP3944=y CONFIG_LEDS_LP55XX_COMMON=y # CONFIG_LEDS_LP5521 is not set # CONFIG_LEDS_LP5523 is not set CONFIG_LEDS_LP5562=y # CONFIG_LEDS_LP8501 is not set CONFIG_LEDS_LP8788=y CONFIG_LEDS_CLEVO_MAIL=y CONFIG_LEDS_PCA955X=y # CONFIG_LEDS_PCA963X is not set # CONFIG_LEDS_WM831X_STATUS is not set # CONFIG_LEDS_WM8350 is not set # CONFIG_LEDS_DA903X is not set CONFIG_LEDS_DA9052=y CONFIG_LEDS_DAC124S085=y CONFIG_LEDS_PWM=y # CONFIG_LEDS_REGULATOR is not set CONFIG_LEDS_BD2802=y # CONFIG_LEDS_INTEL_SS4200 is not set # CONFIG_LEDS_LT3593 is not set CONFIG_LEDS_ADP5520=y CONFIG_LEDS_MC13783=y # CONFIG_LEDS_TCA6507 is not set CONFIG_LEDS_MAX8997=y CONFIG_LEDS_LM355x=y CONFIG_LEDS_OT200=y # CONFIG_LEDS_BLINKM is not set # # LED Triggers # CONFIG_LEDS_TRIGGERS=y CONFIG_LEDS_TRIGGER_TIMER=y # CONFIG_LEDS_TRIGGER_ONESHOT is not set CONFIG_LEDS_TRIGGER_HEARTBEAT=y CONFIG_LEDS_TRIGGER_BACKLIGHT=y CONFIG_LEDS_TRIGGER_CPU=y CONFIG_LEDS_TRIGGER_GPIO=y CONFIG_LEDS_TRIGGER_DEFAULT_ON=y # # iptables trigger is under Netfilter config (LED target) # # CONFIG_LEDS_TRIGGER_TRANSIENT is not set # CONFIG_LEDS_TRIGGER_CAMERA is not set # CONFIG_ACCESSIBILITY is not set # CONFIG_EDAC is not set CONFIG_RTC_LIB=y CONFIG_RTC_CLASS=y # CONFIG_RTC_HCTOSYS is not set # CONFIG_RTC_SYSTOHC is not set CONFIG_RTC_DEBUG=y # # RTC interfaces # # CONFIG_RTC_INTF_SYSFS is not set # CONFIG_RTC_INTF_PROC is not set CONFIG_RTC_INTF_DEV=y CONFIG_RTC_INTF_DEV_UIE_EMUL=y # CONFIG_RTC_DRV_TEST is not set # # I2C RTC drivers # CONFIG_RTC_DRV_88PM860X=y CONFIG_RTC_DRV_88PM80X=y CONFIG_RTC_DRV_DS1307=y CONFIG_RTC_DRV_DS1374=y # CONFIG_RTC_DRV_DS1672 is not set # CONFIG_RTC_DRV_DS3232 is not set CONFIG_RTC_DRV_LP8788=y # CONFIG_RTC_DRV_MAX6900 is not set CONFIG_RTC_DRV_MAX8998=y CONFIG_RTC_DRV_MAX8997=y CONFIG_RTC_DRV_RS5C372=y CONFIG_RTC_DRV_ISL1208=y # CONFIG_RTC_DRV_ISL12022 is not set CONFIG_RTC_DRV_X1205=y # CONFIG_RTC_DRV_PALMAS is not set CONFIG_RTC_DRV_PCF2127=y # CONFIG_RTC_DRV_PCF8523 is not set CONFIG_RTC_DRV_PCF8563=y # CONFIG_RTC_DRV_PCF8583 is not set # CONFIG_RTC_DRV_M41T80 is not set # CONFIG_RTC_DRV_BQ32K is not set CONFIG_RTC_DRV_TPS65910=y CONFIG_RTC_DRV_S35390A=y CONFIG_RTC_DRV_FM3130=y CONFIG_RTC_DRV_RX8581=y CONFIG_RTC_DRV_RX8025=y # CONFIG_RTC_DRV_EM3027 is not set CONFIG_RTC_DRV_RV3029C2=y # # SPI RTC drivers # CONFIG_RTC_DRV_M41T93=y CONFIG_RTC_DRV_M41T94=y CONFIG_RTC_DRV_DS1305=y # CONFIG_RTC_DRV_DS1390 is not set CONFIG_RTC_DRV_MAX6902=y # CONFIG_RTC_DRV_R9701 is not set CONFIG_RTC_DRV_RS5C348=y CONFIG_RTC_DRV_DS3234=y CONFIG_RTC_DRV_PCF2123=y CONFIG_RTC_DRV_RX4581=y # # Platform RTC drivers # CONFIG_RTC_DRV_CMOS=y CONFIG_RTC_DRV_VRTC=y CONFIG_RTC_DRV_DS1286=y CONFIG_RTC_DRV_DS1511=y CONFIG_RTC_DRV_DS1553=y # CONFIG_RTC_DRV_DS1742 is not set CONFIG_RTC_DRV_DA9052=y CONFIG_RTC_DRV_STK17TA8=y CONFIG_RTC_DRV_M48T86=y CONFIG_RTC_DRV_M48T35=y CONFIG_RTC_DRV_M48T59=y # CONFIG_RTC_DRV_MSM6242 is not set # CONFIG_RTC_DRV_BQ4802 is not set # CONFIG_RTC_DRV_RP5C01 is not set # CONFIG_RTC_DRV_V3020 is not set CONFIG_RTC_DRV_DS2404=y # CONFIG_RTC_DRV_WM831X is not set CONFIG_RTC_DRV_WM8350=y CONFIG_RTC_DRV_PCF50633=y CONFIG_RTC_DRV_AB3100=y # # on-CPU RTC drivers # # CONFIG_RTC_DRV_PCAP is not set CONFIG_RTC_DRV_MC13XXX=y CONFIG_RTC_DRV_SNVS=y # CONFIG_RTC_DRV_MOXART is not set # # HID Sensor RTC drivers # # CONFIG_DMADEVICES is not set CONFIG_AUXDISPLAY=y CONFIG_KS0108=y CONFIG_KS0108_PORT=0x378 CONFIG_KS0108_DELAY=2 CONFIG_CFAG12864B=y CONFIG_CFAG12864B_RATE=20 CONFIG_UIO=y CONFIG_UIO_CIF=y CONFIG_UIO_PDRV_GENIRQ=y # CONFIG_UIO_DMEM_GENIRQ is not set # CONFIG_UIO_AEC is not set CONFIG_UIO_SERCOS3=y CONFIG_UIO_PCI_GENERIC=y CONFIG_UIO_NETX=y CONFIG_UIO_MF624=y # CONFIG_VIRT_DRIVERS is not set CONFIG_VIRTIO=y # # Virtio drivers # CONFIG_VIRTIO_PCI=y CONFIG_VIRTIO_BALLOON=y CONFIG_VIRTIO_MMIO=y CONFIG_VIRTIO_MMIO_CMDLINE_DEVICES=y # # Microsoft Hyper-V guest support # # CONFIG_STAGING is not set CONFIG_X86_PLATFORM_DEVICES=y # CONFIG_CHROMEOS_LAPTOP is not set CONFIG_DELL_LAPTOP=y CONFIG_SENSORS_HDAPS=y CONFIG_INTEL_SCU_IPC=y CONFIG_INTEL_SCU_IPC_UTIL=y CONFIG_GPIO_INTEL_PMIC=y CONFIG_INTEL_MID_POWER_BUTTON=y CONFIG_INTEL_MFLD_THERMAL=y # CONFIG_IBM_RTL is not set CONFIG_SAMSUNG_LAPTOP=y CONFIG_GOLDFISH_PIPE=y # # Hardware Spinlock drivers # CONFIG_CLKSRC_I8253=y CONFIG_CLKEVT_I8253=y CONFIG_CLKBLD_I8253=y CONFIG_DW_APB_TIMER=y # CONFIG_MAILBOX is not set CONFIG_IOMMU_SUPPORT=y CONFIG_OF_IOMMU=y # # Remoteproc drivers # CONFIG_REMOTEPROC=y CONFIG_STE_MODEM_RPROC=y # # Rpmsg drivers # CONFIG_PM_DEVFREQ=y # # DEVFREQ Governors # CONFIG_DEVFREQ_GOV_SIMPLE_ONDEMAND=y CONFIG_DEVFREQ_GOV_PERFORMANCE=y # CONFIG_DEVFREQ_GOV_POWERSAVE is not set # CONFIG_DEVFREQ_GOV_USERSPACE is not set # # DEVFREQ Drivers # CONFIG_EXTCON=y # # Extcon Device Drivers # CONFIG_OF_EXTCON=y # CONFIG_EXTCON_GPIO is not set CONFIG_EXTCON_ADC_JACK=y CONFIG_EXTCON_MAX77693=y CONFIG_EXTCON_MAX8997=y CONFIG_EXTCON_ARIZONA=y # CONFIG_EXTCON_PALMAS is not set # CONFIG_MEMORY is not set CONFIG_IIO=y CONFIG_IIO_BUFFER=y CONFIG_IIO_BUFFER_CB=y CONFIG_IIO_KFIFO_BUF=y CONFIG_IIO_TRIGGERED_BUFFER=y CONFIG_IIO_TRIGGER=y CONFIG_IIO_CONSUMERS_PER_TRIGGER=2 # # Accelerometers # # CONFIG_BMA180 is not set # CONFIG_HID_SENSOR_ACCEL_3D is not set # CONFIG_IIO_ST_ACCEL_3AXIS is not set CONFIG_KXSD9=y # # Analog to digital converters # CONFIG_AD_SIGMA_DELTA=y # CONFIG_AD7266 is not set # CONFIG_AD7298 is not set CONFIG_AD7476=y # CONFIG_AD7791 is not set CONFIG_AD7793=y # CONFIG_AD7887 is not set CONFIG_AD7923=y CONFIG_EXYNOS_ADC=y CONFIG_LP8788_ADC=y CONFIG_MAX1363=y # CONFIG_MCP320X is not set # CONFIG_NAU7802 is not set CONFIG_TI_ADC081C=y # # Amplifiers # # CONFIG_AD8366 is not set # # Hid Sensor IIO Common # CONFIG_HID_SENSOR_IIO_COMMON=y CONFIG_HID_SENSOR_IIO_TRIGGER=y CONFIG_HID_SENSOR_ENUM_BASE_QUIRKS=y CONFIG_IIO_ST_SENSORS_I2C=y CONFIG_IIO_ST_SENSORS_SPI=y CONFIG_IIO_ST_SENSORS_CORE=y # # Digital to analog converters # CONFIG_AD5064=y CONFIG_AD5360=y CONFIG_AD5380=y # CONFIG_AD5421 is not set CONFIG_AD5446=y # CONFIG_AD5449 is not set CONFIG_AD5504=y CONFIG_AD5624R_SPI=y CONFIG_AD5686=y CONFIG_AD5755=y # CONFIG_AD5764 is not set CONFIG_AD5791=y # CONFIG_AD7303 is not set CONFIG_MAX517=y CONFIG_MCP4725=y # # Frequency Synthesizers DDS/PLL # # # Clock Generator/Distribution # CONFIG_AD9523=y # # Phase-Locked Loop (PLL) frequency synthesizers # # CONFIG_ADF4350 is not set # # Digital gyroscope sensors # CONFIG_ADIS16080=y # CONFIG_ADIS16130 is not set CONFIG_ADIS16136=y # CONFIG_ADIS16260 is not set CONFIG_ADXRS450=y CONFIG_HID_SENSOR_GYRO_3D=y CONFIG_IIO_ST_GYRO_3AXIS=y CONFIG_IIO_ST_GYRO_I2C_3AXIS=y CONFIG_IIO_ST_GYRO_SPI_3AXIS=y # CONFIG_ITG3200 is not set # # Inertial measurement units # CONFIG_ADIS16400=y # CONFIG_ADIS16480 is not set CONFIG_IIO_ADIS_LIB=y CONFIG_IIO_ADIS_LIB_BUFFER=y CONFIG_INV_MPU6050_IIO=y # # Light sensors # # CONFIG_ADJD_S311 is not set CONFIG_APDS9300=y # CONFIG_HID_SENSOR_ALS is not set CONFIG_SENSORS_LM3533=y CONFIG_SENSORS_TSL2563=y CONFIG_VCNL4000=y # # Magnetometer sensors # CONFIG_AK8975=y CONFIG_HID_SENSOR_MAGNETOMETER_3D=y # CONFIG_IIO_ST_MAGN_3AXIS is not set # # Triggers - standalone # CONFIG_IIO_INTERRUPT_TRIGGER=y CONFIG_IIO_SYSFS_TRIGGER=y # # Pressure sensors # CONFIG_IIO_ST_PRESS=y CONFIG_IIO_ST_PRESS_I2C=y CONFIG_IIO_ST_PRESS_SPI=y # # Temperature sensors # CONFIG_TMP006=y CONFIG_NTB=y CONFIG_VME_BUS=y # # VME Bridge Drivers # CONFIG_VME_CA91CX42=y CONFIG_VME_TSI148=y # # VME Board Drivers # CONFIG_VMIVME_7805=y # # VME Device Drivers # CONFIG_PWM=y CONFIG_PWM_SYSFS=y # CONFIG_PWM_PCA9685 is not set CONFIG_IRQCHIP=y # CONFIG_IPACK_BUS is not set CONFIG_RESET_CONTROLLER=y CONFIG_FMC=y # CONFIG_FMC_FAKEDEV is not set CONFIG_FMC_TRIVIAL=y CONFIG_FMC_WRITE_EEPROM=y CONFIG_FMC_CHARDEV=y # # Firmware Drivers # # CONFIG_EDD is not set # CONFIG_FIRMWARE_MEMMAP is not set CONFIG_DELL_RBU=y CONFIG_DCDBAS=y # CONFIG_DMIID is not set CONFIG_DMI_SYSFS=y CONFIG_ISCSI_IBFT_FIND=y # CONFIG_GOOGLE_FIRMWARE is not set # # File systems # CONFIG_DCACHE_WORD_ACCESS=y # CONFIG_FS_POSIX_ACL is not set CONFIG_FILE_LOCKING=y CONFIG_FSNOTIFY=y # CONFIG_DNOTIFY is not set CONFIG_INOTIFY_USER=y CONFIG_FANOTIFY=y # CONFIG_QUOTA is not set # CONFIG_QUOTACTL is not set CONFIG_AUTOFS4_FS=y # CONFIG_FUSE_FS is not set # # Caches # # CONFIG_FSCACHE is not set # # Pseudo filesystems # CONFIG_PROC_FS=y # CONFIG_PROC_KCORE is not set # CONFIG_PROC_SYSCTL is not set # CONFIG_PROC_PAGE_MONITOR is not set CONFIG_SYSFS=y # CONFIG_TMPFS is not set # CONFIG_HUGETLBFS is not set # CONFIG_HUGETLB_PAGE is not set CONFIG_CONFIGFS_FS=y # CONFIG_MISC_FILESYSTEMS is not set CONFIG_NLS=y CONFIG_NLS_DEFAULT="iso8859-1" CONFIG_NLS_CODEPAGE_437=y CONFIG_NLS_CODEPAGE_737=y CONFIG_NLS_CODEPAGE_775=y # CONFIG_NLS_CODEPAGE_850 is not set # CONFIG_NLS_CODEPAGE_852 is not set CONFIG_NLS_CODEPAGE_855=y # CONFIG_NLS_CODEPAGE_857 is not set CONFIG_NLS_CODEPAGE_860=y CONFIG_NLS_CODEPAGE_861=y # CONFIG_NLS_CODEPAGE_862 is not set CONFIG_NLS_CODEPAGE_863=y CONFIG_NLS_CODEPAGE_864=y CONFIG_NLS_CODEPAGE_865=y CONFIG_NLS_CODEPAGE_866=y CONFIG_NLS_CODEPAGE_869=y CONFIG_NLS_CODEPAGE_936=y # CONFIG_NLS_CODEPAGE_950 is not set CONFIG_NLS_CODEPAGE_932=y CONFIG_NLS_CODEPAGE_949=y CONFIG_NLS_CODEPAGE_874=y # CONFIG_NLS_ISO8859_8 is not set # CONFIG_NLS_CODEPAGE_1250 is not set # CONFIG_NLS_CODEPAGE_1251 is not set # CONFIG_NLS_ASCII is not set CONFIG_NLS_ISO8859_1=y CONFIG_NLS_ISO8859_2=y CONFIG_NLS_ISO8859_3=y CONFIG_NLS_ISO8859_4=y # CONFIG_NLS_ISO8859_5 is not set CONFIG_NLS_ISO8859_6=y # CONFIG_NLS_ISO8859_7 is not set CONFIG_NLS_ISO8859_9=y CONFIG_NLS_ISO8859_13=y CONFIG_NLS_ISO8859_14=y CONFIG_NLS_ISO8859_15=y # CONFIG_NLS_KOI8_R is not set CONFIG_NLS_KOI8_U=y CONFIG_NLS_MAC_ROMAN=y CONFIG_NLS_MAC_CELTIC=y # CONFIG_NLS_MAC_CENTEURO is not set CONFIG_NLS_MAC_CROATIAN=y CONFIG_NLS_MAC_CYRILLIC=y # CONFIG_NLS_MAC_GAELIC is not set CONFIG_NLS_MAC_GREEK=y CONFIG_NLS_MAC_ICELAND=y CONFIG_NLS_MAC_INUIT=y # CONFIG_NLS_MAC_ROMANIAN is not set CONFIG_NLS_MAC_TURKISH=y CONFIG_NLS_UTF8=y # # Kernel hacking # CONFIG_TRACE_IRQFLAGS_SUPPORT=y # # printk and dmesg options # CONFIG_DEFAULT_MESSAGE_LOGLEVEL=4 # # Compile-time checks and compiler options # CONFIG_DEBUG_INFO=y CONFIG_DEBUG_INFO_REDUCED=y # CONFIG_ENABLE_WARN_DEPRECATED is not set CONFIG_ENABLE_MUST_CHECK=y CONFIG_FRAME_WARN=1024 # CONFIG_STRIP_ASM_SYMS is not set CONFIG_READABLE_ASM=y CONFIG_UNUSED_SYMBOLS=y CONFIG_DEBUG_FS=y # CONFIG_HEADERS_CHECK is not set # CONFIG_DEBUG_SECTION_MISMATCH is not set CONFIG_ARCH_WANT_FRAME_POINTERS=y CONFIG_FRAME_POINTER=y # CONFIG_DEBUG_FORCE_WEAK_PER_CPU is not set # CONFIG_MAGIC_SYSRQ is not set CONFIG_DEBUG_KERNEL=y # # Memory Debugging # # CONFIG_DEBUG_PAGEALLOC is not set # CONFIG_DEBUG_OBJECTS is not set CONFIG_SLUB_DEBUG_ON=y CONFIG_SLUB_STATS=y CONFIG_HAVE_DEBUG_KMEMLEAK=y # CONFIG_DEBUG_KMEMLEAK is not set CONFIG_DEBUG_STACK_USAGE=y # CONFIG_DEBUG_VM is not set # CONFIG_DEBUG_VIRTUAL is not set CONFIG_DEBUG_MEMORY_INIT=y CONFIG_DEBUG_HIGHMEM=y CONFIG_HAVE_DEBUG_STACKOVERFLOW=y CONFIG_DEBUG_STACKOVERFLOW=y CONFIG_HAVE_ARCH_KMEMCHECK=y # CONFIG_DEBUG_SHIRQ is not set # # Debug Lockups and Hangs # CONFIG_LOCKUP_DETECTOR=y CONFIG_HARDLOCKUP_DETECTOR=y CONFIG_BOOTPARAM_HARDLOCKUP_PANIC=y CONFIG_BOOTPARAM_HARDLOCKUP_PANIC_VALUE=1 # CONFIG_BOOTPARAM_SOFTLOCKUP_PANIC is not set CONFIG_BOOTPARAM_SOFTLOCKUP_PANIC_VALUE=0 # CONFIG_DETECT_HUNG_TASK is not set # CONFIG_PANIC_ON_OOPS is not set CONFIG_PANIC_ON_OOPS_VALUE=0 CONFIG_SCHED_DEBUG=y CONFIG_SCHEDSTATS=y # CONFIG_TIMER_STATS is not set # # Lock Debugging (spinlocks, mutexes, etc...) # CONFIG_DEBUG_RT_MUTEXES=y CONFIG_DEBUG_PI_LIST=y # CONFIG_RT_MUTEX_TESTER is not set CONFIG_DEBUG_SPINLOCK=y CONFIG_DEBUG_MUTEXES=y CONFIG_DEBUG_WW_MUTEX_SLOWPATH=y CONFIG_DEBUG_LOCK_ALLOC=y CONFIG_PROVE_LOCKING=y CONFIG_LOCKDEP=y # CONFIG_LOCK_STAT is not set # CONFIG_DEBUG_LOCKDEP is not set CONFIG_DEBUG_ATOMIC_SLEEP=y # CONFIG_DEBUG_LOCKING_API_SELFTESTS is not set CONFIG_TRACE_IRQFLAGS=y CONFIG_STACKTRACE=y CONFIG_DEBUG_KOBJECT=y # CONFIG_DEBUG_KOBJECT_RELEASE is not set # CONFIG_DEBUG_BUGVERBOSE is not set CONFIG_DEBUG_WRITECOUNT=y CONFIG_DEBUG_LIST=y # CONFIG_DEBUG_SG is not set # CONFIG_DEBUG_NOTIFIERS is not set # CONFIG_DEBUG_CREDENTIALS is not set # # RCU Debugging # CONFIG_PROVE_RCU=y CONFIG_PROVE_RCU_REPEATEDLY=y # CONFIG_SPARSE_RCU_POINTER is not set # CONFIG_RCU_TORTURE_TEST is not set CONFIG_RCU_CPU_STALL_TIMEOUT=21 CONFIG_RCU_TRACE=y # CONFIG_NOTIFIER_ERROR_INJECTION is not set CONFIG_FAULT_INJECTION=y CONFIG_FAILSLAB=y CONFIG_FAIL_PAGE_ALLOC=y # CONFIG_FAIL_MMC_REQUEST is not set CONFIG_FAULT_INJECTION_DEBUG_FS=y CONFIG_FAULT_INJECTION_STACKTRACE_FILTER=y CONFIG_LATENCYTOP=y CONFIG_ARCH_HAS_DEBUG_STRICT_USER_COPY_CHECKS=y # CONFIG_DEBUG_STRICT_USER_COPY_CHECKS is not set CONFIG_USER_STACKTRACE_SUPPORT=y CONFIG_NOP_TRACER=y CONFIG_HAVE_FUNCTION_TRACER=y CONFIG_HAVE_FUNCTION_GRAPH_TRACER=y CONFIG_HAVE_FUNCTION_GRAPH_FP_TEST=y CONFIG_HAVE_FUNCTION_TRACE_MCOUNT_TEST=y CONFIG_HAVE_DYNAMIC_FTRACE=y CONFIG_HAVE_DYNAMIC_FTRACE_WITH_REGS=y CONFIG_HAVE_FTRACE_MCOUNT_RECORD=y CONFIG_HAVE_SYSCALL_TRACEPOINTS=y CONFIG_HAVE_C_RECORDMCOUNT=y CONFIG_TRACER_MAX_TRACE=y CONFIG_TRACE_CLOCK=y CONFIG_RING_BUFFER=y CONFIG_EVENT_TRACING=y CONFIG_CONTEXT_SWITCH_TRACER=y CONFIG_RING_BUFFER_ALLOW_SWAP=y CONFIG_TRACING=y CONFIG_GENERIC_TRACER=y CONFIG_TRACING_SUPPORT=y CONFIG_FTRACE=y CONFIG_FUNCTION_TRACER=y # CONFIG_IRQSOFF_TRACER is not set CONFIG_SCHED_TRACER=y # CONFIG_FTRACE_SYSCALLS is not set CONFIG_TRACER_SNAPSHOT=y CONFIG_TRACER_SNAPSHOT_PER_CPU_SWAP=y CONFIG_BRANCH_PROFILE_NONE=y # CONFIG_PROFILE_ANNOTATED_BRANCHES is not set # CONFIG_PROFILE_ALL_BRANCHES is not set # CONFIG_STACK_TRACER is not set CONFIG_UPROBE_EVENT=y CONFIG_PROBE_EVENTS=y # CONFIG_DYNAMIC_FTRACE is not set # CONFIG_FUNCTION_PROFILER is not set CONFIG_FTRACE_SELFTEST=y CONFIG_FTRACE_STARTUP_TEST=y CONFIG_EVENT_TRACE_TEST_SYSCALLS=y # CONFIG_MMIOTRACE is not set CONFIG_RING_BUFFER_BENCHMARK=y CONFIG_RING_BUFFER_STARTUP_TEST=y # # Runtime Testing # # CONFIG_TEST_LIST_SORT is not set CONFIG_BACKTRACE_SELF_TEST=y CONFIG_RBTREE_TEST=y # CONFIG_ATOMIC64_SELFTEST is not set CONFIG_TEST_STRING_HELPERS=y CONFIG_TEST_KSTRTOX=y CONFIG_PROVIDE_OHCI1394_DMA_INIT=y CONFIG_DMA_API_DEBUG=y # CONFIG_SAMPLES is not set CONFIG_HAVE_ARCH_KGDB=y # CONFIG_KGDB is not set CONFIG_STRICT_DEVMEM=y # CONFIG_X86_VERBOSE_BOOTUP is not set # CONFIG_EARLY_PRINTK is not set CONFIG_X86_PTDUMP=y # CONFIG_DEBUG_RODATA is not set # CONFIG_DOUBLEFAULT is not set CONFIG_DEBUG_TLBFLUSH=y CONFIG_IOMMU_STRESS=y CONFIG_HAVE_MMIOTRACE_SUPPORT=y CONFIG_IO_DELAY_TYPE_0X80=0 CONFIG_IO_DELAY_TYPE_0XED=1 CONFIG_IO_DELAY_TYPE_UDELAY=2 CONFIG_IO_DELAY_TYPE_NONE=3 CONFIG_IO_DELAY_0X80=y # CONFIG_IO_DELAY_0XED is not set # CONFIG_IO_DELAY_UDELAY is not set # CONFIG_IO_DELAY_NONE is not set CONFIG_DEFAULT_IO_DELAY_TYPE=0 CONFIG_DEBUG_BOOT_PARAMS=y CONFIG_CPA_DEBUG=y CONFIG_OPTIMIZE_INLINING=y # CONFIG_DEBUG_NMI_SELFTEST is not set CONFIG_X86_DEBUG_STATIC_CPU_HAS=y # # Security options # # CONFIG_KEYS is not set # CONFIG_SECURITY_DMESG_RESTRICT is not set # CONFIG_SECURITY is not set CONFIG_SECURITYFS=y CONFIG_DEFAULT_SECURITY_DAC=y CONFIG_DEFAULT_SECURITY="" CONFIG_CRYPTO=y # # Crypto core or helper # CONFIG_CRYPTO_FIPS=y CONFIG_CRYPTO_ALGAPI=y CONFIG_CRYPTO_ALGAPI2=y CONFIG_CRYPTO_AEAD=y CONFIG_CRYPTO_AEAD2=y CONFIG_CRYPTO_BLKCIPHER=y CONFIG_CRYPTO_BLKCIPHER2=y CONFIG_CRYPTO_HASH=y CONFIG_CRYPTO_HASH2=y CONFIG_CRYPTO_RNG=y CONFIG_CRYPTO_RNG2=y CONFIG_CRYPTO_PCOMP=y CONFIG_CRYPTO_PCOMP2=y CONFIG_CRYPTO_MANAGER=y CONFIG_CRYPTO_MANAGER2=y # CONFIG_CRYPTO_MANAGER_DISABLE_TESTS is not set CONFIG_CRYPTO_GF128MUL=y # CONFIG_CRYPTO_NULL is not set CONFIG_CRYPTO_WORKQUEUE=y CONFIG_CRYPTO_CRYPTD=y # CONFIG_CRYPTO_AUTHENC is not set CONFIG_CRYPTO_ABLK_HELPER_X86=y CONFIG_CRYPTO_GLUE_HELPER_X86=y # # Authenticated Encryption with Associated Data # CONFIG_CRYPTO_CCM=y # CONFIG_CRYPTO_GCM is not set CONFIG_CRYPTO_SEQIV=y # # Block modes # CONFIG_CRYPTO_CBC=y CONFIG_CRYPTO_CTR=y CONFIG_CRYPTO_CTS=y CONFIG_CRYPTO_ECB=y CONFIG_CRYPTO_LRW=y CONFIG_CRYPTO_PCBC=y CONFIG_CRYPTO_XTS=y # # Hash modes # # CONFIG_CRYPTO_CMAC is not set CONFIG_CRYPTO_HMAC=y CONFIG_CRYPTO_XCBC=y CONFIG_CRYPTO_VMAC=y # # Digest # CONFIG_CRYPTO_CRC32C=y CONFIG_CRYPTO_CRC32C_INTEL=y # CONFIG_CRYPTO_CRC32 is not set # CONFIG_CRYPTO_CRC32_PCLMUL is not set # CONFIG_CRYPTO_CRCT10DIF is not set # CONFIG_CRYPTO_GHASH is not set CONFIG_CRYPTO_MD4=y CONFIG_CRYPTO_MD5=y CONFIG_CRYPTO_MICHAEL_MIC=y CONFIG_CRYPTO_RMD128=y # CONFIG_CRYPTO_RMD160 is not set CONFIG_CRYPTO_RMD256=y CONFIG_CRYPTO_RMD320=y # CONFIG_CRYPTO_SHA1 is not set CONFIG_CRYPTO_SHA256=y CONFIG_CRYPTO_SHA512=y CONFIG_CRYPTO_TGR192=y CONFIG_CRYPTO_WP512=y # # Ciphers # CONFIG_CRYPTO_AES=y CONFIG_CRYPTO_AES_586=y CONFIG_CRYPTO_AES_NI_INTEL=y CONFIG_CRYPTO_ANUBIS=y CONFIG_CRYPTO_ARC4=y # CONFIG_CRYPTO_BLOWFISH is not set # CONFIG_CRYPTO_CAMELLIA is not set CONFIG_CRYPTO_CAST_COMMON=y CONFIG_CRYPTO_CAST5=y # CONFIG_CRYPTO_CAST6 is not set CONFIG_CRYPTO_DES=y CONFIG_CRYPTO_FCRYPT=y # CONFIG_CRYPTO_KHAZAD is not set # CONFIG_CRYPTO_SALSA20 is not set # CONFIG_CRYPTO_SALSA20_586 is not set # CONFIG_CRYPTO_SEED is not set CONFIG_CRYPTO_SERPENT=y CONFIG_CRYPTO_SERPENT_SSE2_586=y CONFIG_CRYPTO_TEA=y CONFIG_CRYPTO_TWOFISH=y CONFIG_CRYPTO_TWOFISH_COMMON=y CONFIG_CRYPTO_TWOFISH_586=y # # Compression # CONFIG_CRYPTO_DEFLATE=y CONFIG_CRYPTO_ZLIB=y # CONFIG_CRYPTO_LZO is not set # CONFIG_CRYPTO_LZ4 is not set CONFIG_CRYPTO_LZ4HC=y # # Random Number Generation # CONFIG_CRYPTO_ANSI_CPRNG=y CONFIG_CRYPTO_HW=y # CONFIG_CRYPTO_DEV_PADLOCK is not set CONFIG_CRYPTO_DEV_GEODE=y CONFIG_CRYPTO_DEV_HIFN_795X=y # CONFIG_CRYPTO_DEV_HIFN_795X_RNG is not set CONFIG_HAVE_KVM=y CONFIG_VIRTUALIZATION=y CONFIG_BINARY_PRINTF=y # # Library routines # CONFIG_BITREVERSE=y CONFIG_GENERIC_STRNCPY_FROM_USER=y CONFIG_GENERIC_STRNLEN_USER=y CONFIG_GENERIC_NET_UTILS=y CONFIG_GENERIC_FIND_FIRST_BIT=y CONFIG_GENERIC_PCI_IOMAP=y CONFIG_GENERIC_IOMAP=y CONFIG_GENERIC_IO=y CONFIG_PERCPU_RWSEM=y CONFIG_CRC_CCITT=y CONFIG_CRC16=y # CONFIG_CRC_T10DIF is not set CONFIG_CRC_ITU_T=y CONFIG_CRC32=y # CONFIG_CRC32_SELFTEST is not set CONFIG_CRC32_SLICEBY8=y # CONFIG_CRC32_SLICEBY4 is not set # CONFIG_CRC32_SARWATE is not set # CONFIG_CRC32_BIT is not set # CONFIG_CRC7 is not set CONFIG_LIBCRC32C=y # CONFIG_CRC8 is not set CONFIG_ZLIB_INFLATE=y CONFIG_ZLIB_DEFLATE=y CONFIG_LZO_COMPRESS=y CONFIG_LZO_DECOMPRESS=y CONFIG_LZ4HC_COMPRESS=y CONFIG_LZ4_DECOMPRESS=y CONFIG_XZ_DEC=y # CONFIG_XZ_DEC_X86 is not set CONFIG_XZ_DEC_POWERPC=y CONFIG_XZ_DEC_IA64=y CONFIG_XZ_DEC_ARM=y CONFIG_XZ_DEC_ARMTHUMB=y CONFIG_XZ_DEC_SPARC=y CONFIG_XZ_DEC_BCJ=y CONFIG_XZ_DEC_TEST=y CONFIG_REED_SOLOMON=y CONFIG_REED_SOLOMON_DEC16=y CONFIG_BCH=y CONFIG_BCH_CONST_PARAMS=y CONFIG_HAS_IOMEM=y CONFIG_HAS_IOPORT=y CONFIG_HAS_DMA=y CONFIG_CHECK_SIGNATURE=y CONFIG_NLATTR=y CONFIG_ARCH_HAS_ATOMIC64_DEC_IF_POSITIVE=y # CONFIG_AVERAGE is not set # CONFIG_CORDIC is not set CONFIG_DDR=y CONFIG_FONT_SUPPORT=y # CONFIG_FONTS is not set CONFIG_FONT_8x8=y CONFIG_FONT_8x16=y

