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[alsa-devel] [PATCH v2 0/3] Expose more controls from tegra-max98090
by Tomeu Vizoso 21 Feb '15

21 Feb '15

v2: * Added ack tags from Stephen Hello, we need some more controls in userspace so policy can be applied at events such as microphone and headphone jacks being plugged in, to be used by Tegra-based Chromebooks. Regards, Tomeu Tomeu Vizoso (3): ASoC: tegra: Expose Headphones pin to userspace ASoC: tegra: Add sink for the internal mic to tegra_max98090 ASoC: tegra: Add control for the Mic Jack pin .../devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + sound/soc/tegra/tegra_max98090.c | 4 ++++ 2 files changed, 5 insertions(+) -- 1.9.3

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[alsa-devel] [PATCH] ASoC: sta32x: fix register range in regmap.
by Thomas Niederprüm 21 Feb '15

21 Feb '15

The STA32X_AUTO3 is a writable register that currently does not appear in the regmap ranges(neither read nor write). By adding this register to the register ranges there is no gap anymore and the existing register ranges can be joined. This fixes a regression that made it impossible to choose the preset EQ mode set through the STA32X_AUTO3 register. Signed-off-by: Thomas Niederprüm <niederp(a)physik.uni-kl.de> --- I realized that I missed one register in the DT enabling patchset, so here I send a patch to clean up my mess :) Thomas sound/soc/codecs/sta32x.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index bc6edb1..d320227 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -107,13 +107,11 @@ static const struct reg_default sta32x_regs[] = { }; static const struct regmap_range sta32x_write_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_read_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_STATUS), + regmap_reg_range(STA32X_CONFA, STA32X_STATUS), }; static const struct regmap_range sta32x_volatile_regs_range[] = { -- 2.1.1

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[alsa-devel] [Patch V3 1/2] ASoC: max98357a: Add missing header files
by Kenneth Westfield 21 Feb '15

21 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Add missing header files to avoid implicit declarations and indirect inclusions. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- for fix/max98357a branch changes from v2: - split patch into logical changes - rebased to new branch changes from v1: - matched subject line to subsystem style sound/soc/codecs/max98357a.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index f493fb6fd4ea764d617b3d7dd5f3b1ebe1e0fe20..e9e6efbc21dd527299fb459ec01903fa8c24dae2 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,10 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include <linux/module.h> +#include <linux/device.h> +#include <linux/err.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> #define DRV_NAME "max98357a" -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ASoC: cirrus: tlv320aic23 needs I2C
by Arnd Bergmann 21 Feb '15

21 Feb '15

The tlv320aic23 codec is selected by the ep93xx snapper platform, which are missing a dependency on I2C, and that can result in this build error, as found during randconfig builds: .../codecs/tlv320aic23-i2c.c: In function 'tlv320aic23_i2c_probe': .../codecs/tlv320aic23-i2c.c:27:2: error: implicit declaration of function 'i2c_check_functionality' [-Werror=implicit-function-declaration] if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) ^ This adds the missing dependency. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 7b7fbcd49e5e..c7cd60f009e9 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97 config SND_EP93XX_SOC_SNAPPERCL15 tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C select SND_EP93XX_SOC_I2S select SND_SOC_TLV320AIC23_I2C help

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[alsa-devel] [PATCH v2] ASoC: Samsung: add missing I2C/SPI dependencies
by Arnd Bergmann 21 Feb '15

21 Feb '15

A few sound drivers for the samsung platforms are missing dependencies on I2C or SPI, which can lead to build errors like codecs/rt5631.c:1737:1: warning: data definition has no type or storage class 31_i2c_driver); codecs/rt5631.c:1737:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] codecs/rt5631.c:1737:1: warning: parameter names (without types) in function declaration codecs/rt5631.c:1726:26: warning: 'rt5631_i2c_driver' defined but not used [-Wunused-variable] I have gone through all the ones that did not already have an I2C dependency and added the ones that I found missing, namely arndale, odroid-x2, littlemill, bells and speyside and this patch adds all the dependencies. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3cebf6ca03df..0632a36852c8 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -174,7 +174,7 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -189,7 +189,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -206,7 +206,7 @@ config SND_SOC_LOWLAND config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 @@ -223,7 +223,7 @@ config SND_SOC_SNOW config SND_SOC_ODROIDX2 tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S help @@ -231,6 +231,6 @@ config SND_SOC_ODROIDX2 config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631

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[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Ensure DSP controls are always persistent
by Charles Keepax 21 Feb '15

21 Feb '15

From: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> Currently DSP controls are persistent (across DSP On/Off) only if they were set whilst the DSP is off. This change makes the controls persistent irrespective of when they are set. Signed-off-by: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ff67b33..185ee85 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -420,10 +420,9 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, memcpy(ctl->cache, p, ctl->len); - if (!ctl->enabled) { - ctl->set = 1; + ctl->set = 1; + if (!ctl->enabled) return 0; - } return wm_coeff_write_control(kcontrol, p, ctl->len); } -- 1.7.2.5

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[alsa-devel] [PATCH] ALSA: firewire-lib: fix an unexpected byte sequence for micro sign
by Takashi Sakamoto 21 Feb '15

21 Feb '15

The sign for microsecond (U+0085, MICRO SIGN) was encoded to '0x c2 b5' by UTF-8 character encoding scheme. But the byte sequence was converted to '0x c3 82 c2 b5' in a previous commit. As a result, the byte sequence cannot represent microsecond sign in UTF-8 or ASCII. This may confuse developers. This commit replaces the sign to string expression with 'microseconds' to purge superfluous troubles. Fixes: 5c697e5b46ef("ALSA: firewire-lib: remove rx_blocks_for_midi quirk") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 0d58018..7b7b183 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -33,7 +33,7 @@ */ #define MAX_MIDI_RX_BLOCKS 8 -#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 Âµs */ +#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ #define ISO_DATA_LENGTH_SHIFT 16 -- 2.1.0

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[alsa-devel] [PATCH v3 0/5] alsa-lib: audio timestamping evolutions
by Pierre-Louis Bossart 21 Feb '15

21 Feb '15

alsa-lib side of kernel patches already merged on topic/timestamp. Pierre-Louis Bossart (5): sync asound.h with kernel version add support for get/set_audio_htstamp_config pcm: add helper functions to query timestamping capabilities pcm: add support for new STATUS_EXT ioctl test: fix audio_time with new get/set audio_tstamp_config include/pcm.h | 47 ++++- include/sound/asound.h | 37 +++- src/pcm/pcm.c | 73 +++++++- src/pcm/pcm_hw.c | 16 +- test/audio_time.c | 491 +++++++++++++++++++++++++++++++------------------ 5 files changed, 475 insertions(+), 189 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v6 0/7] audio timestamping evolutions
by Pierre-Louis Bossart 21 Feb '15

21 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. v5: no functional change, only addressed Takashi comments on patch 2,3,4 fixed used of bool with true/false and ! in comparisons move trigger_tstamp_pending_update flag to snd_usb_substream, will be used by ALSA core when other devices need it v6: rebased on v3.19-rc7 after merge of initial 4 patches address Takashi's comments: simplified parameter handling, const and zero-clear cleanups in pcm.h, fix git-bisect issue Pierre-Louis Bossart (7): ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: add .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: core: remove .wall_clock ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 66 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 28 ++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 41 +++++- sound/pci/hda/hda_controller.c | 40 ++++-- 7 files changed, 444 insertions(+), 55 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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[alsa-devel] [PATCH 0/4] Internal PCM fixes / cleanups
by Takashi Iwai 20 Feb '15

20 Feb '15

Hi, while reading the internal PCM (BE) handling code again, I noticed a few potential bugs. Here are fix and cleanup patches. Takashi === Takashi Iwai (4): ALSA: pcm: Minor refactoring in snd_pcm_attach_substream() ALSA: pcm: Don't add internal PCMs to PCM device list ALSA: pcm: Don't notify internal PCMs ALSA: pcm: Don't ignore internal PCMs in snd_pcm_dev_disconnect() sound/core/pcm.c | 92 +++++++++++++++++++++++++------------------------------- 1 file changed, 41 insertions(+), 51 deletions(-) -- 2.3.0

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[alsa-devel] [PATCH] ALSA: hda - Avoid unnecessary power-up at mixer amp changes
by Takashi Iwai 20 Feb '15

20 Feb '15

When the mixer amp is touched by control elements, we don't have to power up always; if the codec was suspended at the time, we can just update the amp cache and it's reflected to the hardware upon resume. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0734cb35887d..5e755eb6f19a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2148,11 +2148,10 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_read); static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val, - bool init_only) + bool init_only, bool cache_only) { struct hda_amp_info *info; unsigned int caps; - unsigned int cache_only; if (snd_BUG_ON(mask & ~0xff)) mask &= 0xff; @@ -2170,7 +2169,7 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, return 0; } info->vol[ch] = val; - cache_only = info->head.dirty = codec->cached_write; + info->head.dirty |= cache_only; caps = info->amp_caps; mutex_unlock(&codec->hash_mutex); if (!cache_only) @@ -2194,7 +2193,8 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, direction, idx, mask, val, false); + return codec_amp_update(codec, nid, ch, direction, idx, mask, val, + false, codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); @@ -2241,7 +2241,8 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true); + return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true, + codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init); @@ -2383,8 +2384,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, maxval = get_amp_max_value(codec, nid, dir, 0); if (val > maxval) val = maxval; - return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, - HDA_AMP_VOLMASK, val); + return codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val, + false, !hda_codec_is_power_on(codec)); } /** @@ -2434,14 +2435,12 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); @@ -3109,19 +3108,19 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change = codec_amp_update(codec, nid, 0, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change |= codec_amp_update(codec, nid, 1, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); hda_call_check_power_status(codec, nid); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); -- 2.3.0

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[alsa-devel] [PATCH 0/6] Minor HD-audio cleanups
by Takashi Iwai 20 Feb '15

20 Feb '15

Hi, this is a series of minor cleanups for HD-audio codes. No functional changes but just code refactoring for the further works. The patches are found in topic/hda-cleanup branch of sound git tree, too. Takashi === Takashi Iwai (6): ALSA: hda - Drop hda_bus_template for snd_hda_bus_new() ALSA: hda - Embed struct hda_bus_unsolicited into struct hda_bus ALSA: hda - Introduce azx_has_pm_runtime() macro ALSA: hda - Fold hda_priv.h into hda_controller.h ALSA: hda - Drop azx_mixer_create() ALSA: hda - Split azx_codec_create() to two phases sound/pci/hda/hda_codec.c | 54 +----- sound/pci/hda/hda_codec.h | 22 +-- sound/pci/hda/hda_controller.c | 96 +++++----- sound/pci/hda/hda_controller.h | 397 +++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_i915.c | 2 +- sound/pci/hda/hda_intel.c | 23 ++- sound/pci/hda/hda_intel.h | 2 +- sound/pci/hda/hda_local.h | 17 -- sound/pci/hda/hda_priv.h | 406 ----------------------------------------- sound/pci/hda/hda_tegra.c | 9 +- 10 files changed, 477 insertions(+), 551 deletions(-) delete mode 100644 sound/pci/hda/hda_priv.h -- 2.3.0

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[alsa-devel] Correct 5.1 settings for Asus mobo AT4NM10-I with IDT 92HD73C1X5
by Misan Thropos 19 Feb '15

19 Feb '15

I got it to work with the help of "debianuser" on #alsa@freenode and hdajackretask which can be found in alsa-tools-gui (if you use Debian). Had to change some pin settings from default configuration to make it work. You can get the working configuration from: http://www.alsa-project.org/db/?f=71b06cd49f390e0ff4061d069a2efcb93b3148d2 It enables the S/PDIF digital output too. Maybe you like to take that information for a model for IDT 92HD73C1X5. cheers, Misan

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[alsa-devel] [PATCH 0/19] ASoC: rsnd: move 2nd DMAC from DMAEngine to rsnd driver
by Kuninori Morimoto 19 Feb '15

19 Feb '15

Hi Mark, Arnd, Vinod Cc: Laurent, SH-ML Renesas R-Car sound (= rsnd) needs 2 DMAC which are called as Audio DMAC (= 1st DMAC) and Audio DMAC peri peri (2nd DMAC). And rsnd had assumed that 1st / 2nd DMACs are implemented as DMAEngine. But, in result of DMA ML long term discussion, 2nd DMAC was concluded that it is not a general purpose DMAC (2nd DMAC is for Device to Device inside sound system). Additionally, current DMAEngine can't support Device to Device, and we don't have correct DT bindings for it at this point. So the easiest solution for it is that move it from DMAEngine to rsnd driver. http://thread.gmane.org/gmane.linux.ports.sh.devel/41063/focus=43829 These patches add Audio DMAC peri peri implementation in rsnd driver. I will remove current DMAEngine side Audio DMAC peri peri implement if these are accepted. Arnd, Vinod I'm happy if you can send Ack for these patches 1) - 5) are cleanup / tidyup rsnd driver 6) - 11) adds Audio DMAC peri peri implementation in rsnd driver 12) change dma-names for 1st DMAC on DT 13) - 14) cleanup. these are no longer needed now 15) - 19) DT bindings text update 12) changes dma-names for 1st DMAC. It was assumed that 1st / 2nd DMAC are used as DMAEngine. dma-names had same naming rule for both DMAC to avoid complex DT mapping issue, But now, 2nd DMAC is no longer DMAEngine, we don't need to use this naming rule anymore. Now, no SoC/platform is using DMA for rsnd driver yet in upstream. I believe 12) is no problem for DT. Kuninori Morimoto (19): 1) ASoC: rsnd: remove SH-DMA-BASE specific implementation 2) ASoC: rsnd: remove un-needed parameter from rsnd_dma_init() 3) ASoC: rsnd: remove unused rsnd_dma_available() 4) ASoC: rsnd: remove un-needed parameter from rsnd_dma_quit() 5) ASoC: rsnd: tidyup rsnd_dma_to_mod() macro declaration position 6) ASoC: rsnd: enable to get resource by name 7) ASoC: rsnd: add rsnd_gen_get_phy_addr() to get physical address 8) ASoC: rsnd: add dma.c for Audio DMAC / Audio DMAC peri peri 9) ASoC: rsnd: move rsnd_gen_dma_addr() from gen.c to dma.c 10) ASoC: rsnd: enable to care 1st / 2nd DMAC on rsnd_dma_xxx() 11) ASoC: rsnd: add Audio DMAC peri peri support rework 12) ASoC: rsnd: dma-names cares 1st DMAC only 13) ASoC: rsnd: remove rsnd_dma::addr 14) ASoC: rsnd: remove rsnd_dma::dir 15) ASoC: rsnd: tidyup rcar_sound,ssi sample code 16) ASoC: rsnd: add sample code of rcar_sound,src irq 17) ASoC: rsnd: add sample code of reg-names 18) ASoC: rsnd: add sample code of missing clocks 19) ASoC: rsnd: add sample code of dma entry .../devicetree/bindings/sound/renesas,rsnd.txt | 131 +++-- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 224 +------- sound/soc/sh/rcar/dma.c | 583 ++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 145 +---- sound/soc/sh/rcar/rsnd.h | 50 +- sound/soc/sh/rcar/src.c | 5 +- sound/soc/sh/rcar/ssi.c | 5 +- 8 files changed, 725 insertions(+), 420 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] baytrail-pcm-audio: audio broken on Lenovo Thinkpad Tablet 10
by Sebastien Bourdeauducq 18 Feb '15

18 Feb '15

Hi, the Thinkpad Tablet 10 has an impressive amount of problems under Linux, among which the audio is completely broken. The kernel log is: [ 7.532413] rt5640 i2c-10EC5640:00: Device with ID register 6271 is not rt5640/39 [ 7.908208] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [ 7.908231] platform byt-rt5640: Driver byt-rt5640 requests probe deferral [ 7.931549] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 7.931557] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 [ 7.967040] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 7.967047] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 7.967051] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 7.967212] byt-rt5640 byt-rt5640: ASoC: CODEC DAI rt5640-aif1 not registered [ 7.967248] platform byt-rt5640: Driver byt-rt5640 requests probe deferral I'm running kernel 3.19 with the latest firmware from the linux-firmware repository. Sebastien -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH 0/3] Fixing snd_use_case_get() bugs
by Tanu Kaskinen 17 Feb '15

17 Feb '15

Tanu Kaskinen (3): ucm: fix incorrect error code sign ucm: fix the logic of choosing the default cdev ucm: fix some variable constness issues include/use-case.h | 5 +++-- src/ucm/main.c | 51 ++++++++++++++++++++++++++++++++------------------- 2 files changed, 35 insertions(+), 21 deletions(-) -- 1.9.3

2 4

[alsa-devel] [PATCH] ALSA: usb: Fix support for Denon DA-300USB DAC (ID 154e:1003)
by Takashi Iwai 17 Feb '15

17 Feb '15

From: Frank C Guenther <bugzilla.frnkcg(a)spamgourmet.com> Fix problem where playback of Denon DA-300USB DAC sometimes does not start and leads to error messages like "clock source 41 is not valid, cannot use". Solution: Treat this device the same as other Denon/Marantz devices in sound/usb/quirks.c. Tested with both PCM and DSD formats. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=93261 Signed-off-by: Frank C Guenther <bugzilla.frnkcg(a)spamgourmet.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5c0c05314df0..753a47de8459 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1127,6 +1127,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, int err; switch (subs->stream->chip->usb_id) { + case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ @@ -1206,6 +1207,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { switch (le16_to_cpu(dev->descriptor.idProduct)) { + case 0x1003: /* Denon DA300-USB */ case 0x3005: /* Marantz HD-DAC1 */ case 0x3006: /* Marantz SA-14S1 */ mdelay(20); @@ -1267,6 +1269,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* Denon/Marantz devices with USB DAC functionality */ switch (chip->usb_id) { + case USB_ID(0x154e, 0x1003): /* Denon DA300-USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ if (fp->altsetting == 2) -- 2.3.0

1 0

[alsa-devel] [PATCH] ASoC: Intel: add SNDRV_PCM_INFO_DRAIN_TRIGGER flag
by libin.yang＠intel.com 17 Feb '15

17 Feb '15

From: Libin Yang <libin.yang(a)intel.com> Add SNDRV_PCM_INFO_DRAIN_TRIGGER in snd_pcm_hardware.info to call sst_hsw_stream_set_silence_start() when PCM draining. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index d6fa9d5..7e21e8f 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -91,7 +91,8 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | + SNDRV_PCM_INFO_DRAIN_TRIGGER, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = PAGE_SIZE, -- 1.9.1

3 3

[alsa-devel] [PATCH 40/45] include/uapi/sound/emu10k1.h: include sound/asound.h
by Mikko Rapeli 17 Feb '15

17 Feb '15

Fixes userspace compilation errors like: error: field ‘id’ has incomplete type struct snd_ctl_elem_id id; /* full control ID definition */ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/emu10k1.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index d1bbaf7..ec1535b 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -23,8 +23,7 @@ #define _UAPI__SOUND_EMU10K1_H #include <linux/types.h> - - +#include <sound/asound.h> /* * ---- FX8010 ---- -- 2.1.4

2 1

[alsa-devel] [PATCH 39/45] include/uapi/sound/asequencer.h: include sound/asound.h
by Mikko Rapeli 17 Feb '15

17 Feb '15

Fixes userspace compilation error: error: unknown type name ‘snd_seq_client_type_t’ snd_seq_client_type_t type; /* client type */ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/asequencer.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 09c8a00..5a5fa49 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -22,6 +22,7 @@ #ifndef _UAPI__SOUND_ASEQUENCER_H #define _UAPI__SOUND_ASEQUENCER_H +#include <sound/asound.h> /** version of the sequencer */ #define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) -- 2.1.4

2 1

[alsa-devel] [PATCH 35/45] include/uapi/sound/asound.h: include stdlib.h in userspace
by Mikko Rapeli 17 Feb '15

17 Feb '15

Fixes compiler errors like: error: field ‘trigger_tstamp’ has incomplete type error: invalid application of ‘sizeof’ to incomplete t ype ‘struct timespec’ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/asound.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 941d32f..af156b0 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -25,6 +25,9 @@ #include <linux/types.h> +#ifndef __KERNEL__ +#include <stdlib.h> +#endif /* * protocol version -- 2.1.4

2 1

[alsa-devel] [PATCH 0/3] ALSA: hdspm - Remove various code duplication
by Adrian Knoth 17 Feb '15

17 Feb '15

From: Adrian Knoth <aknoth(a)google.com> Hi! As promised last week, here's a series that removes code duplication in hdspm.c. Adrian Knoth (3): ALSA: hdspm - DRY cleanup in .open callbacks ALSA: hdspm - DRY cleanup in .release callback ALSA: hdspm - DRY cleanup of snd_pcm_ops sound/pci/rme9652/hdspm.c | 141 ++++++++++++---------------------------------- 1 file changed, 35 insertions(+), 106 deletions(-) -- 2.1.0