1 0

[alsa-devel] [PATCH 3/3] ASoC: cs42l73: Add Device Tree support for CS42L73
by Brian Austin 18 Oct '13

18 Oct '13

This patch adds device tree support for the cs42l73 codec. Reset Gpio and Charge Pump Frequency are available Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- .../devicetree/bindings/sound/cs42l73.txt | 22 +++++++++++++++ sound/soc/codecs/cs42l73.c | 31 +++++++++++++++++++--- 2 files changed, 50 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs42l73.txt diff --git a/Documentation/devicetree/bindings/sound/cs42l73.txt b/Documentation/devicetree/bindings/sound/cs42l73.txt new file mode 100644 index 0000000..5a86495 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l73.txt @@ -0,0 +1,22 @@ +CS42L73 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l73" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset_gpio : a GPIO spec for the reset pin. + - chgfreq : Charge Pump Frequency values 0x00-0x0F + + +Example: + +codec: cs42l73@4a { + compatible = "cirrus,cs42l73"; + reg = <0x4a>; + reset_gpio = <&gpio 10 0>; + chgfreq = <0x05>; +}; \ No newline at end of file diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index e3187cc..4607958 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -17,7 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -#include <linux/gpio.h> +#include <linux/of_gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -1414,6 +1414,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, struct cs42l73_private *cs42l73; struct cs42l73_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; + u32 val32; unsigned int devid = 0; unsigned int reg; @@ -1431,9 +1432,26 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, return ret; } - if (pdata) + if (pdata) { cs42l73->pdata = *pdata; - + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l73_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + if (of_property_read_u32(i2c_client->dev.of_node, + "chgfreq", &val32) >= 0) + pdata->chgfreq = val32; + } + pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node, + "reset-gpio", 0); + cs42l73->pdata = *pdata; + } + i2c_set_clientdata(i2c_client, cs42l73); if ( cs42l73->pdata.reset_gpio) { @@ -1493,6 +1511,12 @@ static int cs42l73_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id cs42l73_of_match[] = { + { .compatible = "cirrus,cs42l73", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs42l73_of_match); + static const struct i2c_device_id cs42l73_id[] = { {"cs42l73", 0}, {} @@ -1504,6 +1528,7 @@ static struct i2c_driver cs42l73_i2c_driver = { .driver = { .name = "cs42l73", .owner = THIS_MODULE, + .of_match_table = cs42l73_of_match, }, .id_table = cs42l73_id, .probe = cs42l73_i2c_probe, -- 1.8.4.rc3

1 1

[alsa-devel] [PATCH 1/3] ASoC: cs42l73: Add platform data support for cs42l73 codec
by Brian Austin 18 Oct '13

18 Oct '13

Add support for RST GPIO and Charge Pump Freq in platform data Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs42l73.h | 22 ++++++++++++++++++++ sound/soc/codecs/cs42l73.c | 51 ++++++++++++++++++++++++++++++---------------- sound/soc/codecs/cs42l73.h | 1 + 3 files changed, 56 insertions(+), 18 deletions(-) create mode 100644 include/sound/cs42l73.h diff --git a/include/sound/cs42l73.h b/include/sound/cs42l73.h new file mode 100644 index 0000000..f354be4 --- /dev/null +++ b/include/sound/cs42l73.h @@ -0,0 +1,22 @@ +/* + * linux/sound/cs42l73.h -- Platform data for CS42L73 + * + * Copyright (c) 2012 Cirrus Logic 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. + */ + +#ifndef __CS42L73_H +#define __CS42L73_H + +struct cs42l73_platform_data { + /* RST GPIO */ + unsigned int reset_gpio; + unsigned int chgfreq; + int jack_detection; + unsigned int mclk_freq; +}; + +#endif /* __CS42L73_H */ diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 3b20c86..dfdb715 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -28,6 +29,7 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <sound/cs42l73.h> #include "cs42l73.h" struct sp_config { @@ -35,6 +37,7 @@ struct sp_config { u32 srate; }; struct cs42l73_private { + struct cs42l73_platform_data pdata; struct sp_config config[3]; struct regmap *regmap; u32 sysclk; @@ -310,15 +313,6 @@ static const struct soc_enum ng_delay_enum = SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); -static const char * const charge_pump_freq_text[] = { - "0", "1", "2", "3", "4", - "5", "6", "7", "8", "9", - "10", "11", "12", "13", "14", "15" }; - -static const struct soc_enum charge_pump_enum = - SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, - ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); - static const char * const cs42l73_mono_mix_texts[] = { "Left", "Right", "Mono Mix"}; @@ -511,8 +505,6 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), SOC_ENUM("NG Delay", ng_delay_enum), - SOC_ENUM("Charge Pump Frequency", charge_pump_enum), - SOC_DOUBLE_R_TLV("XSP-IP Volume", CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, attn_tlv), @@ -1367,11 +1359,16 @@ static int cs42l73_probe(struct snd_soc_codec *codec) return ret; } - regcache_cache_only(cs42l73->regmap, true); - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + /* Set Charge Pump Frequency */ + if (cs42l73->pdata.chgfreq) + snd_soc_update_bits(codec, CS42L73_CPFCHC, + CS42L73_CHARGEPUMP_MASK, + cs42l73->pdata.chgfreq << 4); + + /* MCLK1 as master clk */ + cs42l73->mclksel = CS42L73_CLKID_MCLK1; cs42l73->mclk = 0; return ret; @@ -1415,6 +1412,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l73_private *cs42l73; + struct cs42l73_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1426,14 +1424,32 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; } - i2c_set_clientdata(i2c_client, cs42l73); - cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap); if (IS_ERR(cs42l73->regmap)) { ret = PTR_ERR(cs42l73->regmap); dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); return ret; } + + if (pdata) + cs42l73->pdata = *pdata; + + i2c_set_clientdata(i2c_client, cs42l73); + + if (cs42l73->pdata.reset_gpio) { + ret = gpio_request_one(cs42l73->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L73 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l73->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 1); + } + + regcache_cache_bypass(cs42l73->regmap, true); + /* initialize codec */ ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, &reg); devid = (reg & 0xFF) << 12; @@ -1444,7 +1460,6 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_E, &reg); devid |= (reg & 0xF0) >> 4; - if (devid != CS42L73_DEVID) { ret = -ENODEV; dev_err(&i2c_client->dev, @@ -1462,7 +1477,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, dev_info(&i2c_client->dev, "Cirrus Logic CS42L73, Revision: %02X\n", reg & 0xFF); - regcache_cache_only(cs42l73->regmap, true); + regcache_cache_bypass(cs42l73->regmap, false); ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_cs42l73, cs42l73_dai, diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h index f30a4c4..4f83d39 100644 --- a/sound/soc/codecs/cs42l73.h +++ b/sound/soc/codecs/cs42l73.h @@ -159,6 +159,7 @@ #define THMOVLD_115C 2 #define THMOVLD_098C 3 +#define CS42L73_CHARGEPUMP_MASK (0xF0) /* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ #define SP_3ST (1 << 7) -- 1.8.4.rc3

3 7

[alsa-devel] [PATCH] ASoC: kirkwood: add S/PDIF support
by Jean-Francois Moine 18 Oct '13

18 Oct '13

This patch adds S/PDIF input/output for mvebu DT boards. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- Documentation/devicetree/bindings/sound/mvebu-audio.txt | 23 +++++ sound/soc/kirkwood/kirkwood-i2s.c | 128 ++++++++++++++++---------- sound/soc/kirkwood/kirkwood.h | 2 + 3 files changed, 102 insertions(+), 51 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt index f0062c5..c19a5d3 100644 --- a/Documentation/devicetree/bindings/sound/mvebu-audio.txt +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -22,6 +22,27 @@ Required properties: "internal" for the internal clock "extclk" for the external clock +Optional properties: + +- i2s-out: This is a boolean property. If present, the transmitting + function of I2S will be enabled, indicating the I2S is connected + to some IP block, such as an HDMI encoder/display-controller. + +- i2s-in: This is a boolean property. If present, the receiving + function of I2S will be enabled, indicating the I2S is connected + to some IP block, such as an HDMI encoder/display-controller. + +- spdif-out: This is a boolean property. If present, the transmitting + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF out connector/jack on the board or it's connecting to some + other IP block, such as an HDMI encoder/display-controller. + +- spdif-in: This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF in connector/jack on the board. + +* Note: At least one of these four properties should be set in the DT binding. + Example: i2s1: audio-controller@b4000 { @@ -30,4 +51,6 @@ i2s1: audio-controller@b4000 { interrupts = <21>, <22>; clocks = <&gate_clk 13>; clock-names = "internal"; + i2s-out; + spdif-out; }; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d..c62c7fa 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_PLAYCTL_I2S_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -253,12 +257,14 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, writel(value, priv->io + KIRKWOOD_INT_MASK); /* enable playback */ + ctl &= priv->ctl_play_mask; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +278,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +309,8 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -310,6 +319,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, writel(value, priv->io + KIRKWOOD_INT_MASK); /* enable record */ + ctl &= priv->ctl_rec_mask; writel(ctl, priv->io + KIRKWOOD_RECCTL); break; @@ -404,55 +414,75 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; +static const struct snd_soc_component_driver kirkwood_i2s_component = { + .name = DRV_NAME, +}; -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .capture = { +static int kirkwood_register_component(struct device *dev, + struct kirkwood_dma_data *priv) +{ + struct device_node *np = dev->of_node; + uint32_t ctl_play_mask = ~(KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN); + uint32_t ctl_rec_mask = ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); + int ret; + + static const struct snd_soc_pcm_stream stream = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000, .formats = KIRKWOOD_I2S_FORMATS, - }, - .ops = &kirkwood_i2s_dai_ops, -}; + }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .ops = &kirkwood_i2s_dai_ops, -}; + if (np) { + if (of_property_read_bool(np, "i2s-in")) + ctl_rec_mask |= KIRKWOOD_RECCTL_I2S_EN; + if (of_property_read_bool(np, "i2s-out")) + ctl_play_mask |= KIRKWOOD_PLAYCTL_I2S_EN; + if (of_property_read_bool(np, "spdif-in")) + ctl_rec_mask |= KIRKWOOD_RECCTL_SPDIF_EN; + if (of_property_read_bool(np, "spdif-out")) + ctl_play_mask |= KIRKWOOD_PLAYCTL_SPDIF_EN; + } else { + ctl_rec_mask |= KIRKWOOD_RECCTL_I2S_EN; + ctl_play_mask |= KIRKWOOD_PLAYCTL_I2S_EN; + } + priv->ctl_play_mask = ctl_play_mask; + priv->ctl_rec_mask = ctl_rec_mask; + + priv->soc_dai.probe = kirkwood_i2s_probe; + priv->soc_dai.ops = &kirkwood_i2s_dai_ops; + if (ctl_play_mask & (KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN)) { + memcpy(&priv->soc_dai.playback, &stream, + sizeof priv->soc_dai.playback); + if (!IS_ERR(priv->extclk)) + priv->soc_dai.playback.rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT; + } + if (ctl_rec_mask & (KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN)) { + memcpy(&priv->soc_dai.capture, &stream, + sizeof priv->soc_dai.capture); + if (!IS_ERR(priv->extclk)) + priv->soc_dai.capture.rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT; + } -static const struct snd_soc_component_driver kirkwood_i2s_component = { - .name = DRV_NAME, -}; + ret = snd_soc_register_component(dev, &kirkwood_i2s_component, + &priv->soc_dai, 1); + if (ret) + dev_err(dev, "snd_soc_register_component failed\n"); + return ret; +} static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -503,7 +533,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } else { dev_info(&pdev->dev, "found external clock\n"); clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; } } @@ -520,12 +549,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_128; } - err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); - if (err) { - dev_err(&pdev->dev, "snd_soc_register_component failed\n"); + err = kirkwood_register_component(&pdev->dev, priv); + if (err) goto err_component; - } err = snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform); if (err) { diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index f8e1ccc..8f11e03 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -134,6 +134,8 @@ struct kirkwood_dma_data { struct clk *extclk; uint32_t ctl_play; uint32_t ctl_rec; + uint32_t ctl_play_mask, ctl_rec_mask; + struct snd_soc_dai_driver soc_dai; struct snd_pcm_substream *substream_play; struct snd_pcm_substream *substream_rec; int irq; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 11

[alsa-devel] [PATCH] ASoC: simple-card: un-implemented .set_fmt is not error
by Kuninori Morimoto 18 Oct '13

18 Oct '13

Current simple-card returns error if DAI doesn't support .set_fmt callback. But the return value -ENOTSUPP means "not supported", it is not error. This patch avoids such case Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 62befbd..b75c7b1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -29,6 +29,11 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, if (!ret && daifmt) ret = snd_soc_dai_set_fmt(dai, daifmt); + if (ret == -ENOTSUPP) { + dev_warn(dai->dev, "ASoC: set_fmt is not supported\n"); + ret = 0; + } + if (!ret && set->sysclk) ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); -- 1.7.9.5

2 1

[alsa-devel] [Patch v2][ 16/37] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 18 Oct '13

18 Oct '13

Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Sascha Hauer <kernel(a)pengutronix.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: Eric Bénard <eric(a)eukrea.com> Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 23 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 125 +++++++++++++++++--- 3 files changed, 133 insertions(+), 20 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..8791037 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,23 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,eukrea-tlv320" +- model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- audio-codec : The phandle of the tlv320aic23 audio codec. +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,eukrea-tlv320"; + model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,audio-codec = <&tlv320aic23>; + mux-int-port = <2>; + mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f..9c3cd64 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -161,12 +161,15 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || OF depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9a4a0ca..975af57 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -17,6 +17,8 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,24 +28,33 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 +struct eukrea_tlv320 { + char of; +}; + static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct eukrea_tlv320 *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; + /* fsl_ssi lacks the set_fmt ops. */ + if (!priv->of) { + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } } ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | @@ -60,15 +71,18 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, pr_err("%s: failed setting codec sysclk\n", __func__); return ret; } - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); - ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, - SND_SOC_CLOCK_IN); - if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); - return ret; - } + /* fsl_ssi lacks the set_sysclk ops */ + if (!priv->of) { + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + } return 0; } @@ -80,7 +94,6 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, @@ -97,8 +110,56 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct platform_device *ssi_pdev; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np; + struct eukrea_tlv320 *priv; + + priv = devm_kzalloc(&pdev->dev, + sizeof(struct eukrea_tlv320), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (np) { + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "ssi-controller missing or " + "invalid.\n"); + ret = -ENODEV; + goto err; + } + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + eukrea_tlv320_dai.platform_of_node = ssi_np; + priv->of = 1; + } else { + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + } - if (machine_is_eukrea_cpuimx27()) { + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -115,8 +176,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -130,14 +195,29 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,eukrea-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } eukrea_tlv320.dev = &pdev->dev; + snd_soc_card_set_drvdata(&eukrea_tlv320, priv); ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -149,10 +229,17 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,eukrea-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, .remove = eukrea_tlv320_remove,}; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v3] sound: pci: emu10k1: code refactoring
by Geyslan G. Bem 18 Oct '13

18 Oct '13

Partially restructures _snd_emu10k1_audigy_init_efx() and _snd_emu10k1_init_efx() functions. Be noted that the cast is demanded to use '__user'. So, in these cases, avoid patches based on the coccinelle 'drop_kmalloc_cast' semantic patch. Signed-off-by: Geyslan G. Bem <geyslan(a)gmail.com> --- sound/pci/emu10k1/emufx.c | 76 ++++++++++++++++++++++++++++------------------- 1 file changed, 45 insertions(+), 31 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 0275209..1f9c7c4 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1182,15 +1182,20 @@ static int _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) u32 *gpr_map; mm_segment_t seg; - if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL || - (icode->gpr_map = (u_int32_t __user *) - kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || - (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, - sizeof(*controls), GFP_KERNEL)) == NULL) { - err = -ENOMEM; - goto __err; - } + err = -ENOMEM; + icode = kzalloc(sizeof(*icode), GFP_KERNEL); + if (!icode) + return err; + + icode->gpr_map = (u_int32_t __user *) kcalloc(512 + 256 + 256 + 2 * 1024, + sizeof(u_int32_t), GFP_KERNEL); + if (!icode->gpr_map) + goto __err_gpr; + controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, + sizeof(*controls), GFP_KERNEL); + if (!controls) + goto __err_ctrls; + gpr_map = (u32 __force *)icode->gpr_map; icode->tram_data_map = icode->gpr_map + 512; @@ -1741,12 +1746,12 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) emu->support_tlv = 0; /* clear again */ snd_leave_user(seg); - __err: +__err: kfree(controls); - if (icode != NULL) { - kfree((void __force *)icode->gpr_map); - kfree(icode); - } +__err_ctrls: + kfree((void __force *)icode->gpr_map); +__err_gpr: + kfree(icode); return err; } @@ -1813,18 +1818,26 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) u32 *gpr_map; mm_segment_t seg; - if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL) - return -ENOMEM; - if ((icode->gpr_map = (u_int32_t __user *) - kcalloc(256 + 160 + 160 + 2 * 512, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || - (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, - sizeof(struct snd_emu10k1_fx8010_control_gpr), - GFP_KERNEL)) == NULL || - (ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL)) == NULL) { - err = -ENOMEM; - goto __err; - } + err = -ENOMEM; + icode = kzalloc(sizeof(*icode), GFP_KERNEL); + if (!icode) + return err; + + icode->gpr_map = (u_int32_t __user *) kcalloc(256 + 160 + 160 + 2 * 512, + sizeof(u_int32_t), GFP_KERNEL); + if (!icode->gpr_map) + goto __err_gpr; + + controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, + sizeof(struct snd_emu10k1_fx8010_control_gpr), + GFP_KERNEL); + if (!controls) + goto __err_ctrls; + + ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL); + if (!ipcm) + goto __err_ipcm; + gpr_map = (u32 __force *)icode->gpr_map; icode->tram_data_map = icode->gpr_map + 256; @@ -2363,13 +2376,14 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_leave_user(seg); if (err >= 0) err = snd_emu10k1_ipcm_poke(emu, ipcm); - __err: +__err: kfree(ipcm); +__err_ipcm: kfree(controls); - if (icode != NULL) { - kfree((void __force *)icode->gpr_map); - kfree(icode); - } +__err_ctrls: + kfree((void __force *)icode->gpr_map); +__err_gpr: + kfree(icode); return err; } -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: simple-card: un-implemented set_fmt is not error
by Kuninori Morimoto 18 Oct '13

18 Oct '13

Current simple-card returns error if DAI doesn't support .set_fmt callback. But the error is -ENOTSUPP (= not supported), and it is not error. This patch avoids such case Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8c49147..b2fbb70 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -27,6 +27,11 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, if (!ret && daifmt) ret = snd_soc_dai_set_fmt(dai, daifmt); + if (ret == -ENOTSUPP) { + dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n"); + ret = 0; + } + if (!ret && set->sysclk) ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); -- 1.7.9.5

1 0

[alsa-devel] [PATCH 1/3] ALSA: pxa: slightly refactor reset handling
by Dmitry Eremin-Solenikov 18 Oct '13

18 Oct '13

PXA25x also shows some problems when using interrupts during reset handling. Thus do not use interrupts on all pxa kinds (to detect codec ready state). Instead use a common mdelay-loop on all platforms to detect codecs becoming ready. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- sound/arm/pxa2xx-ac97-lib.c | 27 ++++++++++----------------- 1 file changed, 10 insertions(+), 17 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index e6f4633..99a4668 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -117,8 +117,7 @@ static inline void pxa_ac97_warm_pxa25x(void) { gsr_bits = 0; - GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); + GCR |= GCR_WARM_RST; } static inline void pxa_ac97_cold_pxa25x(void) @@ -129,8 +128,6 @@ static inline void pxa_ac97_cold_pxa25x(void) gsr_bits = 0; GCR = GCR_COLD_RST; - GCR |= GCR_CDONE_IE|GCR_SDONE_IE; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); } #endif @@ -149,8 +146,6 @@ static inline void pxa_ac97_warm_pxa27x(void) static inline void pxa_ac97_cold_pxa27x(void) { - unsigned int timeout; - GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -161,29 +156,20 @@ static inline void pxa_ac97_cold_pxa27x(void) udelay(5); clk_disable(ac97conf_clk); GCR = GCR_COLD_RST | GCR_WARM_RST; - timeout = 100; /* wait for the codec-ready bit to be set */ - while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(1); } #endif #ifdef CONFIG_PXA3xx static inline void pxa_ac97_warm_pxa3xx(void) { - int timeout = 100; - gsr_bits = 0; /* Can't use interrupts */ GCR |= GCR_WARM_RST; - while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(1); } static inline void pxa_ac97_cold_pxa3xx(void) { - int timeout = 1000; - /* Hold CLKBPB for 100us */ GCR = 0; GCR = GCR_CLKBPB; @@ -199,14 +185,13 @@ static inline void pxa_ac97_cold_pxa3xx(void) GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); GCR = GCR_WARM_RST | GCR_COLD_RST; - while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(10); } #endif bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) { unsigned long gsr; + unsigned int timeout = 100; #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) @@ -224,6 +209,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) else #endif BUG(); + + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); + gsr = GSR | gsr_bits; if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", @@ -239,6 +228,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) { unsigned long gsr; + unsigned int timeout = 1000; #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) @@ -257,6 +247,9 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) #endif BUG(); + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); + gsr = GSR | gsr_bits; if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", -- 1.8.4.rc3

2 5

[alsa-devel] [PATCH][RFC] ASoC: add default xlate on snd_soc_of_get_dai_name()
by Kuninori Morimoto 18 Oct '13

18 Oct '13

Current snd_soc_of_get_dai_name() needs .of_xlate_dai_name() callback on each component drivers. But required behavior on almost all these drivers is just returns its indexed dai driver's name. This patch adds this feature as default behavior. .of_xlate_dai_name() can overwrite it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This is RFC patch. On ML discussed that snd_soc_of_get_dai_name() needs .of_xlate_dai_name to get/exchange device name from device node ID. But, almost all driver's .of_xlate_dai_name will be same implementation. (return name of indexed dai driver) So, this patch adds it as default behavior. .of_xlate_dai_name() can over write it. This patch is based on mark/topic/component include/sound/soc.h | 4 +++- sound/soc/soc-core.c | 16 ++++++++++++---- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1dd7dc5..ae715ca 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -684,10 +684,12 @@ struct snd_soc_component_driver { struct snd_soc_component { const char *name; int id; - int num_dai; struct device *dev; struct list_head list; + struct snd_soc_dai_driver *dai_drv; + int num_dai; + const struct snd_soc_component_driver *driver; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 711bd36..51ea2c3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4048,6 +4048,7 @@ __snd_soc_register_component(struct device *dev, cmpnt->dev = dev; cmpnt->driver = cmpnt_drv; + cmpnt->dai_drv = dai_drv; cmpnt->num_dai = num_dai; /* @@ -4609,12 +4610,19 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, if (pos->dev->of_node != args.np) continue; - if (!pos->driver->of_xlate_dai_name) { - ret = -ENOSYS; - break; + if (pos->driver->of_xlate_dai_name) { + ret = pos->driver->of_xlate_dai_name(pos, &args, dai_name); + } else { + int id = args.args[0]; + + if (id < 0 || id >= pos->num_dai) { + ret = -EINVAL; + } else { + *dai_name = pos->dai_drv[id].name; + ret = 0; + } } - ret = pos->driver->of_xlate_dai_name(pos, &args, dai_name); break; } mutex_unlock(&client_mutex); -- 1.7.9.5

3 4

[alsa-devel] [PATCH] sound: pci: emu10k1: code refactoring and casting removal
by Geyslan G. Bem 18 Oct '13

18 Oct '13

Partially restructures _snd_emu10k1_audigy_init_efx() and _snd_emu10k1_init_efx() functions. Removes useless casting (void *) from value returned by kcalloc; see Documentation/CodingStyle, Chap 14. Signed-off-by: Geyslan G. Bem <geyslan(a)gmail.com> --- sound/pci/emu10k1/emufx.c | 84 ++++++++++++++++++++++++++--------------------- 1 file changed, 46 insertions(+), 38 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 0275209..afd4691 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1182,15 +1182,20 @@ static int _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) u32 *gpr_map; mm_segment_t seg; - if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL || - (icode->gpr_map = (u_int32_t __user *) - kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || - (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, - sizeof(*controls), GFP_KERNEL)) == NULL) { - err = -ENOMEM; - goto __err; - } + err = -ENOMEM; + icode = kzalloc(sizeof(*icode), GFP_KERNEL); + if (!icode) + return err; + + icode->gpr_map = (__user) kcalloc(512 + 256 + 256 + 2 * 1024, + sizeof(u_int32_t), GFP_KERNEL); + if (!icode->gpr_map) + goto __err_gpr; + controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, + sizeof(*controls), GFP_KERNEL); + if (!controls) + goto __err_ctrls; + gpr_map = (u32 __force *)icode->gpr_map; icode->tram_data_map = icode->gpr_map + 512; @@ -1741,12 +1746,12 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) emu->support_tlv = 0; /* clear again */ snd_leave_user(seg); - __err: +__err: kfree(controls); - if (icode != NULL) { - kfree((void __force *)icode->gpr_map); - kfree(icode); - } +__err_ctrls: + kfree((void __force *)icode->gpr_map); +__err_gpr: + kfree(icode); return err; } @@ -1813,18 +1818,26 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) u32 *gpr_map; mm_segment_t seg; - if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL) - return -ENOMEM; - if ((icode->gpr_map = (u_int32_t __user *) - kcalloc(256 + 160 + 160 + 2 * 512, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || - (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, - sizeof(struct snd_emu10k1_fx8010_control_gpr), - GFP_KERNEL)) == NULL || - (ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL)) == NULL) { - err = -ENOMEM; - goto __err; - } + err = -ENOMEM; + icode = kzalloc(sizeof(*icode), GFP_KERNEL); + if (!icode) + return err; + + icode->gpr_map = (__user) kcalloc(256 + 160 + 160 + 2 * 512, + sizeof(u_int32_t), GFP_KERNEL); + if (!icode->gpr_map) + goto __err_gpr; + + controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, + sizeof(struct snd_emu10k1_fx8010_control_gpr), + GFP_KERNEL); + if (!controls) + goto __err_ctrls; + + ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL); + if (!ipcm) + goto __err_ipcm; + gpr_map = (u32 __force *)icode->gpr_map; icode->tram_data_map = icode->gpr_map + 256; @@ -2335,14 +2348,8 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) for (z = 0; z < 16; z++) OP(icode, &ptr, iACC3, FXBUS2(z), C_00000000, C_00000000, EXTIN(z)); } - - if (gpr > tmp) { - snd_BUG(); - err = -EIO; - goto __err; - } - if (i > SND_EMU10K1_GPR_CONTROLS) { + if (gpr > tmp || i > SND_EMU10K1_GPR_CONTROLS) { snd_BUG(); err = -EIO; goto __err; @@ -2363,13 +2370,14 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_leave_user(seg); if (err >= 0) err = snd_emu10k1_ipcm_poke(emu, ipcm); - __err: +__err: kfree(ipcm); +__err_ipcm: kfree(controls); - if (icode != NULL) { - kfree((void __force *)icode->gpr_map); - kfree(icode); - } +__err_ctrls: + kfree((void __force *)icode->gpr_map); +__err_gpr: + kfree(icode); return err; } -- 1.8.4

4 6

[alsa-devel] [PATCH 1/2] ASoC: dai: Provide interface for setting DMA data at probe time
by Mark Brown 17 Oct '13

17 Oct '13

From: Mark Brown <broonie(a)linaro.org> Allow DMA data to be set at probe time for devices that can do that, avoiding the need to do it every time we start a stream and supporting non-DT dmaengine users using the helpers. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc-dai.h | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1ce3685..800c101 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -286,6 +286,13 @@ static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, dai->capture_dma_data = data; } +static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, + void *playback, void *capture) +{ + dai->playback_dma_data = playback; + dai->capture_dma_data = capture; +} + static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, void *data) { -- 1.8.4.rc3

1 1

[alsa-devel] [PATCH 0/8] i2c: Remove redundant driver field from the i2c_client struct
by Lars-Peter Clausen 17 Oct '13

17 Oct '13

Hi, This series removes the redundant driver field from the i2c_client struct. The field is redundant since the same pointer can be accessed through to_i2c_driver(client->dev.driver). The commit log suggests that the field has been around since forever (since before v2.6.12-rc2) and it looks as if it was simply forgotten to remove it during the conversion of the i2c framework to the generic device driver model. Nevertheless there are a still a few users of the field around. This series first updates all users to use an alternative method of accessing the same data and then the last patch removes the driver field from the i2c_client struct. Note that due to this changes on most architectures neither the code size nor the type of generated instructions will change. This is due to the fact that we aren't really interested in the pointer value itself, but rather want to dereference it to access one of the fields of the struct. offset_of() (and hence to_i2c_driver) works by subtracting a offset from the pointer, so the compiler can internally create the sum of these two offsets and use that to access the field. E.g. on ARM the expression client->driver->command(...) generates ... ldr r3, [r0, #28] ldr r3, [r3, #32] blx r3 ... and the expression to_i2c_driver(client->dev.driver)->command(...) generates ... ldr r3, [r0, #160] ldr r3, [r3, #-4] blx r3 ... Other architectures will generate similar code. The most common pattern is to use the i2c_driver to get to the device_driver struct embedded in it. The same struct can easily be accessed through the device struct embedded in the i2c_client struct. E.g. client->driver->driver.field can be replaced by client->dev.driver->field. Here again the generated code is almost identical and only the offsets differ. E.g. on ARM the expression 'client->driver->driver.owner' generates ldr r3, [r0, #28] ldr r0, [r3, #44] and 'client->dev.driver->owner' generates ldr r3, [r0, #160] ldr r0, [r3, #8] The kernel overall code size is slightly reduced since the code that manages the driver field is removed and of course also the runtime memory footprint of the i2c_client struct is reduced. - Lars Lars-Peter Clausen (8): [media] s5c73m3: Don't use i2c_client->driver [media] exynos4-is: Don't use i2c_client->driver [media] core: Don't use i2c_client->driver drm: encoder_slave: Don't use i2c_client->driver drm: nouveau: Don't use i2c_client->driver ALSA: ppc: keywest: Don't use i2c_client->driver ASoC: imx-wm8962: Don't use i2c_client->driver i2c: Remove redundant 'driver' field from the i2c_client struct drivers/gpu/drm/drm_encoder_slave.c | 8 ++++---- drivers/gpu/drm/nouveau/core/subdev/therm/ic.c | 3 ++- drivers/i2c/i2c-core.c | 21 ++++++++++++--------- drivers/i2c/i2c-smbus.c | 10 ++++++---- drivers/media/i2c/s5c73m3/s5c73m3-core.c | 2 +- drivers/media/platform/exynos4-is/media-dev.c | 6 +++--- drivers/media/v4l2-core/tuner-core.c | 6 +++--- drivers/media/v4l2-core/v4l2-common.c | 10 +++++----- include/linux/i2c.h | 2 -- include/media/v4l2-common.h | 2 +- sound/ppc/keywest.c | 4 ++-- sound/soc/fsl/imx-wm8962.c | 2 +- 12 files changed, 40 insertions(+), 36 deletions(-) -- 1.8.0

7 17

[alsa-devel] [PATCH 0/4] remove devm_pinctrl_get_select_default from drivers
by Wolfram Sang 17 Oct '13

17 Oct '13

The core does this automatically, no need to do this explicitly in drivers. Those patchces should go via the respective subtrees. Thanks! Wolfram Sang (4): drivers/gpu/drm/tilcdc: don't use devm_pinctrl_get_select_default() in probe drivers/mmc/host: don't use devm_pinctrl_get_select_default() in probe drivers/pwm: don't use devm_pinctrl_get_select_default() in probe sound/soc/atmel: don't use devm_pinctrl_get_select_default() in probe drivers/gpu/drm/tilcdc/tilcdc_panel.c | 6 ------ drivers/gpu/drm/tilcdc/tilcdc_slave.c | 6 ------ drivers/gpu/drm/tilcdc/tilcdc_tfp410.c | 6 ------ drivers/mmc/host/mvsdio.c | 7 +------ drivers/mmc/host/omap_hsmmc.c | 7 ------- drivers/mmc/host/sdhci-esdhc-imx.c | 8 -------- drivers/pwm/pwm-tiecap.c | 6 ------ drivers/pwm/pwm-tiehrpwm.c | 6 ------ sound/soc/atmel/atmel_wm8904.c | 8 -------- 9 files changed, 1 insertion(+), 59 deletions(-) -- 1.8.4.rc3