2 4

[alsa-devel] [PATCH v2] ALSA: usb-audio: Don't attempt to get Lifecam HD-5000 sample rate
by Joe Turner 17 Feb '15

17 Feb '15

Adds a quirk to disable the check that the sample rate has been set correctly, as the Lifecam does not support getting the sample rate. This means that we don't need to wait for the USB timeout when attempting to get the sample rate. Waiting for the timeout causes problems in some applications, which give up on the device acquisition process before it has had time to complete, resulting in no sound. Signed-off-by: Joe Turner <joe(a)oampo.co.uk> --- Changes in v2: - Fixed code style, as suggested by Takashi Iwai <tiwai(a)suse.de> sound/usb/clock.c | 5 +++++ sound/usb/quirks.c | 5 +++++ sound/usb/quirks.h | 2 ++ 3 files changed, 12 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 03fed66..9d6e709 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -303,6 +303,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, return err; } + /* Don't check the sample rate for devices which we know don't + * support reading */ + if (snd_usb_get_sample_rate_quirk(chip)) + return 0; + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a739841..50add5e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1111,6 +1111,11 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, } } +int snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) +{ + /* MS Lifecam HD-5000 doesn't support reading the sample rate. */ + return chip->usb_id == USB_ID(0x045E, 0x076D); +} /* Marantz/Denon USB DACs need a vendor cmd to switch * between PCM and native DSD mode diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 1b86238..428c158 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -21,6 +21,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt); +int snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip); + int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp); -- 2.3.0

2 1

[alsa-devel] iMX6 SSI driver: fsl_ssi vs imx-ssi
by Xuebing Wang 17 Feb '15

17 Feb '15

Hi community, I am developing an ASOC codec driver based on iMX6SL platform. I noticed there are fsl_ssi.c and imx-ssi.c in folder sound/soc/fsl. Would you please tell me what are the differences between these 2, and which one should I use? Thanks. -- Xuebing

2 2

[alsa-devel] [PATCH] .gitignore: add test-driver
by Tanu Kaskinen 16 Feb '15

16 Feb '15

Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> --- .gitignore | 1 + 1 file changed, 1 insertion(+) diff --git a/.gitignore b/.gitignore index 05a2459..e1cc163 100644 --- a/.gitignore +++ b/.gitignore @@ -11,6 +11,7 @@ Makefile libtool depcomp ltmain.sh +test-driver version Makefile.in aclocal.m4 -- 1.9.3

2 3

[alsa-devel] [PATCH 0/6] Modernize sequecner device binding
by Takashi Iwai 16 Feb '15

16 Feb '15

Hi, this is a series of patches for 3.21 (or whatever the next-to-next) to sanitze and modernize the device binding code of ALSA sequencer core. Takashi Takashi Iwai (6): ALSA: seq: Move EXPORT_SYMBOL() after each function ALSA: seq: Don't compile snd_seq_device_load_drivers() for built-in ALSA: seq: Rewrite sequencer device binding with standard bus ALSA: seq: Clean up device and driver structs ALSA: seq: Define driver object in each driver ALSA: seq: Drop snd_seq_autoload_lock() and _unlock() include/sound/seq_device.h | 46 +-- include/sound/seq_kernel.h | 6 +- sound/core/seq/oss/seq_oss.c | 22 +- sound/core/seq/oss/seq_oss_synth.c | 6 +- sound/core/seq/oss/seq_oss_synth.h | 4 +- sound/core/seq/seq_device.c | 569 +++++++++---------------------------- sound/core/seq/seq_dummy.c | 6 +- sound/core/seq/seq_midi.c | 36 +-- sound/drivers/opl3/opl3_seq.c | 34 +-- sound/drivers/opl4/opl4_seq.c | 33 +-- sound/isa/sb/emu8000_synth.c | 35 +-- sound/pci/emu10k1/emu10k1_synth.c | 35 +-- 12 files changed, 256 insertions(+), 576 deletions(-) -- 2.3.0

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[alsa-devel] [PATCH] MS Lifecam HD-5000: Don't attempt to get the sample rate
by Joe Turner 14 Feb '15

14 Feb '15

Hi Folks, This is my first time doing anything kernel-related, so apologies in advance for any faux-pas which are to follow. I hope this is at least going to the correct place! Currently after setting the sample rate of USB devices the driver tries to get the sample rate back from the device, to check that the two match up. The Lifecam doesn't support getting the sample rate, and so this operation times-out, and we continue regardless. Because the USB timeout is pretty long this causes issues with some apps, which begin to doubt the ability of the device to produce audio before the timeout is up and misbehave in various ways. See https://www.google.co.uk/search?q=%22cannot+get+freq+at+ep+0x82%22 for a range of reports relating to this issue. This patch adds a quirk so that rather than check the sample rate and timeout, we don't bother checking at all. Hope this all looks okay. Cheers, Joe

2 1

[alsa-devel] [PATCH] ALSA: hda/tegra check correct return value from ioremap_resource
by eliot＠blennerhassett.gen.nz 14 Feb '15

14 Feb '15

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> --- sound/pci/hda/hda_tegra.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 227990b..375e94f 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -329,8 +329,8 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); hda->regs = devm_ioremap_resource(dev, res); - if (IS_ERR(chip->remap_addr)) - return PTR_ERR(chip->remap_addr); + if (IS_ERR(hda->regs)) + return PTR_ERR(hda->regs); chip->remap_addr = hda->regs + HDA_BAR0; chip->addr = res->start + HDA_BAR0; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Add snd_soc_of_parse_audio_routing()
by Nicolin Chen 14 Feb '15

14 Feb '15

This patch adds snd_soc_of_parse_audio_routing() to get dapm routes via Device Tree. Because snd_soc_of_parse_audio_routing() overrides dpam_routes, we use snd_soc_dapm_add_routes() to append the original build-in DPCM routes instead of linking them to dpam_routes directly. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl-asoc-card.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 3f6959c..bdc9671 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -388,6 +388,8 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + snd_soc_dapm_add_routes(&card->dapm, audio_map, ARRAY_SIZE(audio_map)); + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -503,15 +505,19 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dev = &pdev->dev; priv->card.name = priv->name; priv->card.dai_link = priv->dai_link; - priv->card.dapm_routes = audio_map; priv->card.late_probe = fsl_asoc_card_late_probe; - priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + goto asrc_fail; + } + /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; -- 1.9.1

2 4

[alsa-devel] [Patch V4 00/10] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 14 Feb '15

14 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This patch series adds support for I2S audio playback on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This contains an MI2S port, which is what these drivers are configured to use. The LPAIF also contains a DMA engine that is dedicated to moving audio samples into the transmit FIFO of the MI2S port. In addition, there is also low-power memory (LPM) within the audio subsystem, which is used for buffering the audio samples. The development board being used for testing contains the ipq806x SOC and a Maxim max98357a DAC/amp. One bus from the MI2S port of the SOC is connected to the DAC/amp for stereo playback. This bus is configured so that the SOC is bus master and consists of DATA, LRCLK, and BCLK. The DAC/amp does not need MCLK to operate. In addition, a single GPIO pin from the SOC is connected to the same DAC/amp, which gives enable/disable control over the DAC/amp. The specific drivers added are: * a codec DAI driver for controlling the DAC/amp * a CPU DAI driver for controlling the MI2S port * a platform driver for controlling the LPAIF DMA engine These drivers, together, are tied into simple-audio-card to complete the audio implementation. Corresponding additions to the device tree for the ipq806x SOC and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. The LPASS also contains clocks that need to be controlled. Those drivers have been submitted as a separate patch series: [PATCH v3 0/8] qcom audio clock control drivers http://lkml.org/lkml/2015/1/19/656 = Changes since V3 [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085694.h… * Placed the content of the inline functions into the callbacks. * Replaced use of readl/writel register access functions with regmap access functions. Notable exception is the ISR, which uses ioread32/iowrite32. * Rearranged the sequencing of the hardware block enables to fit within the ASoC framework callbacks, while remaining functional. REQ 1: The hardware requires the enable sequence to be: LPAIF-DMA[enable],then LPAIF-MI2S[enable], then DAC-GPIO[enable] REQ 2: The hardware requires the disable sequence to be: DAC-GPIO[disable], then LPAIF-MI2S[disable] * Corrected the implementation of the pointer callback. * Utilize the LPM to buffer audio samples, rather than memory external to LPASS. * Corrected the interrupt clearing in the ISR. * Implemented a default system clock (defined by the simple-card DT node), and optional LPASS DT node modifiers that can alter the system clock in order to expand the range of available bit clock frequencies. * Addressed all of the remaining issues raised by Mark Brown. * General code cleanup. = Changes since V2 [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085186.h… * Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. * Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. * Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. * Removed machine driver and tied DAI drivers to simple-audio-card. * Packaged the build files and Maxim codec files together in one change. * Removed QCOM as vendor from Maxim code and documentation. * Separated the SOC and board definitions into the correct DTS files. * Update device tree documentation to reflect changes. * General code cleanup. = Changes since V1 [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084322.h… * Remove the native LPAIF driver, and move its functionality to the CPU DAI driver. * Add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver). * Use devm_* and dev_* where possible. * ISR only handles relevant DMA channel now. * Update device tree documentation to reflect changes. * General code cleanup. Kenneth Westfield (10): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: max98357a: Document MAX98357A bindings ASoC: qcom: Document LPASS CPU bindings ASoC: codec: Add MAX98357A codec driver ASoC: ipq806x: add LPASS header files ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add LPASS platform driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../devicetree/bindings/sound/max98357a.txt | 14 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 66 +++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 23 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 138 +++++ sound/soc/qcom/Kconfig | 27 + sound/soc/qcom/Makefile | 6 + sound/soc/qcom/lpass-cpu.c | 557 +++++++++++++++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 170 +++++++ sound/soc/qcom/lpass-platform.c | 514 +++++++++++++++++++ sound/soc/qcom/lpass.h | 64 +++ 15 files changed, 1594 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/lpass-cpu.c create mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-platform.c create mode 100644 sound/soc/qcom/lpass.h -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ASoC: Intel: set initial runtime PM status to active for ACPI-enumerated ADSP
by mengdong.lin＠intel.com 14 Feb '15

14 Feb '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The ADSP on Braswell/Baytrail is an ACPI device. This patch sets its initial runtime PM status to active. Otherwise, its initial status is suspended and runtime_suspend ops will not be called after probe and thus cannot further trigger ACPI _PS3 (D3) method to put the device into low power D3cold state. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8a8d56a..d6ea800 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -379,6 +379,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) * initially active. So change the state to active before * enabling the pm */ + + if (!acpi_disabled) + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); if (acpi_disabled) -- 1.9.1

2 1

[alsa-devel] [Patch] ASoC: codec: Add missing deps/hdrs to MAX98357A
by Kenneth Westfield 14 Feb '15

14 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> For the max98357a codec driver: - Add missing build dependancy to GPIOLIB in Kconfig. - Add header files to avoid implicit declarations and indirect inclusions. - Remove use of DRV_NAME constant. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- for topic/max98357a branch sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98357a.c | 23 +++++++++++++++-------- 2 files changed, 16 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6ecac1e4428eafc491d6cfc25110f6b3dbf699ed..0500b3ed5fa5ac4d45663ad659da9db3003bc5a3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333ea29e5a6b49e2a0adb8c58257d34b7508..4ee23fbc4e1237268ae104319a9cfe4fc369934e 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,11 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include <linux/module.h> +#include <linux/device.h> +#include <linux/err.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> #include <sound/soc.h> - -#define DRV_NAME "max98357a" +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -77,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "max98357a", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "max98357a-playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -117,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -125,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -135,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: Add missing error check for devm_kzalloc
by Kiran Padwal 14 Feb '15

14 Feb '15

This patch add a missing check on the return value of devm_kzalloc, which would cause a NULL pointer dereference in a OOM situation. Signed-off-by: Kiran Padwal <kiran.padwal(a)smartplayin.com> --- sound/soc/omap/omap-hdmi-audio.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index ccfb41c..f7eb42a 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) return ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + card->name = devm_kasprintf(dev, GFP_KERNEL, "HDMI %s", dev_name(ad->dssdev)); card->owner = THIS_MODULE; -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Add support for CBS_CFM mode
by Peter Ujfalusi 14 Feb '15

14 Feb '15

Support for setups where codec is bitclock slave and frame master. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 750124030576..c0e485cba9eb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -441,6 +441,18 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); mcasp->bclk_master = 1; break; + case SND_SOC_DAIFMT_CBS_CFM: + /* codec is clock slave and frame master */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 1; + break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); -- 2.3.0

2 1

[alsa-devel] [PATCH 0/4] ASoC: samsung: clean up references to outdated Kconfig symbols
by Paul Bolle 14 Feb '15

14 Feb '15

A few series of ARM cleanups regarding S5PC100 and S5PV210 were included in v3.17. These series removed these five Kconfig symbols: MACH_AQUILA MACH_GONI MACH_SMDKC100 MACH_SMDKC110 MACH_SMDKV210 This series, done on top of next-20150119, removes the remaining references to those five symbols. (The first two patches are actually resends.) Paul Bolle (4): ASoC: samsung: Remove goni or aquila with the WM8994 ASoC: samsung: Remove PCM support for WM8580 on SMDK ASoC: samsung: MACH_SMDKC100 isn't supported anymore ASoC: samsung: remove outdated Kconfig references sound/soc/samsung/Kconfig | 24 +-- sound/soc/samsung/Makefile | 4 - sound/soc/samsung/goni_wm8994.c | 289 ------------------------------------- sound/soc/samsung/smdk_wm8580pcm.c | 176 ---------------------- 4 files changed, 2 insertions(+), 491 deletions(-) delete mode 100644 sound/soc/samsung/goni_wm8994.c delete mode 100644 sound/soc/samsung/smdk_wm8580pcm.c -- 1.9.3

2 5

[alsa-devel] sound/pci/hda/hda_tegra.c
by Eliot Blennerhassett 13 Feb '15

13 Feb '15

Browsing code I came across this... sound/pci/hda/hda_tegra.c static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) ... res = platform_get_resource(pdev, IORESOURCE_MEM, 0); hda->regs = devm_ioremap_resource(dev, res); Checking chip->remap_addr before assignment looks wierd to me, am I missing something? Should it be 'hda->regs' here? if (IS_ERR(chip->remap_addr)) return PTR_ERR(chip->remap_addr); chip->remap_addr = hda->regs + HDA_BAR0; chip->addr = res->start + HDA_BAR0;

2 1

[alsa-devel] [PATCH] ALSA: hda - enable mute led quirk for one more hp machine.
by Hui Wang 13 Feb '15

13 Feb '15

Otherwise, the mute led can't work at all. Tested-by: Taihsiang Ho <taihsiang.ho(a)canonical.com> Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1410704 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- 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 ddb9308..b2b24a8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4937,6 +4937,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY), /* ALC282 */ + SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2236, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), -- 1.9.1

2 1

[alsa-devel] [PATCH 0/5] ALSA: control: improve core codes with intrusion
by Takashi Sakamoto 13 Feb '15

13 Feb '15

This patchset is for improving core control codes. Most of them have any effects for processing, messaging or status codes. This patchset is based on my previous one: [alsa-devel] [PATCH 0/8] ALSA: control: refactoring core codes http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087783.h… Takashi Sakamoto (5): ALSA: control: drop a sub-effect for caller's data ALSA: control: use dev_dbg() for warning to add duplicated controls ALSA: control: fix logic error about control count in a device ALSA: control: improve returned value because of the rest ALSA: control: save stack usage at creating new instance sound/core/control.c | 209 ++++++++++++++++++++++++++++----------------------- 1 file changed, 116 insertions(+), 93 deletions(-) -- 2.1.0

2 11

[alsa-devel] [PATCH 0/4 v3] ASoC: rsnd: tidyup .remove method
by Kuninori Morimoto 13 Feb '15

13 Feb '15

Hi Mark These are v3 of unbind/bind issue fixup patches. v2 patch-set was Kuninori Morimoto (6): 1) ASoC: rsnd: set device data before snd_soc_register_platform/component 2) ASoC: soc-core: indicate unregister debug message once 3) ASoC: soc-core: add snd_soc_remove_card() 4) ASoC: soc-core: deactivate pins in snd_soc_instantiate_card() 5) ASoC: soc-core: call soc_cleanup_card_debugfs() from snd_soc_unregister_card() 6) ASoC: soc-core: call snd_soc_remove_card() when component del Now, 1) / 2) are applied already. So, v3 patches is same as 3) - 6)

3 13

[alsa-devel] davinci-mcasp: extension to use AHCLKX pin as external clock source
by Urs Fässler 12 Feb '15

12 Feb '15

The AHCLKX pin seems not to be supported as external source. The first patch is a general fix to allow external clock. The second allows you to select the AHCLKX pin as clock source. Patch built against v3.19-rc7

2 4

[alsa-devel] Status of ALSA FireWire stack as of Linux 3.19
by Takashi Sakamoto 12 Feb '15

12 Feb '15

Dear all, I'm just a developer, not a maintainer of this stack, but often receive personal messages from users. I think it helpful to explain about its status and known issues here, as of Linux 3.19. == Impovements in 3.19 release * ALSA FireWire stack gain a thlottle for MIDI data rate. No devices can disorder due to MIDI buffer overflow. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084652.h… * ALSA Dice driver becomes to handle more models with 176.4/192.0 kHz support and both of playback/capture for PCM/MIDI substreams. * ALSA firewire-speakers driver was obsoleted by ALSA OXFW driver to handles more models based on OXFW 970/971. * As a result, ALSA supports approximately 140 FireWire models in comparison to Linux 3.15. == Model dependent issues There're some reports about model dependent issues. * PreSonus FP10 (former known as FirePod) * When ALSA BeBoB driver re-establishes connections, this model causes hung-up, then reset automatically. The cause is unknown. * Terratek PHASE 88 Rack FW * https://github.com/takaswie/snd-firewire-improve/issues/8 * According to a reporter, when ALSA BeBoB driver establishes connections according to CMP, this model transfers no packets. The cause is unknown. * Behringer XENIX UFX 1604 * http://sourceforge.net/p/ffado/mailman/message/32846563/ * When ALSA BeBoB driver communicates, this model sometimes misses to transfer responses. The cause is unknown but I guess that this model expects vendor-specific command. * Echo AudioFire 2 * http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080379.html * At least, firmware version 5.7.3 for this model has a quirk about AMDTP processing. It's not compliant to this protocol. Even if ALSA Fireworks driver communicates successfully, ALSA FireWire stack detects protocol error. * Weiss engineering DAC 202 * According to a reporter, when ALSA Dice driver probes this model it's trials are ETIMEDOUT at setting sampling rate. The cause is unknown. == Stack and driver issues There're still some issues about ALSA FireWire stack and drivers. Most of them are written in my report: https://github.com/takaswie/alsa-firewire-report * 9.2 A lack of arrangement for severe latency * 9.3 A lack of delay for timestamp processing * 9.4 A lack of PCM configuration for virtual surround nodes * 9.6 A lack of control elements for ALSA control interface * 9.7 A lack of packet sorting * 9.9 Various results at bus reset * 9.10 A lack of synchronizing several devices on the same bus * Sometimes kernel oops at disconnecting devices during streaming. * I guess three contexts have somewhat race condition: * Application process with snd_pcm_prepare() * Isochronous context * .remove callback by OHCI 1394 host controller driver == The other issues * PulseAudio 6,0 (not released yet) gain multichannel profiles. http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/6.0/ ALSA FireWire sound devices can be available, while in most case the display name is the name of OHCI 3194 host controller. https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1381475 I've posted RFC patch to pulseaudio-discuss but I think it better to touch hardware database which systemd produces. http://lists.freedesktop.org/archives/pulseaudio-discuss/2015-January/02294… == My plan for 3.21 release * Some userspace library is under development for developers of mixer application. * libhinawa http://mailman.alsa-project.org/pipermail/alsa-devel/2015-January/086969.ht… * alsa-gir https://github.com/takaswie/alsa-gir * An offload of control functionality from ALSA OXFW driver to userspace * Currently ALSA OXFW driver includes control functionality in kernel side, while this can be implemented in userspace. This functionaliy utilizes certailn commands which the other drivers can use for the same purpose. * Need to discuss about it, later. * ALSA BeBoB driver improvement * ALSA BeBoB driver's bootup command for M-Audio models depends on little-endiannness machine. I need more time to prepare for big-endianness machine to test my patch for this issue. * Some models expect BridgeCo.-specific command to change sampling rate. Currently ALSA BeBoB driver uses commands in AV/C general specification. Let's start to replace these comands. Regards Takashi Sakamoto