3 7

[alsa-devel] [PATCH 1/3] ALSA: hda - Treat zero connection as non-error
by Takashi Iwai 17 Oct '13

17 Oct '13

The zero-length connection list happens so often on Haswell HDMI, and it results in warning messages like ALSA: hda_codec: invalid CONNECT_LIST verb 5[1]:0 at each time the codec resumes from the power-save, which is fairly annoying. Since this is no real error, make it shown only in the verbose debug mode. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 68801ba..d725bbf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -574,7 +574,7 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, if (range_val) { /* ranges between the previous and this one */ if (!prev_nid || prev_nid >= val) { - snd_printk(KERN_WARNING "hda_codec: " + snd_printdd("hda_codec: " "invalid dep_range_val %x:%x\n", prev_nid, val); continue; -- 1.8.4

2 4

[alsa-devel] [PATCH v3] sound: pci: emu10k1: code refactoring and casting removal
by Geyslan G. Bem 17 Oct '13

17 Oct '13

Partially restructures _snd_emu10k1_audigy_init_efx() and _snd_emu10k1_init_efx() functions. Removes useless casting (void *) from value returned by kcalloc; see Documentation/CodingStyle, Chap 14. Signed-off-by: Geyslan G. Bem <geyslan(a)gmail.com> --- sound/pci/emu10k1/emufx.c | 76 ++++++++++++++++++++++++++++------------------- 1 file changed, 45 insertions(+), 31 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 0275209..6c45714 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1182,15 +1182,20 @@ static int _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) u32 *gpr_map; mm_segment_t seg; - if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL || - (icode->gpr_map = (u_int32_t __user *) - kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || - (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, - sizeof(*controls), GFP_KERNEL)) == NULL) { - err = -ENOMEM; - goto __err; - } + err = -ENOMEM; + icode = kzalloc(sizeof(*icode), GFP_KERNEL); + if (!icode) + return err; + + icode->gpr_map = (__user) kcalloc(512 + 256 + 256 + 2 * 1024, + sizeof(u_int32_t), GFP_KERNEL); + if (!icode->gpr_map) + goto __err_gpr; + controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, + sizeof(*controls), GFP_KERNEL); + if (!controls) + goto __err_ctrls; + gpr_map = (u32 __force *)icode->gpr_map; icode->tram_data_map = icode->gpr_map + 512; @@ -1741,12 +1746,12 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) emu->support_tlv = 0; /* clear again */ snd_leave_user(seg); - __err: +__err: kfree(controls); - if (icode != NULL) { - kfree((void __force *)icode->gpr_map); - kfree(icode); - } +__err_ctrls: + kfree((void __force *)icode->gpr_map); +__err_gpr: + kfree(icode); return err; } @@ -1813,18 +1818,26 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) u32 *gpr_map; mm_segment_t seg; - if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL) - return -ENOMEM; - if ((icode->gpr_map = (u_int32_t __user *) - kcalloc(256 + 160 + 160 + 2 * 512, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || - (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, - sizeof(struct snd_emu10k1_fx8010_control_gpr), - GFP_KERNEL)) == NULL || - (ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL)) == NULL) { - err = -ENOMEM; - goto __err; - } + err = -ENOMEM; + icode = kzalloc(sizeof(*icode), GFP_KERNEL); + if (!icode) + return err; + + icode->gpr_map = (__user) kcalloc(256 + 160 + 160 + 2 * 512, + sizeof(u_int32_t), GFP_KERNEL); + if (!icode->gpr_map) + goto __err_gpr; + + controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, + sizeof(struct snd_emu10k1_fx8010_control_gpr), + GFP_KERNEL); + if (!controls) + goto __err_ctrls; + + ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL); + if (!ipcm) + goto __err_ipcm; + gpr_map = (u32 __force *)icode->gpr_map; icode->tram_data_map = icode->gpr_map + 256; @@ -2363,13 +2376,14 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) snd_leave_user(seg); if (err >= 0) err = snd_emu10k1_ipcm_poke(emu, ipcm); - __err: +__err: kfree(ipcm); +__err_ipcm: kfree(controls); - if (icode != NULL) { - kfree((void __force *)icode->gpr_map); - kfree(icode); - } +__err_ctrls: + kfree((void __force *)icode->gpr_map); +__err_gpr: + kfree(icode); return err; } -- 1.8.4

2 1

[alsa-devel] [PATCH v2 0/3] Enable mute LEDs and mic mute LEDs on Thinkpad
by David Henningsson 17 Oct '13

17 Oct '13

Encouraged by the quick and helpful feedback from both Takashi and Henrique (thanks!), I went to work with v2 of the patch. Changes since v1: - thinkpad-acpi patch: rewritten to follow Takashi's ideas - thinkpad-acpi patch: added documentation update - alsa side: added a separate patch for the new fixup action - alsa side: rewrote patch a little, mostly based on comments from Takashi David Henningsson (3): thinkpad-acpi: Add mute and mic-mute LED functionality ALSA: hda - add HDA_FIXUP_ACT_FREE action ALSA: hda - add connection to thinkpad_acpi to control mute/micmute LEDs Documentation/laptops/thinkpad-acpi.txt | 7 ++- drivers/platform/x86/thinkpad_acpi.c | 92 ++++++++++++++++++++++++++++++- include/linux/thinkpad_acpi.h | 15 +++++ sound/pci/hda/hda_local.h | 1 + sound/pci/hda/patch_conexant.c | 92 ++++++++++++++++++++++++++++++- 5 files changed, 202 insertions(+), 5 deletions(-) create mode 100644 include/linux/thinkpad_acpi.h -- 1.7.9.5

3 8

[alsa-devel] [PATCH] ASoC: wm8962: Move register initialisation to I2C probe()
by Mark Brown 17 Oct '13

17 Oct '13

From: Mark Brown <broonie(a)linaro.org> This is more idiomatic and is required for robust operation since we must ensure that the clocking configuration is valid as rapidly as possible. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8962.c | 150 +++++++++++++++++++++++----------------------- 1 file changed, 75 insertions(+), 75 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 54379ee..2bf9ee7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3242,7 +3242,7 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) } #endif -static void wm8962_set_gpio_mode(struct snd_soc_codec *codec, int gpio) +static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio) { int mask = 0; int val = 0; @@ -3263,8 +3263,8 @@ static void wm8962_set_gpio_mode(struct snd_soc_codec *codec, int gpio) } if (mask) - snd_soc_update_bits(codec, WM8962_ANALOGUE_CLOCKING1, - mask, val); + regmap_update_bits(wm8962->regmap, WM8962_ANALOGUE_CLOCKING1, + mask, val); } #ifdef CONFIG_GPIOLIB @@ -3276,7 +3276,6 @@ static inline struct wm8962_priv *gpio_to_wm8962(struct gpio_chip *chip) static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) { struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); - struct snd_soc_codec *codec = wm8962->codec; /* The WM8962 GPIOs aren't linearly numbered. For simplicity * we export linear numbers and error out if the unsupported @@ -3292,7 +3291,7 @@ static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) return -EINVAL; } - wm8962_set_gpio_mode(codec, offset + 1); + wm8962_set_gpio_mode(wm8962, offset + 1); return 0; } @@ -3376,7 +3375,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = &wm8962->pdata; int i; bool dmicclk, dmicdat; @@ -3409,75 +3407,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) } } - /* SYSCLK defaults to on; make sure it is off so we can safely - * write to registers if the device is declocked. - */ - snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0); - - /* Ensure we have soft control over all registers */ - snd_soc_update_bits(codec, WM8962_CLOCKING2, - WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); - - /* Ensure that the oscillator and PLLs are disabled */ - snd_soc_update_bits(codec, WM8962_PLL2, - WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, - 0); - - /* Apply static configuration for GPIOs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) - if (pdata->gpio_init[i]) { - wm8962_set_gpio_mode(codec, i + 1); - snd_soc_write(codec, 0x200 + i, - pdata->gpio_init[i] & 0xffff); - } - - - /* Put the speakers into mono mode? */ - if (pdata->spk_mono) - snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2, - WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); - - /* Micbias setup, detection enable and detection - * threasholds. */ - if (pdata->mic_cfg) - snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_4, - WM8962_MICDET_ENA | - WM8962_MICDET_THR_MASK | - WM8962_MICSHORT_THR_MASK | - WM8962_MICBIAS_LVL, - pdata->mic_cfg); - - /* Latch volume update bits */ - snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, - WM8962_IN_VU, WM8962_IN_VU); - snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME, - WM8962_IN_VU, WM8962_IN_VU); - snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME, - WM8962_ADC_VU, WM8962_ADC_VU); - snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME, - WM8962_ADC_VU, WM8962_ADC_VU); - snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME, - WM8962_DAC_VU, WM8962_DAC_VU); - snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME, - WM8962_DAC_VU, WM8962_DAC_VU); - snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME, - WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); - snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME, - WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); - snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, - WM8962_HPOUT_VU, WM8962_HPOUT_VU); - snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, - WM8962_HPOUT_VU, WM8962_HPOUT_VU); - - /* Stereo control for EQ */ - snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0); - - /* Don't debouce interrupts so we don't need SYSCLK */ - snd_soc_update_bits(codec, WM8962_IRQ_DEBOUNCE, - WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB | - WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB, - 0); - wm8962_add_widgets(codec); /* Save boards having to disable DMIC when not in use */ @@ -3671,6 +3600,77 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, goto err_enable; } + /* SYSCLK defaults to on; make sure it is off so we can safely + * write to registers if the device is declocked. + */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA, 0); + + /* Ensure we have soft control over all registers */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + + /* Ensure that the oscillator and PLLs are disabled */ + regmap_update_bits(wm8962->regmap, WM8962_PLL2, + WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, + 0); + + /* Apply static configuration for GPIOs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) + if (pdata->gpio_init[i]) { + wm8962_set_gpio_mode(wm8962, i + 1); + regmap_write(wm8962->regmap, 0x200 + i, + pdata->gpio_init[i] & 0xffff); + } + + + /* Put the speakers into mono mode? */ + if (pdata->spk_mono) + regmap_update_bits(wm8962->regmap, WM8962_CLASS_D_CONTROL_2, + WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); + + /* Micbias setup, detection enable and detection + * threasholds. */ + if (pdata->mic_cfg) + regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, + WM8962_MICDET_ENA | + WM8962_MICDET_THR_MASK | + WM8962_MICSHORT_THR_MASK | + WM8962_MICBIAS_LVL, + pdata->mic_cfg); + + /* Latch volume update bits */ + regmap_update_bits(wm8962->regmap, WM8962_LEFT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + regmap_update_bits(wm8962->regmap, WM8962_RIGHT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + regmap_update_bits(wm8962->regmap, WM8962_LEFT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + regmap_update_bits(wm8962->regmap, WM8962_RIGHT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + regmap_update_bits(wm8962->regmap, WM8962_LEFT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + regmap_update_bits(wm8962->regmap, WM8962_RIGHT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + regmap_update_bits(wm8962->regmap, WM8962_SPKOUTL_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + regmap_update_bits(wm8962->regmap, WM8962_SPKOUTR_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + regmap_update_bits(wm8962->regmap, WM8962_HPOUTL_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + regmap_update_bits(wm8962->regmap, WM8962_HPOUTR_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + + /* Stereo control for EQ */ + regmap_update_bits(wm8962->regmap, WM8962_EQ1, + WM8962_EQ_SHARED_COEFF, 0); + + /* Don't debouce interrupts so we don't need SYSCLK */ + regmap_update_bits(wm8962->regmap, WM8962_IRQ_DEBOUNCE, + WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB | + WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB, + 0); + if (wm8962->pdata.in4_dc_measure) { ret = regmap_register_patch(wm8962->regmap, wm8962_dc_measure, -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH] ASoC: wm8962: Move interrupt initalisation to probe()
by Mark Brown 17 Oct '13

17 Oct '13

From: Mark Brown <broonie(a)linaro.org> This is more idiomatic and fixes bugs in the error handling paths. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8962.c | 68 +++++++++++++++++++++++------------------------ 1 file changed, 33 insertions(+), 35 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 11d80f3..54379ee 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct wm8962_pdata *pdata = &wm8962->pdata; - int i, trigger, irq_pol; + int i; bool dmicclk, dmicdat; wm8962->codec = codec; @@ -3506,36 +3506,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962_init_beep(codec); wm8962_init_gpio(codec); - if (wm8962->irq) { - if (pdata->irq_active_low) { - trigger = IRQF_TRIGGER_LOW; - irq_pol = WM8962_IRQ_POL; - } else { - trigger = IRQF_TRIGGER_HIGH; - irq_pol = 0; - } - - snd_soc_update_bits(codec, WM8962_INTERRUPT_CONTROL, - WM8962_IRQ_POL, irq_pol); - - ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq, - trigger | IRQF_ONESHOT, - "wm8962", codec->dev); - if (ret != 0) { - dev_err(codec->dev, "Failed to request IRQ %d: %d\n", - wm8962->irq, ret); - wm8962->irq = 0; - /* Non-fatal */ - } else { - /* Enable some IRQs by default */ - snd_soc_update_bits(codec, - WM8962_INTERRUPT_STATUS_2_MASK, - WM8962_FLL_LOCK_EINT | - WM8962_TEMP_SHUT_EINT | - WM8962_FIFOS_ERR_EINT, 0); - } - } - return 0; } @@ -3544,9 +3514,6 @@ static int wm8962_remove(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int i; - if (wm8962->irq) - free_irq(wm8962->irq, codec); - cancel_delayed_work_sync(&wm8962->mic_work); wm8962_free_gpio(codec); @@ -3619,7 +3586,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, struct wm8962_pdata *pdata = dev_get_platdata(&i2c->dev); struct wm8962_priv *wm8962; unsigned int reg; - int ret, i; + int ret, i, irq_pol, trigger; wm8962 = devm_kzalloc(&i2c->dev, sizeof(struct wm8962_priv), GFP_KERNEL); @@ -3714,6 +3681,37 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, ret); } + if (wm8962->irq) { + if (pdata->irq_active_low) { + trigger = IRQF_TRIGGER_LOW; + irq_pol = WM8962_IRQ_POL; + } else { + trigger = IRQF_TRIGGER_HIGH; + irq_pol = 0; + } + + regmap_update_bits(wm8962->regmap, WM8962_INTERRUPT_CONTROL, + WM8962_IRQ_POL, irq_pol); + + ret = devm_request_threaded_irq(&i2c->dev, wm8962->irq, NULL, + wm8962_irq, + trigger | IRQF_ONESHOT, + "wm8962", &i2c->dev); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request IRQ %d: %d\n", + wm8962->irq, ret); + wm8962->irq = 0; + /* Non-fatal */ + } else { + /* Enable some IRQs by default */ + regmap_update_bits(wm8962->regmap, + WM8962_INTERRUPT_STATUS_2_MASK, + WM8962_FLL_LOCK_EINT | + WM8962_TEMP_SHUT_EINT | + WM8962_FIFOS_ERR_EINT, 0); + } + } + pm_runtime_enable(&i2c->dev); pm_request_idle(&i2c->dev); -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH v5] ALSA: Add SoC on-chip internal memory support for DMA buffer allocation
by Nicolin Chen 17 Oct '13

17 Oct '13

Now it's quite common that an SoC contains its on-chip internal memory. By using this memory space for DMA buffer during audio playback/record, we can shutdown the voltage for external memory to save power. Thus add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v4->v5: * Minimized the OF dependency. v3->v4: * Appropriately placed '#ifdef CONFIG_OF' * Tested by adding '#undef CONFIG_OF' to modified files, and passed compiling. v2->v3: * Moved iram specific mmap procedure out of ifdef ARCH_HAS_DMA_MMAP_COHERENT v1->v2: * Added of_node check * Dropped noops and unused physical addr in snd_free_dev_iram() --- include/sound/memalloc.h | 1 + sound/core/memalloc.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 6 +++++ 3 files changed, 76 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b82..510aec4 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f..c1dd704 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> +#include <linux/genalloc.h> #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> @@ -157,6 +158,63 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal memory + * @dev: DMA device pointer + * @size: number of bytes to allocate from the iram + * @dma: dma-view physical address + * + * Return cpu-view address or NULL indicating allocating failure + * + * This function requires iram phandle provided via of_node + */ +void *snd_malloc_dev_iram(struct device *dev, size_t size, dma_addr_t *dma) +{ + struct gen_pool *pool = NULL; + unsigned long vaddr; + +#ifdef CONFIG_OF + if (!dev->of_node) + return NULL; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); +#endif + if (!pool) + return NULL; + + vaddr = gen_pool_alloc(pool, size); + if (!vaddr) + return NULL; + + *dma = gen_pool_virt_to_phys(pool, vaddr); + + return (void *)vaddr; +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal memory + * @dev: DMA device pointer + * @size: size in bytes of memory to free + * @ptr: cpu-view address returned from snd_malloc_dev_iram + * + * This function requires iram phandle provided via of_node + */ +void snd_free_dev_iram(struct device *dev, size_t size, void *ptr) +{ + struct gen_pool *pool = NULL; + +#ifdef CONFIG_OF + if (!dev->of_node) + return; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); +#endif + if (!pool) + return; + + gen_pool_free(pool, (unsigned long)ptr, size); +} #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +255,14 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + dmab->area = snd_malloc_dev_iram(device, size, &dmab->addr); + if (dmab->area) + break; + /* Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +335,9 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_free_dev_iram(dmab->dev.dev, dmab->bytes, dmab->area); + break; case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..513f095 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3199,6 +3199,12 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) -- 1.8.4

1 0

Re: [alsa-devel] [PATCH] ASoC: fsl_ssi: Fix irq_of_parse_and_map() return value check
by Mark Brown 17 Oct '13

17 Oct '13

On Wed, Oct 02, 2013 at 09:15:22PM -0700, Guenter Roeck wrote: > irq_of_parse_and_map() returns 0 on error, not NO_IRQ. Applied, thanks.

2 2

[alsa-devel] [PATCH v4] ALSA: Add SoC on-chip internal memory support for DMA buffer allocation
by Nicolin Chen 17 Oct '13

17 Oct '13

Now it's quite common that an SoC contains its on-chip internal memory. By using this memory space for DMA buffer during audio playback/record, we can shutdown the voltage for external memory to save power. Thus add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v3->v4: * Appropriately placed '#ifdef CONFIG_OF' * Tested by adding '#undef CONFIG_OF' to modified files, and passed compiling. v2->v3: * Moved iram specific mmap procedure out of ifdef ARCH_HAS_DMA_MMAP_COHERENT v1->v2: * Added of_node check * Dropped noops and unused physical addr in snd_free_dev_iram() --- include/sound/memalloc.h | 5 ++++ sound/core/memalloc.c | 71 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 7 +++++ 3 files changed, 83 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b82..b53443e 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,11 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#ifdef CONFIG_OF +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ +#else +#define SNDRV_DMA_TYPE_DEV_IRAM SNDRV_DMA_TYPE_DEV +#endif /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f..fc0f27b 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> +#include <linux/genalloc.h> #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> @@ -157,6 +158,61 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +#ifdef CONFIG_OF +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal memory + * @dev: DMA device pointer + * @size: number of bytes to allocate from the iram + * @dma: dma-view physical address + * + * Return cpu-view address or NULL indicating allocating failure + * + * This function requires iram phandle provided via of_node + */ +void *snd_malloc_dev_iram(struct device *dev, size_t size, dma_addr_t *dma) +{ + struct gen_pool *pool; + unsigned long vaddr; + + if (!dev->of_node) + return NULL; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + if (!pool) + return NULL; + + vaddr = gen_pool_alloc(pool, size); + if (!vaddr) + return NULL; + + *dma = gen_pool_virt_to_phys(pool, vaddr); + + return (void *)vaddr; +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal memory + * @dev: DMA device pointer + * @size: size in bytes of memory to free + * @ptr: cpu-view address returned from snd_malloc_dev_iram + * + * This function requires iram phandle provided via of_node + */ +void snd_free_dev_iram(struct device *dev, size_t size, void *ptr) +{ + struct gen_pool *pool; + + if (!dev->of_node) + return; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + if (!pool) + return; + + gen_pool_free(pool, (unsigned long)ptr, size); +} +#endif /* CONFIG_OF */ #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +253,16 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA +#ifdef CONFIG_OF + case SNDRV_DMA_TYPE_DEV_IRAM: + dmab->area = snd_malloc_dev_iram(device, size, &dmab->addr); + if (dmab->area) + break; + /* Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; +#endif case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +335,11 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA +#ifdef CONFIG_OF + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_free_dev_iram(dmab->dev.dev, dmab->bytes, dmab->area); + break; +#endif case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..74de2a3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3199,6 +3199,13 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM && + SNDRV_DMA_TYPE_DEV_IRAM != SNDRV_DMA_TYPE_DEV) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) -- 1.8.4

4 6

[alsa-devel] [PATCH] sound: pci: emu10k1: Remove useless casting in emufx.c
by Geyslan G. Bem 17 Oct '13

17 Oct '13

Casting (void *) value returned by kcalloc is useless as mentioned in Documentation/CodingStyle, Chap 14. Signed-off-by: Geyslan G. Bem <geyslan(a)gmail.com> --- sound/pci/emu10k1/emufx.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 0275209..3f51dac 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1183,9 +1183,8 @@ static int _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) mm_segment_t seg; if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL || - (icode->gpr_map = (u_int32_t __user *) - kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || + (icode->gpr_map = kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t), + GFP_KERNEL)) == NULL || (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, sizeof(*controls), GFP_KERNEL)) == NULL) { err = -ENOMEM; @@ -1815,10 +1814,9 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL) return -ENOMEM; - if ((icode->gpr_map = (u_int32_t __user *) - kcalloc(256 + 160 + 160 + 2 * 512, sizeof(u_int32_t), - GFP_KERNEL)) == NULL || - (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, + if ((icode->gpr_map = kcalloc(256 + 160 + 160 + 2 * 512, sizeof(u_int32_t), + GFP_KERNEL)) == NULL || + (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS, sizeof(struct snd_emu10k1_fx8010_control_gpr), GFP_KERNEL)) == NULL || (ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL)) == NULL) { -- 1.8.4

3 3

[alsa-devel] [RFC PATCH 0/2] Enable mute LEDs and mic mute LEDs on Thinkpad
by David Henningsson 16 Oct '13

16 Oct '13

Hi! These two patches bind the thinkpad-acpi and the snd-hda-intel drivers together so that the mute and mic-mute LEDs correspond to the current mute/micmute status of the internal sound card. The patches are against Takashi's sound tree and are working on a Thinkpad X220-tablet, but the patches are otherwise in draft quality. Which means that there are several outstanding issues that would need feedback from upstream. First, I'd like feedback on the connection between the drivers - Takashi suggested that we could bind the drivers together like we've recently done when connecting the snd-hda-intel and the i915 driver, so this follows a similar scheme, with exported symbols from the thinkpad-acpi driver and a symbol_request on the snd-hda-intel side. This also opens up for future improvements so that the hardware mute/volume does not need to be its own card, but instead properly integrate with the output it actually controls on the kernel side. Second, about the LEDs as a security feature. As implemented in these patches there is no userspace interface at all, which would meet Matthew's concern about that. A future improvement could be that an alsamixer control was added, that allows the LED to be turned off, but not on. Which means, if PulseAudio or other daemon knows that recording goes on from another source (e g bluetooth or network connected mic) it could turn the mic mute LED off to indicate that to the user, but the only way to turn the LED on is to actually mute the internal card's hardware. Finally, it is my hope that I will get some quick and constructive feedback so we can have it all in order to be merged in time for the 3.13 kernel merge window. I'll be at Linuxcon Europe next week to answer questions if you have any, and we might also spend some time talking about it on the audio meeting on Monday. Thanks in advance! David Henningsson (2): thinkpad-acpi: Add mute and mic-mute LED functionality ALSA: hda - add connection to thinkpad_acpi to control mute/micmute LEDs drivers/platform/x86/thinkpad_acpi.c | 94 +++++++++++++++++++++++++++++++++- include/linux/thinkpad_acpi.h | 10 ++++ sound/pci/hda/patch_conexant.c | 78 ++++++++++++++++++++++++++++ 3 files changed, 181 insertions(+), 1 deletion(-) create mode 100644 include/linux/thinkpad_acpi.h -- 1.7.9.5

3 7

[alsa-devel] [PATCH 1/6] ASoC: cs4271: Include linux/of.h header
by Sachin Kamat 16 Oct '13

16 Oct '13

'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/cs4271.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index a20f1bb..f6e9534 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -25,6 +25,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> #include <sound/pcm.h> -- 1.7.9.5

2 11

[alsa-devel] [PATCH 1/2] ASoC: rt5640: Power down LDO while suspended
by Mark Brown 16 Oct '13

16 Oct '13

From: Mark Brown <broonie(a)linaro.org> If we have control over the LDO then disable it during suspend; the device is already being put into reset so will be non-functional over suspend anyway and this will save a small amount of power. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/rt5640.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 641eeeb..b0cde92 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1979,12 +1979,20 @@ static int rt5640_suspend(struct snd_soc_codec *codec) rt5640_reset(codec); regcache_cache_only(rt5640->regmap, true); regcache_mark_dirty(rt5640->regmap); + if (gpio_is_valid(rt5640->pdata.ldo1_en)) + gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 0); return 0; } static int rt5640_resume(struct snd_soc_codec *codec) { + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(rt5640->pdata.ldo1_en)) { + gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 1); + msleep(400); + } rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- 1.8.4.rc3

2 2

[alsa-devel] dsp after dmix
by Michael Trimarchi 16 Oct '13

16 Oct '13

Hi all, I have written a plugin to use the dsp of omapl138 and filtering the audio stream. My problem is to apply filtering after dmix plugin. Is there a way to have dsp filtering after alsa mixing? Michael

4 11

[alsa-devel] [PATCH 1/1] ASoC: smdk_wm8994: Add .pm to struct smdk_audio_driver
by Sachin Kamat 16 Oct '13

16 Oct '13

Register PM ops for this driver. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/smdk_wm8994.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 831972d..b072bd1 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -9,6 +9,7 @@ #include "../codecs/wm8994.h" #include <sound/pcm_params.h> +#include <sound/soc.h> #include <linux/module.h> #include <linux/of.h> #include <linux/of_device.h> @@ -206,6 +207,7 @@ static struct platform_driver smdk_audio_driver = { .name = "smdk-audio-wm8894", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), + .pm = &snd_soc_pm_ops, }, .probe = smdk_audio_probe, }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v3] ALSA: Add SoC on-chip internal memory support for DMA buffer allocation
by Nicolin Chen 16 Oct '13

16 Oct '13

Now it's quite common that an SoC contains its on-chip internal memory. By using this memory space for DMA buffer during audio playback/record, we can shutdown the voltage for external memory to save power. Thus add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v2->v3: * Moved iram specific mmap procedure out of ifdef ARCH_HAS_DMA_MMAP_COHERENT v1->v2: * Added of_node check * Dropped noops and unused physical addr in snd_free_dev_iram() --- include/sound/memalloc.h | 5 ++++ sound/core/memalloc.c | 72 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 7 +++++ 3 files changed, 84 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b82..b53443e 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,11 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#ifdef CONFIG_OF +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ +#else +#define SNDRV_DMA_TYPE_DEV_IRAM SNDRV_DMA_TYPE_DEV +#endif /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f..2957938 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> +#include <linux/genalloc.h> #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> @@ -157,6 +158,61 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +#ifdef CONFIG_OF +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal memory + * @dev: DMA device pointer + * @size: number of bytes to allocate from the iram + * @dma: dma-view physical address + * + * Return cpu-view address or NULL indicating allocating failure + * + * This function requires iram phandle provided via of_node + */ +void *snd_malloc_dev_iram(struct device *dev, size_t size, dma_addr_t *dma) +{ + struct gen_pool *pool; + unsigned long vaddr; + + if (!dev->of_node) + return NULL; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + if (!pool) + return NULL; + + vaddr = gen_pool_alloc(pool, size); + if (!vaddr) + return NULL; + + *dma = gen_pool_virt_to_phys(pool, vaddr); + + return (void *)vaddr; +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal memory + * @dev: DMA device pointer + * @size: size in bytes of memory to free + * @ptr: cpu-view address returned from snd_malloc_dev_iram + * + * This function requires iram phandle provided via of_node + */ +void snd_free_dev_iram(struct device *dev, size_t size, void *ptr) +{ + struct gen_pool *pool; + + if (!dev->of_node) + return; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + if (!pool) + return; + + gen_pool_free(pool, (unsigned long)ptr, size); +} +#endif /* CONFIG_OF */ #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +253,17 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: +#ifdef CONFIG_OF + dmab->area = snd_malloc_dev_iram(device, size, &dmab->addr); + if (dmab->area) + break; +#endif + /* + * Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +336,11 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: +#ifdef CONFIG_OF + snd_free_dev_iram(dmab->dev.dev, dmab->bytes, dmab->area); + break; +#endif case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..74de2a3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3199,6 +3199,13 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM && + SNDRV_DMA_TYPE_DEV_IRAM != SNDRV_DMA_TYPE_DEV) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) -- 1.8.4

2 1

[alsa-devel] [PATCH v2 1/2] ALSA: Add SoC on-chip internal memory support for DMA buffer allocation
by Nicolin Chen 16 Oct '13

16 Oct '13

Now it's quite common that an SoC contains its on-chip internal memory. By using this memory space for DMA buffer during audio playback/record, we can shutdown the voltage for external memory to save power. Thus add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v1->v2: * Added of_node check * Dropped noops and unused physical addr in snd_free_dev_iram() --- include/sound/memalloc.h | 5 ++++ sound/core/memalloc.c | 72 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 8 +++++- 3 files changed, 84 insertions(+), 1 deletion(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b82..b53443e 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,11 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#ifdef CONFIG_OF +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ +#else +#define SNDRV_DMA_TYPE_DEV_IRAM SNDRV_DMA_TYPE_DEV +#endif /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f..2957938 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> +#include <linux/genalloc.h> #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> @@ -157,6 +158,61 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +#ifdef CONFIG_OF +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal memory + * @dev: DMA device pointer + * @size: number of bytes to allocate from the iram + * @dma: dma-view physical address + * + * Return cpu-view address or NULL indicating allocating failure + * + * This function requires iram phandle provided via of_node + */ +void *snd_malloc_dev_iram(struct device *dev, size_t size, dma_addr_t *dma) +{ + struct gen_pool *pool; + unsigned long vaddr; + + if (!dev->of_node) + return NULL; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + if (!pool) + return NULL; + + vaddr = gen_pool_alloc(pool, size); + if (!vaddr) + return NULL; + + *dma = gen_pool_virt_to_phys(pool, vaddr); + + return (void *)vaddr; +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal memory + * @dev: DMA device pointer + * @size: size in bytes of memory to free + * @ptr: cpu-view address returned from snd_malloc_dev_iram + * + * This function requires iram phandle provided via of_node + */ +void snd_free_dev_iram(struct device *dev, size_t size, void *ptr) +{ + struct gen_pool *pool; + + if (!dev->of_node) + return; + + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + if (!pool) + return; + + gen_pool_free(pool, (unsigned long)ptr, size); +} +#endif /* CONFIG_OF */ #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +253,17 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: +#ifdef CONFIG_OF + dmab->area = snd_malloc_dev_iram(device, size, &dmab->addr); + if (dmab->area) + break; +#endif + /* + * Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +336,11 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: +#ifdef CONFIG_OF + snd_free_dev_iram(dmab->dev.dev, dmab->bytes, dmab->area); + break; +#endif case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..5578bd0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3201,12 +3201,18 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && - substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { return dma_mmap_coherent(substream->dma_buffer.dev.dev, area, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); + } else if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #elif defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV && !plat_device_is_coherent(substream->dma_buffer.dev.dev)) -- 1.8.4

2 3

[alsa-devel] [patch] ALSA: hdsp - info leak in snd_hdsp_hwdep_ioctl()
by Dan Carpenter 16 Oct '13

16 Oct '13

In GCC the sizeof(hdsp_version) is 8 because there is a 2 byte hole at the end of the struct after ->firmware_rev. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 4f255df..f59a321 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4845,6 +4845,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if ((err = hdsp_get_iobox_version(hdsp)) < 0) return err; } + memset(&hdsp_version, 0, sizeof(hdsp_version)); hdsp_version.io_type = hdsp->io_type; hdsp_version.firmware_rev = hdsp->firmware_rev; if ((err = copy_to_user(argp, &hdsp_version, sizeof(hdsp_version))))

2 1

[alsa-devel] [PATCH 1/2] ALSA: Add SoC on-chip internal memory support for DMA buffer allocation
by Nicolin Chen 16 Oct '13

16 Oct '13

Now it's quite common that an SoC contains its on-chip internal memory. By using this memory space for DMA buffer during audio playback/record, we can shutdown the voltage for external memory to save power. Thus add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- include/sound/memalloc.h | 5 ++++ sound/core/memalloc.c | 71 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 8 +++++- 3 files changed, 83 insertions(+), 1 deletion(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b82..b53443e 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,11 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#ifdef CONFIG_OF +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ +#else +#define SNDRV_DMA_TYPE_DEV_IRAM SNDRV_DMA_TYPE_DEV +#endif /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f..9136d5d 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> +#include <linux/genalloc.h> #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> @@ -157,6 +158,59 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +#ifdef CONFIG_OF +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal memory + * @dev: DMA device pointer + * @size: number of bytes to allocate from the iram + * @dma: dma-view physical address + * + * Return cpu-view address or NULL indicating allocating failure + * + * This function requires iram phandle provided via of_node + */ +void *snd_malloc_dev_iram(struct device *dev, size_t size, dma_addr_t *dma) +{ + struct gen_pool *pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + unsigned long vaddr; + + if (!pool) + return NULL; + + vaddr = gen_pool_alloc(pool, size); + if (!vaddr) + return NULL; + + if (dma) + *dma = gen_pool_virt_to_phys(pool, vaddr); + + return (void *)vaddr; +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal memory + * @dev: DMA device pointer + * @size: size in bytes of memory to free + * @ptr: cpu-view address returned from snd_malloc_dev_iram + * @dma: dma-view address returned from snd_malloc_dev_iram + * + * This function requires iram phandle provided via of_node + */ +void snd_free_dev_iram(struct device *dev, size_t size, void *ptr, dma_addr_t dma) +{ + struct gen_pool *pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + if (!pool) + return; + + gen_pool_free(pool, (unsigned long)ptr, size); + + ptr = NULL; + + if (dma) + dma = 0; +} +#endif /* CONFIG_OF */ #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +251,17 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: +#ifdef CONFIG_OF + dmab->area = snd_malloc_dev_iram(device, size, &dmab->addr); + if (dmab->area) + break; +#endif + /* + * Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +334,12 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: +#ifdef CONFIG_OF + snd_free_dev_iram(dmab->dev.dev, dmab->bytes, + dmab->area, dmab->addr); + break; +#endif case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..5578bd0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3201,12 +3201,18 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && - substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { return dma_mmap_coherent(substream->dma_buffer.dev.dev, area, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); + } else if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #elif defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV && !plat_device_is_coherent(substream->dma_buffer.dev.dev)) -- 1.8.4

2 5

[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: Add an interface to override its functions
by Nicolin Chen 15 Oct '13

15 Oct '13

Each implementation of gerneric pcm dmaengine has been naturally limited by the pre-defined gerneric functions. Thus add an extra interface for user to create a backdoor so that they can override those fixed functions for some uncommon requirements: using on-chip memory for DMA buffers and accordingly different mmap function for example, while at the meantime, they can continue to benifit from the concise and wise generic pcm dmaengine. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- include/sound/dmaengine_pcm.h | 13 +++++++++++++ sound/core/pcm_dmaengine.c | 25 +++++++++++++++++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 30 ++++++++++++++++++------------ 3 files changed, 56 insertions(+), 12 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f11c35c..a6e9d04 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -33,6 +33,18 @@ snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream) return DMA_DEV_TO_MEM; } +struct dmaengine_pcm { + struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; + const struct snd_dmaengine_pcm_config *config; + struct snd_soc_platform platform; + unsigned int flags; +}; + +static inline struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) +{ + return container_of(p, struct dmaengine_pcm, platform); +} + int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd); @@ -123,6 +135,7 @@ struct snd_dmaengine_pcm_config { struct snd_pcm_substream *substream); dma_filter_fn compat_filter_fn; + const struct snd_soc_platform_driver *driver; const struct snd_pcm_hardware *pcm_hardware; unsigned int prealloc_buffer_size; }; diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index aa924d9..d78a67e 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -186,8 +186,14 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + const struct snd_dmaengine_pcm_config *config = pcm->config; int ret; + if (config->driver->ops->trigger) + return config->driver->ops->trigger(substream, cmd); + switch (cmd) { case SNDRV_PCM_TRIGGER_START: ret = dmaengine_pcm_prepare_and_submit(substream); @@ -224,6 +230,13 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + const struct snd_dmaengine_pcm_config *config = pcm->config; + + if (config->driver->ops->pointer) + return config->driver->ops->pointer(substream); + return bytes_to_frames(substream->runtime, prtd->pos); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); @@ -238,11 +251,17 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + const struct snd_dmaengine_pcm_config *config = pcm->config; struct dma_tx_state state; enum dma_status status; unsigned int buf_size; unsigned int pos = 0; + if (config->driver->ops->pointer) + return config->driver->ops->pointer(substream); + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { buf_size = snd_pcm_lib_buffer_bytes(substream); @@ -341,6 +360,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan); int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + const struct snd_dmaengine_pcm_config *config = pcm->config; + + if (config->driver->ops->close) + return config->driver->ops->close(substream); kfree(prtd); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index e29ec3c..6e66c53 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -24,18 +24,6 @@ #include <sound/dmaengine_pcm.h> -struct dmaengine_pcm { - struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; - const struct snd_dmaengine_pcm_config *config; - struct snd_soc_platform platform; - unsigned int flags; -}; - -static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) -{ - return container_of(p, struct dmaengine_pcm, platform); -} - /** * snd_dmaengine_pcm_prepare_slave_config() - Generic prepare_slave_config callback * @substream: PCM substream @@ -75,9 +63,13 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + const struct snd_dmaengine_pcm_config *config = pcm->config; struct dma_slave_config slave_config; int ret; + if (config->driver->ops->hw_params) + return config->driver->ops->hw_params(substream, params); + if (pcm->config->prepare_slave_config) { ret = pcm->config->prepare_slave_config(substream, params, &slave_config); @@ -96,9 +88,13 @@ static int dmaengine_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + const struct snd_dmaengine_pcm_config *config = pcm->config; struct dma_chan *chan = pcm->chan[substream->stream]; int ret; + if (config->driver->ops->open) + return config->driver->ops->open(substream); + ret = snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); if (ret) @@ -118,6 +114,13 @@ static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, static void dmaengine_pcm_free(struct snd_pcm *pcm) { + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct dmaengine_pcm *dma_pcm = soc_platform_to_pcm(rtd->platform); + const struct snd_dmaengine_pcm_config *config = dma_pcm->config; + + if (config->driver->pcm_free) + return config->driver->pcm_free(pcm); + snd_pcm_lib_preallocate_free_for_all(pcm); } @@ -145,6 +148,9 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) unsigned int i; int ret; + if (config->driver->pcm_new) + return config->driver->pcm_new(rtd); + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { substream = rtd->pcm->streams[i].substream; if (!substream) -- 1.8.4

3 8

[alsa-devel] ALSA Driver Source - available outside of kernel?
by Paul Smedley 14 Oct '13

14 Oct '13

Hi All, As you may be aware, alsa is the basis of an audio driver uniaud for the os/2 operating system (http://svn.netlabs.org/uniaud). Up to alsa 1.0.25 I could download just the alsa-driver source in order to update uniaud, I see now the source is part of the kernel Is it possible to download just the alsa-driver 1.0.27.2 source from anywhere, or do i need to extract it from current linux kernel source? We also utilised things like adriver.h that were present to support older linux kernels - do these still exist? I tried mcking around with git and could only find 1.0.25 source. Thanks, Paul

1 0

[alsa-devel] [PATCH 0/3] ASoC: mxs-saif error handling
by Markus Pargmann 14 Oct '13

14 Oct '13

Hi, This series contains error handling patches for mxs-saif. The first patch is a comment about the possible command sequence of trigger functions. Regards, Markus Pargmann Markus Pargmann (3): ASoC: snd_soc_dai_ops trigger function description ASoC: mxs-saif: Store saif state ASoC: mxs-saif: Handle errors in trigger function include/sound/soc-dai.h | 7 +++++++ sound/soc/mxs/mxs-saif.c | 30 +++++++++++++++++++++++++----- sound/soc/mxs/mxs-saif.h | 5 +++++ 3 files changed, 37 insertions(+), 5 deletions(-)

3 6

[alsa-devel] [PATCH 1/2] ASoC: pcm1681: Fix max_register setting
by Axel Lin 14 Oct '13

14 Oct '13

According to the datasheet, the max_register is 13h. ARRAY_SIZE(pcm1681_reg_defaults) + 1 is 18 which is wrong. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 651ce09..c91eba5 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -270,7 +270,7 @@ MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); static const struct regmap_config pcm1681_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1, + .max_register = 0x13, .reg_defaults = pcm1681_reg_defaults, .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), .writeable_reg = pcm1681_writeable_reg, -- 1.8.1.2

2 3

[alsa-devel] tascam us122l patch commit
by James Miller 14 Oct '13

14 Oct '13

Hi, I have just attempted to install the 3.10.10 realtime kernel and I have discovered that the patch l listed in the following webpage : http://www.remastersys.com/forums/index.php?topic=3013.60 doesn't seem to have made it 'upstream' (I am not particularly familiar with the terminology). It's simple to do by hand but I can confirm that I have been using the above patch (VM_IO) on 3.8.13 and 3.10.10 kernels with success - the tascam us122L works fine. Many thanks -- James Stewart Miller Bsc(hons) Psych.

2 2

[alsa-devel] [PATCH] ASoC: tegra: support new register layouts in Tegra124
by Stephen Warren 14 Oct '13

14 Oct '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra124 introduces some small changes to the layout of some registers. Modify the affected drivers to program those registers appropriately based on which SoC they're running on. Tegra124 also introduced some new modules on the AHUB configlink register bus. These will require new entries in configlink_clocks[] in the AHUB driver. However, supporting that change likely relies on switching Tegra to the common reset framework, so I'll defer that change for now. Based-on-work-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Based-on-work-by: Songhee Baek <sbaek(a)nvidia.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_ahub.c | 115 ++++++++++++++++++++++++++++++++----- sound/soc/tegra/tegra30_ahub.h | 38 +++++++++++- sound/soc/tegra/tegra30_i2s.c | 51 +++++++++++----- sound/soc/tegra/tegra30_i2s.h | 7 +++ sound/soc/tegra/tegra_asoc_utils.c | 2 + sound/soc/tegra/tegra_asoc_utils.h | 1 + 6 files changed, 186 insertions(+), 28 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d554d46..bdd19db 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -100,6 +100,7 @@ int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, { int channel; u32 reg, val; + struct tegra30_ahub_cif_conf cif_conf; channel = find_first_zero_bit(ahub->rx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); @@ -123,15 +124,21 @@ int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16; tegra30_apbif_write(reg, val); + cif_conf.threshold = 0; + cif_conf.audio_channels = 2; + cif_conf.client_channels = 2; + cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.expand = 0; + cif_conf.stereo_conv = 0; + cif_conf.replicate = 0; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_RX; + cif_conf.truncate = 0; + cif_conf.mono_conv = 0; + reg = TEGRA30_AHUB_CIF_RX_CTRL + (channel * TEGRA30_AHUB_CIF_RX_CTRL_STRIDE); - val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | - TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; - tegra30_apbif_write(reg, val); + ahub->soc_data->set_audio_cif(ahub->regmap_apbif, reg, &cif_conf); return 0; } @@ -183,6 +190,7 @@ int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, { int channel; u32 reg, val; + struct tegra30_ahub_cif_conf cif_conf; channel = find_first_zero_bit(ahub->tx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); @@ -206,15 +214,21 @@ int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16; tegra30_apbif_write(reg, val); + cif_conf.threshold = 0; + cif_conf.audio_channels = 2; + cif_conf.client_channels = 2; + cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.expand = 0; + cif_conf.stereo_conv = 0; + cif_conf.replicate = 0; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_TX; + cif_conf.truncate = 0; + cif_conf.mono_conv = 0; + reg = TEGRA30_AHUB_CIF_TX_CTRL + (channel * TEGRA30_AHUB_CIF_TX_CTRL_STRIDE); - val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | - TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - tegra30_apbif_write(reg, val); + ahub->soc_data->set_audio_cif(ahub->regmap_apbif, reg, &cif_conf); return 0; } @@ -437,13 +451,21 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { static struct tegra30_ahub_soc_data soc_data_tegra30 = { .clk_list_mask = CLK_LIST_MASK_TEGRA30, + .set_audio_cif = tegra30_ahub_set_cif, }; static struct tegra30_ahub_soc_data soc_data_tegra114 = { .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .set_audio_cif = tegra30_ahub_set_cif, +}; + +static struct tegra30_ahub_soc_data soc_data_tegra124 = { + .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .set_audio_cif = tegra124_ahub_set_cif, }; static const struct of_device_id tegra30_ahub_of_match[] = { + { .compatible = "nvidia,tegra124-ahub", .data = &soc_data_tegra124 }, { .compatible = "nvidia,tegra114-ahub", .data = &soc_data_tegra114 }, { .compatible = "nvidia,tegra30-ahub", .data = &soc_data_tegra30 }, {}, @@ -497,6 +519,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, ahub); + ahub->soc_data = soc_data; ahub->dev = &pdev->dev; ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio"); @@ -669,6 +692,70 @@ static struct platform_driver tegra30_ahub_driver = { }; module_platform_driver(tegra30_ahub_driver); +void tegra30_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf) +{ + unsigned int value; + + value = (conf->threshold << + TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + ((conf->audio_channels - 1) << + TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + ((conf->client_channels - 1) << + TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + (conf->audio_bits << + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) | + (conf->client_bits << + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) | + (conf->expand << + TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) | + (conf->stereo_conv << + TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) | + (conf->replicate << + TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT) | + (conf->direction << + TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) | + (conf->truncate << + TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) | + (conf->mono_conv << + TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT); + + regmap_write(regmap, reg, value); +} +EXPORT_SYMBOL_GPL(tegra30_ahub_set_cif); + +void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf) +{ + unsigned int value; + + value = (conf->threshold << + TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + ((conf->audio_channels - 1) << + TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + ((conf->client_channels - 1) << + TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + (conf->audio_bits << + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) | + (conf->client_bits << + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) | + (conf->expand << + TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) | + (conf->stereo_conv << + TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) | + (conf->replicate << + TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT) | + (conf->direction << + TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) | + (conf->truncate << + TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) | + (conf->mono_conv << + TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT); + + regmap_write(regmap, reg, value); +} +EXPORT_SYMBOL_GPL(tegra124_ahub_set_cif); + MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); MODULE_DESCRIPTION("Tegra30 AHUB driver"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h index 09766cd..d67321d 100644 --- a/sound/soc/tegra/tegra30_ahub.h +++ b/sound/soc/tegra/tegra30_ahub.h @@ -25,16 +25,30 @@ #define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0xf #define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) +#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT 24 +#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0x3f +#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) + /* Channel count minus 1 */ #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 24 #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 7 #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) /* Channel count minus 1 */ +#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 20 +#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 0xf +#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) + +/* Channel count minus 1 */ #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 7 #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) +/* Channel count minus 1 */ +#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 +#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 0xf +#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) + #define TEGRA30_AUDIOCIF_BITS_4 0 #define TEGRA30_AUDIOCIF_BITS_8 1 #define TEGRA30_AUDIOCIF_BITS_12 2 @@ -86,7 +100,7 @@ #define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH1 (TEGRA30_AUDIOCIF_STEREO_CONV_CH1 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) #define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_AVG (TEGRA30_AUDIOCIF_STEREO_CONV_AVG << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) -#define TEGRA30_AUDIOCIF_CTRL_REPLICATE 3 +#define TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT 3 #define TEGRA30_AUDIOCIF_DIRECTION_TX 0 #define TEGRA30_AUDIOCIF_DIRECTION_RX 1 @@ -468,8 +482,30 @@ extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, enum tegra30_ahub_txcif txcif); extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif); +struct tegra30_ahub_cif_conf { + unsigned int threshold; + unsigned int audio_channels; + unsigned int client_channels; + unsigned int audio_bits; + unsigned int client_bits; + unsigned int expand; + unsigned int stereo_conv; + unsigned int replicate; + unsigned int direction; + unsigned int truncate; + unsigned int mono_conv; +}; + +void tegra30_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf); +void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf); + struct tegra30_ahub_soc_data { u32 clk_list_mask; + void (*set_audio_cif)(struct regmap *regmap, + unsigned int reg, + struct tegra30_ahub_cif_conf *conf); /* * FIXME: There are many more differences in HW, such as: * - More APBIF channels. diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 47565fd04..5f20b69 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -30,6 +30,7 @@ #include <linux/io.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> @@ -179,6 +180,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); unsigned int mask, val, reg; int ret, sample_size, srate, i2sclock, bitcnt; + struct tegra30_ahub_cif_conf cif_conf; if (params_channels(params) != 2) return -EINVAL; @@ -217,21 +219,26 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val); - val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | - TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16; + cif_conf.threshold = 0; + cif_conf.audio_channels = 2; + cif_conf.client_channels = 2; + cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.expand = 0; + cif_conf.stereo_conv = 0; + cif_conf.replicate = 0; + cif_conf.truncate = 0; + cif_conf.mono_conv = 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_RX; reg = TEGRA30_I2S_CIF_RX_CTRL; } else { - val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_TX; reg = TEGRA30_I2S_CIF_TX_CTRL; } - regmap_write(i2s->regmap, reg, val); + i2s->soc_data->set_audio_cif(i2s->regmap, reg, &cif_conf); val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) | (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT); @@ -396,9 +403,24 @@ static const struct regmap_config tegra30_i2s_regmap_config = { .cache_type = REGCACHE_RBTREE, }; +static const struct tegra30_i2s_soc_data tegra30_i2s_config = { + .set_audio_cif = tegra30_ahub_set_cif, +}; + +static const struct tegra30_i2s_soc_data tegra124_i2s_config = { + .set_audio_cif = tegra124_ahub_set_cif, +}; + +static const struct of_device_id tegra30_i2s_of_match[] = { + { .compatible = "nvidia,tegra124-i2s", .data = &tegra124_i2s_config }, + { .compatible = "nvidia,tegra30-i2s", .data = &tegra30_i2s_config }, + {}, +}; + static int tegra30_i2s_platform_probe(struct platform_device *pdev) { struct tegra30_i2s *i2s; + const struct of_device_id *match; u32 cif_ids[2]; struct resource *mem, *memregion; void __iomem *regs; @@ -412,6 +434,14 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, i2s); + match = of_match_device(tegra30_i2s_of_match, &pdev->dev); + if (!match) { + dev_err(&pdev->dev, "Error: No device match found\n"); + ret = -ENODEV; + goto err; + } + i2s->soc_data = (struct tegra30_i2s_soc_data *)match->data; + i2s->dai = tegra30_i2s_dai_template; i2s->dai.name = dev_name(&pdev->dev); @@ -539,11 +569,6 @@ static int tegra30_i2s_resume(struct device *dev) } #endif -static const struct of_device_id tegra30_i2s_of_match[] = { - { .compatible = "nvidia,tegra30-i2s", }, - {}, -}; - static const struct dev_pm_ops tegra30_i2s_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend, tegra30_i2s_runtime_resume, NULL) diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index bea23af..4d0b0a3 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -225,7 +225,14 @@ #define TEGRA30_I2S_LCOEF_COEF_MASK_US 0xffff #define TEGRA30_I2S_LCOEF_COEF_MASK (TEGRA30_I2S_LCOEF_COEF_MASK_US << TEGRA30_I2S_LCOEF_COEF_SHIFT) +struct tegra30_i2s_soc_data { + void (*set_audio_cif)(struct regmap *regmap, + unsigned int reg, + struct tegra30_ahub_cif_conf *conf); +}; + struct tegra30_i2s { + const struct tegra30_i2s_soc_data *soc_data; struct snd_soc_dai_driver dai; int cif_id; struct clk *clk_i2s; diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index d173880..1be311c 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -182,6 +182,8 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; else if (of_machine_is_compatible("nvidia,tegra114")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA114; + else if (of_machine_is_compatible("nvidia,tegra124")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA124; else { dev_err(data->dev, "SoC unknown to Tegra ASoC utils\n"); return -EINVAL; diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 19fdcaf..9577121 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -30,6 +30,7 @@ enum tegra_asoc_utils_soc { TEGRA_ASOC_UTILS_SOC_TEGRA20, TEGRA_ASOC_UTILS_SOC_TEGRA30, TEGRA_ASOC_UTILS_SOC_TEGRA114, + TEGRA_ASOC_UTILS_SOC_TEGRA124, }; struct tegra_asoc_utils_data { -- 1.8.1.5

2 1

[alsa-devel] [PATCH V2] sound/soc/fsl: remove leftover release_mem_region
by Wolfram Sang 14 Oct '13

14 Oct '13

When converting this driver to devm_ioremap_resource, the removal of this now unneeded function has been forgotten. Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> --- V2: Remove now unused variable. Thanks to Lars-Peter for pointing out! sound/soc/fsl/imx-ssi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f58bcd8..0272289 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -615,7 +615,6 @@ failed_pcm_dma: failed_pcm_fiq: snd_soc_unregister_component(&pdev->dev); failed_register: - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: snd_soc_set_ac97_ops(NULL); @@ -625,7 +624,6 @@ failed_clk: static int imx_ssi_remove(struct platform_device *pdev) { - struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); imx_pcm_dma_exit(pdev); @@ -636,7 +634,6 @@ static int imx_ssi_remove(struct platform_device *pdev) if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); snd_soc_set_ac97_ops(NULL); -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH v2] sound: soc: fsl: Fix memory leak in imx-audmux.c
by Felipe Pena 14 Oct '13

14 Oct '13

When audmux_clk is used and clk_prepare_enable function succeed, the memory alloc'd to buf variable is leaked Signed-off-by: Felipe Pena <felipensp(a)gmail.com> --- sound/soc/fsl/imx-audmux.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index d3bf71a..ac86993 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -66,13 +66,10 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { ssize_t ret; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; int port = (int)file->private_data; u32 pdcr, ptcr; - if (!buf) - return -ENOMEM; - if (audmux_clk) { ret = clk_prepare_enable(audmux_clk); if (ret) @@ -85,6 +82,10 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (audmux_clk) clk_disable_unprepare(audmux_clk); + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); -- 1.7.10.4

3 4

[alsa-devel] [Trivial PATCH] sound: Remove unnecessary semicolons
by Joe Perches 14 Oct '13

14 Oct '13

These aren't necessary after switch and if blocks. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/oss/sb_ess.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/twl6040.c | 4 ++-- sound/soc/fsl/fsl_spdif.c | 4 ++-- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 6 +++--- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 4 ++-- 12 files changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index c0be085..0e7254b 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -1544,7 +1544,7 @@ static int ess_has_rec_mixer (int submodel) return 1; default: return 0; - }; + } }; #ifdef FKS_LOGGING diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index b46dc9b..9fb03b4 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -671,7 +671,7 @@ static int snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) return err; break; #endif - }; + } if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_SPDIF) { for (i = 0; i < ARRAY_SIZE(snd_vortex_mixer_spdif); i++) { diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index bb53dea..8697ced 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -777,7 +777,7 @@ static int asoc_ssc_init(struct device *dev) if (ret) { dev_err(dev, "Could not register PCM: %d\n", ret); goto err_unregister_dai; - }; + } return 0; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 5f9af1f..49cc5f6 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -328,7 +328,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ak4641->playback_fs = rate; ak4641_set_deemph(codec); - }; + } return 0; } diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index ea141e1..1544330 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -382,7 +382,7 @@ static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai, break; default: return -EINVAL; - }; + } snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 6d31d88..e29cdb7 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -429,7 +429,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, default: dev_err(codec->dev, "Invalid bit width\n"); return -EINVAL; - }; + } ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); if (ret < 0) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3c79dbb..35059a2 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -246,7 +246,7 @@ static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, return priv->dl2_unmuted; default: return 1; - }; + } } /* @@ -1100,7 +1100,7 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i break; default: break; - }; + } } static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e..c0fea02 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -963,7 +963,7 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) @@ -982,7 +982,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config fsl_spdif_regmap_config = { diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 52af7f6..364bf6a 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -297,7 +297,7 @@ static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg) @@ -310,7 +310,7 @@ static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg) @@ -321,7 +321,7 @@ static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra20_i2s_regmap_config = { diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 551b3c9..08bc693 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -213,7 +213,7 @@ static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg) @@ -234,7 +234,7 @@ static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg) @@ -247,7 +247,7 @@ static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra20_spdif_regmap_config = { diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d554d46..805b1f3 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -346,7 +346,7 @@ static bool tegra30_ahub_apbif_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: break; - }; + } if (REG_IN_ARRAY(reg, CHANNEL_CTRL) || REG_IN_ARRAY(reg, CHANNEL_CLEAR) || @@ -381,7 +381,7 @@ static bool tegra30_ahub_apbif_volatile_reg(struct device *dev, return true; default: break; - }; + } if (REG_IN_ARRAY(reg, CHANNEL_CLEAR) || REG_IN_ARRAY(reg, CHANNEL_STATUS) || diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 47565fd04..bd71145 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -369,7 +369,7 @@ static bool tegra30_i2s_wr_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg) @@ -382,7 +382,7 @@ static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra30_i2s_regmap_config = {

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix inverted internal mic not indicated on some machines
by David Henningsson 14 Oct '13

14 Oct '13

The create_bind_cap_vol_ctl does not create any control indicating that an inverted dmic is present. Therefore, create multiple capture volumes in this scenario, so we always have some indication that the internal mic is inverted. This happens on the Lenovo Ideapad U310 as well as the Lenovo Yoga 13 (both are based on the CX20590 codec), but the fix is generic and could be needed for other codecs/machines too. Thanks to Szymon Acedański for the pointer and a draft patch. BugLink: https://bugs.launchpad.net/bugs/1239392 BugLink: https://bugs.launchpad.net/bugs/1227491 Reported-by: Szymon Acedański <accek(a)mimuw.edu.pl> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) I think this fix is safe enough to send to Linus and stable, but will leave the final judgement to Takashi. diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac41e9c..26ad4f0 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3531,7 +3531,7 @@ static int create_capture_mixers(struct hda_codec *codec) if (!multi) err = create_single_cap_vol_ctl(codec, n, vol, sw, inv_dmic); - else if (!multi_cap_vol) + else if (!multi_cap_vol && !inv_dmic) err = create_bind_cap_vol_ctl(codec, n, vol, sw); else err = create_multi_cap_vol_ctl(codec); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] sound/soc/fsl: remove leftover release_mem_region
by Wolfram Sang 13 Oct '13

13 Oct '13

When converting this driver to devm_ioremap_resource, the removal of this now unneeded function has been forgotten. Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> --- sound/soc/fsl/imx-ssi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f58bcd8..b2cc743 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -615,7 +615,6 @@ failed_pcm_dma: failed_pcm_fiq: snd_soc_unregister_component(&pdev->dev); failed_register: - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: snd_soc_set_ac97_ops(NULL); @@ -636,7 +635,6 @@ static int imx_ssi_remove(struct platform_device *pdev) if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); snd_soc_set_ac97_ops(NULL); -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] sound: soc: fsl: Fix memory leak in imx-audmux.c
by Felipe Pena 13 Oct '13

13 Oct '13

When audmux_clk is used and clk_prepare_enable function succeed, the memory alloc'd to buf variable is leaked Signed-off-by: Felipe Pena <felipensp(a)gmail.com> --- sound/soc/fsl/imx-audmux.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index d3bf71a..977759a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -66,19 +66,20 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { ssize_t ret; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; int port = (int)file->private_data; u32 pdcr, ptcr; - if (!buf) - return -ENOMEM; - if (audmux_clk) { ret = clk_prepare_enable(audmux_clk); if (ret) return ret; } + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); -- 1.7.10.4

1 0

[alsa-devel] Recording devices opened mono still sending stereo causing corrupt audio
by Sean M. Pappalardo - D.J. Pegasus 12 Oct '13

12 Oct '13

(Re-sending including alsa-devel and more information.) Hello there. I'm a developer for Mixxx, the cross-platform open-source DJ software which uses PortAudio and we've been trying for a long time to track down the cause of "horrible mic sound" as documented in our bug: https://bugs.launchpad.net/mixxx/+bug/900364 The resulting recorded audio sounds about 50-60% lower pitch with crackles and it only happens with the ALSA API on Linux and only with certain (mostly Intel-based) audio devices. I have just discovered by testing in Audacity (which also uses PA) that the problem occurs when PA (and therefore ALSA) is asked to open just one channel of a sound input device yet the device (or ALSA) seems to stream two anyway. I have been able to reproduce the problem in Audacity (which also uses PA) when selecting the ALSA API and mono recording, but it works correctly when I change it to stereo recording. I'm suspecting there's a miscommunication between PA and ALSA or ALSA and the sound hardware where either the lower level is saying its input(s) can only be opened in stereo and the higher level (or the application above it) ignores that, or the lower level says it's fine opening mono but then misbehaves. How can I find out which, whether it's ALSA, PA, or the sound hardware at fault? Sincerely, Sean M. Pappalardo "D.J. Pegasus" Mixxx Developer - Controller Specialist

5 6

[alsa-devel] [PATCH] ASoC: rcar: add ID check on rsnd_dai_get()
by Kuninori Morimoto 11 Oct '13

11 Oct '13

checking id in rsnd_dai_get() is good idea Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 28c24fc..b234ed6 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -318,6 +318,9 @@ int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id) { + if ((id < 0) || (id >= rsnd_dai_nr(priv))) + return NULL; + return priv->rdai + id; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rcar: fixup generation checker
by Kuninori Morimoto 11 Oct '13

11 Oct '13

Current rcar is using rsnd_is_gen1/gen2() to checking its IP generation, but it needs data mask. This patch fixes it up. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/rsnd.h | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index fe66533..fb0a312 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -68,6 +68,7 @@ struct rsnd_scu_platform_info { * * A : generation */ +#define RSND_GEN_MASK (0xF << 0) #define RSND_GEN1 (1 << 0) /* fixme */ #define RSND_GEN2 (2 << 0) /* fixme */ diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3868aaf..3b71c3f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -216,8 +216,8 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1) -#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2) +#define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1) +#define rsnd_is_gen2(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2) /* * R-Car ADG -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rcar: fixup rsnd_platform_call() return value
by Kuninori Morimoto 11 Oct '13

11 Oct '13

Un-implemented platform callback is not error. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index fc83f0f..28c24fc 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -103,7 +103,7 @@ * rsnd_platform functions */ #define rsnd_platform_call(priv, dai, func, param...) \ - (!(priv->info->func) ? -ENODEV : \ + (!(priv->info->func) ? 0 : \ priv->info->func(param)) /* -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH v4 0/5] dma: mxs-dma bugfixes and cleanup
by Fabio Estevam 11 Oct '13

11 Oct '13

Hi Markus, [Also including alsa-devel and some other folks involved with mxs audio] On Fri, Oct 11, 2013 at 9:12 AM, Markus Pargmann <mpa(a)pengutronix.de> wrote: > Hi, > > v4 adds another patch to increase sound stability. Patch 3 ("dma: mxs-dma: Fix > channel reset hardware bug") decreases the number of channel stalls, but after > hours of repeated playback the bug still appears. The new patch ("dma: mxs: Use > semaphores for cyclic DMA") replaces reset channel usage with hardware > semaphore counter mechanism. How is the bug you are trying to solve manifest and how to reproduce it? I have been investigating an issue about sound stability on mx28 as well. The usecase I am doing is: while true do arecord -D hw:0,1 -d 10 -f S16_LE -r 44100 -c 2 | aplay sleep 2 done (Record 10 seconds, sleep and keep in this loop forever). The problem I see is that randomly the captured audio becomes extremely noisy and loud (throughout the whole 10 seconds) and then in the subsequent loop it becomes normal. This is very random and it happens at a 1 or 2% failure rate. I tested your v3, but it did not help to fix this particular issue. I also tried experimenting with the SAIF IOMUX settings: index cda19c8..463fa92 100644 --- a/arch/arm/boot/dts/imx28.dtsi +++ b/arch/arm/boot/dts/imx28.dtsi @@ -545,9 +545,9 @@ MX28_PAD_SAIF0_BITCLK__SAIF0_BITCLK MX28_PAD_SAIF0_SDATA0__SAIF0_SDATA0 >; - fsl,drive-strength = <MXS_DRIVE_12mA>; + fsl,drive-strength = <MXS_DRIVE_4mA>; fsl,voltage = <MXS_VOLTAGE_HIGH>; - fsl,pull-up = <MXS_PULL_ENABLE>; + fsl,pull-up = <MXS_PULL_DISABLE>; }; saif0_pins_b: saif0@1 { @@ -557,9 +557,9 @@ MX28_PAD_SAIF0_BITCLK__SAIF0_BITCLK MX28_PAD_SAIF0_SDATA0__SAIF0_SDATA0 >; - fsl,drive-strength = <MXS_DRIVE_12mA>; + fsl,drive-strength = <MXS_DRIVE_4mA>; fsl,voltage = <MXS_VOLTAGE_HIGH>; - fsl,pull-up = <MXS_PULL_ENABLE>; + fsl,pull-up = <MXS_PULL_DISABLE>; }; saif1_pins_a: saif1@0 { @@ -567,9 +567,9 @@ fsl,pinmux-ids = < MX28_PAD_SAIF1_SDATA0__SAIF1_SDATA0 >; - fsl,drive-strength = <MXS_DRIVE_12mA>; + fsl,drive-strength = <MXS_DRIVE_4mA>; fsl,voltage = <MXS_VOLTAGE_HIGH>; - fsl,pull-up = <MXS_PULL_ENABLE>; + fsl,pull-up = <MXS_PULL_DISABLE>; }; and my tests are showing a much better stability. Not able to get this problem after several hours. Regards, Fabio Estevam