1 0

[alsa-devel] [Patch V2] ASoC: max98357a: Add missing deps and headers
by Kenneth Westfield 12 Feb '15

12 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Add missing build dependancy to GPIOLIB in Kconfig. Add header files to avoid implicit declarations and indirect inclusions. Remove use of DRV_NAME constant. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- for topic/max98357a branch changes from v1: - matched subject line to subsystem style sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98357a.c | 23 +++++++++++++++-------- 2 files changed, 16 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6ecac1e4428eafc491d6cfc25110f6b3dbf699ed..0500b3ed5fa5ac4d45663ad659da9db3003bc5a3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333ea29e5a6b49e2a0adb8c58257d34b7508..4ee23fbc4e1237268ae104319a9cfe4fc369934e 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,11 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include <linux/module.h> +#include <linux/device.h> +#include <linux/err.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> #include <sound/soc.h> - -#define DRV_NAME "max98357a" +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -77,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "max98357a", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "max98357a-playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -117,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -125,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -135,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [Patch V5 00/12] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 12 Feb '15

12 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This patch series adds support for I2S audio playback on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This contains an MI2S port, which is what these drivers are configured to use. The LPAIF also contains a DMA engine that is dedicated to moving audio samples into the transmit FIFO of the MI2S port. The development board being used for testing contains the ipq806x SOC and a Maxim max98357a DAC/amp. One bus from the MI2S port of the SOC is connected to the DAC/amp for stereo playback. This bus is configured so that the SOC is bus master and consists of DATA, LRCLK, and BCLK. The DAC/amp does not need MCLK to operate. In addition, a single GPIO pin from the SOC is connected to the same DAC/amp, which gives enable/disable control over the DAC/amp. The specific drivers added are: * a codec DAI driver for controlling the DAC/amp * a CPU DAI driver for controlling the MI2S port * a platform driver for controlling the LPAIF DMA engine * a machine driver that instantiates a dai-link for playback Corresponding additions to the device tree for the ipq806x SOC and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. The LPASS also contains clocks that need to be controlled. Those drivers have been submitted as a separate patch series: [PATCH v3 0/8] qcom audio clock control drivers http://lkml.org/lkml/2015/1/19/656 = Changes since V4 [Patch V4 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087499.h… * Replaced simple-card with a machine driver to resolve the system clock configuration, rather than having the CPU DAI driver do it. * Added header files to avoid indirect header dependencies and implicit forward declarations. * Tweaked the ISR to match the conventions of the surrounding code. * Removed the usage of the low-power memory as it is not needed. * Removed the use of the DRV_NAME constant. * Added explicit dependancy on gpiolib for the codec driver. * Moved the MODULE_DEVICE_TABLE macro inside the CONFIG_OF conditional. * Modified the documentation to reflect the changes. * General code cleanup. = Changes since V3 [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085694.h… * Placed the content of the inline functions into the callbacks. * Replaced use of readl/writel register access functions with regmap access functions. Notable exception is the ISR, which uses ioread32/iowrite32. * Rearranged the sequencing of the hardware block enables to fit within the ASoC framework callbacks, while remaining functional. REQ 1: The hardware requires the enable sequence to be: LPAIF-DMA[enable],then LPAIF-MI2S[enable], then DAC-GPIO[enable] REQ 2: The hardware requires the disable sequence to be: DAC-GPIO[disable], then LPAIF-MI2S[disable] * Corrected the implementation of the pointer callback. * Utilize the LPM to buffer audio samples, rather than memory external to LPASS. * Corrected the interrupt clearing in the ISR. * Implemented a default system clock (defined by the simple-card DT node), and optional LPASS DT node modifiers that can alter the system clock in order to expand the range of available bit clock frequencies. * Addressed all of the remaining issues raised by Mark Brown. * General code cleanup. = Changes since V2 [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085186.h… * Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. * Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. * Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. * Removed machine driver and tied DAI drivers to simple-audio-card. * Packaged the build files and Maxim codec files together in one change. * Removed QCOM as vendor from Maxim code and documentation. * Separated the SOC and board definitions into the correct DTS files. * Update device tree documentation to reflect changes. * General code cleanup. = Changes since V1 [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084322.h… * Remove the native LPAIF driver, and move its functionality to the CPU DAI driver. * Add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver). * Use devm_* and dev_* where possible. * ISR only handles relevant DMA channel now. * Update device tree documentation to reflect changes. * General code cleanup. Kenneth Westfield (12): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: max98357a: Document MAX98357A bindings ASoC: qcom: Document LPASS CPU bindings ASoC: qcom: Document IPQ806X-MAX89357A bindings ASoC: codec: Add MAX98357A codec driver ASoC: ipq806x: add LPASS header files ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add LPASS platform driver ASoC: ipq806x: Add @max98357a machine driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../devicetree/bindings/sound/max98357a.txt | 14 + .../bindings/sound/qcom,ipq806x-max98357a.txt | 24 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 38 ++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 23 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 145 ++++++ sound/soc/qcom/Kconfig | 26 + sound/soc/qcom/Makefile | 11 + sound/soc/qcom/ipq806x-max98357a.c | 171 +++++++ sound/soc/qcom/lpass-cpu.c | 481 +++++++++++++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 172 +++++++ sound/soc/qcom/lpass-platform.c | 526 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 51 ++ 17 files changed, 1697 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,ipq806x-max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/ipq806x-max98357a.c create mode 100644 sound/soc/qcom/lpass-cpu.c create mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-platform.c create mode 100644 sound/soc/qcom/lpass.h -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 14

[alsa-devel] [PATCH] ASoC: max98357a: Fix missing include
by Vincent Stehlé 12 Feb '15

12 Feb '15

This fixes the following compilation errors: sound/soc/codecs/max98357a.c: In function ‘max98357a_daiops_trigger’: sound/soc/codecs/max98357a.c:30:3: error: implicit declaration of function ‘gpiod_set_value’ [-Werror=implicit-function-declaration] sound/soc/codecs/max98357a.c: In function ‘max98357a_codec_probe’: sound/soc/codecs/max98357a.c:55:2: error: implicit declaration of function ‘devm_gpiod_get’ [-Werror=implicit-function-declaration] sound/soc/codecs/max98357a.c:61:2: error: implicit declaration of function ‘gpiod_direction_output’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors Signed-off-by: Vincent Stehlé <vincent.stehle(a)laposte.net> Cc: Kenneth Westfield <kwestfie(a)codeaurora.org> Cc: Mark Brown <broonie(a)kernel.org> --- Hi, Those compilation errors can be seen for example with hppa allmodconfig. Best regards, V. sound/soc/codecs/max98357a.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333..f493fb6 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/soc.h> #define DRV_NAME "max98357a" -- 2.1.4

2 1

[alsa-devel] [PATCH 0/3] ucm: miscellaneous fixes
by Tanu Kaskinen 11 Feb '15

11 Feb '15

While working on PulseAudio's UCM code, I noticed strange use of const in snd_use_case_get(). The third patch fixes that. While working on the third patch, I also noticed strange code in execute_sequence(). The first patch fixes one obvious bug, but there was another issue which I didn't know how to fix, so I just added a FIXME comment instead (that's just as an RFC, I hope the discussion will reveal what the proper fix should be). Tanu Kaskinen (3): ucm: fix variable mixup ucm: add a FIXME comment ucm: fix inappropriate use of const include/use-case.h | 2 +- src/ucm/main.c | 34 +++++++++++++++++++++------------- 2 files changed, 22 insertions(+), 14 deletions(-) -- 1.9.3

3 11

[alsa-devel] [patch] ALSA: seq: potential out of bounds in do_control()
by Dan Carpenter 11 Feb '15

11 Feb '15

Smatch complains that "control" is user specifigy and needs to be capped. The call tree to understand this warning is quite long. snd_seq_write() <-- get the event from the user snd_seq_client_enqueue_event() snd_seq_deliver_event() deliver_to_subscribers() snd_seq_deliver_single_event() snd_opl3_oss_event_input() snd_midi_process_event() do_control() Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- I have spent some time reviewing this code, but I may have missed something where we verify that control is in bounds. I'm not very familiar with this code and the call tree is fairly long. diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 9b6470c..7ba9373 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -269,6 +269,9 @@ do_control(struct snd_midi_op *ops, void *drv, struct snd_midi_channel_set *chse { int i; + if (control >= ARRAY_SIZE(chan->control)) + return; + /* Switches */ if ((control >=64 && control <=69) || (control >= 80 && control <= 83)) { /* These are all switches; either off or on so set to 0 or 127 */

3 3

[alsa-devel] [PATCH v2] ALSA: line6: Improve line6_read/write_data() interfaces
by Chris Rorvick 11 Feb '15

11 Feb '15

The address cannot be negative so make it unsigned. Also, an unsigned int is always sufficient for the length, so no need to overdo it with a size_t. Finally, add in range checks to see if the values passed in actually fit where they are used. Signed-off-by: Chris Rorvick <chris(a)rorvick.com> --- This was dropped from an earlier series ("Cleanup reads/writes to Line 6 device memory".) This version leaves the arguements wide (within reason) and does range checking. Regards, Chris sound/usb/line6/driver.c | 14 ++++++++++---- sound/usb/line6/driver.h | 8 ++++---- 2 files changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 99b63a7..4f82b57 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -302,14 +302,17 @@ static void line6_data_received(struct urb *urb) /* Read data from device. */ -int line6_read_data(struct usb_line6 *line6, int address, void *data, - size_t datalen) +int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, + unsigned datalen) { struct usb_device *usbdev = line6->usbdev; int ret; unsigned char len; unsigned count; + if (address > 0xffff || datalen > 0xff) + return EINVAL; + /* query the serial number: */ ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, @@ -370,14 +373,17 @@ EXPORT_SYMBOL_GPL(line6_read_data); /* Write data to device. */ -int line6_write_data(struct usb_line6 *line6, int address, void *data, - size_t datalen) +int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, + unsigned datalen) { struct usb_device *usbdev = line6->usbdev; int ret; unsigned char status; int count; + if (address > 0xffff || datalen > 0xffff) + return EINVAL; + ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, 0x0022, address, data, datalen, diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 5d20294..7da643e 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -147,8 +147,8 @@ struct usb_line6 { extern char *line6_alloc_sysex_buffer(struct usb_line6 *line6, int code1, int code2, int size); -extern int line6_read_data(struct usb_line6 *line6, int address, void *data, - size_t datalen); +extern int line6_read_data(struct usb_line6 *line6, unsigned address, + void *data, unsigned datalen); extern int line6_read_serial_number(struct usb_line6 *line6, u32 *serial_number); extern int line6_send_raw_message_async(struct usb_line6 *line6, @@ -161,8 +161,8 @@ extern void line6_start_timer(struct timer_list *timer, unsigned long msecs, void (*function)(unsigned long), unsigned long data); extern int line6_version_request_async(struct usb_line6 *line6); -extern int line6_write_data(struct usb_line6 *line6, int address, void *data, - size_t datalen); +extern int line6_write_data(struct usb_line6 *line6, unsigned address, + void *data, unsigned datalen); int line6_probe(struct usb_interface *interface, const struct usb_device_id *id, -- 2.1.0

2 2

[alsa-devel] Memory and Cycles overhead in <snd_soc_init>
by Bhaskara rao Budiredla 11 Feb '15

11 Feb '15

Hi - May I know the purpose of the call to snd_soc_init( ) from kernel_init thread? snd_soc_init( ) is trying to register a dummy platform device "snd-soc-dummy" containing (dummy_codec/dummy_platform) and a sound card device. Neither of those devices are added to platform bus before they are making a call to platform_driver_register( ). Due to that platform_match( ) will fail and they will not be added to platform bus . Thus, snd_soc_init( ) is not doing any useful task during kernel boot time. Could someone help me if my understanding is correct? If I am correct, shall we remove the call to snd_soc_init( ) in kernel_init thread to save boot time and memory? Otherwise please suggest me where those dummy devices and sound card gets added to platform bus and the purpose of creating them? Thanks, Bhaskara

2 3

[alsa-devel] [PATCH 0/6] Cleanup reads/writes to Line 6 device memory
by Chris Rorvick 11 Feb '15

11 Feb '15

The primary fix here is to throttle the requests for status after a read or write operation. In some cases I was seeing more than 1000 before the device was ready with a response. The remainder are smaller improvements related to the read/write logic. All of this seems pretty straightforward with the possible exception of changing the returned error from EINVAL to EIO. The EINVAL seems wrong to me but maybe not? Regards, Chris Chris Rorvick (6): ALSA: line6: Improve line6_read/write_data() interfaces ALSA: line6: Add delay before reading status ALSA: line6: Return error if device not responding ALSA: line6: Return EIO if read/write not successful ALSA: line6: Use explicit type for serial number ALSA: line6: toneport: Use explicit type for firmware version sound/usb/line6/driver.c | 51 +++++++++++++++++++++++++++++++++------------- sound/usb/line6/driver.h | 10 ++++----- sound/usb/line6/pod.c | 4 ++-- sound/usb/line6/toneport.c | 4 ++-- 4 files changed, 46 insertions(+), 23 deletions(-) -- 2.1.0

2 13

[alsa-devel] [PATCH v3 kernel 1/3] snd_pcm_start_at and friends.
by Tim Cussins 11 Feb '15

11 Feb '15

We introduce the kernel-side of the START_AT ioctl. struct runtime is updated to hold information about the currently active start_at timer, if any. This facilitates cancellation via snd_pcm_start_at_abort(), and querying via snd_pcm_status(). struct snd_start_at holds a startat operation and its arguments. Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 0e88e7a..2943e1a 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -421,7 +421,10 @@ struct snd_pcm_status { snd_pcm_state_t suspended_state; /* suspended stream state */ __u32 reserved_alignment; /* must be filled with zero */ struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + int startat_pending; /* 1 if a start_at timer is pending, 0 otherwise */ + int startat_clock_type; /* start_at clock type, if pending */ + struct timespec startat_start_time; /* start_at start time, if pending */ + unsigned char reserved[48-(2*sizeof(struct timespec))]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -473,6 +476,34 @@ enum { SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; +enum { + SNDRV_PCM_STARTAT_OP_SET = 0, + SNDRV_PCM_STARTAT_OP_CANCEL, + SNDRV_PCM_STARTAT_OP_STATUS, + SNDRV_PCM_STARTAT_OP_LAST = SNDRV_PCM_STARTAT_OP_STATUS, +}; + +enum { + SNDRV_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY = 0, + SNDRV_PCM_STARTAT_CLOCK_TYPE_MONOTONIC, + SNDRV_PCM_STARTAT_CLOCK_TYPE_LINK, + SNDRV_PCM_STARTAT_CLOCK_TYPE_LAST = SNDRV_PCM_STARTAT_CLOCK_TYPE_LINK, +}; + +struct snd_start_at { + int op; /* startat operation to be performed */ + union { /* fields for setting a startat timer */ + struct { + int clock_type; /* clock type e.g. SNDRV_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY */ + struct timespec start_time; /* start time */ + } set; + struct { + int clock_type; + struct timespec current_time; + } status; + } args; +}; + /* channel positions */ enum { SNDRV_CHMAP_UNKNOWN = 0, @@ -551,6 +582,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct snd_start_at) + /***************************************************************************** * * -- 1.9.1

3 10

[alsa-devel] [PATCH] ASoC: Intel: Clean data after SST fw fetch
by libin.yang＠intel.com 11 Feb '15

11 Feb '15

From: Libin Yang <libin.yang(a)intel.com> The BDW audio firmware DSP manages the DMA and the DMA cannot be stopped exactly at the end of the playback stream. This means stale samples may be played at PCM stop unless the driver copies silence to the subsequent periods. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 28 +++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 9 ++++++ sound/soc/intel/sst-haswell-pcm.c | 65 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 102 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 0ab1309..394af56 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -31,6 +31,7 @@ #include <linux/dma-mapping.h> #include <linux/debugfs.h> #include <linux/pm_runtime.h> +#include <sound/asound.h> #include "sst-haswell-ipc.h" #include "sst-dsp.h" @@ -242,6 +243,9 @@ struct sst_hsw_stream { u32 (*notify_position)(struct sst_hsw_stream *stream, void *data); void *pdata; + /* record the fw read position when playback */ + snd_pcm_uframes_t old_position; + bool play_silence; struct list_head node; }; @@ -1347,6 +1351,30 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) return 0; } +snd_pcm_uframes_t sst_hsw_stream_get_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->old_position; +} + +void sst_hsw_stream_set_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, snd_pcm_uframes_t val) +{ + stream->old_position = val; +} + +bool sst_hsw_stream_get_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->play_silence; +} + +void sst_hsw_stream_set_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, bool val) +{ + stream->play_silence = val; +} + /* Stream Information - these calls could be inline but we want the IPC ABI to be opaque to client PCM drivers to cope with any future ABI changes */ int sst_hsw_mixer_get_info(struct sst_hsw *hsw) diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index c1ad901..8580960 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -20,6 +20,7 @@ #include <linux/types.h> #include <linux/kernel.h> #include <linux/platform_device.h> +#include <sound/asound.h> #define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 @@ -425,6 +426,14 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); +snd_pcm_uframes_t sst_hsw_stream_get_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +void sst_hsw_stream_set_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, snd_pcm_uframes_t val); +bool sst_hsw_stream_get_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +void sst_hsw_stream_set_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, bool val); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 78fa01b..d6fa9d5 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -36,6 +36,11 @@ #define HSW_PCM_COUNT 6 #define HSW_VOLUME_MAX 0x7FFFFFFF /* 0dB */ +#define SST_OLD_POSITION(d, r, o) ((d) + \ + frames_to_bytes(r, o)) +#define SST_SAMPLES(r, x) (bytes_to_samples(r, \ + frames_to_bytes(r, (x)))) + /* simple volume table */ static const u32 volume_map[] = { HSW_VOLUME_MAX >> 30, @@ -577,23 +582,34 @@ static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); struct hsw_pcm_data *pcm_data; + struct sst_hsw_stream *sst_stream; struct sst_hsw *hsw = pdata->hsw; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; int dai; dai = mod_map[rtd->cpu_dai->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; + sst_stream = pcm_data->stream; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_hsw_stream_set_silence_start(hsw, sst_stream, false); sst_hsw_stream_resume(hsw, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_hsw_stream_set_silence_start(hsw, sst_stream, false); sst_hsw_stream_pause(hsw, pcm_data->stream, 0); break; + case SNDRV_PCM_TRIGGER_DRAIN: + pos = runtime->control->appl_ptr % runtime->buffer_size; + sst_hsw_stream_set_old_position(hsw, pcm_data->stream, pos); + sst_hsw_stream_set_silence_start(hsw, sst_stream, true); + break; default: break; } @@ -607,13 +623,62 @@ static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data) struct snd_pcm_substream *substream = pcm_data->substream; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_hsw *hsw = pdata->hsw; u32 pos; + snd_pcm_uframes_t position = bytes_to_frames(runtime, + sst_hsw_get_dsp_position(hsw, pcm_data->stream)); + unsigned char *dma_area = runtime->dma_area; + snd_pcm_uframes_t dma_frames = + bytes_to_frames(runtime, runtime->dma_bytes); + snd_pcm_uframes_t old_position; + ssize_t samples; pos = frames_to_bytes(runtime, (runtime->control->appl_ptr % runtime->buffer_size)); dev_vdbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + /* SST fw don't know where to stop dma + * So, SST driver need to clean the data which has been consumed + */ + if (dma_area == NULL || dma_frames <= 0 + || (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + || !sst_hsw_stream_get_silence_start(hsw, stream)) { + snd_pcm_period_elapsed(substream); + return pos; + } + + old_position = sst_hsw_stream_get_old_position(hsw, stream); + if (position > old_position) { + if (position < dma_frames) { + samples = SST_SAMPLES(runtime, position - old_position); + snd_pcm_format_set_silence(runtime->format, + SST_OLD_POSITION(dma_area, + runtime, old_position), + samples); + } else + dev_err(rtd->dev, "PCM: position is wrong\n"); + } else { + if (old_position < dma_frames) { + samples = SST_SAMPLES(runtime, + dma_frames - old_position); + snd_pcm_format_set_silence(runtime->format, + SST_OLD_POSITION(dma_area, + runtime, old_position), + samples); + } else + dev_err(rtd->dev, "PCM: dma_bytes is wrong\n"); + if (position < dma_frames) { + samples = SST_SAMPLES(runtime, position); + snd_pcm_format_set_silence(runtime->format, + dma_area, samples); + } else + dev_err(rtd->dev, "PCM: position is wrong\n"); + } + sst_hsw_stream_set_old_position(hsw, stream, position); + /* let alsa know we have play a period */ snd_pcm_period_elapsed(substream); return pos; -- 1.9.1

4 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix the incorrect limitation of the bit clock rate
by Nicolin Chen 11 Feb '15