3 2

[alsa-devel] [PATCH 1/5] ARM: Kconfig: Move if ARCH_S3C64XX statement to mach-s3c64xx/Kconfig
by Tomasz Figa 11 Oct '13

11 Oct '13

All other platforms have this condition checked inside their own Kconfig files, so for consistency this patch makes it this way for mach-s3c64xx as well. Signed-off-by: Tomasz Figa <tomasz.figa(a)gmail.com> --- arch/arm/Kconfig | 2 -- arch/arm/mach-s3c64xx/Kconfig | 4 ++++ 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/arch/arm/Kconfig b/arch/arm/Kconfig index b766dad..dc51f8a 100644 --- a/arch/arm/Kconfig +++ b/arch/arm/Kconfig @@ -995,9 +995,7 @@ source "arch/arm/mach-sti/Kconfig" source "arch/arm/mach-s3c24xx/Kconfig" -if ARCH_S3C64XX source "arch/arm/mach-s3c64xx/Kconfig" -endif source "arch/arm/mach-s5p64x0/Kconfig" diff --git a/arch/arm/mach-s3c64xx/Kconfig b/arch/arm/mach-s3c64xx/Kconfig index bd14e3a..0e23910 100644 --- a/arch/arm/mach-s3c64xx/Kconfig +++ b/arch/arm/mach-s3c64xx/Kconfig @@ -3,6 +3,8 @@ # # Licensed under GPLv2 +if ARCH_S3C64XX + # temporary until we can eliminate all drivers using it. config PLAT_S3C64XX bool @@ -322,3 +324,5 @@ config MACH_S3C64XX_DT board. Note: This is under development and not all peripherals can be supported with this machine file. + +endif -- 1.8.3.2

5 9

[alsa-devel] [PATCH] pcm: use the same set of uint_XX types everywhere
by John Spencer 11 Oct '13

11 Oct '13

previously, pcm.h prototyped the functions snd_pcm_format_silence* with u_int_XX types (legacy BSD types from sys/types.h), while the implementation in pcm_misc.c used the C99 stdint.h types uint_XX. this caused compilation errors on systems where the typedef for u_int_XX and uint_XX is not exactly the same. we fix it by using the C99 stdint.h types everywhere. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/pcm.h | 9 +++++---- src/pcm/pcm_misc.c | 42 +++++++++++++++++++++--------------------- 2 files changed, 26 insertions(+), 25 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..cd847e4 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -33,6 +33,7 @@ extern "C" { #endif +#include <stdint.h> /** * \defgroup PCM PCM Interface * See the \ref pcm page for more details. @@ -1048,10 +1049,10 @@ int snd_pcm_format_width(snd_pcm_format_t format); /* in bits */ int snd_pcm_format_physical_width(snd_pcm_format_t format); /* in bits */ snd_pcm_format_t snd_pcm_build_linear_format(int width, int pwidth, int unsignd, int big_endian); ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples); -u_int8_t snd_pcm_format_silence(snd_pcm_format_t format); -u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format); -u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format); -u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format); +uint8_t snd_pcm_format_silence(snd_pcm_format_t format); +uint16_t snd_pcm_format_silence_16(snd_pcm_format_t format); +uint32_t snd_pcm_format_silence_32(snd_pcm_format_t format); +uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format); int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int samples); snd_pcm_sframes_t snd_pcm_bytes_to_frames(snd_pcm_t *pcm, ssize_t bytes); diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index d52160c..2a8be21 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -363,7 +363,7 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) * \param format Sample format * \return silence 64 bit word */ -u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) +uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S8: @@ -437,7 +437,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { float f[2]; - u_int64_t i; + uint64_t i; } u; u.f[0] = u.f[1] = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -450,7 +450,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { double f; - u_int64_t i; + uint64_t i; } u; u.f = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -463,7 +463,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { float f[2]; - u_int64_t i; + uint64_t i; } u; u.f[0] = u.f[1] = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -476,7 +476,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { double f; - u_int64_t i; + uint64_t i; } u; u.f = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -509,10 +509,10 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) * \param format Sample format * \return silence 32 bit word */ -u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format) +uint32_t snd_pcm_format_silence_32(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 32); - return (u_int32_t)snd_pcm_format_silence_64(format); + return (uint32_t)snd_pcm_format_silence_64(format); } /** @@ -520,10 +520,10 @@ u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format) * \param format Sample format * \return silence 16 bit word */ -u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format) +uint16_t snd_pcm_format_silence_16(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 16); - return (u_int16_t)snd_pcm_format_silence_64(format); + return (uint16_t)snd_pcm_format_silence_64(format); } /** @@ -531,10 +531,10 @@ u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format) * \param format Sample format * \return silence 8 bit word */ -u_int8_t snd_pcm_format_silence(snd_pcm_format_t format) +uint8_t snd_pcm_format_silence(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 8); - return (u_int8_t)snd_pcm_format_silence_64(format); + return (uint8_t)snd_pcm_format_silence_64(format); } /** @@ -550,7 +550,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; switch (snd_pcm_format_physical_width(format)) { case 4: { - u_int8_t silence = snd_pcm_format_silence_64(format); + uint8_t silence = snd_pcm_format_silence_64(format); unsigned int samples1; if (samples % 2 != 0) return -EINVAL; @@ -559,13 +559,13 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 8: { - u_int8_t silence = snd_pcm_format_silence_64(format); + uint8_t silence = snd_pcm_format_silence_64(format); memset(data, silence, samples); break; } case 16: { - u_int16_t silence = snd_pcm_format_silence_64(format); - u_int16_t *pdata = (u_int16_t *)data; + uint16_t silence = snd_pcm_format_silence_64(format); + uint16_t *pdata = (uint16_t *)data; if (! silence) memset(data, 0, samples * 2); else { @@ -575,8 +575,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 24: { - u_int32_t silence = snd_pcm_format_silence_64(format); - u_int8_t *pdata = (u_int8_t *)data; + uint32_t silence = snd_pcm_format_silence_64(format); + uint8_t *pdata = (uint8_t *)data; if (! silence) memset(data, 0, samples * 3); else { @@ -595,8 +595,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 32: { - u_int32_t silence = snd_pcm_format_silence_64(format); - u_int32_t *pdata = (u_int32_t *)data; + uint32_t silence = snd_pcm_format_silence_64(format); + uint32_t *pdata = (uint32_t *)data; if (! silence) memset(data, 0, samples * 4); else { @@ -606,8 +606,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 64: { - u_int64_t silence = snd_pcm_format_silence_64(format); - u_int64_t *pdata = (u_int64_t *)data; + uint64_t silence = snd_pcm_format_silence_64(format); + uint64_t *pdata = (uint64_t *)data; if (! silence) memset(data, 0, samples * 8); else { -- 1.7.3.4

4 6

[alsa-devel] [PATCH] ALSA: hda - Sony VAIO Pro 13 (haswell) now has a working headset jack
by David Henningsson 11 Oct '13

11 Oct '13

Just got the positive confirmation from a tester: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1227093/comments/28 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79e6fe7..bf313be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3761,6 +3761,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { } }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: mc13783: add mixer controls
by Steffen Trumtrar 11 Oct '13

11 Oct '13

Add more controls to the alsa mixer infrastructure. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> --- Changes in v2: - Fix some control names - remove controls that belong to DAPM sound/soc/codecs/mc13783.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index eedbf05..2b62737 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -541,8 +541,26 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + + SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), + SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), + SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), + + SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + + SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) -- 1.8.4.rc3

2 2

[alsa-devel] [PATCH v2 0/2] Sony vaio pro 13 (haswell)
by David Henningsson 11 Oct '13

11 Oct '13

Changes since v1: - Mic patch now applies to linux-stable branch linux-3.11.y cleanly - The two patches are now not connected to each other. I'll send a patch that connects them if/when we get some confirmation that this actually is a working headset mic pin. David Henningsson (2): ALSA: hda - Fix microphone for Sony VAIO Pro 13 (Haswell model) ALSA: hda - Add a headset mic model for ALC269 and friends Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 24 ++++++++++++++++++++++++ 2 files changed, 25 insertions(+) -- 1.7.9.5

2 3

[alsa-devel] [PATCH] local.h: include sys/types.h to fix issues with pcm.h
by John Spencer 11 Oct '13

11 Oct '13

sys/types.h is required for the u_int_XX types used by pcm.h. since a change in pcm.h is not desired, we add the inclusion to the header that includes pcm.h during build. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/local.h | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/include/local.h b/include/local.h index f0db188..9464efa 100644 --- a/include/local.h +++ b/include/local.h @@ -31,6 +31,7 @@ #include <endian.h> #include <stdarg.h> #include <sys/poll.h> +#include <sys/types.h> #include <errno.h> #include <linux/types.h> #include <linux/ioctl.h> -- 1.7.3.4

2 1

Re: [alsa-devel] Fast Track Ultra 8r...
by Felix Homann 11 Oct '13

11 Oct '13

Am 10.10.2013 19:41 schrieb "Petri Järvisalo" <petri.jarvisalo(a)gmail.com>: > > When plugging in the device at other OS, the internal routing for inputs is disabled and software routes straight 1 = 1 to 8 = 8, like this, and there is no effects applied. Those defaults are the worst I can think of for *my* use case. That's why I haven't implemented them. I do fully appreciate that those would be sane defaults for many people. I wouldn't mind anyone implementing them in Alsa. It's just, well, I won't ;-) When I started working on the FTU mixer support I didn't think about the defaults at all. The default we have right now are the ones you get without explicitly specifying any. But in fact, now I completely fell in love with these insane defaults: They give me an instant alert that I haven't restored my personal 'right' settings! That's of extreme value for my purposes as I use the device mainly on live occasions and the 'wrong' settings certainly cannot go unnoticed in the soundcheck. So again, anybody interested is invited to implement new defaults for the FTU. I won' t object any choice, how could I? But I won't do it myself since I'm personally very happy with the current state. Regards, Felix

3 4

[alsa-devel] ALSA: HDA Control and node attachment
by Jason Steigler 10 Oct '13

10 Oct '13

Hello, I was wondering if it's possible from user-space to change which node a control is attached to. In older ALSA (1.0.23), I had used the '3stack-6ch-dig' model to get 6 channel audio for an "Realtek ALC662 rev 1" codec. I had configured the front panel mic (node 0x19) to be an output using hda-verb that could be controlled by the Center and LFE mixer controls. The board also has jacks on the back and one of those would match the audio coming out the front. When I change the volume or mute the controls, they both were affected (which I expected/wanted). With newer ALSA (1.0.24 with debian Wheezy's stock kernel), the model '3stack-6ch-dig' doesn't exist, but the codec can be configured to 6 channels. I have made the same adjustments with hda-verb to make the node an output. The volume control works for the 'front' and 'rear' jacks, but only muting works on the 'rear' jack. I noticed that the control nodes for Center and LFE are not the same between the 2 setups. In the older version, they were tied to node 0x0e (Audio Mixer), and with the newer version node 0x18 (which looks like the 'Rear Mic' jack). I am not knowledgeable of the details of the HDA codec (yet), and am mostly curious if they can be changed in user-space or if I'll have to start learning the HDA codec code and try to find a way to move the controls around. I tried using hda-jack-retask and could assign the mic nodes as outputs, but wasn't sure if it would allow me to move controls on the HDA graph. Thanks in advance! Jason

1 0

[alsa-devel] Fast Track Ultra 8r...
by Petri Järvisalo 10 Oct '13

10 Oct '13

Hello, I'm new here and somewhat understand how to script php and python ... and very unskilled with linux... anyway i noticed that the default alsa mixer values for fast track ultra 8r are way off the optimal when the sound card is first time plugged in. What I expected was clear stereo/surround sound, what i got was distorted mono sound with reverb! Far away from optimal settings! So, I stated to seek knowledge on web, and finally stumbled upon a homepage that had some info about the Linux support. What i found out was that the sound card has internal routing system, zero latency and a effect processor.Also i found out that the routing works by controlling the input/output levels at alsa mixer. The problem, by default all alsa mixer values are assigned to 100%, this means the internal routing is routing everything to everything distorting the sound, and making the output mono! The website I found had suggestion and a bash script to correct the internal routing: http://heikki.ketoharju.info/2013/03/linux-and-fasttrackultra/#comment-989 alas, the main article is a good read too :) Well the script did not work for me, so i changed the settings manually to my alsa mixer, which took quite some time, and it works like a charm. Anyway my best wish would be to get a fix for this sound card, so it would "just work" when plugged in. Thanks for any info on how i could get this "patch" to be widely available so others could benefit it too. Best regards, Petri

2 1

[alsa-devel] [PATCH] pcm.h: include sys/types.h, necessary for u_int_XX types
by John Spencer 10 Oct '13

10 Oct '13

fixes compilation on musl libc. --- include/pcm.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..7ea9e84 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -29,6 +29,8 @@ #ifndef __ALSA_PCM_H #define __ALSA_PCM_H +#include <sys/types.h> + #ifdef __cplusplus extern "C" { #endif -- 1.7.3.4

2 4

[alsa-devel] Internal Mic Boost channel is unused
by Nathanael D. Noblet 10 Oct '13

10 Oct '13

Hello, I have an issue I'd like to help completely solve. Unfortunately I only have a tenuous grasp of what is actually going on so some of this could be wrong/include errors. Here's what I know. 1) I have the following hardware (alsa-info attached). 2) The internal microphone requires that the mic boost channel be something other than 0 to function properly. 3) Changing what pulseaudio expects things to be labelled causes the problem to go away. (via the attached patch to /usr/share/pulseaudio/paths/analog-input-mic.conf). From the discussion with David Henningsson (diwic on IRC). It seems that pulse doesn't expect a Mic Boost channel to be used with internal microphones. As such to fix this particular hardware, the driver would need to make the internal mic boost be labelled "Internal Mic Boost" as opposed to Mic Boost, which is then is used for both external and internal mics. I'd love to be able to help any way I can to get these fixed once and for all. Let me know if you need more information or what not. -- Nathanael d. Noblet t 403.875.4613

3 13

[alsa-devel] [PATCH 1/2] ALSA: hda - Add a headset mic model for ALC269 and friends
by David Henningsson 10 Oct '13

10 Oct '13

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index f911e36..ba143cd 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -28,6 +28,7 @@ ALC269/270/275/276/28x/29x alc269-dmic Enable ALC269(VA) digital mic workaround alc271-dmic Enable ALC271X digital mic workaround inv-dmic Inverted internal mic workaround + headset-mic Indicates a combined headset (mic+headphone jack) lenovo-dock Enables docking station I/O for some Lenovos dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ed9deb6..8cb3350 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2819,6 +2819,15 @@ static void alc269_fixup_hweq(struct hda_codec *codec, alc_write_coef_idx(codec, 0x1e, coef | 0x80); } +static void alc269_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; +} + static void alc271_fixup_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3516,6 +3525,7 @@ enum { ALC271_FIXUP_DMIC, ALC269_FIXUP_PCM_44K, ALC269_FIXUP_STEREO_DMIC, + ALC269_FIXUP_HEADSET_MIC, ALC269_FIXUP_QUANTA_MUTE, ALC269_FIXUP_LIFEBOOK, ALC269_FIXUP_AMIC, @@ -3614,6 +3624,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_stereo_dmic, }, + [ALC269_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + }, [ALC269_FIXUP_QUANTA_MUTE] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_quanta_mute, @@ -3979,6 +3993,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"}, {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC269_FIXUP_HEADSET_MIC, .name = "headset-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ALSA: hda - Distinguish between the headset mic pin and the TRS mic
by David Henningsson 10 Oct '13

10 Oct '13

On the Acer AO722, pin 0x1a represents the mic part of the TRRS headset jack, and pin 0x1b represents the TRS mic jack. Mark pin 0x1a as headset mic will help us know which one is which. Reported-by: Nathanael D. Noblet <nathanael(a)gnat.ca> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ec68eae..7f30c9f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3231,6 +3231,8 @@ enum { CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, + CXT_FIXUP_HEADSET_MIC_PIN, + CXT_FIXUP_HEADSET_MIC, CXT_FIXUP_GPIO1, }; @@ -3306,6 +3308,15 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec, } } +static void cxt_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; +} + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { @@ -3376,6 +3387,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_headphone_mic, }, + [CXT_FIXUP_HEADSET_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_HEADSET_MIC, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x03a1903c }, /* use as headset mic, with jack detect */ + { } + } + }, + [CXT_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headset_mic, + }, [CXT_FIXUP_GPIO1] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -3395,6 +3419,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1025, 0x0598, "Acer AO722", CXT_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), -- 1.7.9.5

1 0

[alsa-devel] [PATCHv2] pcm.h: include sys/types.h, necessary for u_int_XX types
by John Spencer 10 Oct '13

10 Oct '13

fixes compilation on musl libc. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/pcm.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..7ea9e84 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -29,6 +29,8 @@ #ifndef __ALSA_PCM_H #define __ALSA_PCM_H +#include <sys/types.h> + #ifdef __cplusplus extern "C" { #endif -- 1.7.3.4

1 0

[alsa-devel] [RFC] ASoC: pcm: Trigger all commands on error
by Markus Pargmann 10 Oct '13

10 Oct '13

If a start command is triggered and an error occures in the first called function, e.g. DMA, all other functions are not executed and the error value is returned immediately. In case of a start command, the calling function will call undo_start, calling this trigger function with the stop command. It will call stop for each function. But only the first function was started previously. The other functions may fail in the assumption that a stop command always comes after a start command. As the API does not specify the behaviour of trigger functionpointers, I think this should be fixed in the function calling the trigger functionpointers. This patch changes the behaviour. The trigger function calls all functionpointers independent of their returncodes. The first error-code is returned. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- sound/soc/soc-pcm.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6..23fc25b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -616,26 +616,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; + int ret = 0; + int err = 0; if (codec_dai->driver->ops->trigger) { ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + if (!err && ret) + err = ret; } if (platform->driver->ops && platform->driver->ops->trigger) { ret = platform->driver->ops->trigger(substream, cmd); - if (ret < 0) - return ret; + if (!err && ret) + err = ret; } if (cpu_dai->driver->ops->trigger) { ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; + if (!err && ret) + err = ret; } - return 0; + + return err; } static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, -- 1.8.4.rc3

2 13

[alsa-devel] [PATCH] ASoC: mc13783: add mixer controls
by Steffen Trumtrar 10 Oct '13

10 Oct '13

Add more kcontrols for the alsa mixer infrastructure. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> --- sound/soc/codecs/mc13783.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index eedbf05..fafe999 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -534,6 +534,30 @@ static struct snd_soc_dapm_route mc13783_routes[] = { static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", "Mono", "Mono Mix"}; +static const char * const mc13783_alsp[] = {"Off", "Codec", "Right"}; + +static const char * const mc13783_ahs[] = {"Codec", "Mixer"}; + +static const struct soc_enum mc13783_enum_asp = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 4, ARRAY_SIZE(mc13783_alsp), + mc13783_alsp); + +static const struct soc_enum mc13783_enum_alsp = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 7, ARRAY_SIZE(mc13783_alsp), + mc13783_alsp); + +static const struct soc_enum mc13783_enum_ahs = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 11, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_arxout = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 17, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_codec = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 4, ARRAY_SIZE(mc13783_alsp), + mc13783_alsp); + static const struct soc_enum mc13783_enum_3d_mixer = SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer), mc13783_3d_mixer); @@ -541,8 +565,32 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + SOC_ENUM("Earpiece Switch", mc13783_enum_asp), + SOC_ENUM("Loudspeaker Switch", mc13783_enum_alsp), + + SOC_ENUM("Headset Switch", mc13783_enum_ahs), + SOC_SINGLE("Headset Amp Right Switch", MC13783_AUDIO_RX0, 9, 1, 0), + SOC_SINGLE("Headset Amp Left Switch", MC13783_AUDIO_RX0, 10, 1, 0), + + SOC_ENUM("Line out Amp Switch", mc13783_enum_arxout), + SOC_SINGLE("Line out Amp Right Switch", MC13783_AUDIO_RX0, 15, 1, 0), + SOC_SINGLE("Line out Amp Left Switch", MC13783_AUDIO_RX0, 16, 1, 0), + + SOC_SINGLE("Codec Capture Mix Switch", MC13783_AUDIO_RX0, 21, 1, 0), + SOC_SINGLE("PCM Capture Mix Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mix Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("Codec Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("Codec Capture Switch", MC13783_AUDIO_RX1, 0, 1, 0), + + SOC_SINGLE("PGA Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("PGA Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) -- 1.8.4.rc3

3 5

[alsa-devel] [PATCH] ALSA: hda - Enable surround speakers (when line out is also present)
by David Henningsson 10 Oct '13

10 Oct '13

In the case where we have both line out and more than stereo speakers, the speaker DACs will end up in extra_out_nid. In fact, AFAIU, speakers are the only ones that can end up in extra_out_nid, and if we have several of those, they should be surround outputs rather than copy front. BugLink: https://bugs.launchpad.net/bugs/1236965 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) Notes: * Thanks to Raymond Yau for the pointer * Not tested (yet), except in hda-emu * Not sure if this should be sent to stable or not, given that it is a change of behaviour * Alsainfo available here: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1236965/+attachm… diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5b6c4e3..68801ba 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5395,11 +5395,6 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, mout->hp_out_nid[i], stream_tag, 0, format); - for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) - if (!mout->no_share_stream && mout->extra_out_nid[i]) - snd_hda_codec_setup_stream(codec, - mout->extra_out_nid[i], - stream_tag, 0, format); /* surrounds */ for (i = 1; i < mout->num_dacs; i++) { @@ -5410,6 +5405,20 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format); } + + /* extra surrounds */ + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) { + int ch = 0; + if (!mout->extra_out_nid[i]) + break; + if (chs >= (i + 1) * 2) + ch = i * 2; + else if (!mout->no_share_stream) + break; + snd_hda_codec_setup_stream(codec, mout->extra_out_nid[i], + stream_tag, ch, format); + } + return 0; } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare); -- 1.7.9.5

2 1

Re: [alsa-devel] Cannot load snd-usb-audio on 3.12-rc2
by Takashi Iwai 10 Oct '13

10 Oct '13

At Mon, 23 Sep 2013 16:41:07 -0700, Sarah Sharp wrote: > > The snd-usb-audio driver no longer loads properly on 3.12-rc2 when I > plug in my USB headset. It worked fine on 3.11-rc4 (not sure about > vanilla 3.11). > > Trying to manually load the driver fails: > > sarah@xanatos:~$ sudo modprobe snd-usb-audio > FATAL: Error inserting snd_usb_audio > (/lib/modules/3.12.0-rc2/kernel/sound/usb/snd-usb-audio.ko): Invalid argument > > dmesg shows: > > [ 44.231668] snd_usb_audio: `-2' invalid for parameter `index' There is no corresponding code change in the USB audio driver side, so I guess it must be in the module parameter parser that got broken. Does the patch below fix? thanks, Takashi --- diff --git a/kernel/params.c b/kernel/params.c index 81c4e78..c00d5b5 100644 --- a/kernel/params.c +++ b/kernel/params.c @@ -254,11 +254,11 @@ int parse_args(const char *doing, STANDARD_PARAM_DEF(byte, unsigned char, "%hhu", unsigned long, kstrtoul); -STANDARD_PARAM_DEF(short, short, "%hi", long, kstrtoul); +STANDARD_PARAM_DEF(short, short, "%hi", long, kstrtol); STANDARD_PARAM_DEF(ushort, unsigned short, "%hu", unsigned long, kstrtoul); -STANDARD_PARAM_DEF(int, int, "%i", long, kstrtoul); +STANDARD_PARAM_DEF(int, int, "%i", long, kstrtol); STANDARD_PARAM_DEF(uint, unsigned int, "%u", unsigned long, kstrtoul); -STANDARD_PARAM_DEF(long, long, "%li", long, kstrtoul); +STANDARD_PARAM_DEF(long, long, "%li", long, kstrtol); STANDARD_PARAM_DEF(ulong, unsigned long, "%lu", unsigned long, kstrtoul); int param_set_charp(const char *val, const struct kernel_param *kp)

2 2

[alsa-devel] firewire amdtp details
by Uli Franke 09 Oct '13

09 Oct '13

Hi Clemens One of the reasons why we are working on snd_dice is because we experienced (rare but unacceptable nevertheless) audio dropouts with the original driver. The DICE PLL lost lock in such cases. Investigations showed that this varies with the hardware/temperature/sun spots/snowfall. This led me to the conclusion that the device buffering/SYT-timing must be somehow close to its limits (late/early). In such a case just little node clock differences (induced by temperature and alike) can cause the system to drop out of the PLL's locking range. During capture mode SYT implementation/review I re-read the SYT code and found the following: from amdtp_stream_set_parameters: /* default buffering in the device */ s->transfer_delay = TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE; if (s->flags & CIP_BLOCKING) /* additional buffering needed to adjust for no-data packets */ s->transfer_delay += TICKS_PER_SECOND * amdtp_syt_intervals[sfc] / rate; which sets a global transfer delay. But later in calculate_syt I get the impression that (not exactly) the same operations are performed again accidentally: if (syt_offset < TICKS_PER_CYCLE) { syt_offset += TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE; if (s->flags & CIP_BLOCKING) syt_offset += s->transfer_delay; syt = (cycle + syt_offset / TICKS_PER_CYCLE) << 12; syt += syt_offset % TICKS_PER_CYCLE; return syt & 0xffff; } When I interpret the code from set_parameters as some sort of precomputation, the code in calculate_syt should be something like this: if (syt_offset < TICKS_PER_CYCLE) { syt_offset += s->transfer_delay; syt = (cycle + syt_offset / TICKS_PER_CYCLE) << 12; syt += syt_offset % TICKS_PER_CYCLE; return syt & 0xffff; } This could explain that the stream gets delayed too much - resulting in late timing (to be verified though). Therefore I'm a bit curious if you could enlighten me about this as s->transfer_delay isn't used in any other place. Additionally I appreciate any justifications for the subtraction of TICKS_PER_CYCLE from the global transfer delay. This would help me to get a better understanding of snd_dice's SYT mechanics. Regards Uli

2 1

[alsa-devel] [PATCH] ASoC: dapm: Use async I/O for DAPM sequences
by Mark Brown 09 Oct '13

09 Oct '13

From: Mark Brown <broonie(a)linaro.org> Within a DAPM sequence we normally don't care about when exactly a register write has completed so long as they happen in the order we requested. This means that we can issue most of the writes we do asynchronously which should maximise the ability of the underlying frameworks to keep the hardware busy, providing a small performance improvement on some systems. We currently ensure that all writes are completed both when changing to a different device and when calling into the regulator and clock frameworks. This should ensure that the previous ordering is maintained. We also ensure that writes are completed prior to calling into widget event functions since some event functions implement delays. This should be improved in future so that widgets can disable this sync in order to add extra writes. Signed-off-by: Mark Brown <broonie(a)linaro.org> xvdfvsd --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 25b9554..905e4c5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -409,6 +409,12 @@ static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) mutex_unlock(&w->platform->mutex); } +static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec && dapm->codec->using_regmap) + regmap_async_complete(dapm->codec->control_data); +} + static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { @@ -417,8 +423,9 @@ static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, int ret; if (w->codec && w->codec->using_regmap) { - ret = regmap_update_bits_check(w->codec->control_data, - reg, mask, value, &change); + ret = regmap_update_bits_check_async(w->codec->control_data, + reg, mask, value, + &change); if (ret != 0) return ret; } else { @@ -1201,6 +1208,8 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, { int ret; + soc_dapm_async_complete(w->dapm); + if (SND_SOC_DAPM_EVENT_ON(event)) { if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, false); @@ -1234,6 +1243,8 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, if (!w->clk) return -EIO; + soc_dapm_async_complete(w->dapm); + #ifdef CONFIG_HAVE_CLK if (SND_SOC_DAPM_EVENT_ON(event)) { return clk_prepare_enable(w->clk); @@ -1426,6 +1437,7 @@ static void dapm_seq_check_event(struct snd_soc_card *card, if (w->event && (w->event_flags & event)) { pop_dbg(w->dapm->dev, card->pop_time, "pop test : %s %s\n", w->name, ev_name); + soc_dapm_async_complete(w->dapm); trace_snd_soc_dapm_widget_event_start(w, event); ret = w->event(w, NULL, event); trace_snd_soc_dapm_widget_event_done(w, event); @@ -1498,6 +1510,7 @@ static void dapm_seq_run(struct snd_soc_card *card, struct list_head *list, int event, bool power_up) { struct snd_soc_dapm_widget *w, *n; + struct snd_soc_dapm_context *d; LIST_HEAD(pending); int cur_sort = -1; int cur_subseq = -1; @@ -1528,6 +1541,9 @@ static void dapm_seq_run(struct snd_soc_card *card, cur_subseq); } + if (cur_dapm && w->dapm != cur_dapm) + soc_dapm_async_complete(cur_dapm); + INIT_LIST_HEAD(&pending); cur_sort = -1; cur_subseq = INT_MIN; @@ -1586,6 +1602,10 @@ static void dapm_seq_run(struct snd_soc_card *card, cur_dapm->seq_notifier(cur_dapm, i, cur_subseq); } + + list_for_each_entry(d, &card->dapm_list, list) { + soc_dapm_async_complete(d); + } } static void dapm_widget_update(struct snd_soc_card *card) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH] ASoC: dapm: Run clock and regulator events separately to other supplies
by Mark Brown 09 Oct '13

09 Oct '13

From: Mark Brown <broonie(a)linaro.org> In order to avoid trying to use an external clock or supply for an on-chip supply prior to it being enabled move the clock and regulator supply events to a separate step in DAPM sequencing from normal supply events. This should have minimal practical impact since these widgets are sorted using SND_SOC_NOPM which is a negative value and hence sorted separately to any real register writes, though it may be relevant if supplies have event callbacks only. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-dapm.c | 50 +++++++++++++++++++++++++------------------------- 1 file changed, 25 insertions(+), 25 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 905e4c5..eb26d5a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -59,31 +59,31 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, - [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, [snd_soc_dapm_clock_supply] = 1, - [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_supply] = 2, + [snd_soc_dapm_micbias] = 3, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai_in] = 3, - [snd_soc_dapm_dai_out] = 3, - [snd_soc_dapm_aif_in] = 3, - [snd_soc_dapm_aif_out] = 3, - [snd_soc_dapm_mic] = 4, - [snd_soc_dapm_mux] = 5, - [snd_soc_dapm_virt_mux] = 5, - [snd_soc_dapm_value_mux] = 5, - [snd_soc_dapm_dac] = 6, - [snd_soc_dapm_switch] = 7, - [snd_soc_dapm_mixer] = 7, - [snd_soc_dapm_mixer_named_ctl] = 7, - [snd_soc_dapm_pga] = 8, - [snd_soc_dapm_adc] = 9, - [snd_soc_dapm_out_drv] = 10, - [snd_soc_dapm_hp] = 10, - [snd_soc_dapm_spk] = 10, - [snd_soc_dapm_line] = 10, - [snd_soc_dapm_kcontrol] = 11, - [snd_soc_dapm_post] = 12, + [snd_soc_dapm_dai_in] = 4, + [snd_soc_dapm_dai_out] = 4, + [snd_soc_dapm_aif_in] = 4, + [snd_soc_dapm_aif_out] = 4, + [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_mux] = 6, + [snd_soc_dapm_virt_mux] = 6, + [snd_soc_dapm_value_mux] = 6, + [snd_soc_dapm_dac] = 7, + [snd_soc_dapm_switch] = 8, + [snd_soc_dapm_mixer] = 8, + [snd_soc_dapm_mixer_named_ctl] = 8, + [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_adc] = 10, + [snd_soc_dapm_out_drv] = 11, + [snd_soc_dapm_hp] = 11, + [snd_soc_dapm_spk] = 11, + [snd_soc_dapm_line] = 11, + [snd_soc_dapm_kcontrol] = 12, + [snd_soc_dapm_post] = 13, }; static int dapm_down_seq[] = { @@ -109,10 +109,10 @@ static int dapm_down_seq[] = { [snd_soc_dapm_dai_in] = 10, [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, - [snd_soc_dapm_clock_supply] = 12, - [snd_soc_dapm_regulator_supply] = 12, [snd_soc_dapm_supply] = 12, - [snd_soc_dapm_post] = 13, + [snd_soc_dapm_clock_supply] = 13, + [snd_soc_dapm_regulator_supply] = 13, + [snd_soc_dapm_post] = 14, }; static void pop_wait(u32 pop_time) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Use module_usb_driver
by Sachin Kamat 09 Oct '13

09 Oct '13

module_usb_driver makes code simpler by removing the boilerplate. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/usb/card.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9d9de8d..d979050 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -751,15 +751,4 @@ static struct usb_driver usb_audio_driver = { .supports_autosuspend = 1, }; -static int __init snd_usb_audio_init(void) -{ - return usb_register(&usb_audio_driver); -} - -static void __exit snd_usb_audio_cleanup(void) -{ - usb_deregister(&usb_audio_driver); -} - -module_init(snd_usb_audio_init); -module_exit(snd_usb_audio_cleanup); +module_usb_driver(usb_audio_driver); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add fixup for ASUS N56VZ
by Takashi Iwai 08 Oct '13

08 Oct '13

ASUS N56VZ needs a fixup for the bass speaker pin, which was already provided via model=asus-mode4. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=841645 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52c26d3..ed9deb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4596,6 +4596,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- 1.8.4

1 0

Re: [alsa-devel] [Alsa-user] hdmi sound rerouted?
by Clemens Ladisch 08 Oct '13

08 Oct '13

Dagg Stompler wrote: >> daggs(a)gmx.com wrote: >>> card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] >>> card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] >>> card 2: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] >> >> The second one is called "Generic" because your kernel does not (yet) >> know the hardware. >> >>> 01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Cedar HDMI Audio [Radeon HD 5400/6300 Series] >>> 06:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] RV635 HDMI Audio [Radeon HD 3600 Series] >> >> These are the HDMI outputs of *two* graphics cards. The first one's >> model is so new that the driver doesn't know it. > > the 3600 doesn't have hdmi socket so I think it is irrelevant for my issue. The RV635 chip does support HDMI outputs (even if none are actually connected), so the corresponding PCI device shows up here. >>> If you update the kernel, you will get two devices named "HDA ATI HDMI". > > upgrade to where? Oops. Even the latest driver lacks the PCI IDs of many HDMI HDA devices (aa50/aa58/aa60/aa68/aa80/aa88/aa90/aa98: Cypress/Juniper/Redwood/Cedar/ Cayman+Antilles/Barts/Turks/Caicos). Alex, are these devices that require disabling snooping? Regards, Clemens

4 5

[alsa-devel] ALSA: snd-usb-usx2y: remove bogus frame checks
by Guido Aulisi 08 Oct '13

08 Oct '13

Hi, I did some more testing on my US122: I used my patch (see below) and it worked (with default nrpack=4), but I got "should not be here with counts=42" errors, then after some time the kernel crashed, but I couldn't copy the trace. Then I set nrpack=1 and I got no "should not be here..." and it was more stable, no crash. I only checked playback, not recording, nor MIDI, with aplay I played some songs for about one hour... I hope this can be useful. But I think there are other bugs in this driver. Ciao Guido Aulisi The patch is: diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..cd16fcc 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -328,7 +328,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) + if (likely((urb->start_frame & 0x03FF) == (usX2Y->wait_iso_frame & 0x03FF))) subs->completed_urb = urb; else { usX2Y_error_sequence(usX2Y, subs, urb);

2 1

[alsa-devel] Can't get capturing from RME Hammerfall MADI card to work: sound loops & silcence
by Niels Laukens 08 Oct '13

08 Oct '13

Hi, I'm trying to capture audio from an RME Hammerfall MADI card (PCI ID 10ee:3fc6). The setup previously worked fine on Windows, so I'm currently assuming that the card and the input signal is OK. I wrote a very simple ALSA-application that simply opens the device, sets the access format to NONINTERLEAVED (which seems to be the only option the card likes), selects 32bit sampling (although the audio is 24bit, ALSA rejects this option), 48kHz sampling and 64 channels (also, the only option the card accepts). Next, it loops with snd_pcm_readn() to get 100ms worth of audio, and dumps channel 10 to file. (Full source in attachement) The result is strange for various reasons: * The second half of every buffer returned by snd_pcm_readn() is always zero's. * The call seems to return way to fast: I would expect to get 48000 frames/second, but judging from the timing of the snd_pcm_readn() call, it's more like 1536000 frames/second (i.e. 32 fold faster). * When playing back the resulting file, I do recognize some music, but it sounds like it's the same part over and over again. Also, restarting the capture program minutes later, returns the same (at least as far as my ears & brain can recall) fragment of audio. What am I doing wrong here? (Samples of the PCM-file are available upon request) I've also tried a related test with `arecord`, with similar but different results: * The audio was also interspersed with silence * The capture duration did not match the requested '-d' option: the recording was a factor of 2 shorter than requested. * There was more audio in the recording: if I manually cut out the silent parts, the audio sounded reasonable. What am I missing here? Is this the right place to ask? Or should this go to the RME forum? Thanks in advance, Niels

6 5

[alsa-devel] Audio Miniconference 2013 schedule - Edinburgh 21st October
by Mark Brown 07 Oct '13

07 Oct '13

The audio mini-summit in Edinburgh is now less than a month away so it seems like a good time to start pulling together a schedule. Looking at the topics people mentioned when filling out the registration form there were three major topics that people wanted to cover: - Compressed audio review: tinycompress, integration into userspace, driver upstreaming. - UCM and other use case management status & review. - Effects offload to DSPs with dynamically instantiatable algorithms, and general dynamic DSP firmware handling. Plus a few things that only one or two people brought up: - Process stuff (the list, bug tracking and so on). - Some relatively specific hardware support topics: - HDMI hotplug handling throughout the stack - Mic status LED handling throughout the stack. - Code cleanups: - Moving away from the custom ALSA device registration infrastructure. - General code modernisation. - General routing control in userspace - UIs, automation. The first question is if there is anything important that is missing from the above list - please speak up soon if there is! It'd also be good to have volunteers to lead the discussions in each area. In terms of format I'd propose that we go through the list above in roughly that order, having someone lead discussion on the topic. Does this sound good to people? As previously mentioned it'd be good if anyone planning to attend could fill in the form at: https://docs.google.com/forms/d/105uav9tkwwkDU4P1T3IOWjfsa3kgtDbFAJOlHgBQR5… to help with planning & coordination.

8 13

[alsa-devel] [PATCH v2] ALSA: snd-usb-usx2y: remove bogus frame checks
by Daniel Mack 07 Oct '13

07 Oct '13

The frame check in i_usX2Y_urb_complete() and i_usX2Y_usbpcm_urb_complete() is bogus and produces false positives as described in this LAU thread: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 This patch removes the check code entirely. Cc: fzu(a)wemgehoertderstaat.de Reported-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 22 +++------------------- sound/usb/usx2y/usx2yhwdeppcm.c | 7 +------ 2 files changed, 4 insertions(+), 25 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..6234a51 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index f2a1acd..814d0e8 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -244,13 +244,8 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + subs->completed_urb = urb; capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE]; capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; -- 1.8.3.1

4 10

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Connect 'Left Line1R Mux' and 'Right Line1L Mux'
by Peter Ujfalusi 07 Oct '13

07 Oct '13

The two paths were not connected in the DAPM route causing the associated routes to be non working and the following warnings printed in the logs: tlv320aic3x-codec 0-001b: ASoC: mux Right Line1L Mux has no paths tlv320aic3x-codec 0-001b: ASoC: mux Left Line1R Mux has no paths Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 892c108..a160b7d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -678,6 +678,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Left Input */ {"Left Line1L Mux", "single-ended", "LINE1L"}, {"Left Line1L Mux", "differential", "LINE1L"}, + {"Left Line1R Mux", "single-ended", "LINE1R"}, + {"Left Line1R Mux", "differential", "LINE1R"}, {"Left Line2L Mux", "single-ended", "LINE2L"}, {"Left Line2L Mux", "differential", "LINE2L"}, @@ -694,6 +696,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, {"Right Line1R Mux", "differential", "LINE1R"}, + {"Right Line1L Mux", "single-ended", "LINE1L"}, + {"Right Line1L Mux", "differential", "LINE1L"}, {"Right Line2R Mux", "single-ended", "LINE2R"}, {"Right Line2R Mux", "differential", "LINE2R"}, -- 1.8.3.2

2 1

Re: [alsa-devel] [PATCH] sound: usb: support wireless devices in snd_usb_parse_datainterval
by Takashi Iwai 07 Oct '13

07 Oct '13

At Tue, 1 Oct 2013 14:32:57 -0500, Thomas Pugliese wrote: > > Hi, > This patch adds support for dev speed USB_SPEED_WIRELESS in > snd_usb_parse_datainterval which allows the usb sound core to create > ISO urbs with the correct number and size of buffers. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Thanks, applied this one, too. Takashi > --- > sound/usb/helper.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/helper.c b/sound/usb/helper.c > index 6209024..51ed1ac 100644 > --- a/sound/usb/helper.c > +++ b/sound/usb/helper.c > @@ -118,6 +118,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, > { > switch (snd_usb_get_speed(chip->dev)) { > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > if (get_endpoint(alts, 0)->bInterval >= 1 && > get_endpoint(alts, 0)->bInterval <= 4) > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-usb" 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] sound: usb: add support for wireless USB devices
by Takashi Iwai 07 Oct '13

07 Oct '13

At Tue, 1 Oct 2013 14:32:15 -0500, Thomas Pugliese wrote: > > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Thanks, applied now. (BTW, please add the maintainers explicitly to Cc at the next time.) Takashi > --- > sound/usb/card.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/card.c b/sound/usb/card.c > index 64952e2..ef76aa0 100644 > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -349,6 +349,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, > case USB_SPEED_LOW: > case USB_SPEED_FULL: > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > break; > default: > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-usb" 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

[alsa-devel] [PATCH 1/3] ASoC: dapm: Add support for virtual mixer controls
by Lars-Peter Clausen 07 Oct '13

07 Oct '13

This patch adds support for virtual DAPM mixer controls. They are similar to virtual DAPM enums. There is no hardware register backing the control, so changing the control's value wont have any direct effect on the hardware. But it still influences the DAPM graph by causing the path it sits on to be connected or disconnected. This in turn can cause power changes for some of the widgets on the DAPM graph, which will then modify the hardware state. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 4 ++++ include/sound/soc.h | 3 ++- sound/soc/soc-dapm.c | 45 +++++++++++++++++++++++++++------------------ 3 files changed, 33 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 27a72d5..2037c45 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -286,6 +286,8 @@ struct device; .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } +#define SOC_DAPM_SINGLE_VIRT(xname, max) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0) #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -300,6 +302,8 @@ struct device; .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array) #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ diff --git a/include/sound/soc.h b/include/sound/soc.h index 6dd329a..af1678c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1049,7 +1049,8 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift; + int reg, rreg; + unsigned int shift, rshift; unsigned int invert:1; unsigned int autodisable:1; }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c17c14c..25b9554 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -499,18 +499,22 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrol_news[i].private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = soc_widget_read(w, reg); - val = (val >> shift) & mask; - if (invert) - val = max - val; + if (reg != SND_SOC_NOPM) { + val = soc_widget_read(w, reg); + val = (val >> shift) & mask; + if (invert) + val = max - val; + p->connect = !!val; + } else { + p->connect = 0; + } - p->connect = !!val; } break; case snd_soc_dapm_mux: { @@ -2791,7 +2795,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2804,7 +2808,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, kcontrol->id.name); mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (dapm_kcontrol_is_powered(kcontrol)) + if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) val = (snd_soc_read(codec, reg) >> shift) & mask; else val = dapm_kcontrol_get_value(kcontrol); @@ -2835,7 +2839,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2857,19 +2861,24 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm_kcontrol_set_value(kcontrol, val); + change = dapm_kcontrol_set_value(kcontrol, val); - mask = mask << shift; - val = val << shift; + if (reg != SND_SOC_NOPM) { + mask = mask << shift; + val = val << shift; + + change = snd_soc_test_bits(codec, reg, mask, val); + } - change = snd_soc_test_bits(codec, reg, mask, val); if (change) { - update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + if (reg != SND_SOC_NOPM) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - card->update = &update; + card->update = &update; + } soc_dapm_mixer_update_power(card, kcontrol, connect); -- 1.8.0

3 10

[alsa-devel] [PATCH 0/7] ALSA: hda - hdmi: Various channel mapping fixes
by Anssi Hannula 07 Oct '13

07 Oct '13

Hi! Apparently there were several more bugs in the HDMI channel mapping code, so here you go. I haven't marked any for stable since these don't really affect the regular layouts that much and the patches aren't that trivial. If you want to pick some for stable, my suggestions are 1, 5, 6, maybe 3 (those I initially marked for stable until I changed my mind). Anssi Hannula (7): ALSA: hda - hdmi: Fix reported channel map on common default layouts ALSA: hda - hdmi: Fix incorrect default channel mapping for unusual CAs ALSA: hda - hdmi: Fix programmed active channel count ALSA: hda - hdmi: Fix unused slots being enabled in manual and non-PCM mappings ALSA: hda - hdmi: Fix channel maps with less common speakers ALSA: hda - hdmi: Fix available channel maps missing from TLV ALSA: hda - hdmi: Tweak debug messages to be more useful sound/pci/hda/patch_hdmi.c | 146 +++++++++++++++++++++-------------- 1 file changed, 90 insertions(+), 56 deletions(-) -- Anssi Hannula

2 8

[alsa-devel] [PATCH] ASoC: dapm: Ignore VMID widgets for target bias
by Lars-Peter Clausen 07 Oct '13

07 Oct '13

VMID widgets behave very similar to signal generator widgets. Both are always considered to be powered up. This means that we need to ignore the VMID widgets in the same way as signal generator widgets when calculating the DAPM context's target bias level. Otherwise the presence of a VMID widget, regardless whether it is on an active path or not, will cause the DAPM context to be powered up. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c17c14c..177f8a1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1840,6 +1840,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) */ switch (w->id) { case snd_soc_dapm_siggen: + case snd_soc_dapm_vmid: break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: -- 1.8.0

2 1

[alsa-devel] [PATCH 0/2] ASoC: fix sound on mx31moboard
by Philippe Rétornaz 07 Oct '13

07 Oct '13

Hello Theses two patches are necessary to get back the sound working on mx31moboards. The first one fix imx-ssi to work even if only the dma is available. The second one fix the audmux settings on mx31moboard boards. Thanks, Philippe Philippe Rétornaz (2): ASoC: fsl: imx-ssi: fix probe on imx31 ASoC: fsl: Fix sound on mx31moboard sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-ssi.c | 23 ++++++++++++----------- sound/soc/fsl/imx-ssi.h | 2 ++ 3 files changed, 15 insertions(+), 12 deletions(-) -- 1.7.9.5

3 9

[alsa-devel] [PATCH 3/4] ESI W192M: Add sampling rate control.
by klem.dev＠gmail.com 07 Oct '13

07 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Add sampling rate control for adc/dac in Ego Sys Waveterminal 192M. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 83 insertions(+) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index d559916..cf2b6a5 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <linux/slab.h> #include <sound/tlv.h> #include "ice1712.h" @@ -69,6 +70,37 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned char *change) +{ + unsigned char new, old; + int idx; + + /*stac9460 1*/ + for (idx = 0; idx < 7; idx++) { + if (change[idx]) { + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx] = (new != old); + if (change[idx]) + stac9460_put(ice, idx, new); + } + } + + /*stac9460 2*/ + for (idx = 0; idx < 3; idx++) { + if (change[idx + 7]) { + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx + 7] = (new != old); + if (change[idx + 7]) + stac9460_2_put(ice, idx, new); + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, @@ -97,10 +129,13 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct mutex *mute_mutex = ice->spec; unsigned char new, old; int id, idx; int change; + mutex_lock(mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; old = stac9460_get(ice, idx); @@ -128,6 +163,9 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, stac9460_2_put(ice, idx - 6, new); } } + + mutex_unlock(mute_mutex); + return change; } @@ -381,6 +419,41 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned char changed[10]; + struct mutex *mute_mutex = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(mute_mutex); + /* we have to remember current mute status for each DAC */ + stac9460_dac_mute_all(ice, 0, changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, changed); + mutex_unlock(mute_mutex); +} + + static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -479,9 +552,11 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct mutex *mute_mutex; /*WTM 192M*/ ice->num_total_dacs = 8; @@ -489,12 +564,20 @@ static int wtm_init(struct snd_ice1712 *ice) ice->force_rdma1 = 1; ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ + mute_mutex = kzalloc(sizeof(*mute_mutex), GFP_KERNEL); + if (!mute_mutex) + return -ENOMEM; + ice->spec = mute_mutex; + mutex_init(mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH 1/4] ESI W192M: Update ESI W192M driver.
by klem.dev＠gmail.com 07 Oct '13

07 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Update Waveterminal 192M driver to support alsa changes that happened since last driver commit. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 37 ++++++++++++++++++++++--------------- 1 file changed, 22 insertions(+), 15 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..b6b650c 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -463,7 +463,7 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, (unsigned short)-1 }; @@ -473,9 +473,10 @@ static int wtm_init(struct snd_ice1712 *ice) ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); @@ -485,19 +486,25 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x67, /* 49MHz crystal, mpu401, + * spdif-in+ 1 stereo ADC, + * 4 stereo DACs + */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /* out-en, out-int, spdif-in */ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, /* GPIO20: 0 = CD drive dig. input + * passthrough, + * 1 = SPDIF-OUT from ice1724 + */ }; -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix mono speakers and headset mic on Dell Vostro 5470
by David Henningsson 07 Oct '13

07 Oct '13

On this machine, DAC on node 0x03 seems to give mono output. Also, it needs additional patches for headset mic support. It supports CTIA style headsets only. Alsa-info available at the bug link below. Cc: stable(a)kernel.org (v3.10+) BugLink: https://bugs.launchpad.net/bugs/1236228 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0e303b9..52c26d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3496,6 +3496,15 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, } } +static void alc290_fixup_mono_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + /* Remove DAC node 0x03, as it seems to be + giving mono output */ + snd_hda_override_wcaps(codec, 0x03, 0); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3522,6 +3531,7 @@ enum { ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASUS_X101_FUNC, @@ -3535,6 +3545,7 @@ enum { ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, ALC283_FIXUP_INT_MIC, + ALC290_FIXUP_MONO_SPEAKERS, }; static const struct hda_fixup alc269_fixups[] = { @@ -3712,6 +3723,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC269_FIXUP_DELL3_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, [ALC269_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -3804,6 +3824,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC290_FIXUP_MONO_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc290_fixup_mono_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3845,6 +3871,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.7.9.5

2 1

[alsa-devel] [PATCH]: remove unnecessary/obsolete compat/hsearch_r.c
by Waldemar Brodkorb 07 Oct '13

07 Oct '13

The usage of hsearch functions where removed long time ago in commit 273d115de05574251bdd661747ecb68449a5cf1d. This patch highly increases portability for non-glibc systems. Remove the complete compat directory as requested by Takashi Iwai. Signed-off-by: Waldemar Brodkorb <wbx(a)openadk.org> --- configure.in | 4 +- src/Makefile.am | 7 +- src/compat/Makefile.am | 8 -- src/compat/hsearch_r.c | 236 ------------------------------------------------ 4 files changed, 3 insertions(+), 252 deletions(-) delete mode 100644 src/compat/Makefile.am delete mode 100644 src/compat/empty.c delete mode 100644 src/compat/hsearch_r.c diff --git a/configure.in b/configure.in index 00fff2b..35fd89f 100644 --- a/configure.in +++ b/configure.in @@ -64,8 +64,6 @@ AC_HEADER_TIME dnl Checks for library functions. AC_PROG_GCC_TRADITIONAL -AC_CHECK_FUNC([hsearch_r], [HAVE_HSEARCH_R=yes]) -AM_CONDITIONAL([ALSA_HSEARCH_R], [test "x$HAVE_HSEARCH_R" != xyes]) AC_CHECK_FUNCS([uselocale]) SAVE_LIBRARY_VERSION @@ -659,7 +657,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/pcm/Makefile src/pcm/scopes/Makefile \ src/rawmidi/Makefile src/timer/Makefile \ src/hwdep/Makefile src/seq/Makefile src/ucm/Makefile \ - src/compat/Makefile src/alisp/Makefile \ + src/alisp/Makefile \ src/conf/Makefile src/conf/alsa.conf.d/Makefile \ src/conf/cards/Makefile \ src/conf/pcm/Makefile \ diff --git a/src/Makefile.am b/src/Makefile.am index 08f482a..8f789fe 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -46,8 +46,8 @@ if BUILD_ALISP SUBDIRS += alisp libasound_la_LIBADD += alisp/libalisp.la endif -SUBDIRS += compat conf -libasound_la_LIBADD += compat/libcompat.la @ALSA_DEPLIBS@ +SUBDIRS += conf +libasound_la_LIBADD += @ALSA_DEPLIBS@ libasound_la_LDFLAGS = -version-info $(COMPATNUM) $(VSYMS) $(SYMFUNCS) $(LDFLAGS_NOUNDEFINED) @@ -84,9 +84,6 @@ ucm/libucm.la: instr/libinstr.la: $(MAKE) -C instr libinstr.la -compat/libcompat.la: - $(MAKE) -C compat libcompat.la - alisp/libalisp.la: $(MAKE) -C alisp libalisp.la diff --git a/src/compat/Makefile.am b/src/compat/Makefile.am deleted file mode 100644 index 01f54fc..0000000 --- a/src/compat/Makefile.am +++ /dev/null @@ -1,8 +0,0 @@ -noinst_LTLIBRARIES = libcompat.la -EXTRA_libcompat_la_SOURCES = hsearch_r.c - -if ALSA_HSEARCH_R -libcompat_la_SOURCES = empty.c hsearch_r.c -else -libcompat_la_SOURCES = empty.c -endif diff --git a/src/compat/empty.c b/src/compat/empty.c deleted file mode 100644 index e69de29..0000000 diff --git a/src/compat/hsearch_r.c b/src/compat/hsearch_r.c deleted file mode 100644 index 96ceac1..0000000 --- a/src/compat/hsearch_r.c +++ /dev/null @@ -1,236 +0,0 @@ -/* Copyright (C) 1993, 1995, 1996, 1997 Free Software Foundation, Inc. - This file is part of the GNU C Library. - Contributed by Ulrich Drepper <drepper(a)gnu.ai.mit.edu>, 1993. - - The GNU C Library is free software; you can redistribute it and/or - modify it under the terms of the GNU Lesser General Public License as - published by the Free Software Foundation; either version 2.1 of the - License, or (at your option) any later version. - - The GNU C Library 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 - Lesser General Public License for more details. - - You should have received a copy of the GNU Lesser General Public - License along with the GNU C Library; see the file COPYING.LIB. If not, - write to the Free Software Foundation, Inc., 59 Temple Place - Suite 330, - Boston, MA 02111-1307, USA. */ - -#include <errno.h> -#include <malloc.h> -#include <string.h> - -#define __USE_GNU -#ifndef __set_errno -#define __set_errno(e) errno = (e) -#endif -#include <search.h> - -/* [Aho,Sethi,Ullman] Compilers: Principles, Techniques and Tools, 1986 - [Knuth] The Art of Computer Programming, part 3 (6.4) */ - - -/* The reentrant version has no static variables to maintain the state. - Instead the interface of all functions is extended to take an argument - which describes the current status. */ -typedef struct _ENTRY -{ - unsigned int used; - ENTRY entry; -} -_ENTRY; - - -/* For the used double hash method the table size has to be a prime. To - correct the user given table size we need a prime test. This trivial - algorithm is adequate because - a) the code is (most probably) called a few times per program run and - b) the number is small because the table must fit in the core */ -static int -isprime (unsigned int number) -{ - /* no even number will be passed */ - unsigned int div = 3; - - while (div * div < number && number % div != 0) - div += 2; - - return number % div != 0; -} - - -/* Before using the hash table we must allocate memory for it. - Test for an existing table are done. We allocate one element - more as the found prime number says. This is done for more effective - indexing as explained in the comment for the hsearch function. - The contents of the table is zeroed, especially the field used - becomes zero. */ -int -hcreate_r (nel, htab) - size_t nel; - struct hsearch_data *htab; -{ - /* Test for correct arguments. */ - if (htab == NULL) - { - __set_errno (EINVAL); - return 0; - } - - /* There is still another table active. Return with error. */ - if (htab->table != NULL) - return 0; - - /* Change nel to the first prime number not smaller as nel. */ - nel |= 1; /* make odd */ - while (!isprime (nel)) - nel += 2; - - htab->size = nel; - htab->filled = 0; - - /* allocate memory and zero out */ - htab->table = (_ENTRY *) calloc (htab->size + 1, sizeof (_ENTRY)); - if (htab->table == NULL) - return 0; - - /* everything went alright */ - return 1; -} - - -/* After using the hash table it has to be destroyed. The used memory can - be freed and the local static variable can be marked as not used. */ -void -hdestroy_r (htab) - struct hsearch_data *htab; -{ - /* Test for correct arguments. */ - if (htab == NULL) - { - __set_errno (EINVAL); - return; - } - - if (htab->table != NULL) - /* free used memory */ - free (htab->table); - - /* the sign for an existing table is an value != NULL in htable */ - htab->table = NULL; -} - - -/* This is the search function. It uses double hashing with open addressing. - The argument item.key has to be a pointer to an zero terminated, most - probably strings of chars. The function for generating a number of the - strings is simple but fast. It can be replaced by a more complex function - like ajw (see [Aho,Sethi,Ullman]) if the needs are shown. - - We use an trick to speed up the lookup. The table is created by hcreate - with one more element available. This enables us to use the index zero - special. This index will never be used because we store the first hash - index in the field used where zero means not used. Every other value - means used. The used field can be used as a first fast comparison for - equality of the stored and the parameter value. This helps to prevent - unnecessary expensive calls of strcmp. */ -int -hsearch_r (item, action, retval, htab) - ENTRY item; - ACTION action; - ENTRY **retval; - struct hsearch_data *htab; -{ - unsigned int hval; - unsigned int count; - unsigned int len = strlen (item.key); - unsigned int idx; - - /* Compute an value for the given string. Perhaps use a better method. */ - hval = len; - count = len; - while (count-- > 0) - { - hval <<= 4; - hval += item.key[count]; - } - - /* First hash function: simply take the modulo but prevent zero. */ - hval %= htab->size; - if (hval == 0) - ++hval; - - /* The first index tried. */ - idx = hval; - - if (htab->table[idx].used) - { - /* Further action might be required according to the action value. */ - unsigned hval2; - - if (htab->table[idx].used == hval - && strcmp (item.key, htab->table[idx].entry.key) == 0) - { - if (action == ENTER) - htab->table[idx].entry.data = item.data; - - *retval = &htab->table[idx].entry; - return 1; - } - - /* Second hash function, as suggested in [Knuth] */ - hval2 = 1 + hval % (htab->size - 2); - - do - { - /* Because SIZE is prime this guarantees to step through all - available indexes. */ - if (idx <= hval2) - idx = htab->size + idx - hval2; - else - idx -= hval2; - - /* If we visited all entries leave the loop unsuccessfully. */ - if (idx == hval) - break; - - /* If entry is found use it. */ - if (htab->table[idx].used == hval - && strcmp (item.key, htab->table[idx].entry.key) == 0) - { - if (action == ENTER) - htab->table[idx].entry.data = item.data; - - *retval = &htab->table[idx].entry; - return 1; - } - } - while (htab->table[idx].used); - } - - /* An empty bucket has been found. */ - if (action == ENTER) - { - /* If table is full and another entry should be entered return - with error. */ - if (action == ENTER && htab->filled == htab->size) - { - __set_errno (ENOMEM); - *retval = NULL; - return 0; - } - - htab->table[idx].used = hval; - htab->table[idx].entry = item; - - ++htab->filled; - - *retval = &htab->table[idx].entry; - return 1; - } - - __set_errno (ESRCH); - *retval = NULL; - return 0; -} -- 1.7.9.5

2 1

[alsa-devel] ALSA lib pcm_hw.c:1466:(_snd_pcm_hw_open) Unknown field chmap
by Alexander Stein 07 Oct '13

07 Oct '13

Hi, I get this error message from several games I got recently. There is either not sound at all or it is choppy. This was with alsa-lib-1.0.27.1. So I downgraded to 1.0.26 and everything was fine. I compiled alsa-lib-1.0.27.1 myself and started them using LD_LIBRARY_PATH: everything worked. So I assumed there is something in the config files which has changed. Comparing them I noticed some changes in /usr/share/alsa/cards/Audigy2.conf. Those lines were added: chmap [ "UNKNOWN" "FL,FR" ] chmap [ "UNKNOWN" "RL,RR" ] chmap [ "UNKNOWN" "FC,LFE" ] chmap [ "UNKNOWN" "SL,SR" ] See alsa-lib commits 5ba11d69dafbd4502469fd957b4580169779664c and ec7acfc408856cfe71a122e3b5387463d7b3d919. Updating to 1.0.27.1 again and commenting those lines out everything is ok now. So I'm wondering what is the reason for this error message? How to fix that? Thanks, Alexander PS: my sound card: 03:07.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value 1102:0008

2 1

[alsa-devel] [PATCH - Ego Sys Waveterminal 192M 1/1] Update driver to support alsa changes happened since last driver commit. Add TLV support for dB scales. Add sampling rate control.
by klem.dev＠gmail.com 05 Oct '13

05 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..286be47 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -4,8 +4,8 @@ * Lowlevel functions for Ego Sys Waveterminal 192M * * Copyright (c) 2006 Guedez Clement <klem.dev(a)gmail.com> - * Some functions are taken from the Prodigy192 driver - * source + * Thanks to the developers of the Prodigy192 driver on which + * I based this driver * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -25,15 +25,25 @@ +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/slab.h> #include <sound/core.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +#include <sound/tlv.h> + +struct wtm_spec { + struct ak4114 *ak4114; + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* @@ -68,6 +78,35 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned char *change) +{ + unsigned char new, old; + int idx; + + /*stac9460 1*/ + for (idx = 0; idx < 7; idx++) { + if (change[idx]) { + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx] = (new != old); + if (change[idx]) + stac9460_put(ice, idx, new); + } + } + + /*stac9460 2*/ + for (idx = 0; idx < 3; idx++) { + if (change[idx + 7]) { + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx + 7] = (new != old); + if (change[idx + 7]) + stac9460_2_put(ice, idx, new); + } + } +} + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, @@ -96,10 +135,14 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char new, old; int id, idx; int change; + mutex_lock(&spec->mute_mutex); + + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; old = stac9460_get(ice, idx); @@ -127,6 +170,7 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, stac9460_2_put(ice, idx - 6, new); } } + mutex_unlock(&spec->mute_mutex); return change; } @@ -175,7 +219,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; - nvol = ucontrol->value.integer.value[0] & 0x7f; + nvol = ucontrol->value.integer.value[0]; tmp = stac9460_get(ice, idx); ovol = 0x7f - (tmp & 0x7f); change = (ovol != nvol); @@ -186,7 +230,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, } else { id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); idx = id + STAC946X_LF_VOLUME; - nvol = ucontrol->value.integer.value[0] & 0x7f; + nvol = ucontrol->value.integer.value[0]; if (id < 6) tmp = stac9460_get(ice, idx); else @@ -339,7 +383,24 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, * MIC / LINE switch fonction */ -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Line In", "Mic" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items + - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -353,7 +414,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -369,7 +430,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) @@ -380,6 +441,46 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned char changed[10]; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + stac9460_dac_mute_all(ice, 0, changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, changed); + mutex_unlock(&spec->mute_mutex); +} + + + +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ @@ -390,15 +491,19 @@ static struct snd_kcontrol_new stac9640_controls[] = { .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +524,14 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,18 +543,40 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; /*INIT*/ +static void stac9460_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = entry->private_data; + int reg, val; + /* registers 0x0 - 0x14 */ + for (reg = 0; reg <= 0x15; reg++) { + val = stac9460_get(ice, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); + } +} + +static void stac9460_proc_init(struct snd_ice1712 *ice) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ice->card, "stac9460_codec", &entry)) + snd_info_set_text_ops(entry, ice, stac9460_proc_regs_read); +} + + static int wtm_add_controls(struct snd_ice1712 *ice) { unsigned int i; @@ -458,46 +588,64 @@ static int wtm_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; } + stac9460_proc_init(ice); return 0; } static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; + /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x67, /* 49MHz crystal, mpu401, + * spdif-in+ 1 stereo ADC, + * 4 stereo DACs + */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /* out-en, out-int, spdif-in */ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, /* GPIO20: 0 = CD drive dig. input + * passthrough, + * 1 = SPDIF-OUT from ice1724 + */ }; -- 1.7.10.4

3 2

[alsa-devel] [PATCH 4/4] ESI W192M: Add text "Line in/Mic".
by klem.dev＠gmail.com 05 Oct '13

05 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Add text "Line in/Mic" for selecting input gain state for Ego Sys Waveterminal 192M. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index cf2b6a5..464cca5 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -377,8 +377,23 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Line In", "Mic" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items + - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -392,7 +407,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -408,7 +423,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH 2/4] ESI W192M: Add TLV Support.
by klem.dev＠gmail.com 05 Oct '13

05 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Add TLV support for control value in dB scale for Ego Sys Waveterminal 192M. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index b6b650c..d559916 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <sound/tlv.h> #include "ice1712.h" #include "envy24ht.h" @@ -380,6 +381,10 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ @@ -394,11 +399,14 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +427,14 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +446,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; -- 1.8.4.rc3

1 0

[alsa-devel] [PING!] Re: HDMI audio is not working on my new desktop
by Alexander E. Patrakov 05 Oct '13

05 Oct '13

2013/8/14 Alexander E. Patrakov <patrakov(a)gmail.com>: > Hello. > > I built a new desktop PC based on the Gigabyte H87N-WIFI motherboard. > It has one DVI and two HDMI outputs. However, I cannot get any sound > over HDMI. The same OS image works perfectly on my laptop that also > has an HDMI output. > > This is not a pulseaudio problem, as I don't get any sound even if I > test with this command: > > pasuspender -- speaker-test -c2 -D hdmi:0,2 # or, for that matter, 0,0 or 0,1 > > Alsa-info.sh has uploaded its output to > http://www.alsa-project.org/db/?f=1ad7192a15ff4a70b5ddb2579476dbb98adbd929 > > Please note this error in dmesg, it may be relevant: > > [ 1.803595] hda_codec: invalid CONNECT_LIST verb 5[1]:0 > [ 1.803719] hda_codec: invalid CONNECT_LIST verb 7[1]:0 As nobody replied, I have converted this into a formal bug report: https://bugzilla.kernel.org/show_bug.cgi?id=60769 Haswell people, _please_ comment on the bug. -- Alexander E. Patrakov

3 9

[alsa-devel] Microphone does not longer work on Lenovo N581
by Manuel Reimer 05 Oct '13

05 Oct '13

Hello, I'm using Archlinux and so usually run always on the latest kernel. But unfortunately somewhere with Kernel 3.9 my Intel sound got broken. Playback works, but no microphone is found. So I keep downgrading to 3.8.8 for some months, now. Can someone please help me with finding a solution for this? Is it really possible that the alsa driver "snd_had_intel" was broken between 3.8 and 3.9 and noone found this bug, so far? Thank you very much in advance Greetings, Manuel

3 4

[alsa-devel] [PATCH 1/6] ASoC: cs4271: Include linux/of.h header
by Sachin Kamat 04 Oct '13

04 Oct '13

'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/cs4271.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index a20f1bb..f6e9534 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -25,6 +25,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> #include <sound/pcm.h> -- 1.7.9.5

2 9

[alsa-devel] us122l
by James Miller 04 Oct '13

04 Oct '13

please ignore my previous email -- James Stewart Miller Bsc(hons) Psych.