11 Feb '15

According to i.MX Reference Manual, the bit-clock frequency generated by SSI must be never greater than 1/5 of the peripheral clock frequency. This peripheral clock, however, is not baudclk but the IPG clock (i.e. ssi_private->clk in the fsl_ssi driver). So this patch just simply fixes the incorrect limitation applied to the bit clock (baudclk) rate. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 059496e..d7365c5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - /* The bclk rate must be smaller than 1/5 sysclk rate */ - if (factor * (i + 1) < 5) - continue; - tmprate = freq * factor * (i + 2); if (baudclk_is_used) @@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + /* + * Hardware limitation: The bclk rate must be + * never greater than 1/5 IPG clock rate + */ + if (clkrate * 5 > clk_get_rate(ssi_private->clk)) + continue; + clkrate /= factor; afreq = clkrate / (i + 1); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Add the chip type to distinguish the setting of the clock source
by Oder Chiou 11 Feb '15

11 Feb '15

There is only one clock source in the rt5676, so the chip type is added to distinguish the setting of the clock source in the VAD function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 23 +++++++++++++++++------ sound/soc/codecs/rt5677.h | 6 ++++++ 2 files changed, 23 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb87..3aae8a4 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -718,11 +718,19 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, - RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, - RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); + if (rt5677->type == RT5677) { + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | + RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | + RT5677_DSP_CLK_SRC_BYPASS); + } else { + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_DSP_CLK_SRC_MASK, + RT5677_DSP_CLK_SRC_BYPASS); + } regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); rt5677_set_dsp_mode(codec, true); @@ -4733,7 +4741,8 @@ static const struct regmap_config rt5677_regmap = { }; static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", 0 }, + { "rt5677", RT5677 }, + { "rt5676", RT5676 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -4850,6 +4859,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5677); + rt5677->type = id->driver_data; + if (pdata) rt5677->pdata = *pdata; diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index c0a625f..07df96b 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1665,6 +1665,11 @@ enum { RT5677_IRQ_JD3, }; +enum rt5677_type { + RT5677, + RT5676, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1681,6 +1686,7 @@ struct rt5677_priv { int pll_in; int pll_out; int pow_ldo2; /* POW_LDO2 pin */ + enum rt5677_type type; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [GIT PULL]: Add sst audio firmware for Braswell platforms
by Vinod Koul 11 Feb '15

11 Feb '15

Hello Linux firmware maintainers, Please PULL to receive the Intel audio firmware for audio DSP in Braswell platforms. This firmware is using the same license as the Baytrail firmware already available in linux-firmware git. The following changes since commit b794c5039dcf0b7ebfeb58929d035f7a1d4c80dd: Merge branch 'marvell-mwifiex' (2015-01-27 10:42:47 -0500) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git BSW for you to fetch changes up to f692aba54c8c80cb27a3eb5388594a55a30504ea: linux-firmware: add sst audio firmware for braswell platforms (2015-02-02 21:12:22 -0800) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: add sst audio firmware for braswell platforms WHENCE | 8 ++++++++ intel/fw_sst_22a8.bin | Bin 0 -> 647390 bytes 2 files changed, 8 insertions(+) create mode 100644 intel/fw_sst_22a8.bin Thanks -- ~Vinod

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[alsa-devel] Crackly playback, Roland AIRA TR-8 USB audio almost works with ALSA . . .
by David W. 10 Feb '15

10 Feb '15

Dear ALSA developers, On 30/12/14 19:39, David W. wrote: > On 15/12/14 23:37, David W. wrote: >> On 07/07/14 07:31, Clemens Ladisch wrote: >>> >>> Apparently, this device needs some vendor-specific magic to enable >>> recording. >>> This AIRA TR-8 USB audio device, which fails with an 'input/output error' after ~10 seconds for 'aplay' and 'arecord', seems to require operation in implicit feedback mode: the following patch suggested by Daniel Mack solves the 'input/output' error for 'aplay', but the playback is noisy and corrupted and the kernel complains of 'rogue URB xfers' (see below). Given this information and the quote below from lsusb etc., I am hoping someone would be able to suggest a further patch to get playback and recording working :-) I can patch, compile and test recent linux kernels (whatever is in: https://www.archlinux.org/packages/?sort=&repo=Testing&q=linux) Here's the patch from Daniel which helped: diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0d8aba5..e35575b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -346,6 +346,15 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x0582, 0x017c): + ep = 0x85; + iface = usb_ifnum_to_if(dev, 3); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && Here's an example of the kind of complaint that the 3.18.2-2-ARCH kernel, with the above patch applied, reports: Jan 12 23:18:20 dwmobile kernel: WARNING: CPU: 2 PID: 1889 at drivers/usb/core/urb.c:450 usb_submit_urb+0x265/0x5f0 [usbcore]() Jan 12 23:18:20 dwmobile kernel: usb 2-1.1: BOGUS urb xfer, pipe 0 != type 1 Jan 12 23:18:20 dwmobile kernel: Modules linked in: snd_usb_audio snd_usbmidi_lib snd_rawmidi snd_seq_device ctr ccm sha256_ssse3 sha256_generic d Jan 12 23:18:20 dwmobile kernel: snd_pcm snd_timer video mei_me battery ac snd intel_agp button intel_gtt acpi_cpufreq shpchp mei soundcore proce Jan 12 23:18:20 dwmobile kernel: CPU: 2 PID: 1889 Comm: aplay Not tainted 3.18.2-2-custom #1 Jan 12 23:18:20 dwmobile kernel: Hardware name: System76, Inc. Pangolin Performance /W76x/M77xCUH Jan 12 23:18:20 dwmobile kernel: 0000000000000000 000000005911a000 ffff88022988faf8 ffffffff8154f134 Jan 12 23:18:20 dwmobile kernel: 0000000000000000 ffff88022988fb50 ffff88022988fb38 ffffffff81072bc1 Jan 12 23:18:20 dwmobile kernel: ffff88022988fb28 ffff88022f1aba00 0000000000000020 0000000000000003 Jan 12 23:18:20 dwmobile kernel: Call Trace: Jan 12 23:18:20 dwmobile kernel: [<ffffffff8154f134>] dump_stack+0x4e/0x71 Jan 12 23:18:20 dwmobile kernel: [<ffffffff81072bc1>] warn_slowpath_common+0x81/0xa0 Jan 12 23:18:20 dwmobile kernel: [<ffffffff81072c35>] warn_slowpath_fmt+0x55/0x70 Jan 12 23:18:20 dwmobile kernel: [<ffffffffa009b005>] ? usb_enable_endpoint+0x85/0x90 [usbcore] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0098e75>] usb_submit_urb+0x265/0x5f0 [usbcore] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa08d2c65>] snd_usb_endpoint_start+0x125/0x350 [snd_usb_audio] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa08dac89>] start_endpoints+0xb9/0x1a0 [snd_usb_audio] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa08dcbad>] snd_usb_pcm_prepare+0x17d/0x560 [snd_usb_audio] Jan 12 23:18:20 dwmobile kernel: [<ffffffff810d7af6>] ? current_fs_time+0x16/0x60 Jan 12 23:18:20 dwmobile kernel: [<ffffffffa028627b>] snd_pcm_do_prepare+0x1b/0x30 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0285e4f>] snd_pcm_action_single+0x2f/0x70 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0285f06>] snd_pcm_action_nonatomic+0x76/0x80 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0288c18>] snd_pcm_common_ioctl1+0x688/0xbc0 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0289268>] snd_pcm_playback_ioctl1+0x118/0x280 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0289404>] snd_pcm_playback_ioctl+0x34/0x40 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffff811e23e0>] do_vfs_ioctl+0x2d0/0x4b0 Jan 12 23:18:20 dwmobile kernel: [<ffffffff811cf2ac>] ? vfs_write+0x18c/0x200 Jan 12 23:18:20 dwmobile kernel: [<ffffffff811e2641>] SyS_ioctl+0x81/0xa0 Jan 12 23:18:20 dwmobile kernel: [<ffffffff81554ca9>] system_call_fastpath+0x12/0x17 Jan 12 23:18:20 dwmobile kernel: ---[ end trace e6531ceb4fd4f2e5 ]--- Jan 12 23:18:20 dwmobile kernel: ------------[ cut here ]------------ >> >> amidi -l >> >> Dir Device Name >> IO hw:0,0,0 USB X-Session MIDI 1 >> I hw:0,0,1 USB X-Session MIDI 2 >> IO hw:2,0,0 TR-8 MIDI 1 # this device >> IO hw:2,0,1 TR-8 MIDI 2 # this device >> >> >> ./alsacap -d hw:2,0 >> >> *** Exploring configuration space of device 'hw:1,0' for playback *** >> 4 channels >> Sampling rate 96000 Hz >> Sample formats: S32_LE >> Significant bits: 32 >> >> >> Bus 002 Device 005: ID 0582:017c Roland Corp. >> Device Descriptor: >> bLength 18 >> bDescriptorType 1 >> bcdUSB 2.00 >> bDeviceClass 255 Vendor Specific Class >> bDeviceSubClass 0 >> bDeviceProtocol 255 >> bMaxPacketSize0 64 >> idVendor 0x0582 Roland Corp. >> idProduct 0x017c >> bcdDevice 1.00 >> iManufacturer 1 Roland >> iProduct 2 TR-8 >> iSerial 0 >> bNumConfigurations 1 >> Configuration Descriptor: >> bLength 9 >> bDescriptorType 2 >> wTotalLength 176 >> bNumInterfaces 4 >> bConfigurationValue 1 >> iConfiguration 0 >> bmAttributes 0xc0 >> Self Powered >> MaxPower 0mA >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 0 >> bAlternateSetting 0 >> bNumEndpoints 0 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 255 Vendor Specific Subclass >> bInterfaceProtocol 0 >> iInterface 0 >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 1 >> bAlternateSetting 0 >> bNumEndpoints 0 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 2 >> bInterfaceProtocol 2 >> iInterface 0 >> # <06 24 F1 02 <inputs> <outputs>> from patch comment by Clemens Ladisch >> ** UNRECOGNIZED: 06 24 f1 01 00 00 >> # Audio OUT 4 channels? >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 1 >> bAlternateSetting 1 >> bNumEndpoints 1 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 2 >> bInterfaceProtocol 2 >> iInterface 0 >> # seen in other Roland devices >> # http://ubuntuforums.org/archive/index.php/t-1905531.html >> ** UNRECOGNIZED: 07 24 01 01 00 01 00 >> ** UNRECOGNIZED: 0b 24 02 01 04 04 18 01 00 77 01 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x0d EP 13 OUT >> bmAttributes 5 >> Transfer Type Isochronous # audio >> Synch Type Asynchronous >> Usage Type Data >> # 4x 24-bit 96kHz == 56 bytes per 4 channels (or 32-bit?) >> wMaxPacketSize 0x00e0 1x 224 bytes >> bInterval 1 >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 2 >> bAlternateSetting 0 >> bNumEndpoints 0 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 2 >> bInterfaceProtocol 1 >> iInterface 0 >> # Audio IN: 14 channels? >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 2 >> bAlternateSetting 1 >> bNumEndpoints 1 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 2 >> bInterfaceProtocol 1 >> iInterface 0 >> ** UNRECOGNIZED: 07 24 01 07 00 01 00 >> ** UNRECOGNIZED: 0b 24 02 01 0e 04 18 01 00 77 01 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x8e EP 14 IN >> bmAttributes 37 >> Transfer Type Isochronous >> Synch Type Asynchronous >> Usage Type Implicit feedback Data >> # 14 channels x 56 bytes per transfer 96kHz, 24-bit? >> wMaxPacketSize 0x0310 1x 784 bytes >> bInterval 1 >> # MIDI? >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 3 >> bAlternateSetting 0 >> bNumEndpoints 2 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 3 >> bInterfaceProtocol 0 >> iInterface 0 >> # <06 24 F1 02 <inputs> <outputs>> 2 in 2 out >> ** UNRECOGNIZED: 06 24 f1 02 02 02 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x03 EP 3 OUT >> bmAttributes 2 >> Transfer Type Bulk >> Synch Type None >> Usage Type Data >> wMaxPacketSize 0x0200 1x 512 bytes >> bInterval 1 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x84 EP 4 IN >> bmAttributes 2 >> Transfer Type Bulk >> Synch Type None >> Usage Type Data >> wMaxPacketSize 0x0200 1x 512 bytes >> bInterval 0 >> # MIDI: same endpoints needs Bulk AND Interrupt packet types? >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 3 >> bAlternateSetting 1 >> bNumEndpoints 2 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 3 >> bInterfaceProtocol 0 >> iInterface 0 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x03 EP 3 OUT >> bmAttributes 3 >> Transfer Type Interrupt >> Synch Type None >> Usage Type Data >> wMaxPacketSize 0x0200 1x 512 bytes >> bInterval 1 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x85 EP 5 IN >> bmAttributes 3 >> Transfer Type Interrupt >> Synch Type None >> Usage Type Data >> wMaxPacketSize 0x0200 1x 512 bytes >> bInterval 1 >> Device Qualifier (for other device speed): >> bLength 10 >> bDescriptorType 6 >> bcdUSB 2.00 >> bDeviceClass 255 Vendor Specific Class >> bDeviceSubClass 0 >> bDeviceProtocol 255 >> bMaxPacketSize0 64 >> bNumConfigurations 1 >> can't get debug descriptor: Resource temporarily unavailable >> Device Status: 0x0001 >> Self Powered >> >> >> >> A summary of some USB packets during idle, start playback, stop >> playback, start recording, stop recording. Playback and record was >> probably only using a pair of stereo channels IIRC (maybe even one mono >> channel). I could do the same with all channels if that would be of any use. >> >> >> - at all times, polling seems to be 'Bulk or Interrupt Transfer' to >> Device Object FFFFFA800BBF1C00h (00000094) with I/O Request Packet >> FFFFFA800B97B830h >> >> >> - start playback: 'Select Interface', 'Ifc 1 / Alt 1' and 'Ifc 2 / Alt >> 1', Device Object: FFFFFA800BBF1C00h, I/O Request Packet: FFFFFA800778E780h >> >> 1x 'Create' using I/O Request Packet: FFFFFA800C7A2480h to Device >> Objects: FFFFFA800C060C20h (00000095) and FFFFFA800BE30600h (00000096) >> >> >> - during playback: lots of 'Isoch Transfer' to FFFFFA800BBF1C00h with >> nine different I/O Request Packets >> >> >> - stop playback: 'Select Interface', 'Ifc 1 / Alt 0' and 'Ifc 1 / Alt >> 0', Device Object: FFFFFA800BBF1C00h; I/O Request Packet: >> FFFFFA8007552810h >> >> 1x 'Close' using I/O Request Packet: FFFFFA800C7A2480h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h; some 'Cleanups' that >> return 'Invalid Device Request' >> >> >> - start recording: 'Select Interface', 'Ifc 1 / Alt 1' and 'Ifc 2 / Alt >> 1', Device Object: FFFFFA800BBF1C00h, I/O Request Packet: FFFFFA800778E780h >> >> 5x 'Create' using I/O Request Packet: FFFFFA800C0E9C60h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> 2x 'Create' using I/O Request Packet: FFFFFA800C60A010h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> 1x 'Create' using I/O Request Packet: FFFFFA800754F010h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> 2x 'Close' using I/O Request Packet: FFFFFA800C0E9C60h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> >> >> - during recording: lots of 'Isoch Transfer' to FFFFFA800BBF1C00h with >> nine different I/O Request Packets >> >> >> - stop recording: 'Select Interface', 'Ifc 1 / Alt 0' and 'Ifc 1 / Alt >> 0', Device Object: FFFFFA800BBF1C00h; I/O Request Packet: >> FFFFFA8007552810h >> >> 4x 'Close' using I/O Request Packet: FFFFFA8007543C60h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> >>

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[alsa-devel] [RFC PATCH 0/3]
by Yakir Yang 10 Feb '15

10 Feb '15

RK3288 hdmi eye-diagram test would fail when pixel clock is 148.5MHz, and single-ended test would failed when display mode is 74.25MHz. - Fix some code style, leave space for next patches. - For hdmi eye-diagram test, we turn on the Transmitter Trailer-B and improve slopeboost to 25%-30% decrease. - For hdmi single-ended test, we set CKLVL & TXLVL to 17 when pixel clock is 74.25MHz, keep CKLVL & TXLVL to 13 when pixel clock is 148.5MHz. Yakir Yang (3): drm: bridge/dw_hdmi: fixed codec style drm: bridge/dw_hdmi_rockchip: improve hdmi eye-diagram test drm: bridge/dw_hdmi: improve hdmi single-end test drivers/gpu/drm/bridge/dw_hdmi.c | 20 ++++++++++---------- drivers/gpu/drm/imx/dw_hdmi-imx.c | 12 ++++++------ drivers/gpu/drm/rockchip/dw_hdmi-rockchip.c | 14 +++++++------- include/drm/bridge/dw_hdmi.h | 5 +++-- 4 files changed, 26 insertions(+), 25 deletions(-) -- 1.7.9.5

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[alsa-devel] [RESEND PATCH 0/6] Add support for the WM8280 and WM8281
by Charles Keepax 10 Feb '15

10 Feb '15

This set of patches adds support for the WM8280 and WM8281 codecs to the Wolfson Arizona drivers. Just another resend on this series, if it would expedite the process I would be happy if we could just merge the MFD patches then I can push the other patches through their respective trees once that hits mainline, but at the moment all the patches are gated by a build dependency on the MFD patches. Thanks, Charles Richard Fitzgerald (6): mfd: arizona: Add support for WM8280/WM8281 Documentation: devicetree: arizona: Add bindings for WM8280 regulator: arizona-micsupp: Add support for WM8280/WM8281 gpio: arizona: Add support for WM8280/WM8281 extcon: arizona: Add support for WM8280/WM8281 ASoC: arizona: Add support for WM8280/WM8281 Documentation/devicetree/bindings/mfd/arizona.txt | 15 +++++++++++---- drivers/extcon/extcon-arizona.c | 1 + drivers/gpio/gpio-arizona.c | 1 + drivers/mfd/Kconfig | 5 +++-- drivers/mfd/arizona-core.c | 15 +++++++++++++-- drivers/mfd/arizona-i2c.c | 2 ++ drivers/mfd/arizona-irq.c | 1 + drivers/mfd/arizona-spi.c | 2 ++ drivers/regulator/arizona-micsupp.c | 1 + include/linux/mfd/arizona/core.h | 1 + sound/soc/codecs/arizona.c | 2 ++ 11 files changed, 38 insertions(+), 8 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH] ALSA: hda - Add docking station support for another HP machine
by David Henningsson 10 Feb '15

10 Feb '15

From: TienFu Chen <tienfu.chen(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1412800 Signed-off-by: TienFu Chen <tienfu.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- 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 0403061..ddb9308 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4950,6 +4950,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS), + SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC280_FIXUP_HP_DOCK_PINS), SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: atmel_ssc_dai: Allow more rates
by Peter Rosin 10 Feb '15

10 Feb '15

From: Peter Rosin <peda(a)axentia.se> When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to the SSC master clock, divided by either of 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? The divider of 2, 3 or 6 is selected based on the Serial Clock Ratio Considerations section from the SSC documentation: The Transmitter and the Receiver can be programmed to operate with the clock signals provided on either the TK or RK pins. This allows the SSC to support many slave-mode data transfers. In this case, the maximum clock speed allowed on the RK pin is: - Peripheral clock divided by 2 if Receiver Frame Synchro is input - Peripheral clock divided by 3 if Receiver Frame Synchro is output In addition, the maximum clock speed allowed on the TK pin is: - Peripheral clock divided by 6 if Transmit Frame Synchro is input - Peripheral clock divided by 2 if Transmit Frame Synchro is output Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 111 +++++++++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 108 insertions(+), 4 deletions(-) Changes since v2: - Killed the 500ppm reduction. Just trust the nominal clock rates of the given clocks. - Don't multiply the given SSC clk rate by two to get to the SSC master clock rate. That factor was the result of some confusion on my part. The SSC clk *is* the SSC master clock. Changes since v1: - I have checked all Atmel datasheets I could get my hands on (and that seemed relevant), and in every instance where they have described the SSC, the description have the exact rate limitations on the RK and TK pin that I have implemented here. - Improved the comments. - Rebased on top of the latest patches from Bo Chen. diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 379ac2a6ab16..6b8e64899205 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,94 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +288,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +296,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +309,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +884,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +903,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: omap: Kconfig: Support for omap5-uevm analog audio
by Peter Ujfalusi 10 Feb '15

10 Feb '15

The analog audio is supported via omap-abe-twl6040 machine driver on omap5-uevm. Update the Kconfig file to reflect this and select the Palmas clock driver which is providing the 32K clock for audio on omap5-uevm. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a2cd3486ac55..e7c78b0406b5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,17 +100,19 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC + select COMMON_CLK_PALMAS if SOC_OMAP5 help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports: - SDP4430/Blaze boards - PandaBoard (4430) - PandaBoardES (4460) + - omap5-uevm (5432) config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" -- 2.2.2