1 0

[alsa-devel] tascam us122l
by James Miller 04 Oct '13

04 Oct '13

Hi, I've just checked the kernel source for the tascam us122l (us122l.c us122l.h) and it has changed completely since the 3.4 kernel. I'm having a stack trace problem when trying to use jack with us122l. The card is recognised and I am using the .asoundrc from pub.briata.org as I always have done. When the us122l.c got broken (around about kernel version 3.6 to 3.9) I was able to get it working by recompiling the kernel after amending the us122l.c source code by changing a flag that was pushed in to a vm struct. In the new code I can't find any reference to the vm operation and it seems that the code has changed completely, I am going to try and revert to the 3.4 us122l.c to see if that will work, but I'd be grateful if someone could look into this. Thanks James -- James Stewart Miller Bsc(hons) Psych.

1 0

Re: [alsa-devel] [PATCH] sound: usb: support wireless devices in snd_usb_parse_datainterval
by Greg KH 04 Oct '13

04 Oct '13

On Tue, Oct 01, 2013 at 02:32:57PM -0500, Thomas Pugliese wrote: > Hi, > This patch adds support for dev speed USB_SPEED_WIRELESS in > snd_usb_parse_datainterval which allows the usb sound core to create > ISO urbs with the correct number and size of buffers. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Acked-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org>

1 0

Re: [alsa-devel] [PATCH] sound: usb: add support for wireless USB devices
by Greg KH 04 Oct '13

04 Oct '13

On Tue, Oct 01, 2013 at 02:32:15PM -0500, Thomas Pugliese wrote: > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Acked-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org>

1 0

[alsa-devel] snd_device_name_hint segfaults
by Forest Bond 03 Oct '13

03 Oct '13

Hi, I'm seeing segfaults with VLC similar to those reported here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/059168.ht… This is on Ubuntu 12.04 with the latest alsa-lib package (1.0.25-1ubuntu10.2) and the following patches also applied: * 2cfc8b9b44a8e493c41b3d63d5a00b306a18a5ed snd_pcm_direct_parse_open_conf(): use thread-safe getgrnam_r() * 44c1a623dd1fc9e831616b663bebc54ca98df994 Add snd_lib_error_set_local() to install a thread-local error handler. * 25dbb102810b31c02358904d70d53c960fb0a10e snd_device_name_hint(): do not change the global error handler. * f49b2dc522a2564315c76d075203b15a39941e8a snd_device_name_hint(): do not use global snd_config. * 7f2b2c8c1650a1883b48abfcdb455138943854f9 conf: Fix a memory access violation resulting from improper error propogation So I think there are still issues when calling snd_device_name_hint(-1, ...). Here is the backtrace: #0 0x003f0e78 in __strcpy_chk () from /lib/i386-linux-gnu/libc.so.6 #1 0x0625084b in strcpy (__src=0x0, __dest=0xb6025e80 "pcm.(null)") at /usr/include/i386-li nux-gnu/bits/string3.h:105 #2 try_config (config=0xaf1c4df0, list=0xb52fe804, base=0x323e962 "pcm", name=0x0) at nameh int.c:243 #3 0x06251bc5 in add_software_devices (list=0xb52fe804, config=0xaf1c4df0) at namehint.c:51 2 #4 snd_device_name_hint (card=-1, iface=<optimized out>, hints=0xb52fe90c) at namehint.c:59 1 #5 0x0323c9bb in GetDevices (obj=0xb6271700, item=0x0, prefs_dev=<optimized out>) at alsa.c :721 #6 0x0323d4c0 in Probe (dev=0xb6011df8 "default", obj=0xb6271700) at alsa.c:161 #7 Open (obj=0xb6271700) at alsa.c:539 #8 0x0a6c73d0 in generic_start (func=0x323cd10, ap=0xb52fed98 "\273") at modules/modules.c:422 #9 0x0a6c7a3a in vlc_module_load (p_this=0xb6271700, psz_capability=0xa70b959 "audio output", psz_name=<optimized out>, b_strict=false, probe=0xa6c73c0 <generic_start>) at modules/modules.c:347 #10 0x0a6c8002 in module_need (obj=0xb6271700, cap=0xa70b959 "audio output", name=0xa70d2bd "$aout", strict=false) at modules/modules.c:437 #11 0x0a6b1242 in aout_OutputNew (p_aout=0xb6271700, p_format=0xb52feeac) at audio_output/output.c:59 #12 0x0a6ad77f in aout_DecNew (p_aout=0xb6271700, p_format=0xb52feeac, p_replay_gain=0xaef02a14, p_request_vout=0xb52feed0) at audio_output/dec.c:112 #13 0x0a67066e in aout_new_buffer (p_dec=0xaef02870, i_samples=1024) at input/decoder.c:2286 #14 0x0a672690 in decoder_NewAudioBuffer (p_decoder=0xaef02870, i_size=1024) at input/decoder.c:213 #15 0x04032f8c in DecodeBlock (p_dec=0xaef02870, pp_block=0xb52ff05c) at faad.c:464 #16 0x0a670891 in DecoderDecodeAudio (p_dec=0xaef02870, p_block=0xaef03000) at input/decoder.c:1291 #17 0x0a6716f5 in DecoderProcessAudio (b_flush=false, p_block=0xaef03000, p_dec=0xaef02870) at input/decoder.c:1936 #18 DecoderProcess (p_dec=0xaef02870, p_block=<optimized out>) at input/decoder.c:2059 #19 0x0a671959 in DecoderThread (p_data=0xaef02870) at input/decoder.c:938 #20 0x001fbd4c in start_thread () from /lib/i386-linux-gnu/libpthread.so.0 #21 0x003ddace in clone () from /lib/i386-linux-gnu/libc.so.6 As a work around I will probably patch VLC to not call snd_device_name_hint this way. I see this has been done for other applications: http://code.google.com/p/chromium/issues/detail?id=95797 http://code.mythtv.org/trac/ticket/10809 But I wanted to try diagnosing the issue first. Thoughts? Thanks, Forest -- Forest Bond http://www.alittletooquiet.net http://www.rapidrollout.com

4 6

[alsa-devel] [PATCH] ASoC: omap: Fix incorrect ARM dependency
by Jarkko Nikula 03 Oct '13

03 Oct '13

Commit b0e0a4d ("ASoC: omap: Enable COMPILE_TEST build for DT platforms") added two incorrect CONFIG_ARCH_ARM dependencies making impossible to select audio support for Nokia RX-51. Fix this by using correct CONFIG_ARM. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- This is a fix for 3.12-rc. I.e. not for next merge window. --- sound/soc/omap/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index daa78a0..4a07f71 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST) + depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST) select SND_DMAENGINE_PCM config SND_OMAP_SOC_DMIC @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) + depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ALSA: snd-usb-usx2y: fix start frame mask
by Daniel Mack 03 Oct '13

03 Oct '13

This patch is based on a change proposed by Guido Aulisi in a discussion on LAU which can be followed here: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 Cc: fzu(a)wemgehoertderstaat.de Original-code-by: Guido Aulisi <guido.aulisi(a)gmail.com> Tested-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..cd16fcc 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -328,7 +328,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) + if (likely((urb->start_frame & 0x03FF) == (usX2Y->wait_iso_frame & 0x03FF))) subs->completed_urb = urb; else { usX2Y_error_sequence(usX2Y, subs, urb); -- 1.8.3.1

3 4

[alsa-devel] [PATCH] ASoC: davinci-mcasp: add support for suspend and resume
by Daniel Mack 03 Oct '13

03 Oct '13

When the system returns from suspend, it looses its configuration. Most of it is restored by running a normal audio stream startup, but the DAI format is left unset as that's configured on the audio device creation. Hence, it suffices here to care for the registers which are touched by davinci_mcasp_set_dai_fmt() and restore them when the system is resumed. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/davinci/davinci-mcasp.c | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 12 ++++++++++++ 2 files changed, 51 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 591f853..5023da7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1272,12 +1272,51 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int davinci_mcasp_suspend(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + a->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); + a->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); + a->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); + a->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); + a->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); + a->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); + a->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); + + return 0; +} + +static int davinci_mcasp_resume(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, a->context.txfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, a->context.rxfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, a->context.txfmt); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, a->context.rxfmt); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, a->context.aclkxctl); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, a->context.aclkrctl); + mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, a->context.pdir); + + return 0; +} +#endif + +SIMPLE_DEV_PM_OPS(davinci_mcasp_pm_ops, + davinci_mcasp_suspend, + davinci_mcasp_resume); + static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, + .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a9ac0c1..a2e27e1 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -43,6 +43,18 @@ struct davinci_audio_dev { /* McASP FIFO related */ u8 txnumevt; u8 rxnumevt; + +#ifdef CONFIG_PM_SLEEP + struct { + u32 txfmtctl; + u32 rxfmtctl; + u32 txfmt; + u32 rxfmt; + u32 aclkxctl; + u32 aclkrctl; + u32 pdir; + } context; +#endif }; #endif /* DAVINCI_MCASP_H */ -- 1.8.3.1

3 2

[alsa-devel] [PATCH 1/3] ASoC: tas5086: move two variables into private struct
by Daniel Mack 03 Oct '13

03 Oct '13

We need to access the charge_period and start_mid_z values from other places later, so move them to the private struct. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/tas5086.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 6d31d88..31b5868 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -244,6 +244,8 @@ struct tas5086_private { unsigned int mclk, sclk; unsigned int format; bool deemph; + unsigned int charge_period; + unsigned int pwm_start_mid_z; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -720,13 +722,15 @@ static const int tas5086_charge_period[] = { static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - int charge_period = 1300000; /* hardware default is 1300 ms */ - u8 pwm_start_mid_z = 0; int i, ret; + priv->pwm_start_mid_z = 0; + priv->charge_period = 1300000; /* hardware default is 1300 ms */ + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; - of_property_read_u32(of_node, "ti,charge-period", &charge_period); + of_property_read_u32(of_node, "ti,charge-period", + &priv->charge_period); for (i = 0; i < 6; i++) { char name[25]; @@ -735,7 +739,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) "ti,mid-z-channel-%d", i + 1); if (of_get_property(of_node, name, NULL) != NULL) - pwm_start_mid_z |= 1 << i; + priv->pwm_start_mid_z |= 1 << i; } } @@ -744,25 +748,25 @@ static int tas5086_probe(struct snd_soc_codec *codec) * configure 'part 1' of the PWM starts to use Mid-Z, and tell * all configured mid-z channels to start start under 'part 1'. */ - if (pwm_start_mid_z) + if (priv->pwm_start_mid_z) regmap_write(priv->regmap, TAS5086_PWM_START, TAS5086_PWM_START_MIDZ_FOR_START_1 | - pwm_start_mid_z); + priv->pwm_start_mid_z); /* lookup and set split-capacitor charge period */ - if (charge_period == 0) { + if (priv->charge_period == 0) { regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); } else { i = index_in_array(tas5086_charge_period, ARRAY_SIZE(tas5086_charge_period), - charge_period); + priv->charge_period); if (i >= 0) regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, i + 0x08); else dev_warn(codec->dev, "Invalid split-cap charge period of %d ns.\n", - charge_period); + priv->charge_period); } /* enable factory trim */ -- 1.8.3.1

2 3

[alsa-devel] [PATCH] ARM: omap: edma: add suspend suspend/resume hooks
by Daniel Mack 03 Oct '13

03 Oct '13

This patch adds support for edma driver. I know the intermediate code in arch/arm/common is going to be purged eventually when the real dma-engine driver lands, so I don't know whether it's reasonable to include my patch after all. It might just server as template for the suspend/resume functions in the final driver, or maybe anyone wants to include that intermediate hack for the time being. Successfully tested on an AM335x board with davinci-mcasp enabled audio. A patch to make the latter work after suspend has been posted on alsa-devel. Thanks, Dainel Daniel Mack (1): ARM: omap: edma: add suspend suspend/resume hooks arch/arm/common/edma.c | 133 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 133 insertions(+) -- 1.8.3.1

2 2

[alsa-devel] [PATCH] ALSA: snd-usb-usx2y: remove bogus frame check
by Daniel Mack 02 Oct '13

02 Oct '13

The frame check in i_usX2Y_urb_complete() is bogus and produces false positives as described in this LAU thread: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 This code removes the check code entirely. Cc: fzu(a)wemgehoertderstaat.de Reported-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 22 +++------------------- 1 file changed, 3 insertions(+), 19 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..6234a51 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; -- 1.8.3.1

2 2

[alsa-devel] [PATCH] amixer: fix indentation when printing container TLV contents
by Anssi Hannula 02 Oct '13

02 Oct '13

decode_tlv() adds indentation in the beginning, with the expectation that the TLV will be printed on the line afterwards in a switch-case. However, in the case of a container TLV the switch-case simply adds another level of indentation and calls decode_tlv() for the inner TLVs. This causes the first inner TLV to be printed with too much indentation and double '|'. Fix that by printing "container" and a newline for container TLVs, so that the result is as follows: : values=0,0,0,0,0,0,0,0 | container | chmap-variable=FL,FR | chmap-variable=FL,FR,LFE | chmap-variable=FL,FR,FC Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- amixer/amixer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/amixer/amixer.c b/amixer/amixer.c index b83b0c3..cf82892 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -460,6 +460,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ } switch (type) { case SND_CTL_TLVT_CONTAINER: + printf("container\n"); size += sizeof(unsigned int) -1; size /= sizeof(unsigned int); while (idx < size) { -- 1.8.1.5

2 1

[alsa-devel] alsa-lib 1.0.27.2 and musl
by Waldemar Brodkorb 02 Oct '13

02 Oct '13

Hi Developers, I am trying to cross-compile alsa-lib with a musl¹ libc based toolchain. Unfortunately src/compat/hsearch_r.c is failing to compile, because of some glibc specific macros (like _P) normally defined in sys/cdefs.h. After grepping through the code I can not find any consumer for any functions implemented in hsearch_r.c. When I just comment out the code, everything compiles fine. Seems that that usage of hsearch functions was removed in the code long time ago. commit 273d115de05574251bdd661747ecb68449a5cf1d Author: Jaroslav Kysela <perex(a)perex.cz> Date: Sat Jul 15 10:20:32 2000 +0000 Removed old mixer API functions. Added simple control layer (experimental). Can it just be removed? thanks for any advice or comment. best regards Waldemar ¹ http://www.musl-libc.org/

3 2

Re: [alsa-devel] [PATCH] sound: usb: add support for wireless USB devices
by Daniel Mack 02 Oct '13

02 Oct '13

On 01.10.2013 21:32, Thomas Pugliese wrote: > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. Interesting. What device are you testing this with? Daniel > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> > --- > sound/usb/card.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/card.c b/sound/usb/card.c > index 64952e2..ef76aa0 100644 > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -349,6 +349,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, > case USB_SPEED_LOW: > case USB_SPEED_FULL: > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > break; > default: >

2 1

[alsa-devel] snd-usb-usx2y (Tascam US-122) sequence errors
by Dr Nicholas J Bailey 02 Oct '13

02 Oct '13

I'm on debian jessie, Linux arial 3.10-3-686-pae #1 SMP Debian 3.10.11-1 (2013-09-10) i686 GNU/Linux with a Tascam us-122 that has recently stopped working. When I try to capture with anything, there is a few hundred ms of audio then the capture process stalls and the following dmesg appears: [ 462.596296] Sequence Error!(hcd_frame=0 ep=8in;wait=1024,frame=0). [ 462.596296] Most probably some urb of usb-frame 1024 is still missing. [ 462.596296] Cause could be too long delays in usb-hcd interrupt handling. [ 477.959230] Sequence Error!(hcd_frame=0 ep=8in;wait=1024,frame=0). [ 477.959230] Most probably some urb of usb-frame 1024 is still missing. [ 477.959230] Cause could be too long delays in usb-hcd interrupt handling. Before this kernel, the box worked OK. The only think that leads me to believe it might be my fault is that for a very long time, I've had to plug the box in: [ 445.529271] usb 2-1.6: new full-speed USB device number 5 using ehci-pci [ 445.622076] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8006 [ 445.622086] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 then unplug it and plug it in again: [ 445.972743] usb 2-1.6: USB disconnect, device number 5 [ 447.621519] usb 2-1.6: new full-speed USB device number 6 using ehci-pci [ 447.714462] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8007 [ 447.714471] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 [ 447.834852] usbcore: registered new interface driver snd-usb-usx2y [ 455.194314] usb 2-1.6: USB disconnect, device number 6 [ 457.487433] usb 2-1.6: new full-speed USB device number 7 using ehci-pci [ 457.579960] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8006 [ 457.579969] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 to get the firmware to load [ 457.888884] usb 2-1.6: USB disconnect, device number 7 [ 459.379782] usb 2-1.6: new full-speed USB device number 8 using ehci-pci [ 459.472516] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8007 [ 459.472526] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 (green USB light comes on here). I'm not sure about how to start debugging this; there are many things that could be wrong. If it will help, I'll try and file a bug report; if not please tell me to shut up (prefereably after pointing me at the solution :) ) Nick/.

2 3

Re: [alsa-devel] Realtek ALC286 SONY Vaio Pro 13 Ubuntu 13.10
by David Henningsson 02 Oct '13

02 Oct '13

On 10/02/2013 04:13 AM, Yuval Levy wrote: > On 10/01/2013 02:28 AM, David Henningsson wrote: >> Ok. If things work without extra drivers from Sony or Realtek, then it >> should be possible to get things working in Linux too without extra >> information from these parties... > > I have added this information to the bug report > <https://bugs.launchpad.net/bugs/1228689> > where I also post my comments so that they are more persistent and > available to other users with similar issues. > > >> Ok - given that 0x21 works as expected you should only need to do this >> for pin 0x18. > > All my testing was with doing this for both pins. I only noticed now > that I do not need to do this for pin 0x18. I will try to o this only > for pin 0x21 and reboot tomorrow. Ok, cool. Notice that I wrote "do this for pin 0x18 only, not for pin 0x21", where as you replied "I'll do this for pin 0x21 only, not for pin 0x18". 0x18 is for input switching, 0x21 is for output switching. If you remove jack detection, like you did, you will need to do this switching manually, e g in pavucontrol or gnome sound settings. That is, when you plug your headphone in, you need to manually select "Headphone" too, if you don't, the result will probably be that you hear sound in both headphones and speakers. Anyway, let's guess that the high pitched sound, as well as the lack of headset microphone input, is because your headset is OMTP and that the Vaio only supports CTIA. That's my *guess*, but it could be wrong, and it would be good to have this confirmed with a CTIA headset. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

2 2

[alsa-devel] Problems with i.MX233 driver in 3.12rc2
by Trevor Page 01 Oct '13

01 Oct '13

Hello, First post to this list - I hope this is the appropriate place to ask for this assistance. I'm using the Olinuxino Maxi board with the i.MX233 processor. I'm using the 3.12rc2 kernel at the moment because this has recently introduced the drivers for this processor's audio. I understand that the drivers for this chip's audio are: sound/soc/codecs/mxs-builtin-codec.c sound/soc/mxs/mxs-builtin-audio.c sound/soc/mxs/mxs-builtin-dai.c sound/soc/mxs/mxs-builtin-pcm.c My application for the board is a simple C program that uses the Linphone library to make VOIP calls. The problem I am having is that audio usually works (playback and capture) perfectly fine from boot, but at some point it may suddenly go 'dead'. At this point, there is no longer any audio from the headphone socket. I have not yet pinpointed exactly what it takes to trigger the fault, but I have seen it happen once when the call was put on hold while a Linphone call was in progress. Once the audio output is dead, these are some further observations: - When a Linphone call is received, I still hear a click from the headphone output when the call ends. - I have confirmed that the DAC / HP channels are turned up and unmuted. - If I run speaker-test when the audio is working, speaker-test periodically prints time duration information (presumably on each pass of its internal loop), like "0 - Front Left", "Time per period ...". When the audio is dead, speaker-test just freezes on "0 - Front Left" forever, as if some call is either blocking or returning to say it is busy. This might hopefully therefore be a useful point to start debugging. - I have only ever managed to restore audio using a reboot. I would like to have a go at debugging this. My problem though is that although I'm experienced with C and microprocessor programming, compiling and debugging embedded Linux is new ground for me. I'd be grateful for any pointers on things I could try in order to gain an insight into what's going wrong, and tips on how I could debug the drivers. So far my plan for today is to build the kernel again with the i.MX233 audio drivers as external modules so that I can hopefully play with them a bit and compile them separately. (I assume the development and debug process for audio drivers like this is to compile them as .ko modules and modprobe them in). However without having any prior knowledge of Linux drivers or the ALSA / ASoC framework I'm not sure if I'm going to get very far. Regards, Trevor.

3 2

[alsa-devel] [PATCH] ASoC: kirkwood: fix compilation warning in kirkwood_dma_open
by Vladimir Murzin 01 Oct '13

01 Oct '13

writel() supposes the first argument of type unsigned int. This fix the warning: sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_open': sound/soc/kirkwood/kirkwood-dma.c:164:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] Signed-off-by: Vladimir Murzin <murzin.v(a)gmail.com> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index b238434..1135c30 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -161,7 +161,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) * Enable Error interrupts. We're only ack'ing them but * it's useful for diagnostics */ - writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); + writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); } dram = mv_mbus_dram_info(); -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoC: fsl: imx-sgtl5000: Add .remove back
by Fabio Estevam 01 Oct '13

01 Oct '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Commit e8f00c1b01 (Merge remote-tracking branch 'asoc/fix/fsl' into asoc-devm) fixed a conflict, but missed to add the .remove function back,which causes the following build warning: sound/soc/fsl/imx-sgtl5000.c:185:12: warning: 'imx_sgtl5000_remove' defined but not used [-Wunused-function] Fix the warning by adding the .remove function back. Reported-by: Olof Johansson <olof(a)lixom.net> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 6f4bdc8..ed6ba1e 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -204,6 +204,7 @@ static struct platform_driver imx_sgtl5000_driver = { .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, + .remove = imx_sgtl5000_remove, }; module_platform_driver(imx_sgtl5000_driver); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] snd_tlv_convert_from_dB: fix decreasing gain across entries
by Stephen Warren 01 Oct '13

01 Oct '13

From: Stephen Warren <swarren(a)nvidia.com> Currently, for a TLV consisting of TLV_DB_SCALE_ITEMs, if e.g. alsamixer calls snd_mixer_selem_set_playback_dB() with a value that is in-between two TLV_DB_SCALE_ITEMs, and xdir is negative, the selected raw hardware value is the minimum in the first range above that value, rather than the maximum in the last range below that value. The user-visible symptom is that in alsamixer, pressing the down key to reduce the value sticks at certain points, and cannot be incrementally reduced any further, although directly selecting a much lower value (e.g. by pressing 0..9) works as expected. This is triggered e.g. by sound/soc/codec/max98090.c's max98090_hp_tlv[]. Fix this by checking whether xdir is positive or not, rather than checking whether it has a non-zero value. The code to select the previous range's max value is already present. This matches how xdir is used in other parts of the code. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- src/control/tlv.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/tlv.c b/src/control/tlv.c index 6b0b9f4..b08d887 100644 --- a/src/control/tlv.c +++ b/src/control/tlv.c @@ -312,7 +312,7 @@ int snd_tlv_convert_from_dB(unsigned int *tlv, long rangemin, long rangemax, submin, submax, db_gain, value, xdir); else if (db_gain < dbmin) { - *value = xdir || pos == 2 ? submin : prev_submax; + *value = xdir > 0 || pos == 2 ? submin : prev_submax; return 0; } prev_submax = submax; -- 1.8.1.5

2 1

[alsa-devel] [PATCH] amixer: actually print all TLVs in a container TLV
by Anssi Hannula 01 Oct '13

01 Oct '13

Due to a wrong indexing of tlv[] decode_tlv() always skips to the end of the container TLV since the used tlv[1] contains the container TLV size instead of the intended element size. This causes, for example, only the first HDMI channel map TLV to be shown. Fix the indexing to actually use the element size. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- amixer/amixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index 7727943..b83b0c3 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -468,7 +468,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ return; } decode_tlv(spaces + 2, tlv + idx, tlv[idx+1] + 8); - idx += 2 + (tlv[1] + sizeof(unsigned int) - 1) / sizeof(unsigned int); + idx += 2 + (tlv[idx+1] + sizeof(unsigned int) - 1) / sizeof(unsigned int); } break; case SND_CTL_TLVT_DB_SCALE: -- 1.8.1.5

2 1

Re: [alsa-devel] Bug: Conexant CX20588, Kernel 3.11, Win8 changes GPIO settings
by Takashi Iwai 01 Oct '13

01 Oct '13

At Mon, 30 Sep 2013 21:00:38 +0200, Christopher K. wrote: > > Works fine. Thanks for your support. I will now dive into the code to > be able to send you a patch the next time something is wrong :) OK, applied it now. thanks, Takashi > > > Chris > > On Mon, Sep 30, 2013 at 12:19 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Fri, 27 Sep 2013 22:38:22 +0200, > > Christopher wrote: > >> > >> I attached the alsa-info. This issue seems to be specific to the driver > >> provided by Acer (with thx, equalizer etc.) > >> Thanks for your help. > > > > Thanks. Could you check whether the patch below works? > > > > > > Takashi > > > > --- > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Fix GPIO for Acer Aspire 3830TG > > > > Acer Aspire 3830TG seems requiring GPIO bit 0 as the primary mute > > control. When a machine is booted after Windows 8, the GPIO pin is > > turned off and it results in the silent output. > > > > This patch adds the manual fixup of GPIO bit 0 for this model. > > > > Reported-by: Christopher <DIDI2002(a)web.de> > > Cc: <stable(a)vger.kernel.org> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/patch_conexant.c | 11 +++++++++++ > > 1 file changed, 11 insertions(+) > > > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > > index 4edd2d0..ec68eae 100644 > > --- a/sound/pci/hda/patch_conexant.c > > +++ b/sound/pci/hda/patch_conexant.c > > @@ -3231,6 +3231,7 @@ enum { > > CXT_FIXUP_INC_MIC_BOOST, > > CXT_FIXUP_HEADPHONE_MIC_PIN, > > CXT_FIXUP_HEADPHONE_MIC, > > + CXT_FIXUP_GPIO1, > > }; > > > > static void cxt_fixup_stereo_dmic(struct hda_codec *codec, > > @@ -3375,6 +3376,15 @@ static const struct hda_fixup cxt_fixups[] = { > > .type = HDA_FIXUP_FUNC, > > .v.func = cxt_fixup_headphone_mic, > > }, > > + [CXT_FIXUP_GPIO1] = { > > + .type = HDA_FIXUP_VERBS, > > + .v.verbs = (const struct hda_verb[]) { > > + { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, > > + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, > > + { 0x01, AC_VERB_SET_GPIO_DATA, 0x01 }, > > + { } > > + }, > > + }, > > }; > > > > static const struct snd_pci_quirk cxt5051_fixups[] = { > > @@ -3384,6 +3394,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { > > > > static const struct snd_pci_quirk cxt5066_fixups[] = { > > SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), > > + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), > > SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), > > SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), > > SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), > > -- > > 1.8.4 > > >

1 0

Re: [alsa-devel] Realtek ALC286 SONY Vaio Pro 13 Ubuntu 13.10
by David Henningsson 01 Oct '13

01 Oct '13

On 10/01/2013 08:13 AM, Yuval Levy wrote: > Thank you for the quick reply. > > On 09/30/2013 03:50 AM, David Henningsson wrote: >> An important distinction is here: to they work in Windows without >> specific Realtek drivers installed, or do you require Realtek's Windows >> drivers for this support? > > I am 99% certain that yes. I am no Windows expert. After taking > delivery of the laptop and ascertaining that the functionality that I > need works, I have backed up SONY's original stuff and stored it away. > Far away. I have reformatted the SDD, leaving 30 GB for a Windows > install (because unfortunately, the law school's exam system requires > either Windows or Mac). I reinstalled Windows from Microsoft's media. > I have not installed any other driver myself, but I am not sure if > Windows downloads drivers from the internet. It definitely did download > drivers for my Brother printer, automatically and without asking for > permission, after it discovered it on the network. I booted into > Windows before writing this email and looked in the Control Panel. It > says the driver for the two microsoft and for the loudspeakers is Microsoft. Ok. If things work without extra drivers from Sony or Realtek, then it should be possible to get things working in Linux too without extra information from these parties... >> Second, to do some actual triaging. I see you have been running >> hda-jack-sense-test. Could you run it all three scenarios: >> 1) nothing connected >> 2) headphone (without mic) connected >> 3) headset (heaphone + mic combo) connected > > Done and added to the bug report. 0x21 works as you expected. 0x18 not. > > As you suggested I installed alsa-tools-gui and did the advance override > with hdajackretask: > * selected codec Realtek ALC286 > * clicked Advance override in the Options on the right hand side > * clicked Override under pins 0x18 and 0x21 > * changed Jack detection from Present to Not present for both pins Ok - given that 0x21 works as expected you should only need to do this for pin 0x18. > * clicked Apply now > > An error message appeared: Failed to create file > '/home/yuv/.pulse/client.conf.3K083W': No such file or directory > > maybe it should save to /home/yuv/.config/pulse/client.conf.3K083W? > ^^^^^^^ > I am sorry I do not know how to fix this, can you help? Oh. That's fixed upstream [1], but apparently not in Ubuntu yet. Any of these workarounds should do it: 1) use the "boot override" instead (and reboot your computer to test) 2) create a symlink from .pulse to .config/pulse 3) or copy everything from .config/pulse to .pulse (but then you have to restart your computer afterwards, so PulseAudio will look in .pulse instead). -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic [1] http://git.alsa-project.org/?p=alsa-tools.git;a=commitdiff;h=772fbde91c00c8…

1 0

[alsa-devel] [PATCH v2] ASoC: imx-sgtl5000: do not use devres on a foreign device
by Philipp Zabel 01 Oct '13

01 Oct '13

Calling devm_clk_get with any device pointer other than our own confuses devres. Use clk_get instead. This avoids hitting the following warning in the imx-sgtl5000 error path: imx-sgtl5000 sound.12: snd_soc_register_card failed (-517) platform sound.12: Driver imx-sgtl5000 requests probe deferral ------------[ cut here ]------------ WARNING: CPU: 0 PID: 75 at drivers/base/dd.c:272 driver_probe_device+0x194/0x218() Modules linked in: snd_soc_sgtl5000(+) snd_soc_imx_sgtl5000 coda snd_soc_imx_audmux imx_sdma snd_soc_fsl_spdif snd_soc_fsl_ssi CPU: 0 PID: 75 Comm: udevd Not tainted 3.11.0-rc6+ #4682 Backtrace: [<80010bc4>] (dump_backtrace+0x0/0x10c) from [<80010d60>] (show_stack+0x18/0x1c) r6:00000110 r5:00000009 r4:00000000 r3:00000000 [<80010d48>] (show_stack+0x0/0x1c) from [<804f0764>] (dump_stack+0x20/0x28) [<804f0744>] (dump_stack+0x0/0x28) from [<8001a4a4>] (warn_slowpath_common+0x6c/0x8c) [<8001a438>] (warn_slowpath_common+0x0/0x8c) from [<8001a4e8>] (warn_slowpath_null+0x24/0x2c) r8:7f032000 r7:7f02f93c r6:cf8eaa54 r5:cf8eaa20 r4:80728a0c [<8001a4c4>] (warn_slowpath_null+0x0/0x2c) from [<80286bdc>] (driver_probe_device+0x194/0x218) [<80286a48>] (driver_probe_device+0x0/0x218) from [<80286cf4>] (__driver_attach+0x94/0x98) r7:00000000 r6:cf8eaa54 r5:7f02f93c r4:cf8eaa20 [<80286c60>] (__driver_attach+0x0/0x98) from [<802851c8>] (bus_for_each_dev+0x5c/0x90) r6:80286c60 r5:7f02f93c r4:00000000 r3:cf8ef03c [<8028516c>] (bus_for_each_dev+0x0/0x90) from [<80286654>] (driver_attach+0x24/0x28) r6:806d0424 r5:cf16a580 r4:7f02f93c [<80286630>] (driver_attach+0x0/0x28) from [<802861e4>] (bus_add_driver+0xdc/0x234) [<80286108>] (bus_add_driver+0x0/0x234) from [<802871d4>] (driver_register+0x80/0x154) r8:7f032000 r7:00000001 r6:7f02fa68 r5:7f02fa74 r4:7f02f93c [<80287154>] (driver_register+0x0/0x154) from [<8033c278>] (i2c_register_driver+0x34/0xbc) [<8033c244>] (i2c_register_driver+0x0/0xbc) from [<7f032018>] (sgtl5000_i2c_driver_init+0x18/0x24 [snd_soc_sgtl5000]) r5:7f02fa74 r4:cfb7ff48 [<7f032000>] (sgtl5000_i2c_driver_init+0x0/0x24 [snd_soc_sgtl5000]) from [<80008738>] (do_one_initcall+0xf4/0x150) [<80008644>] (do_one_initcall+0x0/0x150) from [<80053f64>] (load_module+0x174c/0x1db4) [<80052818>] (load_module+0x0/0x1db4) from [<800546ac>] (SyS_init_module+0xe0/0xf4) [<800545cc>] (SyS_init_module+0x0/0xf4) from [<8000e540>] (ret_fast_syscall+0x0/0x30) r6:00005b22 r5:00afed68 r4:00000000 ---[ end trace b24c5c3bb145dbdd ]--- Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> Reviewed-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Rebased onto asoc/for-next, adding back imx_sgtl5000_remove. --- sound/soc/fsl/imx-sgtl5000.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 78f86d8..c49e4cb 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); goto fail; @@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: + if (!IS_ERR(data->codec_clk)) + clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); if (codec_np) @@ -180,6 +182,15 @@ fail: return ret; } +static int imx_sgtl5000_remove(struct platform_device *pdev) +{ + struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + + clk_put(data->codec_clk); + + return 0; +} + static const struct of_device_id imx_sgtl5000_dt_ids[] = { { .compatible = "fsl,imx-audio-sgtl5000", }, { /* sentinel */ } -- 1.8.4.rc3

3 2

[alsa-devel] [PATCH 2/2] [RESEND] snd-ac97-codec: Use complete ID when the codec is not known
by Ondrej Zary 30 Sep '13

30 Sep '13

Currently, if the codec ID is not known, the last byte of the ID is used as name. It's not helpful at all, use full codec ID instead. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ac97/ac97_codec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index bf578ba2..30a40a8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1830,7 +1830,7 @@ void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name, int m pid->patch(ac97); } } else - sprintf(name + strlen(name), " id %x", id & 0xff); + sprintf(name + strlen(name), " id %x", id); } /** -- Ondrej Zary

2 2

[alsa-devel] [PATCH] ASoC: generic: Add generic DT based simple codec
by Jean-Francois Moine 30 Sep '13

30 Sep '13

This patch adds a simple sound codec which is described by the DT. This codec may be used when no specific codec action is needed. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- .../devicetree/bindings/sound/simple-codec.txt | 103 +++++++++++++ sound/soc/generic/Kconfig | 6 + sound/soc/generic/Makefile | 2 + sound/soc/generic/simple-codec.c | 197 +++++++++++++++++++++++++ 4 files changed, 3088 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-codec.txt b/Documentation/devicetree/bindings/sound/simple-codec.txt new file mode 100644 index 0000000..75be747 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-codec.txt @@ -0,0 +1,103 @@ +Device-Tree bindings for the simple codec + +Required properties: +- compatible: should be "linux,simple-codec". +- dai-name: name of the codec + +Optional properties: +- capture: information about capture +- playback: information about playback +At least one of the 'capture' or 'playback' nodes must be present. + +Child 'capture' and 'playback' required properties: +- stream-name: name of the stream +- formats: list of the supported formats (see below) +- rates: list of the supported rates (see below) +- #channels: minimum and maximum numbers of channels + +Formats: + "s8" + "u8" + "s16_le" + "s16_be" + "u16_le" + "u16_be" + "s24_le" + "s24_be" + "u24_le" + "u24_be" + "s32_le" + "s32_be" + "u32_le" + "u32_be" + "float_le" + "float_be" + "float64_le" + "float64_be" + "iec958_subframe_le" + "iec958_subframe_be" + "mu_law" + "a_law" + "ima_adpcm" + "mpeg" + "gsm" + "special" + "s24_3l" + "s24_3be" + "u24_3le" + "u24_3be" + "s20_3le" + "s20_3be" + "u20_3le" + "u20_3be" + "s18_3le" + "s18_3b" + "u18_3le" + "u18_3be" + "g723_24" + "g723_24_1b" + "g723_40" + "g723_40_1b" + "dsd_u8" + "dsd_u16_le" + +Rates: + 5512 + 8000 + 11025 + 16000 + 22050 + 32000 + 44100 + 48000 + 64000 + 88200 + 96000 + 176400 + 192000 + +Examples: + + spdifo: spdif-transmitter { + compatible = "linux,simple-codec"; + dai-name = "dit-hifi"; + playback { + stream-name = "S/PDIF Playback"; + formats = "s16_le"; + rates = <8000 11025 16000 22050 32000 44100 + 48000 64000 88200 96000>; + #channels = <1 384>; + }; + }; + + hdmio: hdmi-transmitter { + compatible = "linux,simple-codec"; + dai-name = "hdmi-hifi"; + playback { + stream-name = "HDMI Playback"; + formats = "s16_le", "s24_le"; + rates = <32000 44100 48000 88200 96000 + 176400 192000>; + #channels = <2 8>; + }; + }; diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 610f612..d9221539 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -2,3 +2,9 @@ config SND_SIMPLE_CARD tristate "ASoC Simple sound card support" help This option enables generic simple sound card support + +config SND_SIMPLE_CODEC + tristate "ASoC Simple sound codec support" + depends on OF + help + This option enables generic simple sound codec support diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9c3b246..f94620e 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,3 +1,5 @@ snd-soc-simple-card-objs := simple-card.o +snd-soc-simple-codec-objs := simple-codec.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_SIMPLE_CODEC) += snd-soc-simple-codec.o diff --git a/sound/soc/generic/simple-codec.c b/sound/soc/generic/simple-codec.c new file mode 100644 index 0000000..217cd0e --- /dev/null +++ b/sound/soc/generic/simple-codec.c @@ -0,0 +1,197 @@ +/* + * ALSA SoC simple codec driver + * + * This driver is used by controllers which do not need any codec + * (s/pdif, hdmi..). + * Its parameters are built from a DT description. + * + * Copyright: (C) 2013 Jean-François Moine <moinejf(a)free.fr> + * + * 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/module.h> +#include <sound/soc.h> +#include <linux/of.h> + +#define DRV_NAME "simple-codec" + +static struct snd_soc_codec_driver soc_codec_simple_codec; + +static char *format_tb[] = { + "s8", "u8", "s16_le", "s16_be", + "u16_le", "u16_be", "s24_le", "s24_be", + "u24_le", "u24_be", "s32_le", "s32_be", + "u32_le", "u32_be", "float_le", "float_be", + + "float64_le", "float64_be", "iec958_subframe_le", "iec958_subframe_be", + "mu_law", "a_law", "ima_adpcm", "mpeg", + "gsm", "", "", "", + "", "", "", "special", + + "s24_3l", "s24_3be", "u24_3le", "u24_3be", + "s20_3le", "s20_3be", "u20_3le", "u20_3be", + "s18_3le", "s18_3b", "u18_3le", "u18_3be", + "g723_24", "g723_24_1b", "g723_40", "g723_40_1b", + + "dsd_u8", "dsd_u16_le", +}; + +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#error "Change this table" +#endif +static u32 rate_tb[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 48000, 64000, 88200, 96000, 176400, 192000 }; + +static int get_format(u64 *p_formats, char *p) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(format_tb); i++) { + if (strcmp(format_tb[i], p) == 0) { + *p_formats |= 1 << i; + return 0; + } + } + return -EINVAL; +} + +static int get_rate(u32 *p_rates, u32 val) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rate_tb); i++) { + if (rate_tb[i] == val) { + *p_rates |= 1 << i; + return 0; + } + } + return -EINVAL; +} + +static int stream_populate(struct device *dev, + struct snd_soc_pcm_stream *stream, + struct device_node *np) +{ + u64 formats; + u32 rates, val; + char *p; + int i, ret; + + ret = of_property_read_string(np, "stream-name", &stream->stream_name); + if (ret) { + dev_err(dev, "Failed to parse stream-name string\n"); + return ret; + } + + i = 0; + formats = 0; + for (;;) { + ret = of_property_read_string_index(np, "formats", i, + (const char **) &p); + if (ret) + break; + ret = get_format(&formats, p); + if (ret) { + dev_err(dev, "Bad codec format\n"); + return ret; + } + i++; + } + stream->formats = formats; + + i = 0; + rates = 0; + for (;;) { + ret = of_property_read_u32_index(np, "rates", i, &val); + if (ret) + break; + ret = get_rate(&rates, val); + if (ret) { + dev_err(dev, "Bad rate\n"); + return ret; + } + i++; + } + stream->rates = rates; + + ret = of_property_read_u32_index(np, "#channels", 0, &val); + if (ret) { + dev_err(dev, "Bad min channel\n"); + return ret; + } + stream->channels_min = val; + ret = of_property_read_u32_index(np, "#channels", 1, &val); + if (ret) { + dev_err(dev, "Bad max channel\n"); + return ret; + } + stream->channels_max = val; + + return 0; +} + +static int simple_codec_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct device_node *np = dev->of_node; + struct device_node *np2; + struct snd_soc_dai_driver *dai; + int ret; + + dai = devm_kzalloc(dev, sizeof *dai, GFP_KERNEL); + if (!dai) + return -ENOMEM; + + ret = of_property_read_string(np, "dai-name", &dai->name); + if (ret) { + dev_err(dev, "Failed to parse dai-name\n"); + return ret; + } + + np2 = of_find_node_by_name(np, "capture"); + if (np2) { + ret = stream_populate(dev, &dai->capture, np2); + if (ret) + return ret; + } + np2 = of_find_node_by_name(np, "playback"); + if (np2) { + ret = stream_populate(dev, &dai->playback, np2); + if (ret) + return ret; + } + + return snd_soc_register_codec(dev, &soc_codec_simple_codec, + dai, 1); +} + +static int simple_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static const struct of_device_id simple_codec_of_match[] = { + { .compatible = "linux,simple-codec", }, + { } +}; +MODULE_DEVICE_TABLE(of, simple_codec_of_match); + +static struct platform_driver simple_codec_driver = { + .probe = simple_codec_probe, + .remove = simple_codec_remove, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = simple_codec_of_match, + }, +}; + +module_platform_driver(simple_codec_driver); + +MODULE_AUTHOR("Jean-Francois Moine <moinejf(a)free.fr>"); +MODULE_DESCRIPTION("simple codec driver"); +MODULE_LICENSE("GPL"); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

5 12

[alsa-devel] [PATCH] ASoC: pcm: Remove extra spaces from dev_ prints
by Jarkko Nikula 30 Sep '13

30 Sep '13

dev_ prints are already prefixed by ": " before format string so there is no need for extra spaces. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6..d449872 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -721,7 +721,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - dev_dbg(fe->dev, " connected new DPCM %s path %s %s %s\n", + dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", be->dai_link->name); @@ -749,7 +749,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, if (dpcm->fe == fe) continue; - dev_dbg(fe->dev, " reparent %s path %s %s %s\n", + dev_dbg(fe->dev, "reparent %s path %s %s %s\n", stream ? "capture" : "playback", dpcm->fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -773,7 +773,7 @@ static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) continue; - dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n", + dev_dbg(fe->dev, "freed DSP %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -2116,7 +2116,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } -- 1.8.4.rc3

2 1

[alsa-devel] [RFC] ALSA: Reduce delay for a blocking pcm_drain.
by Dylan Reid 30 Sep '13

30 Sep '13

This patch addresses two issues related to calling snd_pcm_drain. If no_period_wakeup is set, then snd_pcm_drain would wait MAX_SCHEDULE_TIMEOUT jiffies, without a wakeup pending this will leave the calling task blocked indefinitely. Instead block for double the buffer size played back at a low sample rate (4k chosen arbitrarily). Also if the stream is running with period wakeups but with a long period, the delay could be seconds. If only a small part of the buffer is being used, this is unnecessary. Instead wait for the remaining samples to play out, plus one millisecond. This allows systems that fill the buffer only part way to still do a blocking drain before closing to avoid losing samples. Change-Id: I84ee1a52d9bdf80ea6065fb9533e565d3e6c8021 Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- This can be worked around in user space as well by using a nonblocking drain and sleeping based on how many samples are queued. Would that be the right thing to do or is this (or something similar) worth fixing here. To test this I opened a pcm and set no period wakeups, played a few samples and then called drain, which hung indefinitely. Using a USB headset with a max period size of 128k but only 15ms of the buffer filled, having the drain time shorter without losing samples makes a blocking call feasible. --- sound/core/pcm_native.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f4aaf5a..a5f0695 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1481,7 +1481,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } for (;;) { - long tout; + long tout, tout_ms; struct snd_pcm_runtime *to_check; if (signal_pending(current)) { result = -ERESTARTSYS; @@ -1505,16 +1505,13 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, snd_pcm_stream_unlock_irq(substream); up_read(&snd_pcm_link_rwsem); snd_power_unlock(card); - if (runtime->no_period_wakeup) - tout = MAX_SCHEDULE_TIMEOUT; - else { - tout = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - tout = max(t, tout); - } - tout = msecs_to_jiffies(tout * 1000); + tout_ms = runtime->buffer_size * 2 * 1000 / 4000; + if (runtime->rate) { + snd_pcm_sframes_t hw_avail; + hw_avail = snd_pcm_playback_hw_avail(runtime); + tout_ms = 1 + hw_avail * 1000 / runtime->rate; } + tout = msecs_to_jiffies(tout_ms); tout = schedule_timeout_interruptible(tout); snd_power_lock(card); down_read(&snd_pcm_link_rwsem); -- 1.8.1.3.605.g02339dd

3 3

[alsa-devel] [PATCH] Enable internal mic on a Thinkpad machine
by David Henningsson 30 Sep '13

30 Sep '13

From: Kailang <kailang(a)realtek.com> More thorough testing showed that these verbs were necessary to improve quality of the internal mic. Patch originally from Realtek. Cc: stable(a)kernel.org (3.11) BugLink: https://bugs.launchpad.net/bugs/1231931 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc07d36..c4d54a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3531,6 +3531,7 @@ enum { ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, + ALC283_FIXUP_INT_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -3790,6 +3791,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc282_fixup_asus_tx300, }, + [ALC283_FIXUP_INT_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x1a}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0011}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3874,7 +3885,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), -- 1.7.9.5

2 2

Re: [alsa-devel] pcm_rate API question
by Rob Sykes 30 Sep '13

30 Sep '13

> From: Rob Sykes <aquegg(a)yahoo.co.uk> >To: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> >Sent: Friday, 27 September 2013, 16:28 >Subject: Re: [alsa-devel] pcm_rate API question > >> From: Clemens Ladisch <clemens(a)ladisch.de> >> To: Rob Sykes <aquegg(a)yahoo.co.uk>; "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> >> Cc: >> Sent: Friday, 27 September 2013, 13:38 >> Subject: Re: [alsa-devel] pcm_rate API question >> >> Rob Sykes wrote: >>> void (*convert_s16)(void *obj, int16_t *dst, unsigned int >> dst_frames, >>> const int16_t *src, unsigned int src_frames); >>> >>> is the plugin obliged always to consume src_frames and generate dst_frames >> samples? >> >> Yes. src_frames and dst_frames are the period sizes of the respective >> devices; their values will never change, and they define the actual >> conversion ratio. > >Thanks, Clemens. Typically there is some buffering/delay within a resampler -- how does the pcm_rate API allow for this? E.g. from the libsamplerate FAQ: > >Q6 : I'm use the SRC_SINC_* converters and up-sampling by a ratio of 2. I reset the converter and put in 1000 samples and I expect to get 2000 samples out, but I'm getting less than that. Why? >The short answer is that there is a transport delay inside the converter itself. Long answer follows. By way of example, the first time you call src_process() you might only get 1900 samples out. > However, after that first call all subsequent calls will probably get you about 2000 samples out for every 1000 samples you put in. Also, just noticed that when performing a rate conversion from 44100 -> 48000, the buffer sizes used by alsa are 5512 and 6000, however the actual ratio for the required conversion is 5512.5 to 6000. If, as you say above, the buffer sizes define the conversion ratio, then this seems to imply that alsa is unable to perform accurate rate conversion for one of the most common uses! — am I missing something here? A pointer to the API documentation, or perhaps some relevant threads, would be very useful. Cheers, Rob

2 3

Re: [alsa-devel] [RFC] ALSA: Reduce delay for a blocking pcm_drain.
by Raymond Yau 30 Sep '13

30 Sep '13

> > This can be worked around in user space as well by using a nonblocking drain and > sleeping based on how many samples are queued. Would that be the right thing to > do or is this (or something similar) worth fixing here. > > To test this I opened a pcm and set no period wakeups, played a few samples and > then called drain, which hung indefinitely. Using a USB headset with a > max period size of 128k but only 15ms of the buffer filled, having the > drain time shorter without losing samples makes a blocking call > feasible. > Do you mean that you tested on a patched version of USB audio driver since only three drivers support SNDRV_PCM_INFO_NO_PERIOD_WAKEUP http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound?qt=gr…

2 1

Re: [alsa-devel] Bug: Conexant CX20588, Kernel 3.11, Win8 changes GPIO settings
by Takashi Iwai 30 Sep '13

30 Sep '13

At Fri, 27 Sep 2013 22:38:22 +0200, Christopher wrote: > > I attached the alsa-info. This issue seems to be specific to the driver > provided by Acer (with thx, equalizer etc.) > Thanks for your help. Thanks. Could you check whether the patch below works? Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix GPIO for Acer Aspire 3830TG Acer Aspire 3830TG seems requiring GPIO bit 0 as the primary mute control. When a machine is booted after Windows 8, the GPIO pin is turned off and it results in the silent output. This patch adds the manual fixup of GPIO bit 0 for this model. Reported-by: Christopher <DIDI2002(a)web.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4edd2d0..ec68eae 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3231,6 +3231,7 @@ enum { CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, + CXT_FIXUP_GPIO1, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3375,6 +3376,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_headphone_mic, }, + [CXT_FIXUP_GPIO1] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x01 }, + { } + }, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3384,6 +3394,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), -- 1.8.4

1 0

[alsa-devel] [PATCH 1/2] [RESEND] snd-ac97-codec: Add ID for TI TLV320AIC27 codec
by Ondrej Zary 30 Sep '13

30 Sep '13

Add 0x54584e03 ID for TI TLV320AIC27 AC'97 codec according to datasheet: http://www.ti.com/lit/ds/slas253a/slas253a.pdf The weird thing is that the chip is physically marked 320AD91. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ac97/ac97_codec.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 445ca48..bf578ba2 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,6 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] +{ 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -- Ondrej Zary

2 1

[alsa-devel] [PATCH] caiaq: LED support for Maschine Controller
by Hannes Gräuler 30 Sep '13

30 Sep '13

This patch adds LED support for the Native Instruments Maschine Controller. It adds ALSA controls for dimming the LEDs of all buttons and the backlight of the two displays. Signed-off-by: Hannes Gräuler <hgraeule(a)uos.de> --- sound/usb/caiaq/control.c | 92 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/caiaq/device.c | 25 +++++++++++++ sound/usb/caiaq/device.h | 5 +++ 3 files changed, 122 insertions(+) diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index ae6b50f..f65fc09 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -28,6 +28,7 @@ #include "control.h" #define CNT_INTVAL 0x10000 +#define MASCHINE_BANK_SIZE 32 static int control_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -105,6 +106,10 @@ static int control_put(struct snd_kcontrol *kcontrol, USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1)) cmd = EP1_CMD_DIMM_LEDS; + if (cdev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) + cmd = EP1_CMD_DIMM_LEDS; + if (pos & CNT_INTVAL) { int i = pos & ~CNT_INTVAL; @@ -121,6 +126,20 @@ static int control_put(struct snd_kcontrol *kcontrol, usb_sndbulkpipe(cdev->chip.dev, 8), cdev->ep8_out_buf, sizeof(cdev->ep8_out_buf), &actual_len, 200); + } else if (cdev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) { + + int bank = 0; + int offset = 0; + + if (i >= MASCHINE_BANK_SIZE) { + bank = 0x1e; + offset = MASCHINE_BANK_SIZE; + } + + snd_usb_caiaq_send_command_bank(cdev, cmd, bank, + cdev->control_state + offset, + MASCHINE_BANK_SIZE); } else { snd_usb_caiaq_send_command(cdev, cmd, cdev->control_state, sizeof(cdev->control_state)); @@ -490,6 +509,74 @@ static struct caiaq_controller kontrols4_controller[] = { { "LED: FX2: Mode", 133 | CNT_INTVAL }, }; +static struct caiaq_controller maschine_controller[] = { + { "LED: Pad 1", 3 | CNT_INTVAL }, + { "LED: Pad 2", 2 | CNT_INTVAL }, + { "LED: Pad 3", 1 | CNT_INTVAL }, + { "LED: Pad 4", 0 | CNT_INTVAL }, + { "LED: Pad 5", 7 | CNT_INTVAL }, + { "LED: Pad 6", 6 | CNT_INTVAL }, + { "LED: Pad 7", 5 | CNT_INTVAL }, + { "LED: Pad 8", 4 | CNT_INTVAL }, + { "LED: Pad 9", 11 | CNT_INTVAL }, + { "LED: Pad 10", 10 | CNT_INTVAL }, + { "LED: Pad 11", 9 | CNT_INTVAL }, + { "LED: Pad 12", 8 | CNT_INTVAL }, + { "LED: Pad 13", 15 | CNT_INTVAL }, + { "LED: Pad 14", 14 | CNT_INTVAL }, + { "LED: Pad 15", 13 | CNT_INTVAL }, + { "LED: Pad 16", 12 | CNT_INTVAL }, + + { "LED: Mute", 16 | CNT_INTVAL }, + { "LED: Solo", 17 | CNT_INTVAL }, + { "LED: Select", 18 | CNT_INTVAL }, + { "LED: Duplicate", 19 | CNT_INTVAL }, + { "LED: Navigate", 20 | CNT_INTVAL }, + { "LED: Pad Mode", 21 | CNT_INTVAL }, + { "LED: Pattern", 22 | CNT_INTVAL }, + { "LED: Scene", 23 | CNT_INTVAL }, + + { "LED: Shift", 24 | CNT_INTVAL }, + { "LED: Erase", 25 | CNT_INTVAL }, + { "LED: Grid", 26 | CNT_INTVAL }, + { "LED: Right Bottom", 27 | CNT_INTVAL }, + { "LED: Rec", 28 | CNT_INTVAL }, + { "LED: Play", 29 | CNT_INTVAL }, + { "LED: Left Bottom", 32 | CNT_INTVAL }, + { "LED: Restart", 33 | CNT_INTVAL }, + + { "LED: Group A", 41 | CNT_INTVAL }, + { "LED: Group B", 40 | CNT_INTVAL }, + { "LED: Group C", 37 | CNT_INTVAL }, + { "LED: Group D", 36 | CNT_INTVAL }, + { "LED: Group E", 39 | CNT_INTVAL }, + { "LED: Group F", 38 | CNT_INTVAL }, + { "LED: Group G", 35 | CNT_INTVAL }, + { "LED: Group H", 34 | CNT_INTVAL }, + + { "LED: Auto Write", 42 | CNT_INTVAL }, + { "LED: Snap", 43 | CNT_INTVAL }, + { "LED: Right Top", 44 | CNT_INTVAL }, + { "LED: Left Top", 45 | CNT_INTVAL }, + { "LED: Sampling", 46 | CNT_INTVAL }, + { "LED: Browse", 47 | CNT_INTVAL }, + { "LED: Step", 48 | CNT_INTVAL }, + { "LED: Control", 49 | CNT_INTVAL }, + + { "LED: Top Button 1", 57 | CNT_INTVAL }, + { "LED: Top Button 2", 56 | CNT_INTVAL }, + { "LED: Top Button 3", 55 | CNT_INTVAL }, + { "LED: Top Button 4", 54 | CNT_INTVAL }, + { "LED: Top Button 5", 53 | CNT_INTVAL }, + { "LED: Top Button 6", 52 | CNT_INTVAL }, + { "LED: Top Button 7", 51 | CNT_INTVAL }, + { "LED: Top Button 8", 50 | CNT_INTVAL }, + + { "LED: Note Repeat", 58 | CNT_INTVAL }, + + { "Backlight Display", 59 | CNT_INTVAL } +}; + static int add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *cdev) { @@ -553,6 +640,11 @@ int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *cdev) ret = add_controls(kontrols4_controller, ARRAY_SIZE(kontrols4_controller), cdev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + ret = add_controls(maschine_controller, + ARRAY_SIZE(maschine_controller), cdev); + break; } return ret; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 1a61dd1..bc55f70 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -235,6 +235,31 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, cdev->ep1_out_buf, len+1, &actual_len, 200); } +int snd_usb_caiaq_send_command_bank(struct snd_usb_caiaqdev *cdev, + unsigned char command, + unsigned char bank, + const unsigned char *buffer, + int len) +{ + int actual_len; + struct usb_device *usb_dev = cdev->chip.dev; + + if (!usb_dev) + return -EIO; + + if (len > EP1_BUFSIZE - 2) + len = EP1_BUFSIZE - 2; + + if (buffer && len > 0) + memcpy(cdev->ep1_out_buf+2, buffer, len); + + cdev->ep1_out_buf[0] = command; + cdev->ep1_out_buf[1] = bank; + + return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), + cdev->ep1_out_buf, len+2, &actual_len, 200); +} + int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, int rate, int depth, int bpp) { diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index ad102fa..ab0f752 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -128,5 +128,10 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, unsigned char command, const unsigned char *buffer, int len); +int snd_usb_caiaq_send_command_bank(struct snd_usb_caiaqdev *cdev, + unsigned char command, + unsigned char bank, + const unsigned char *buffer, + int len); #endif /* CAIAQ_DEVICE_H */ -- 1.8.4

3 2

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

30 Sep '13

The following changes since commit 4a10c2ac2f368583138b774ca41fac4207911983: Linux 3.12-rc2 (2013-09-23 15:41:09 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-rc2 for you to fetch changes up to 267666ea3bd6e804c2c8bbafbe628026be963d06: Merge remote-tracking branch 'asoc/fix/max98095' into asoc-linus (2013-09-27 14:56:24 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.12 Nothing too exciting here, all driver specific except for the fix from Liam for DPCM systems which have both front and back end DAIs which is not yet used by anything in mainline. ---------------------------------------------------------------- Dan Carpenter (3): ASoC: max98095: a couple array underflows ASoC: ab8500-codec: info leak in anc_status_control_put() ASoC: 88pm860x: array overflow in snd_soc_put_volsw_2r_st() Liam Girdwood (1): ASoC: core: Only add platform DAI widgets once. Mark Brown (6): Merge remote-tracking branch 'asoc/fix/88pm860x' into asoc-linus Merge remote-tracking branch 'asoc/fix/ab8500' into asoc-linus Merge remote-tracking branch 'asoc/fix/bfin' into asoc-linus Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/max98095' into asoc-linus Philipp Zabel (2): ASoC: imx-sgtl5000: do not use devres on a foreign device ASoC: imx-sgtl5000: Fix uninitialized pointer use in error path Valentin Ilie (1): ASoC: blackfin: Add missing break statement to bf6xx sound/soc/blackfin/bf6xx-i2s.c | 1 + sound/soc/codecs/88pm860x-codec.c | 3 +++ sound/soc/codecs/ab8500-codec.c | 7 ++++++- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/fsl/imx-sgtl5000.c | 7 +++++-- sound/soc/soc-core.c | 1 - 6 files changed, 17 insertions(+), 6 deletions(-)

2 1

Re: [alsa-devel] Realtek ALC286 SONY Vaio Pro 13 Ubuntu 13.10
by David Henningsson 30 Sep '13

30 Sep '13

(Adding alsa-devel to CC) On 09/28/2013 05:53 AM, Yuval Levy wrote: > Sirs: > > I have been pointed to your patch to implement headset jack > functionality for some Dell hw [1] after reporting and investigating > soud support issues [2] with a SONY Vaio Pro 13 (Realtek ALC286). This different headset mode support is likely unrelated to your problem. And they don't cover your codec (ALC286), so you would need another set of verbs too, things that only Realtek knows. > Output to internal loudspeakers and HDMI works as expected, but neither > internal microphone nor the headset work (and they do work in Windows, > so it is not a hardware issue). An important distinction is here: to they work in Windows without specific Realtek drivers installed, or do you require Realtek's Windows drivers for this support? Second, to do some actual triaging. I see you have been running hda-jack-sense-test. Could you run it all three scenarios: 1) nothing connected 2) headphone (without mic) connected 3) headset (heaphone + mic combo) connected ...and submit the result? In ideal circumstances, 0x21 should report "Yes" in scenario 2 and 3, and 0x18 should only report yes in scenario 3. If this is not the result you get, try using hda-jack-retask to turn off the jack detection: in Ubuntu 13.10 just install the alsa-tools-gui package (no need to use my ppa). Click "advanced override", find pins 0x18 or 0x21, change "jack detection" from "yes" to "no" and click apply. Now you should be able to manually choose between internal and external microphone, as well as between headphones as speakers. See if this helps. > As you can see in the bug thread, I have been helped in the diagnosis by > a very friendly expert, but it seems that we have reached the limit and > without support from Realtek or Sony the issue will go unresolved. I'm afraid Sony is not a company that actively supports Linux for the desktop machines, sorry. Canonical, which seems to have the most contracts for preloading Linux these days, has a list of certified hardware here: http://www.ubuntu.com/certification/ > Would you kindly help me find out what need to be done for the Realtek > ALC286 in the SONY Vaio Pro 13 to be supported in Ubuntu Linux? > > Many thanks, > Yuv > > [1] > <https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p…> > [2] <https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1228689> > -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

1 0

[alsa-devel] [PATCH] ASoC: si476x: Remove custom register I/O implementation
by Mark Brown 28 Sep '13

28 Sep '13

From: Mark Brown <broonie(a)linaro.org> The current si476x I/O implementation wraps the regmap for the core with functions that make the register map cache only when the device is powered down. This implementation appears to be incomplete since there is no code to synchronise the cache so writes done while the core is powered down will be ignored, the device will only be configured if it is powered. A better and more idiomatic approach would be to have the MFD manage the cache, making the device cache only when it powers things down. This also allows ASoC to use the standard regmap helpers for the device which helps remove the ASoC custom ones so do convert to do that. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/si476x.c | 46 +--------------------------------------------- 1 file changed, 1 insertion(+), 45 deletions(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 38f3b10..ec29d26 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -60,48 +60,6 @@ enum si476x_pcm_format { SI476X_PCM_FORMAT_S24_LE = 6, }; -static unsigned int si476x_codec_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - int err; - unsigned int val; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_read(core->regmap, reg, &val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - if (err < 0) - return err; - - return val; -} - -static int si476x_codec_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val) -{ - int err; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_write(core->regmap, reg, val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - return err; -} - static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), @@ -239,7 +197,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, static int si476x_codec_probe(struct snd_soc_codec *codec) { - codec->control_data = i2c_mfd_cell_to_core(codec->dev); + codec->control_data = dev_get_regmap(codec->dev.parent); return 0; } @@ -268,8 +226,6 @@ static struct snd_soc_dai_driver si476x_dai = { static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, - .read = si476x_codec_read, - .write = si476x_codec_write, .dapm_widgets = si476x_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), .dapm_routes = si476x_dapm_routes, -- 1.8.4.rc3

2 12