2 1

[alsa-devel] [PATCH] ALSA: control: confirm to return all identical information in 'activate' event data
by Takashi Sakamoto 10 Feb '15

10 Feb '15

When event originator doesn't set numerical ID in identical information, the event data includes no numerical ID, thus userspace applications cannot identify the control just by unique ID in event data. This commit fix this bug so as the event data includes all of identical information. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 6a72b3e..884fddd 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -587,6 +587,7 @@ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, } ret = 1; unlock: + *id = kctl->id; up_write(&card->controls_rwsem); if (ret > 0) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, id); -- 2.1.0

2 5

Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Takashi Iwai 10 Feb '15

10 Feb '15

At Mon, 9 Feb 2015 12:38:00 +0100 (CET), Jan Engelhardt wrote: > > > On Monday 2015-02-09 11:29, Takashi Iwai wrote: > >> > >> On 3.19-rc3+ [eb74926920cfa756087a82e0b081df837177cb95] I confirm > >> all the volume bars/switches that went missing for me are back again. > >> > >> As said, all is fine here. > > > >Sorry, I've forgotten this pending issue. > >Could you confirm that the patch below works for you? Then I'll queue > >up for 3.20-rc1 and backport appropriately. > > I have no problems with 3.19-rc6. > Your patch does not appear to break anything, but it does not appear > to fix something, either. It does change the controls again though. > I suggest not to apply it. > > http://picpaste.de/3_19_rc6-NgtzogKW.png > http://picpaste.de/3_19_rc6_YOURPATCH-y6IcoXpE.png OK, if 3.19 works as is, then there is no reason to change there. Takashi

2 1

[alsa-devel] [PATCH lib] control: complement comment for subscribe API
by Takashi Sakamoto 09 Feb '15

09 Feb '15

From: Takashi Sakamoto <mocchi(a)Mocchi64.miraclelinux.com> When minus value is given as a parameter, the API returns whether the caller has already subscribed or not. This commit adds a comment about this so as better documentation. Signed-off-by: Takashi Sakamoto <mocchi(a)Mocchi64.miraclelinux.com> --- src/control/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/control.c b/src/control/control.c index dd428a1..4a28cf6 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -213,7 +213,7 @@ int snd_ctl_poll_descriptors_revents(snd_ctl_t *ctl, struct pollfd *pfds, unsign /** * \brief Ask to be informed about events (poll, #snd_async_add_ctl_handler, #snd_ctl_read) * \param ctl CTL handle - * \param subscribe 0 = unsubscribe, 1 = subscribe + * \param subscribe 0 = unsubscribe, 1 = subscribe, -1 = check subscribe or not * \return 0 on success otherwise a negative error code */ int snd_ctl_subscribe_events(snd_ctl_t *ctl, int subscribe) -- 2.1.0

2 3

[alsa-devel] [PATCH] ALSA: control: fix failure to return new numerical ID in 'replace' event data
by Takashi Sakamoto 09 Feb '15

09 Feb '15

In 'replace' event data, numerical ID of control is always invalid. This commit fix this bug so as the event data has renewed numerical ID for control. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 884fddd..ab74b7e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -440,6 +440,7 @@ add: card->controls_count += kcontrol->count; kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + id = kcontrol->id; count = kcontrol->count; up_write(&card->controls_rwsem); for (idx = 0; idx < count; idx++, id.index++, id.numid++) -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: control: evaluate macro instead of numerical value
by Takashi Sakamoto 09 Feb '15

09 Feb '15

SNDRV_CTL_TLV_OP_XXX is defined but not used in core code. Instead, raw numerical value is evaluated. This commit replaces these values to these macros for better looking. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 8aae6ea..6a72b3e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1077,7 +1077,7 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol, int change = 0; void *new_data; - if (op_flag > 0) { + if (op_flag == SNDRV_CTL_TLV_OP_WRITE) { if (size > 1024 * 128) /* sane value */ return -EINVAL; @@ -1333,9 +1333,12 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, goto __kctl_end; } vd = &kctl->vd[tlv.numid - kctl->id.numid]; - if ((op_flag == 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) == 0) || - (op_flag > 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) == 0) || - (op_flag < 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND) == 0)) { + if ((op_flag == SNDRV_CTL_TLV_OP_READ && + (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) == 0) || + (op_flag == SNDRV_CTL_TLV_OP_WRITE && + (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) == 0) || + (op_flag == SNDRV_CTL_TLV_OP_CMD && + (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND) == 0)) { err = -ENXIO; goto __kctl_end; } @@ -1352,7 +1355,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, return 0; } } else { - if (op_flag) { + if (op_flag != SNDRV_CTL_TLV_OP_READ) { err = -ENXIO; goto __kctl_end; } -- 2.1.0

2 1

[alsa-devel] [PATCH v5 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 09 Feb '15

09 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. v5: no functional change, only addressed Takashi comments on patch 2,3,4 fixed used of bool with true/false and ! in comparisons move trigger_tstamp_pending_update flag to snd_usb_substream, will be used by ALSA core when other devices need it Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 66 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 36 +++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 58 ++++++++- sound/pci/hda/hda_controller.c | 43 +++++-- sound/usb/card.h | 2 + sound/usb/pcm.c | 9 ++ 9 files changed, 479 insertions(+), 59 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

2 17

[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Allow more rates
by Peter Rosin 09 Feb '15

09 Feb '15

From: Peter Rosin <peda(a)axentia.se> When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to within 500ppm of the SSC master clock, possibly divided by 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? The divider of 2, 3 or 6 is selected based on the Serial Clock Ratio Considerations section from the SSC documentation: The Transmitter and the Receiver can be programmed to operate with the clock signals provided on either the TK or RK pins. This allows the SSC to support many slave-mode data transfers. In this case, the maximum clock speed allowed on the RK pin is: - Peripheral clock divided by 2 if Receiver Frame Synchro is input - Peripheral clock divided by 3 if Receiver Frame Synchro is output In addition, the maximum clock speed allowed on the TK pin is: - Peripheral clock divided by 6 if Transmit Frame Synchro is input - Peripheral clock divided by 2 if Transmit Frame Synchro is output Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 113 +++++++++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 110 insertions(+), 4 deletions(-) Changes since v1: - I have checked all Atmel datasheets I could get my hands on (and that seemed relevant), and in every instance where they have described the SSC, the description have the exact rate limitations on the RK and TK pin that I have implemented here. - Improved the comments. - Rebased on top of the latest patches from Bo Chen. One thing remains a bit unclear, and that is the 500ppm deduction. Is that really warranted? The number was just pulled out of my hat... Cheers, Peter diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 379ac2a6ab16..767f65bab25d 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,96 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + /* Take away 500ppm, just to be on the safe side. */ + t.max -= t.max / 2000; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +298,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk) * 2; /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +311,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +886,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +905,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); -- 1.7.10.4

4 14

[alsa-devel] [patch] ALSA: riptide: off by one in snd_riptide_joystick_probe()
by Dan Carpenter 09 Feb '15

09 Feb '15

There is an off by one bug in snd_riptide_joystick_probe() because we do the "dev++" toward the start of the function instead of waiting until the end. The impact of this bug is: 1) A static checker warning. 2) If you connect 32 joysticks to your computer they will fail to load with a different error message than intended. 3) If you pass a module option to specify a non-default joystick port then the parameter array has to be shifted one element. The first two are not serious. For the third one, it's probably too late to change it since the bug was introduced in 2009. Either no one noticed and we can leave it alone, or if they did notice, they probably have implemented work arounds so we can't change it now without confusing them. So instead of fixing it the "theoretically correct way", I have decided to just work around it by making the joystick_port[] one element larger. This silences the static checker warning and doesn't affect user space. Fixes: db1005ec6ff8 ('ALSA: riptide - Fix joystick resource handling') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 29f2827..324e34c 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -125,7 +125,11 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; #ifdef SUPPORT_JOYSTICK -static int joystick_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x200 }; +/* + * The joystick_port accounting is off by one but I'm afraid to change it for + * fear of breaking userspace. + */ +static int joystick_port[SNDRV_CARDS + 1] = { [0 ... (SNDRV_CARDS)] = 0x200 }; #endif static int mpu_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x330 }; static int opl3_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x388 };

2 4

Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Takashi Iwai 09 Feb '15

09 Feb '15

At Sat, 10 Jan 2015 13:10:30 +0100 (CET), Jan Engelhardt wrote: > > > On Wednesday 2014-12-31 09:51, Takashi Iwai wrote: > >> > > >> >Ah, I see you disable the automute. So the headphone plug shouldn't > >> >matter. > >> > >> Changing the "Auto-Mute" control does not do anything; the system always > >> exhibits Automute behavior (if that's a bug, it's one which I did not > >> care about so far). > > > >Oh, that's really interesting. I didn't expect such a > >hardware-controlled automute nowadays. We need to investigate this > >more. > >[...] > >[Volumes] On 3.19, this was renamed differently, so that "PCM" can > >appear again by a softvol item like before. > > On 3.19-rc3+ [eb74926920cfa756087a82e0b081df837177cb95] I confirm > all the volume bars/switches that went missing for me are back again. > > In particular, 3.19 shows a new "Speaker+LO" no-switch volume bar > control, and a "Speaker" no-volume switch control. Works for me, > though I wonder if the change was necessary. (In 3.17, the switch and > volume were combined in one control.) > > As said, all is fine here. Sorry, I've forgotten this pending issue. Could you confirm that the patch below works for you? Then I'll queue up for 3.20-rc1 and backport appropriately. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix lost sound on Thinkpad X240 The commit [7c21539c45ea: ALSA: hda - Add ultra dock support for Thinkpad X240] caused a regression, the sound got lost. The machine seems requiring the fixed binding between DACs and pins, and the commit above resulted in the change of binding. As a fix, this patch adds the binding list to ensure to connect to the correct DAC. Reported-by: Jan Engelhardt <jengelh(a)inai.de> Fixes: 7c21539c45ea ('ALSA: hda - Add ultra dock support for Thinkpad X240.') Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 040306194e6d..dba31bafcd1c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4226,6 +4226,22 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, } } +/* TP X240 seems to need fixed DAC binding; otherwise sound gets lost */ +static void alc_fixup_tpx240_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + static hda_nid_t preferred_pairs[] = { + 0x14, 0x03, + 0x15, 0x02, + 0x16, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + spec->gen.preferred_dacs = preferred_pairs; + } +} + static void alc283_hp_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) { @@ -4411,6 +4427,7 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_TPX240_DOCK, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -4856,6 +4873,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC292_FIXUP_TPX240_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpx240_dacs, + .chained = true, + .chain_id = ALC292_FIXUP_TPT440_DOCK, + }, [ALC283_FIXUP_BXBT2807_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5026,7 +5049,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPX240_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.2.2

2 1

[alsa-devel] Can we remove un-used Sound driver ?
by Kuninori Morimoto 09 Feb '15

09 Feb '15

Hi SH ML, ALSA ML We have below sound drivers in upstream kernel. But, these are not used from upstreamed board. And, these are for very old SoC. I guess Renesas doesn't/can't support these SoC/board anymore. Can we remove these sound drivers ? Because maintenance cost for these is not zero... - ${LINUX}/sound/soc/sh/ssi.c - SH7760/SH7780 - CONFIG_SND_SOC_SH4_SSI - ${LINUX}/linux/sound/soc/sh/hac.c - SH7760/SH7780 - CONFIG_SND_SOC_SH4_HAC - ${LINUX}/sound/soc/sh/sh7760-ac97.c - SH7760 - CONFIG_SND_SH7760_AC97 - ${LINUX}/sound/soc/sh/dma-sh7760.c - SH7760 - CONFIG_SND_SOC_PCM_SH7760 - ${LINUX}/sound/soc/sh/siu_dai.c *1 ${LINUX}/sound/soc/sh/siu_pcm.c ${LINUX}/sound/soc/sh/siu.h - SH7343, SH7722 - CONFIG_SND_SOC_SH4_SIU - ${LINUX}/sound/soc/sh/migor.c *2 - SH7722 - CONFIG_SND_SIU_MIGOR *1 it is registered from ${LINUX}/arch/sh/kernel/cpu/sh4a/setup-sh7722.c but, no one enabled CONFIG_SND_SOC_SH4_SIU in defconfig *2 it uses *1, but no one enabled CONFIG_SND_SIU_MIGOR Best regards --- Kuninori Morimoto

4 12

[alsa-devel] [PATCH 0/2] ASoC: rsnd: tidyup .remove method
by Kuninori Morimoto 09 Feb '15

09 Feb '15

Hi Mark These patches call missing remove/unregister functions on rsnd driver. Kuninori Morimoto (2): ASoC: rsnd: call missing snd_ctl_remove() ASoC: rsnd: call missing snd_soc_unregiter_component/platform() sound/soc/sh/rcar/core.c | 14 +++++++++++++- sound/soc/sh/rcar/dvc.c | 15 +++++++++++++++ sound/soc/sh/rcar/rsnd.h | 5 +++++ 3 files changed, 33 insertions(+), 1 deletion(-) BTW, we found ALSA SoC bind/unbind issue on latest upstream kernel. ALSA SoC is using like this. [CPU] - [CARD] - [CODEC] If we unbind CPU here, it still has CARD infomation. And, kernel doesn't call initialize function when re-bind, and it will panic when playback ------------------------- In our case, CPU : rsnd CARD : simple-card CODEC : ak4642 Current list / # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: rsnddai0ak4642h [rsnd-dai.0-ak4642-hifi], device 0: rsnd-dai.0-ak4642-hifi ak4642-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Unbind CPU echo "ec500000.rcar_sound" > /sys/bus/platform/drivers/rcar_sound/unbind But, it still has unbinded card / # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: rsnddai0ak4642h [rsnd-dai.0-ak4642-hifi], device 0: rsnd-dai.0-ak4642-hifi ak4642-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Re-bind. .probe function was called, but no mapping message "asoc-simple-card sound: ak4642-hifi <-> ec500000.rcar_sound mapping ok" echo "ec500000.rcar_sound" > /sys/bus/platform/drivers/rcar_sound/bind rcar_sound ec500000.rcar_sound: probed Kernel panic when playback after rebind / # aplay /home/xxx.wav ASoC: ak4642-hifi <-> No matching rates Unable to handle kernel paging request at virtual address ee196100 pgd = eebbc000 [ee196100] *pgd=6e01141e(bad) Internal error: Oops: 8000000d [#1] SMP ARM CPU: 0 PID: 936 Comm: aplay Tainted: G W 3.19.0-rc6-02875-gf8eac9a #1541 Hardware name: Generic R8A7790 (Flattened Device Tree) task: edc2adc0 ti: ee08a000 task.ti: ee08a000 PC is at 0xee196100 LR is at soc_pcm_open+0x6e8/0x7ac pc : [<ee196100>] lr : [<c0386e18>] psr: a00f0013 sp : ee08bc90 ip : ee08bc90 fp : ee08bcec r10: eeb30800 r9 : eeae7a00 r8 : ffffffea r7 : eeb30800 r6 : ee947480 r5 : ee328a00 r4 : edbf5010 r3 : ee196100 r2 : ee0dc290 r1 : ee947480 r0 : ee328a00 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c5307d Table: 6ebbc06a DAC: 00000015 Process aplay (pid: 936, stack limit = 0xee08a238) Stack: (0xee08bc90 to 0xee08c000) bc80: 00000001 00000002 edbf5010 00000010 bca0: ee91d520 edbf501c 0000000c 00000000 00000000 00000001 ffffffff eeae7a00 bcc0: ee08bcf4 00000000 ee08bd10 eea3ad00 ee328ce8 ee328d00 00000000 00000001 bce0: ee08bd0c ee08bcf0 c0376a60 c038673c c0045098 ee328a00 00000000 ee328c00 bd00: ee08bd54 ee08bd10 c0376b38 c0376a18 ee08bd2c 00000000 edc2adc0 c0045098 bd20: ee328d04 ee328d04 c024dd10 ee328c00 eea3ad00 eebb3638 00000000 c0661848 bd40: eea3ad00 eea3ad00 ee08bd6c ee08bd58 c0376d04 c0376ab4 c04f179c eea3ad00 bd60: ee08bd8c ee08bd70 c0367f18 c0376ccc c0367e9c 00000000 eebb3638 ee13bec0 bd80: ee08bdbc ee08bd90 c00c8b94 c0367ea8 00000000 00000010 eea3ad00 eebb3638 bda0: ee08be88 00000000 c00c8a5c eea3ad08 ee08bde4 ee08bdc0 c00c303c c00c8a68 bdc0: ee08bed0 eea3ad00 ee08be88 00000000 ee3f0000 ee08be84 ee08bdf4 ee08bde8 bde0: c00c3ff8 c00c2ea0 ee08be6c ee08bdf8 c00d0c78 c00c3fb8 00000101 ee4264c8 be00: ee03e890 ee425770 eebb26b8 00000026 00000000 ee425770 ee08bee8 00000000 be20: 00000802 ee08bf5c 00000000 00000000 ee08a000 eebb3638 ee03e890 ee425770 be40: c00cef24 ee08bed0 eea3ad00 ee08be90 ee08bf5c ee3f0000 ee08be88 00000000 be60: ee08bec4 ee08be70 c00d10d4 c00d0400 ee08be84 ee9a1380 ee08bf0c ee08be88 be80: c036cddc 00000000 ee03e890 ee4274c8 ee08bec4 ee08bea0 c01aa17c ee08bf5c bea0: 00000001 ee3f0000 ffffff9c c000eb24 ee08a000 00000000 ee08bf4c ee08bec8 bec0: c00d1474 c00d0f00 00000041 ee08bed8 ee03e890 ee4274c8 474edc20 00000008 bee0: ee3f0019 eea3a700 ee03e7d0 ee4264c8 eebb3638 00000101 00000000 00000034 bf00: 00000000 00000000 00000000 c00de444 00080802 00080802 be8b4ea4 ee3f0000 bf20: ffffff9c c000eb24 ee08a000 00000000 00080802 00000004 ee3f0000 ffffff9c bf40: ee08bf94 ee08bf50 c00c4430 c00d144c c00d38e0 c00def38 eea3a700 00000802 bf60: be8b0000 00000026 00000100 00000001 be8b466c be8b4ea4 00000002 00000005 bf80: c000eb24 ee08a000 ee08bfa4 ee08bf98 c00c44f8 c00c431c 00000000 ee08bfa8 bfa0: c000e980 c00c44dc be8b466c be8b4ea4 be8b4794 00080802 be8b4628 00080802 bfc0: be8b466c be8b4ea4 00000002 00000005 001255f8 00129264 000dd264 00124478 bfe0: 00000000 be8b4634 00049257 00083036 600f0030 be8b4794 03433c61 0717419f Backtrace: [<c0386730>] (soc_pcm_open) from [<c0376a60>] (snd_pcm_open_substream+0x54/0x9c) r10:00000001 r9:00000000 r8:ee328d00 r7:ee328ce8 r6:eea3ad00 r5:ee08bd10 r4:00000000 [<c0376a0c>] (snd_pcm_open_substream) from [<c0376b38>] (snd_pcm_open+0x90/0x1b4) r5:ee328c00 r4:00000000 [<c0376aa8>] (snd_pcm_open) from [<c0376d04>] (snd_pcm_playback_open+0x44/0x64) r10:eea3ad00 r9:eea3ad00 r8:c0661848 r7:00000000 r6:eebb3638 r5:eea3ad00 r4:ee328c00 [<c0376cc0>] (snd_pcm_playback_open) from [<c0367f18>] (snd_open+0x7c/0x90) r5:eea3ad00 r4:c04f179c [<c0367e9c>] (snd_open) from [<c00c8b94>] (chrdev_open+0x138/0x164) r6:ee13bec0 r5:eebb3638 r4:00000000 r3:c0367e9c [<c00c8a5c>] (chrdev_open) from [<c00c303c>] (do_dentry_open.isra.17+0x1a8/0x2b8) r9:eea3ad08 r8:c00c8a5c r7:00000000 r6:ee08be88 r5:eebb3638 r4:eea3ad00 [<c00c2e94>] (do_dentry_open.isra.17) from [<c00c3ff8>] (vfs_open+0x4c/0x50) r9:ee08be84 r8:ee3f0000 r7:00000000 r6:ee08be88 r5:eea3ad00 r4:ee08bed0 [<c00c3fac>] (vfs_open) from [<c00d0c78>] (do_last.isra.47+0x884/0xb00) [<c00d03f4>] (do_last.isra.47) from [<c00d10d4>] (path_openat+0x1e0/0x54c) r10:00000000 r9:ee08be88 r8:ee3f0000 r7:ee08bf5c r6:ee08be90 r5:eea3ad00 r4:ee08bed0 [<c00d0ef4>] (path_openat) from [<c00d1474>] (do_filp_open+0x34/0x80) r10:00000000 r9:ee08a000 r8:c000eb24 r7:ffffff9c r6:ee3f0000 r5:00000001 r4:ee08bf5c [<c00d1440>] (do_filp_open) from [<c00c4430>] (do_sys_open+0x120/0x1c0) r7:ffffff9c r6:ee3f0000 r5:00000004 r4:00080802 [<c00c4310>] (do_sys_open) from [<c00c44f8>] (SyS_open+0x28/0x2c) r9:ee08a000 r8:c000eb24 r7:00000005 r6:00000002 r5:be8b4ea4 r4:be8b466c [<c00c44d0>] (SyS_open) from [<c000e980>] (ret_fast_syscall+0x0/0x34) Code: 00000000 00000000 00000000 00000000 (0000004f) ---[ end trace bd44069e52a540ae ]--- Segmentation fault Best regards --- Kuninori Morimoto

2 28

[alsa-devel] [GIT PULL] ASoC updates for v3.20
by Mark Brown 09 Feb '15

09 Feb '15

2 1

[alsa-devel] [PATCH] ASoC: rt5670: Set use_single_rw flag for regmap
by Bard Liao 09 Feb '15

09 Feb '15

RT5670 doesn't support auto incrementing writes so driver should set the use_single_rw flag for regmap. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0632b74..6aa4ab2 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2614,6 +2614,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { static const struct regmap_config rt5670_regmap = { .reg_bits = 8, .val_bits = 16, + .use_single_rw = true, .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * RT5670_PR_SPACING), .volatile_reg = rt5670_volatile_register, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ASoC: rt286: Add rt288 codec support
by Bard Liao 09 Feb '15

09 Feb '15

This patch adds support for rt288 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 81bdd276..7a72eac 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -34,6 +34,7 @@ #include "rt286.h" #define RT286_VENDOR_ID 0x10ec0286 +#define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { struct regmap *regmap; @@ -1168,6 +1169,7 @@ static const struct regmap_config rt286_regmap = { static const struct i2c_device_id rt286_i2c_id[] = { {"rt286", 0}, + {"rt288", 0}, {} }; MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); @@ -1210,7 +1212,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_read(rt286->regmap, RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID) { + if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %x is not rt286\n", ret); return -ENODEV; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: max98357a: Fix build in !CONFIG_OF case
by Mark Brown 09 Feb '15

09 Feb '15

Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 98b9153..1806333 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -120,6 +120,7 @@ static const struct of_device_id max98357a_device_id[] = { { .compatible = "maxim," DRV_NAME, }, {} }; +MODULE_DEVICE_TABLE(of, max98357a_device_id); #endif static struct platform_driver max98357a_platform_driver = { @@ -135,4 +136,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, max98357a_device_id); -- 2.1.4

1 0

[alsa-devel] [RESEND PATCH V3 13/15] ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update
by Sylwester Nawrocki 09 Feb '15

09 Feb '15

Clock related properties are added to the Exynos4 I2S device nodes so they can be referred to as clock providers. Missing i2s_opclk1 clock is added to the I2S0 node and clock properties are added to the MAX98090 codec node to allow it to control/read frequency of the MCLK clock directly. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- arch/arm/boot/dts/exynos4.dtsi | 6 ++++++ arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 8 ++++++-- 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/arch/arm/boot/dts/exynos4.dtsi b/arch/arm/boot/dts/exynos4.dtsi index b8168f1..38d8f68 100644 --- a/arch/arm/boot/dts/exynos4.dtsi +++ b/arch/arm/boot/dts/exynos4.dtsi @@ -61,6 +61,8 @@ reg = <0x03830000 0x100>; clocks = <&clock_audss EXYNOS_I2S_BUS>; clock-names = "iis"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk0"; dmas = <&pdma0 12>, <&pdma0 11>, <&pdma0 10>; dma-names = "tx", "rx", "tx-sec"; samsung,idma-addr = <0x03000000>; @@ -372,6 +374,8 @@ reg = <0x13960000 0x100>; clocks = <&clock CLK_I2S1>; clock-names = "iis"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk1"; dmas = <&pdma1 12>, <&pdma1 11>; dma-names = "tx", "rx"; status = "disabled"; @@ -382,6 +386,8 @@ reg = <0x13970000 0x100>; clocks = <&clock CLK_I2S2>; clock-names = "iis"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk2"; dmas = <&pdma0 14>, <&pdma0 13>; dma-names = "tx", "rx"; status = "disabled"; diff --git a/arch/arm/boot/dts/exynos4412-odroid-common.dtsi b/arch/arm/boot/dts/exynos4412-odroid-common.dtsi index 3fbf588..c26b9fb 100644 --- a/arch/arm/boot/dts/exynos4412-odroid-common.dtsi +++ b/arch/arm/boot/dts/exynos4412-odroid-common.dtsi @@ -7,6 +7,7 @@ * published by the Free Software Foundation. */ +#include <dt-bindings/sound/samsung-i2s.h> #include <dt-bindings/input/input.h> #include "exynos4412.dtsi" @@ -37,8 +38,9 @@ pinctrl-names = "default"; status = "okay"; clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_DOUT_AUD_BUS>; - clock-names = "iis", "i2s_opclk0"; + <&clock_audss EXYNOS_DOUT_AUD_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; }; sound: sound { @@ -373,6 +375,8 @@ reg = <0x10>; interrupt-parent = <&gpx0>; interrupts = <0 0>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + clock-names = "mclk"; }; }; -- 1.7.9.5

2 7

[alsa-devel] [PATCH 0/3] ASoC: atmel_ssc_dai: fix and refactor
by Bo Shen 09 Feb '15

09 Feb '15

This patch set do following things: - Fix the DSP mode A setting. - Refactor the startup and shutdown function. - Only clean PDC when use PDC. Bo Shen (3): ASoC: atmel_ssc_dai: fix the setting for DSP mode ASoC: atmel_ssc_dai: refactor the startup and shutdown ASoC: atmel_ssc_dai: only clean PDC when using PDC sound/soc/atmel/atmel_ssc_dai.c | 54 +++++++++++++++++++---------------------- 1 file changed, 25 insertions(+), 29 deletions(-) -- 2.3.0.rc0

2 5

[alsa-devel] [PATCH] ALSA: control: obsolete 'already present' messaging because it's caller's fault
by Takashi Sakamoto 08 Feb '15

08 Feb '15

When drivers or userspace applications add a duplicated control, snd module outputs error message and return -EBUSY. This state is a bug in driver or application, therefore the messaging should be done by them, not by snd module. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index d482997..8906b66 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -357,18 +357,10 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) down_write(&card->controls_rwsem); if (snd_ctl_find_id(card, &id)) { - up_write(&card->controls_rwsem); - dev_err(card->dev, "control %i:%i:%i:%s:%i is already present\n", - id.iface, - id.device, - id.subdevice, - id.name, - id.index); err = -EBUSY; goto error; } if (snd_ctl_find_hole(card, kcontrol->count) < 0) { - up_write(&card->controls_rwsem); err = -ENOMEM; goto error; } @@ -388,6 +380,7 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) return 0; error: + up_write(&card->controls_rwsem); snd_ctl_free_one(kcontrol); return err; } -- 2.1.0

2 2

[alsa-devel] [PATCH v2] ALSA: control: fix failure to return numerical ID in 'add' event
by Takashi Sakamoto 08 Feb '15

08 Feb '15

Currently when adding a new control, the assigned numerical ID is not set for event data, thus userspace applications cannot realize it just by event data. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 60caba1..8aae6ea 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -373,6 +373,7 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) card->controls_count += kcontrol->count; kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + id = kcontrol->id; count = kcontrol->count; up_write(&card->controls_rwsem); for (idx = 0; idx < count; idx++, id.index++, id.numid++) -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: control: fix failure to return numerical ID in event data
by Takashi Sakamoto 08 Feb '15

08 Feb '15

Currently an instance of control has zero as its member for numerical ID even if any IDs are assigned to. According to this bug, any userspace applications cannot identify controls by the ID when handling any events. On the other hand, the other members such as name are still valid, therefore applications can identify controls without fixing this. This is not preferable because the ID has an advantage to allow userspace applications to distinguish each controls by itself, without any combinations. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 60caba1..d482997 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -352,6 +352,9 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) if (id.index > UINT_MAX - kcontrol->count) goto error; + /* Any ID numbers are not assigned to this control yet. */ + id.numid = 0; + down_write(&card->controls_rwsem); if (snd_ctl_find_id(card, &id)) { up_write(&card->controls_rwsem); @@ -369,12 +372,17 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) err = -ENOMEM; goto error; } - list_add_tail(&kcontrol->list, &card->controls); - card->controls_count += kcontrol->count; + + /* Now this control is allowed to have ID number. */ kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; - count = kcontrol->count; + card->controls_count += kcontrol->count; + list_add_tail(&kcontrol->list, &card->controls); + up_write(&card->controls_rwsem); + + id = kcontrol->id; + count = kcontrol->count; for (idx = 0; idx < count; idx++, id.index++, id.numid++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); return 0; -- 2.1.0

2 2

[alsa-devel] Some questions about userspace control elements
by Takashi Sakamoto 08 Feb '15

08 Feb '15

Dear all, I have some questions about ALSA control interface. Currently I'm working for writing control applications in userspace. The application adds some control elements to a certain control device, then execute interprocess communication via the device by ALSA control interface. To help this work, I'm preparing for alsa-gir, with partly implementation of ALSA Control/Timer/Sequencer interface. (Currently I have no plan to support the other interfaces, such as PCM.) https://github.com/takaswie/alsa-gir Then I have six questions. 1.The 'numid' is unique number for each element or not. 2.The 'numid' is always zero in SND_CTL_EVENT_MASK_ADD event. 3.The way to change permissions of each control element. 4.Added elements are not removed when closing fd. 5.Processes can delete elements which the other processes added. 6.The operation to replace element generates separate events. In detail: 1.The 'numid' is unique number for each element or not? I want to use it to indicate each element, if possible. But there're more members in snd_ctl_elem_id structure. 2.The 'numid' can be notified in SND_CTL_EVENT_MASK_ADD event? As long as I test, it's always zero when userspace applications add any elements. Is it an expected behaviour? Or a bug? 3.The way to change permissions of each control element. Is it possible after adding the elements? If possible, how to notify it in the other processes? 4.Any elements added by a process are not removed at closing snd_ctl_t. They're remained after closing the process' snd_ctl_t. Is it an expected behaviour? Or a bug? 5.Processes can delete elements which the other processes added. Is it an expected behaviour? Or a bug? 6.The operation to replace elements generates separate events. ADD/REMOVE events occurs sequencially. Userspace application cannot distinguish replacement from adding/removing, therefore cannot process appropriate action. Is it an expected behaviour? Or a bug? Regards Takashi Sakamoto

2 8

[alsa-devel] [PATCH v2 0/3] More Line 6 cleanup
by Chris Rorvick 08 Feb '15

08 Feb '15

Changes in v2: * return true/false instead of 1/0 * do not include spaces in driver name * drop patch altering struct names (unnecessary) Chris Rorvick (3): ALSA: line6: Add toneport_has_source_select() ALSA: line6: Pass toneport pointer to toneport_has_led() ALSA: line6: Pass driver name to line6_probe() sound/usb/line6/driver.c | 3 ++- sound/usb/line6/driver.h | 3 +-- sound/usb/line6/pod.c | 2 +- sound/usb/line6/podhd.c | 2 +- sound/usb/line6/toneport.c | 53 +++++++++++++++++++++++++--------------------- sound/usb/line6/variax.c | 2 +- 6 files changed, 35 insertions(+), 30 deletions(-) -- 2.1.0

2 4

[alsa-devel] [PATCH] ASoC: Intel: sst: Fix firmware name size handling
by Christian Engelmayer 08 Feb '15

08 Feb '15

Function sst_acpi_probe() uses plain strcpy for setting member firmware_name of a struct intel_sst_drv from member firmware of a struct sst_machines. Thereby the destination array has got a length of 20 byte while the source may hold 32 byte. Since eg. commit 64b9c90b8600 ("ASoC: Intel: Fix BYTCR firmware name") increased strings from "fw_sst_0f28.bin" to "intel/fw_sst_0f28.bin" there is an actual possibility that the 20 byte array at the end of struct intel_sst_drv is overrun. Thus increase the size of the destination and use the same define for both structs. Detected by Coverity CID 1260087. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- Compile tested only. Applies against linux-next. --- sound/soc/intel/sst/sst.h | 3 ++- sound/soc/intel/sst/sst_acpi.c | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 7f4bbfc..562bc48 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -58,6 +58,7 @@ enum sst_algo_ops { #define SST_BLOCK_TIMEOUT 1000 #define FW_SIGNATURE_SIZE 4 +#define FW_NAME_SIZE 32 /* stream states */ enum sst_stream_states { @@ -426,7 +427,7 @@ struct intel_sst_drv { * Holder for firmware name. Due to async call it needs to be * persistent till worker thread gets called */ - char firmware_name[20]; + char firmware_name[FW_NAME_SIZE]; }; /* misc definitions */ diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 43bc1c4..0542442 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -47,7 +47,7 @@ struct sst_machines { char board[32]; char machine[32]; void (*machine_quirk)(void); - char firmware[32]; + char firmware[FW_NAME_SIZE]; struct sst_platform_info *pdata; }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: sirf: atlas7: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 07 Feb '15

07 Feb '15

sound/soc/sirf/atlas7-iacc.c:625:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Rongjun Ying <rongjun.ying(a)csr.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- atlas7-iacc.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/sirf/atlas7-iacc.c +++ b/sound/soc/sirf/atlas7-iacc.c @@ -622,7 +622,6 @@ static const struct dev_pm_ops atlas7_ia static struct platform_driver atlas7_iacc_driver = { .driver = { .name = "sirf-atlas7-iacc", - .owner = THIS_MODULE, .of_match_table = atlas7_iacc_of_match, .pm = &atlas7_iacc_pm_ops, },

1 0

[alsa-devel] [PATCH 0/4] More Line 6 cleanup
by Chris Rorvick 07 Feb '15

07 Feb '15

Not much to say about these other than what's in the commit log. I was looking into what appeared to be a regression that was only reproducible outside of QEMU, but it turned out to be me pulling in the test/snd-device changes but testing against vanilla v3.19-rc7 ALSA modules. Fixing my process. :-) Regards, Chris Chris Rorvick (4): ALSA: line6: Add toneport_has_source_select() ALSA: line6: Pass toneport pointer to toneport_has_led() ALSA: line6: Pass driver name to line6_probe() ALSA: line6: Remove `usb_' prefix from structs sound/usb/line6/capture.c | 2 +- sound/usb/line6/driver.c | 45 +++++++++++----------- sound/usb/line6/driver.h | 25 ++++++------ sound/usb/line6/midi.c | 24 +++++------- sound/usb/line6/midi.h | 6 +-- sound/usb/line6/pcm.c | 5 +-- sound/usb/line6/pcm.h | 4 +- sound/usb/line6/playback.c | 2 +- sound/usb/line6/pod.c | 52 ++++++++++++------------- sound/usb/line6/podhd.c | 7 ++-- sound/usb/line6/toneport.c | 94 ++++++++++++++++++++++++---------------------- sound/usb/line6/variax.c | 41 ++++++++++---------- 12 files changed, 151 insertions(+), 156 deletions(-) -- 2.1.0

2 11

[alsa-devel] [Intel HDA][ALC3226] Too much white noise from microphone
by Shawn Starr 07 Feb '15

07 Feb '15

Hello ALSA devs, I spoke to patrakov on Freenode from the #Alsa IRC channel and they mentioned that the noise background I am experiencing is not normal. I tried two different headsets, both headsets have two separate jacks, the same result. I will test any patches/quirks for kernel as needed. Here is the alsa-info: http://www.alsa-project.org/db/?f=29d3c45e2d66c58be03a10bd0186067a9d735a11 Here is a 10 second recording: http://turl.ca/zrhbcs Here is the amixer information: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Speaker',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Speaker+LO',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 65 [75%] [-16.50dB] Front Right: Playback 65 [75%] [-16.50dB] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [on] Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [20.00dB] Front Right: 2 [67%] [20.00dB] Simple mixer control 'Line Out',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 63 [100%] [30.00dB] [on] Front Right: Capture 63 [100%] [30.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Speaker Only' 'Line Out+Speaker' Item0: 'Line Out+Speaker' Simple mixer control 'Dock Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [off] Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Dock Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right

3 5

[alsa-devel] [PATCH v2 0/3] ASoC: add audio card creation from graph of ports in DT
by Jean-Francois Moine 07 Feb '15

07 Feb '15

This patchset adds a way to build audio cards from the description of a graph of ports in a devicetree. v2: - move the soc-core and dt-card from the patchset "ASoC: tda998x: add a codec to the HDMI transmitter" - add DT documentation (Mark Brown) Jean-Francois Moine (3): ASoC: core: export snd_soc_get_dai_name Documentation: of: Document audio graph bindings ASoC: add generic dt-card support .../devicetree/bindings/sound/audio-interfaces.txt | 101 ++++++++ include/sound/soc.h | 2 + sound/soc/generic/Kconfig | 2 + sound/soc/generic/Makefile | 2 + sound/soc/generic/dt-card.c | 275 +++++++++++++++++++++ sound/soc/soc-core.c | 5 +- 6 files changed, 385 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-interfaces.txt create mode 100644 sound/soc/generic/dt-card.c -- 2.1.4

3 20

[alsa-devel] [PATCH v3 0/15] Those patches is used for dw_hdmi audio support
by Yakir Yang 07 Feb '15

07 Feb '15

We found Designware hdmi driver only support audio clock config, we can not play sound through it. To add Designware HDMI Audio support, we make those patch set: 1): fixed dw_hdmi irq bug, add irq control to suspend/resume interfaces. 2): add suspend/resume callback for dw_hdmi rockchip driver. 3): Warp irq control in functions. 4): Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() 5): Adjust n/cts config order. 6): Set ncts_atomic_write & cts_manual, according to dw_hdmi document. 7): Add audio support for more display resolutions(eg. 800x600). 8): Add audio support for No-CEA display resolutions. 9): Add Audio Sample Channel Status config interfaces to dw_hdmi driver. 10): add audio clock control interfaces o dw_hdmi driver. 11): creat "dw_hdmi-audio" platform device in dw_hdmi driver. 12): add codec driver for hdmi audio, callback dw_hdmi audio config functions. 13): add sound driver for hdmi audio, creat hdmi audio sound card. 14): add dt-bings file and add hdmi_audio node to corresponding dt file. Changes in v3: - Clear Hotplug interrupts before dw_hdmi_fb_register - Wrap irq control in fucntions - Setting the .pm member instead of suspend/resume - Add ID registers parse and record - Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() - Only adjust the n/cts setting order - Set ncts_atomic_write & cts_manual - Determine whether sample channel should set by desig_id. - Delete hdmi_audio_config interface and modify audio clock control functions - Remove audio_config & get_connect_status callback functions and add write/read/mod register callback functions - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Add irq control to suspend/resume interfaces - Add suspend/resume support for dw_hdmi_rockchip driver - add more n/cts combinations for more display resolutions - enable audio support for No-CEA display mode - Add audio sample channel status setting - Add audio config interfaces to dw_hdmi driver - Update the audio control interfaces - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Yakir Yang (15): drm: bridge/dw_hdmi: add irq control to suspend/resume drm: bridge/dw_hdmi: wrap irq control in fucntions drm: rockchip/dw_hdmi_rockchip: add resume/suspend support drm: bridge/dw_hdmi: add indentification registers parse and record drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() drm: bridge/dw_hdmi: adjust n/cts setting order drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio support for No-CEA display resolutions drm: bridge/dw_hdmi: add audio sample channel status setting drm: bridge/dw_hdmi: add audio clock control interfaces drm: bridge/dw_hdmi: creat dw-hdmi-audio platform device ASoC: dw-hdmi-audio: add codec driver for dw hdmi audio ASoC: rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for Rockchip dw-hdmi-audio .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/dw_hdmi.c | 361 ++++++++++++++++++--- drivers/gpu/drm/bridge/dw_hdmi.h | 49 +++ drivers/gpu/drm/rockchip/dw_hdmi-rockchip.c | 16 + include/drm/bridge/dw_hdmi.h | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 333 +++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 79 +++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 168 ++++++++++ 12 files changed, 1013 insertions(+), 39 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

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[alsa-devel] [PATCH 1/4] ASoC: samsung: Remove goni or aquila with the WM8994
by Paul Bolle 07 Feb '15

07 Feb '15

Commit 28c8331d386a ("ARM: S5PV210: Remove support for board files") removed the Kconfig symbols MACH_GONI and MACH_AQUILA. As a result the dependencies of SND_SOC_GONI_AQUILA_WM8994 can never be met. So remove the unbuildable "SoC I2S Audio support for AQUILA/GONI - WM8994". Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Suggested by Arnd Bergmann (see https://lkml.org/lkml/2014/9/4/381 ). Previously submitted in https://lkml.org/lkml/2014/9/18/128 , but in an awkward way. So this is basically a resend. sound/soc/samsung/Kconfig | 11 -- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/goni_wm8994.c | 289 ---------------------------------------- 3 files changed, 302 deletions(-) delete mode 100644 sound/soc/samsung/goni_wm8994.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fc67f97f19f6..da6d0229cfa2 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -146,17 +146,6 @@ config SND_SOC_SMARTQ select SND_SAMSUNG_I2S select SND_SOC_WM8750 -config SND_SOC_GONI_AQUILA_WM8994 - tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" - depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA) - depends on I2C=y - select SND_SAMSUNG_I2S - select MFD_WM8994 - select SND_SOC_WM8994 - help - Say Y if you want to add support for SoC audio on goni or aquila - with the WM8994. - config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" depends on SND_SOC_SAMSUNG diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 31e3dba7e3b5..052fe71be518 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -35,7 +35,6 @@ snd-soc-smdk-wm8994-objs := smdk_wm8994.o snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o -snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o @@ -63,7 +62,6 @@ obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o -obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c deleted file mode 100644 index fad56b9e7369..000000000000 --- a/sound/soc/samsung/goni_wm8994.c +++ /dev/null @@ -1,289 +0,0 @@ -/* - * goni_wm8994.c - * - * Copyright (C) 2010 Samsung Electronics Co.Ltd - * Author: Chanwoo Choi <cw00.choi(a)samsung.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> -#include <mach/gpio-samsung.h> - -#include "../codecs/wm8994.h" - -#define MACHINE_NAME 0 -#define CPU_VOICE_DAI 1 - -static const char *aquila_str[] = { - [MACHINE_NAME] = "aquila", - [CPU_VOICE_DAI] = "aquila-voice-dai", -}; - -static struct snd_soc_card goni; -static struct platform_device *goni_snd_device; - -/* 3.5 pie jack */ -static struct snd_soc_jack jack; - -/* 3.5 pie jack detection DAPM pins */ -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, { - .pin = "Headset Stereophone", - .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | - SND_JACK_AVOUT, - }, -}; - -/* 3.5 pie jack detection gpios */ -static struct snd_soc_jack_gpio jack_gpios[] = { - { - .gpio = S5PV210_GPH0(6), - .name = "DET_3.5", - .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | - SND_JACK_AVOUT, - .debounce_time = 200, - }, -}; - -static const struct snd_soc_dapm_widget goni_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Ext Left Spk", NULL), - SND_SOC_DAPM_SPK("Ext Right Spk", NULL), - SND_SOC_DAPM_SPK("Ext Rcv", NULL), - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Main Mic", NULL), - SND_SOC_DAPM_MIC("2nd Mic", NULL), - SND_SOC_DAPM_LINE("Radio In", NULL), -}; - -static const struct snd_soc_dapm_route goni_dapm_routes[] = { - {"Ext Left Spk", NULL, "SPKOUTLP"}, - {"Ext Left Spk", NULL, "SPKOUTLN"}, - - {"Ext Right Spk", NULL, "SPKOUTRP"}, - {"Ext Right Spk", NULL, "SPKOUTRN"}, - - {"Ext Rcv", NULL, "HPOUT2N"}, - {"Ext Rcv", NULL, "HPOUT2P"}, - - {"Headset Stereophone", NULL, "HPOUT1L"}, - {"Headset Stereophone", NULL, "HPOUT1R"}, - - {"IN1RN", NULL, "Headset Mic"}, - {"IN1RP", NULL, "Headset Mic"}, - - {"IN1RN", NULL, "2nd Mic"}, - {"IN1RP", NULL, "2nd Mic"}, - - {"IN1LN", NULL, "Main Mic"}, - {"IN1LP", NULL, "Main Mic"}, - - {"IN2LN", NULL, "Radio In"}, - {"IN2RN", NULL, "Radio In"}, -}; - -static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); - snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); - - if (machine_is_aquila()) { - snd_soc_dapm_nc_pin(dapm, "SPKOUTRN"); - snd_soc_dapm_nc_pin(dapm, "SPKOUTRP"); - } - - /* Headset jack detection */ - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, - &jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); - if (ret) - return ret; - - ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); - if (ret) - return ret; - - return 0; -} - -static int goni_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int pll_out = 24000000; - int ret = 0; - - /* set the codec FLL */ - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, - params_rate(params) * 256); - if (ret < 0) - return ret; - - /* set the codec system clock */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, - params_rate(params) * 256, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops goni_hifi_ops = { - .hw_params = goni_hifi_hw_params, -}; - -static int goni_voice_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int pll_out = 24000000; - int ret = 0; - - if (params_rate(params) != 8000) - return -EINVAL; - - /* set the codec FLL */ - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, - params_rate(params) * 256); - if (ret < 0) - return ret; - - /* set the codec system clock */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, - params_rate(params) * 256, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_dai_driver voice_dai = { - .name = "goni-voice-dai", - .id = 0, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static const struct snd_soc_component_driver voice_component = { - .name = "goni-voice", -}; - -static struct snd_soc_ops goni_voice_ops = { - .hw_params = goni_voice_hw_params, -}; - -static struct snd_soc_dai_link goni_dai[] = { -{ - .name = "WM8994", - .stream_name = "WM8994 HiFi", - .cpu_dai_name = "samsung-i2s.0", - .codec_dai_name = "wm8994-aif1", - .platform_name = "samsung-i2s.0", - .codec_name = "wm8994-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .init = goni_wm8994_init, - .ops = &goni_hifi_ops, -}, { - .name = "WM8994 Voice", - .stream_name = "Voice", - .cpu_dai_name = "goni-voice-dai", - .codec_dai_name = "wm8994-aif2", - .codec_name = "wm8994-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_IB_IF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &goni_voice_ops, -}, -}; - -static struct snd_soc_card goni = { - .name = "goni", - .owner = THIS_MODULE, - .dai_link = goni_dai, - .num_links = ARRAY_SIZE(goni_dai), - - .dapm_widgets = goni_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(goni_dapm_widgets), - .dapm_routes = goni_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(goni_dapm_routes), -}; - -static int __init goni_init(void) -{ - int ret; - - if (machine_is_aquila()) { - voice_dai.name = aquila_str[CPU_VOICE_DAI]; - goni_dai[1].cpu_dai_name = aquila_str[CPU_VOICE_DAI]; - goni.name = aquila_str[MACHINE_NAME]; - } else if (!machine_is_goni()) - return -ENODEV; - - goni_snd_device = platform_device_alloc("soc-audio", -1); - if (!goni_snd_device) - return -ENOMEM; - - /* register voice DAI here */ - ret = devm_snd_soc_register_component(&goni_snd_device->dev, - &voice_component, &voice_dai, 1); - if (ret) { - platform_device_put(goni_snd_device); - return ret; - } - - platform_set_drvdata(goni_snd_device, &goni); - ret = platform_device_add(goni_snd_device); - - if (ret) - platform_device_put(goni_snd_device); - - return ret; -} - -static void __exit goni_exit(void) -{ - platform_device_unregister(goni_snd_device); -} - -module_init(goni_init); -module_exit(goni_exit); - -/* Module information */ -MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)"); -MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); -MODULE_LICENSE("GPL"); -- 1.9.3

2 1

[alsa-devel] [PATCH 4/4] ASoC: samsung: remove outdated Kconfig references
by Paul Bolle 07 Feb '15

07 Feb '15

The Kconfig symbols MACH_SMDKC110 and MACH_SMDKV210 were removed in v3.17. But the Kconfig entries for SND_SOC_SAMSUNG_SMDK_WM8580 and SND_SOC_SAMSUNG_SMDK_WM9713 still reference them. Perhaps these references to those removed symbols should be changed into references to an existing symbol (such as MACH_S5PV210_DT). But apparently no one noticed that these references were pointless ever since v3.17. So it seems they can simply be removed. Let's do that now. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 5f983d064920..01980bb2c6e7 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && MACH_SMDK6410 depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S @@ -134,7 +134,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && MACH_SMDK6410 select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.9.3

2 1

[alsa-devel] [PATCH 3/4] ASoC: samsung: MACH_SMDKC100 isn't supported anymore
by Paul Bolle 07 Feb '15

07 Feb '15

Commit b8529ec1c1b0 ("ARM: S5PC100: no more support S5PC100 SoC") removed Kconfig symbol MACH_SMDKC100. But there are still two references to that symbol left. These are now pointless. Remove them too. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Apparently Kukjin had a fixup queued (see http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081482.… ). But that fixup never entered the tree, so let's try again. sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3be649098b30..5f983d064920 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S @@ -134,7 +134,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.9.3

2 1

[alsa-devel] [PATCH 2/4] ASoC: samsung: Remove PCM support for WM8580 on SMDK
by Paul Bolle 07 Feb '15

07 Feb '15

Commit 28c8331d386a ("ARM: S5PV210: Remove support for board files") removed the Kconfig symbols MACH_SMDKC110 and MACH_SMDKV210. As a result the dependencies of SND_SOC_SMDK_WM8580_PCM can never be met. So remove the unbuildable "SoC PCM Audio support for WM8580 on SMDK". Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Suggested by Arnd Bergmann (see https://lkml.org/lkml/2014/9/4/381 ). Previously submitted in https://lkml.org/lkml/2014/9/18/147 , but again in an awkward way. So this is a resend too. sound/soc/samsung/Kconfig | 9 -- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/smdk_wm8580pcm.c | 176 ------------------------------------- 3 files changed, 187 deletions(-) delete mode 100644 sound/soc/samsung/smdk_wm8580pcm.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index da6d0229cfa2..3be649098b30 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -153,15 +153,6 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. -config SND_SOC_SMDK_WM8580_PCM - tristate "SoC PCM Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110) - depends on REGMAP_I2C - select SND_SOC_WM8580 - select SND_SAMSUNG_PCM - help - Say Y if you want to add support for SoC audio on the SMDK. - config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 052fe71be518..2b32c371da55 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -36,7 +36,6 @@ snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-smdk-spdif-objs := smdk_spdif.o -snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o snd-soc-tobermory-objs := tobermory.o @@ -62,7 +61,6 @@ obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o -obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c deleted file mode 100644 index 6deec5234c92..000000000000 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ /dev/null @@ -1,176 +0,0 @@ -/* - * sound/soc/samsung/smdk_wm8580pcm.c - * - * Copyright (c) 2011 Samsung Electronics Co. Ltd - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/pcm.h> - -#include <asm/mach-types.h> - -#include "../codecs/wm8580.h" -#include "dma.h" -#include "pcm.h" - -/* - * Board Settings: - * o '1' means 'ON' - * o '0' means 'OFF' - * o 'X' means 'Don't care' - * - * SMDK6410 Base B/D: CFG1-0000, CFG2-1111 - * SMDKC110, SMDKV210: CFGB11-100100, CFGB12-0000 - */ - -#define SMDK_WM8580_EXT_OSC 12000000 -#define SMDK_WM8580_EXT_MCLK 4096000 -#define SMDK_WM8580_EXT_VOICE 2048000 - -static unsigned long mclk_freq; -static unsigned long xtal_freq; - -/* - * If MCLK clock directly gets from XTAL, we don't have to use PLL - * to make MCLK, but if XTAL clock source connects with other codec - * pin (like XTI), we should have to set codec's PLL to make MCLK. - * Because Samsung SoC does not support pcmcdclk output like I2S. - */ - -static int smdk_wm8580_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int rfs, ret; - - switch (params_rate(params)) { - case 8000: - break; - default: - printk(KERN_ERR "%s:%d Sampling Rate %u not supported!\n", - __func__, __LINE__, params_rate(params)); - return -EINVAL; - } - - rfs = mclk_freq / params_rate(params) / 2; - - if (mclk_freq == xtal_freq) { - ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_MCLK, - mclk_freq, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_MCLK); - if (ret < 0) - return ret; - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_PLLA, - mclk_freq, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_PLLA); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, - xtal_freq, mclk_freq); - if (ret < 0) - return ret; - } - - /* Set PCM source clock on CPU */ - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX, - mclk_freq, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Set SCLK_DIV for making bclk */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops smdk_wm8580_pcm_ops = { - .hw_params = smdk_wm8580_pcm_hw_params, -}; - -#define SMDK_DAI_FMT (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | \ - SND_SOC_DAIFMT_CBS_CFS) - -static struct snd_soc_dai_link smdk_dai[] = { - { - .name = "WM8580 PAIF PCM RX", - .stream_name = "Playback", - .cpu_dai_name = "samsung-pcm.0", - .codec_dai_name = "wm8580-hifi-playback", - .platform_name = "samsung-audio", - .codec_name = "wm8580.0-001b", - .dai_fmt = SMDK_DAI_FMT, - .ops = &smdk_wm8580_pcm_ops, - }, { - .name = "WM8580 PAIF PCM TX", - .stream_name = "Capture", - .cpu_dai_name = "samsung-pcm.0", - .codec_dai_name = "wm8580-hifi-capture", - .platform_name = "samsung-pcm.0", - .codec_name = "wm8580.0-001b", - .dai_fmt = SMDK_DAI_FMT, - .ops = &smdk_wm8580_pcm_ops, - }, -}; - -static struct snd_soc_card smdk_pcm = { - .name = "SMDK-PCM", - .owner = THIS_MODULE, - .dai_link = smdk_dai, - .num_links = 2, -}; - -/* - * After SMDKC110 Base Board's Rev is '0.1', 12MHz External OSC(X1) - * is absent (or not connected), so we connect EXT_VOICE_CLK(OSC4), - * 2.0484Mhz, directly with MCLK both Codec and SoC. - */ -static int snd_smdk_probe(struct platform_device *pdev) -{ - int ret = 0; - - xtal_freq = SMDK_WM8580_EXT_OSC; - mclk_freq = SMDK_WM8580_EXT_MCLK; - - if (machine_is_smdkc110() || machine_is_smdkv210()) - xtal_freq = mclk_freq = SMDK_WM8580_EXT_VOICE; - - smdk_pcm.dev = &pdev->dev; - ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); - - return ret; -} - -static struct platform_driver snd_smdk_driver = { - .driver = { - .name = "samsung-smdk-pcm", - }, - .probe = snd_smdk_probe, -}; - -module_platform_driver(snd_smdk_driver); - -MODULE_AUTHOR("Sangbeom Kim, <sbkim73(a)samsung.com>"); -MODULE_DESCRIPTION("ALSA SoC SMDK WM8580 for PCM"); -MODULE_LICENSE("GPL"); -- 1.9.3

2 1

[alsa-devel] [PATCH v2] soc: intel: sst-haswell-ipc: Remove unused functions
by Rickard Strandqvist 07 Feb '15

07 Feb '15

Removes some functions that are not used anywhere: sst_hsw_stream_unmute() sst_hsw_stream_mute() msg_set_stage_type() sst_hsw_dx_get_state() sst_hsw_stream_set_write_position() sst_hsw_stream_get_vol_reg() sst_hsw_stream_get_peak_reg() sst_hsw_stream_get_pointer_reg() sst_hsw_stream_get_read_reg() sst_hsw_stream_get_mixer_id() sst_hsw_stream_get_hw_id() sst_hsw_mixer_set_volume_curve() sst_hsw_mixer_unmute() sst_hsw_mixer_mute() sst_hsw_stream_set_volume_curve() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/intel/sst-haswell-ipc.c | 168 ------------------------------------- sound/soc/intel/sst-haswell-ipc.h | 31 ------- 2 files changed, 199 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c482..d693d4a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -337,12 +337,6 @@ static inline u32 msg_get_stage_type(u32 msg) return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; } -static inline u32 msg_set_stage_type(u32 msg, u32 type) -{ - return (msg & ~IPC_STG_TYPE_MASK) + - (type << IPC_STG_TYPE_SHIFT); -} - static inline u32 msg_get_stream_id(u32 msg) { return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; @@ -969,45 +963,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, } /* Mixer Controls */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, - &stream->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - stream->mute[channel] = 1; - return 0; -} - -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) - -{ - int ret; - - stream->mute[channel] = 0; - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, - stream->mute_volume[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - return 0; -} - int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume) { @@ -1021,17 +976,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream return 0; } -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - stream->vol_req.curve_duration = curve_duration; - stream->vol_req.curve_type = curve; - - return 0; -} - /* stream volume */ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) @@ -1083,42 +1027,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, return 0; } -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, - &hsw->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 1; - return 0; -} - -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, - hsw->mixer_info.volume_register_address[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 0; - return 0; -} - int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume) { @@ -1132,16 +1040,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, return 0; } -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - hsw->curve_duration = curve_duration; - hsw->curve_type = curve; - - return 0; -} - /* global mixer volume */ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume) @@ -1434,48 +1332,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) /* Stream Information - these calls could be inline but we want the IPC ABI to be opaque to client PCM drivers to cope with any future ABI changes */ -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.stream_hw_id; -} - -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.mixer_hw_id; -} - -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.read_position_register_address; -} - -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.presentation_position_register_address; -} - -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.peak_meter_register_address[channel]; -} - -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.volume_register_address[channel]; -} - int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; @@ -1598,30 +1454,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, return ppos; } -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position) -{ - u32 header; - int ret; - - trace_stream_write_position(stream->reply.stream_hw_id, position); - - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | - IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); - header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); - header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); - header |= (stage_id << IPC_STG_ID_SHIFT); - stream->wpos.position = position; - - ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, - sizeof(stream->wpos)); - if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", - stream->reply.stream_hw_id, position); - - return ret; -} - /* physical BE config */ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 138e894..c1ad901 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -376,32 +376,17 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, u32 create_channel_map(enum sst_hsw_channel_config config); /* Stream Mixer Controls - */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); - int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve); - /* Global Mixer Controls - */ -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); - int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume); int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve); - /* Stream API */ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), @@ -440,18 +425,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ @@ -466,8 +439,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position); u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw, struct sst_hsw_stream *stream); u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, @@ -481,8 +452,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, /* DX Config */ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source); /* init */ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); -- 1.7.10.4

2 1

[alsa-devel] [PATCH v3 alsa-lib 1/3] snd_pcm_start_at and friends.
by Tim Cussins 06 Feb '15

06 Feb '15

Upcoming audio hardware has the ability to begin playback when the value of a high-precision free-running counter matches some programmable value. This patch exposes such capability in a general way, allowing for several clock sources, including both system clocks and audio hardware clocks. We define 3 new methods that operate on a snd_pcm_t, and 1 that operates on a snd_pcm_status_t. - snd_pcm_start_at() allows client code to delegate the starting of the stream to the kernel. - snd_pcm_start_at_abort() allows client code to cancel a previous call to snd_pcm_start_at(). - snd_pcm_start_at_gettime() allows client code to query the current time for a given clock. - snd_pcm_status_get_startat_state() allows client code to extract information about any pending start_at timer from a snd_pcm_status_t object. We define a new enum: snd_pcm_startat_clock_type_t, which reinforces a clean boundary between the capabilities of the timestamping system and the start_at system. The clock and audio hw sources are grouped together because the difference is irrelevant to the use of snd_pcm_start_at(). Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/pcm.h b/include/pcm.h index 0655e7f..4f4834b 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -330,6 +330,17 @@ typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; +/** PCM start_at clock types */ +typedef enum _snd_pcm_startat_clock_type { + /** POSIX CLOCK_REALTIME equivalent */ + SND_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY = 0, + /** POSIX CLOCK_MONOTONIC equivalent */ + SND_PCM_STARTAT_CLOCK_TYPE_MONOTONIC, + /** Default link clock */ + SND_PCM_STARTAT_CLOCK_TYPE_LINK, + SND_PCM_STARTAT_CLOCK_TYPE_LAST = SND_PCM_STARTAT_CLOCK_TYPE_LINK, +} snd_pcm_startat_clock_type_t; + /** Unsigned frames quantity */ typedef unsigned long snd_pcm_uframes_t; /** Signed frames quantity */ @@ -478,6 +489,9 @@ int snd_pcm_prepare(snd_pcm_t *pcm); int snd_pcm_reset(snd_pcm_t *pcm); int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status); int snd_pcm_start(snd_pcm_t *pcm); +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, const snd_htimestamp_t* start_time); +int snd_pcm_start_at_abort(snd_pcm_t *pcm); +int snd_pcm_start_at_gettime(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, snd_htimestamp_t* current_time); int snd_pcm_drop(snd_pcm_t *pcm); int snd_pcm_drain(snd_pcm_t *pcm); int snd_pcm_pause(snd_pcm_t *pcm, int enable); @@ -984,6 +998,7 @@ snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_overrange(const snd_pcm_status_t *obj); +void snd_pcm_status_get_startat_state(const snd_pcm_status_t *obj, int *pending, snd_pcm_startat_clock_type_t *clock_type, snd_htimestamp_t *start_time); /** \} */ diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index e74e02f..f2cbd8e 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1085,6 +1085,74 @@ int snd_pcm_start(snd_pcm_t *pcm) } /** + * \brief Start a PCM at a specified point in the future + * \param pcm PCM handle + * \param clock_type Specifies the start_at clock with which to interpret \p start_time + * \param start_time Absolute time at which to start the stream + * \return 0 on success otherwise a negative error code + * \retval -ENOSYS operation not supported for the current timestamp type + * \retval -EINVAL timespec, tstamp_class or tstamp_type is invalid + * \retval -ETIME requested start_time cannot be satisfied + * + * This method is non-blocking: It establishes an appropriate timer in the kernel + * that will start the stream on expiry. + * + * Any pending timer is unconditionally cancelled. + */ +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, const snd_htimestamp_t *start_time) +{ + assert(pcm); + assert(start_time); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at) { + return pcm->fast_ops->start_at(pcm->fast_op_arg, clock_type, start_time); + } + return -EINVAL; +} + +/** + * \brief Abort the pending PCM start_at timer + * \param pcm PCM handle + * \return 0 on success otherwise a negative error code + * + * If a start_at timer is pending, it is cancelled. + */ +int snd_pcm_start_at_abort(snd_pcm_t *pcm) +{ + assert(pcm); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at_abort) { + return pcm->fast_ops->start_at_abort(pcm->fast_op_arg); + } + return -EINVAL; +} + +/** + * \brief Get current time for a given start_at clock + * \param pcm PCM handle + * \param clock_type Start_at clock type e.g. SND_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY + * \return 0 on success otherwise a negative error code + */ +int snd_pcm_start_at_gettime(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, snd_htimestamp_t *current_time) +{ + assert(pcm); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at_gettime) { + return pcm->fast_ops->start_at_gettime(pcm->fast_op_arg, clock_type, current_time); + } + return -EINVAL; +} + +/** * \brief Stop a PCM dropping pending frames * \param pcm PCM handle * \return 0 on success otherwise a negative error code @@ -6380,6 +6448,20 @@ snd_pcm_uframes_t snd_pcm_status_get_overrange(const snd_pcm_status_t *obj) } /** + * \brief Get status of a pending start_at timer, if any. + * \param pending output param: 1 if start_at timer pending, 0 otherwise + * \param clock_type output param: if pending == 1, is updated with current start_at timer clock_type + * \param start_time output param: if pending == 1, is updated with current start_at timer start_time + */ +void snd_pcm_status_get_startat_state(const snd_pcm_status_t *obj, int *pending, snd_pcm_startat_clock_type_t *clock_type, snd_htimestamp_t *start_time) +{ + assert(obj); + *pending = obj->start_at_pending; + *clock_type = obj->start_at_clock_type; + *start_time = obj->start_at_start_time; +} + +/** * \brief get size of #snd_pcm_info_t * \return size in bytes */ -- 1.9.1

1 2

[alsa-devel] [PATCH 2/2] davinci-mcasp: add option to use AHCLKX pin as clock in
by Urs Fässler 06 Feb '15

06 Feb '15

Signed-off-by: Urs Fässler <urs.fassler(a)bytesatwork.ch> --- .../bindings/sound/davinci-mcasp-audio.txt | 1 + sound/soc/davinci/davinci-mcasp.c | 26 +++++++++++++++++++--- 2 files changed, 24 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 46bc982..22a91d2 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -39,6 +39,7 @@ Optional properties: please refer to pinctrl-bindings.txt - fck_parent : Should contain a valid clock name which will be used as parent for the McASP fck +- ahclkx-pin : if set use AHCLKX pin for clock in instead of ACLKX Example: diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 396cdec..3730288 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -85,6 +85,7 @@ struct davinci_mcasp { int sysclk_freq; bool bclk_master; + bool ahclkx_pin; /* McASP FIFO related */ u8 txnumevt; @@ -429,7 +430,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ - mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + if (mcasp->ahclkx_pin) + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); @@ -441,7 +447,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ - mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + if (mcasp->ahclkx_pin) + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); @@ -542,7 +553,12 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); if (mcasp->bclk_master) { - mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + if (mcasp->ahclkx_pin) + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + else + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } else { @@ -1340,6 +1356,8 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( if (ret >= 0) pdata->sram_size_capture = val; + pdata->clk_input_pin = of_property_read_bool(np, "ahclkx-pin"); + return pdata; nodata: @@ -1412,6 +1430,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; } + mcasp->ahclkx_pin = pdata->clk_input_pin; + mcasp->op_mode = pdata->op_mode; /* sanity check for tdm slots parameter */ if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) { -- 2.1.4

2 1

[alsa-devel] [PATCH v3] ASoC: rt286: add jack detection enable/disable APIs and export
by Jie Yang 06 Feb '15

06 Feb '15

Some platforms, e.g. WSB, don't need jack detection when system is in Suspend, for power save reason. Here add jack detection enable/disable APIs and export them, when disabled, it will ignore interrupt, and disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reviewed-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 29 +++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 2 ++ 2 files changed, 31 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index d5cc1bc..1a0c250f 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -402,6 +402,33 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(rt286_mic_detect); +int rt286_enable_jack_detection(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (!jack) + return -EINVAL; + rt286->jack = jack; + if (jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_enable_jack_detection); + +int rt286_disable_jack_detection(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (!jack) + return -EINVAL; + rt286->jack = NULL; + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_disable_jack_detection); + static int is_mclk_mode(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -1012,6 +1039,8 @@ static irqreturn_t rt286_irq(int irq, void *data) bool mic = false; int status = 0; + if (!rt286->jack) + return IRQ_HANDLED; rt286_jack_detect(rt286, &hp, &mic); /* Clear IRQ */ diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index 7130edb..d793185 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -200,6 +200,8 @@ enum { }; int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int rt286_enable_jack_detection(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int rt286_disable_jack_detection(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #endif /* __RT286_H__ */ -- 1.9.1

3 3

[alsa-devel] [PATCH 1/2] davinci-mcasp: use bclk_master flag to decide if the SOC is clock master
by Urs Fässler 06 Feb '15

06 Feb '15

Signed-off-by: Urs Fässler <urs.fassler(a)bytesatwork.ch> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 30b94d4..396cdec 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -541,7 +541,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - if (dir == SND_SOC_CLOCK_OUT) { + if (mcasp->bclk_master) { mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Set up GPIO for Toshiba Satellite S50D
by Takashi Iwai 05 Feb '15

05 Feb '15

Toshiba Satellite S50D laptop with an IDT codec uses the GPIO4 (0x10) as the master EAPD. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=915858 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 605d14003d25..6d36c5b78805 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -99,6 +99,7 @@ enum { STAC_HP_ENVY_BASS, STAC_HP_BNB13_EQ, STAC_HP_ENVY_TS_BASS, + STAC_92HD83XXX_GPIO10_EAPD, STAC_92HD83XXX_MODELS }; @@ -2141,6 +2142,19 @@ static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, spec->headset_jack = 1; } +static void stac92hd83xxx_fixup_gpio10_eapd(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = + spec->gpio_data = 0x10; + spec->eapd_switch = 0; +} + static const struct hda_verb hp_bnb13_eq_verbs[] = { /* 44.1KHz base */ { 0x22, 0x7A6, 0x3E }, @@ -2656,6 +2670,10 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { {} }, }, + [STAC_92HD83XXX_GPIO10_EAPD] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_gpio10_eapd, + }, }; static const struct hda_model_fixup stac92hd83xxx_models[] = { @@ -2861,6 +2879,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x148a, "HP Mini", STAC_92HD83XXX_HP_LED), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_HP, "HP", STAC_92HD83XXX_HP), + SND_PCI_QUIRK(PCI_VENDOR_ID_TOSHIBA, 0xfa91, + "Toshiba Satellite S50D", STAC_92HD83XXX_GPIO10_EAPD), {} /* terminator */ }; -- 2.2.2

1 0

[alsa-devel] ASoC fix for v3.19
by Mark Brown 05 Feb '15

05 Feb '15

The following changes since commit 97fa99a3b8aa83c8ff17ef380dd377efac373812: MAINTAINERS: ASoC: add maintainer for Intel BDW/HSW ASoC driver (2015-02-02 12:00:20 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-intel-v3.19-rc7 for you to fetch changes up to 5c2b06369dafd796ebb4f17dab543d3da500245e: ASoC: Intel: fix sst firmware path for cht-bsw-rt5672 (2015-02-05 20:16:21 +0000) ---------------------------------------------------------------- ASoC: Fix for Intel firmware name Another one liner that arrived after the earlier pull request. There's a trivial conflict with my -next branch, I'll send a pull request with that resolution and some other new stuff before Monday. ---------------------------------------------------------------- Kevin Strasser (1): ASoC: Intel: fix sst firmware path for cht-bsw-rt5672 sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

2 1

[alsa-devel] ASoC AC'97 fix for v3.19
by Mark Brown 05 Feb '15

05 Feb '15

The following changes since commit 97bf6af1f928216fd6c5a66e8a57bfa95a659672: Linux 3.19-rc1 (2014-12-20 17:08:50 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-ac97-v3.19-rc7 for you to fetch changes up to f8d71be5553a3fbc20363243ecfc11dcdd1e18fe: ASoC: wm97xx: Reset AC'97 device before registering it (2015-01-26 19:15:06 +0000) ---------------------------------------------------------------- ASoC: AC'97 fixes These are rather too large for this late in the release cycle but they're clear, well understood and have been tested to fix a regression which was introduced for v3.19. The details are all in Lars' changelog and they've been cooking in -next for a while, to a large extent out of conservatism about the size. ---------------------------------------------------------------- Lars-Peter Clausen (2): ASoC: Add support for allocating AC'97 device before registering it ASoC: wm97xx: Reset AC'97 device before registering it include/sound/soc.h | 1 + sound/soc/codecs/wm9705.c | 16 ++++++++++------ sound/soc/codecs/wm9712.c | 12 ++++++++---- sound/soc/codecs/wm9713.c | 12 ++++++++---- sound/soc/soc-ac97.c | 36 ++++++++++++++++++++++++++++++------ 5 files changed, 57 insertions(+), 20 deletions(-)

2 1

[alsa-devel] ASoC fixes for v3.19
by Mark Brown 05 Feb '15

05 Feb '15

The following changes since commit e36f014edff70fc02b3d3d79cead1d58f289332e: Linux 3.19-rc7 (2015-02-01 20:07:21 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v3.19-rc7 for you to fetch changes up to c503ca52a57cfe24f8937ca145a96ba94f6bb112: Merge remote-tracking branch 'asoc/fix/samsung' into asoc-linus (2015-02-05 19:36:29 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.19 A few last minute fixes for v3.19, all driver specific. None of them stand out particularly - it's all the standard people who are affected will care stuff. The Samsung fix is a DT only fix for the audio controller, it's being merged via the ASoC tree due to process messups (the submitter sent it at the end of a tangentally related series rather than separately to the ARM folks) in order to make sure that it gets to people sooner. ---------------------------------------------------------------- Bo Shen (2): ASoC: atmel_ssc_dai: fix start event for I2S mode ASoC: atmel_ssc_dai: fix the setting for DSP mode Eric Nelson (1): ASoC: sgtl5000: add delay before first I2C access Filip Brozovic (1): ASoC: sgtl5000: Use shift mask when setting codec mode Jarkko Nikula (1): ASoC: rt5640: Add RT5642 ACPI ID for Intel Baytrail Jie Yang (2): ASoC: Intel: Used lock version to update shim registers MAINTAINERS: ASoC: add maintainer for Intel BDW/HSW ASoC driver Manuel Lauss (1): ASoC: wm8731: init mutex in i2c init path Mark Brown (7): Merge remote-tracking branch 'asoc/fix/atmel' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5640' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8731' into asoc-linus Merge remote-tracking branch 'asoc/fix/samsung' into asoc-linus Peter Ujfalusi (1): ASoC: tlv320aic3x: Fix data delay configuration Sylwester Nawrocki (1): ARM: dts: Fix I2S1, I2S2 compatible for exynos4 SoCs MAINTAINERS | 10 ++++++++++ arch/arm/boot/dts/exynos4.dtsi | 4 ++-- sound/soc/atmel/atmel_ssc_dai.c | 24 +++++++----------------- sound/soc/codecs/rt5640.c | 1 + sound/soc/codecs/sgtl5000.c | 13 ++++++++----- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm8731.c | 2 ++ sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 8 files changed, 33 insertions(+), 27 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: Intel: fix sst firmware path
by Kevin Strasser 05 Feb '15

05 Feb '15

All sst firmware is provided under the intel directory of the linux-firmware tree. By default this directory structure is kept when installing on a target system. Change the path to expect a default linux-firmware installation. Signed-off-by: Kevin Strasser <kevin.strasser(a)linux.intel.com> --- sound/soc/intel/sst/sst_acpi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 21b22e6..378ef3c 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -343,16 +343,16 @@ static int sst_acpi_remove(struct platform_device *pdev) } static struct sst_machines sst_acpi_bytcr[] = { - {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + {"10EC5640", "T100", "bytt100_rt5640", NULL, "intel/fw_sst_0f28.bin", &byt_rvp_platform_data }, {}, }; /* Cherryview-based platforms: CherryTrail and Braswell */ static struct sst_machines sst_acpi_chv[] = { - {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "fw_sst_22a8.bin", + {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, - {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "fw_sst_22a8.bin", + {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, {}, }; -- 1.9.1

5 9

[alsa-devel] [PATCH 1/2] ASoC: rt286: Fix potencial crash in jd function
by Bard Liao 05 Feb '15

05 Feb '15

We assign rt286->codec in rt286_probe. If rt286_jack_detect is invoked before rt286_probe, rt286->codec will be NULL and cause a kernel panic. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2eafbcf..3d69cb4 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -306,6 +306,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) *hp = false; *mic = false; + if (!rt286->codec) + return -EINVAL; if (rt286->pdata.cbj_en) { regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); *hp = buf & 0x80000000; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2] ASoC: rt286: add jack detection enable/disable APIs and export
by Jie Yang 05 Feb '15

05 Feb '15

Some platforms, e.g. WSB, don't need jack detection when system is in Suspend, for power save reason. Here add jack detection enable/disable APIs and export them, when disabled, it will ignore interrupt, and disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reviewed-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 24 ++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 2 ++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8104d22..9ea01d0 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -400,6 +400,30 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(rt286_mic_detect); +int rt286_enable_jack_detection(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + enable_irq(rt286->i2c->irq); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_enable_jack_detection); + +int rt286_disable_jack_detection(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + disable_irq(rt286->i2c->irq); + + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_disable_jack_detection); + static int is_mclk_mode(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index b539b73..c958182 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -193,6 +193,8 @@ enum { }; int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int rt286_enable_jack_detection(struct snd_soc_codec *codec); +int rt286_disable_jack_detection(struct snd_soc_codec *codec); #endif /* __RT286_H__ */ -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: Intel:Add Cherrytrail & Braswell machine driver cht_bsw_rt5645
by yang.a.fang＠intel.com 05 Feb '15

05 Feb '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5645 codec Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/Kconfig | 11 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/cht_bsw_rt5645.c | 317 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 330 insertions(+) create mode 100644 sound/soc/intel/cht_bsw_rt5645.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f06fcf1..12093fd 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -110,3 +110,14 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH platforms with RT5672 audio codec. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5645_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5645 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5645 audio codec. + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index e928ec3..a8e53c4 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -28,6 +28,7 @@ snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -35,6 +36,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c new file mode 100644 index 0000000..7a6f1e5 --- /dev/null +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -0,0 +1,317 @@ +/* + * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5645 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Fang, Yang A <yang.a.fang(a)intel.com> + * N,Harshapriya <harshapriya.n(a)intel.com> + * This file is modified from cht_bsw_rt5672.c + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5645.h" +#include "sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "rt5645-aif1" + +struct cht_mc_private { + struct snd_soc_jack hp_jack; + struct snd_soc_jack mic_jack; +}; + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + + codec_dai = cht_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); + return -EIO; + } + if (!SND_SOC_DAPM_EVENT_OFF(event)) + return 0; + + /* Set codec sysclk source to its internal clock because codec PLL will + * be off when idle and MCLK will also be off by ACPI when codec is + * runtime suspended. Codec needs clock for jack detection and button + * press. + */ + snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK, + 0, SND_SOC_CLOCK_IN); + + return 0; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, + params_rate(params) * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + + /* Select clk_i2s1_asrc as ASRC clock source */ + snd_soc_write(codec, RT5645_ASRC_2, 0x1111); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + ret = snd_soc_jack_new(codec, "Headphone Jack", + SND_JACK_HEADPHONE, + &ctx->hp_jack); + if (ret) { + dev_err(runtime->dev, "Headphone jack creation failed\n"); + return ret; + } + + ret = snd_soc_jack_new(codec, "Mic Jack", + SND_JACK_MICROPHONE, + &ctx->mic_jack); + if (ret) { + dev_err(runtime->dev, "Mic jack creation failed\n"); + return ret; + } + + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack); + + return ret; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* CODEC<->CODEC link */ + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5645:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "chtrt5645", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct cht_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5645", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver"); +MODULE_AUTHOR("Fang, Yang A,N,Harshapriya"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5645"); -- 1.7.9.5

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[alsa-devel] [PATCH v3] ASoC: Intel: add a status for runtime suspend/resume
by Jie Yang 05 Feb '15

05 Feb '15

For runtime suspend/resume, it is some different with suspend/resume, e.g. codec power supply won't be switch off, codec jack detection still working(to wake up system from Jack event), won't call call snd_soc_suspend/resume, etc. So here, we add a platform PM status, HSW_PM_STATE_RTD3, to make the status more clear, when in idle, it will enter this status, to transfer from HSW_PM_STATE_RTD3 to HSW_PM_STATE_D3, we will do those extra jobs, and vice versa for resuming. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 70 ++++++++++++++++++++------------------- 1 file changed, 36 insertions(+), 34 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index ad7f4a5..78fa01b 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -119,8 +119,9 @@ struct hsw_pcm_data { }; enum hsw_pm_state { - HSW_PM_STATE_D3 = 0, - HSW_PM_STATE_D0 = 1, + HSW_PM_STATE_D0 = 0, + HSW_PM_STATE_RTD3 = 1, + HSW_PM_STATE_D3 = 2, }; /* private data for the driver */ @@ -1035,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev) struct hsw_priv_data *pdata = dev_get_drvdata(dev); struct sst_hsw *hsw = pdata->hsw; - if (pdata->pm_state == HSW_PM_STATE_D3) + if (pdata->pm_state >= HSW_PM_STATE_RTD3) return 0; sst_hsw_dsp_runtime_suspend(hsw); sst_hsw_dsp_runtime_sleep(hsw); - pdata->pm_state = HSW_PM_STATE_D3; + pdata->pm_state = HSW_PM_STATE_RTD3; return 0; } @@ -1051,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev) struct sst_hsw *hsw = pdata->hsw; int ret; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_RTD3) return 0; ret = sst_hsw_dsp_load(hsw); @@ -1091,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev) struct hsw_pcm_data *pcm_data; int i, err; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_D3) return; err = sst_hsw_dsp_load(hsw); @@ -1139,41 +1140,42 @@ static int hsw_pcm_prepare(struct device *dev) if (pdata->pm_state == HSW_PM_STATE_D3) return 0; - /* suspend all active streams */ - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + else if (pdata->pm_state == HSW_PM_STATE_D0) { + /* suspend all active streams */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; + dev_dbg(dev, "suspending pcm %d\n", i); + snd_pcm_suspend_all(pcm_data->hsw_pcm); + + /* We need to wait until the DSP FW stops the streams */ + msleep(2); + } - if (!pcm_data->substream) - continue; - dev_dbg(dev, "suspending pcm %d\n", i); - snd_pcm_suspend_all(pcm_data->hsw_pcm); + /* preserve persistent memory */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; - /* We need to wait until the DSP FW stops the streams */ - msleep(2); + dev_dbg(dev, "saving context pcm %d\n", i); + err = sst_module_runtime_save(pcm_data->runtime, + &pcm_data->context); + if (err < 0) + dev_err(dev, "failed to save context for PCM %d\n", i); + } + /* enter D3 state and stall */ + sst_hsw_dsp_runtime_suspend(hsw); + /* put the DSP to sleep */ + sst_hsw_dsp_runtime_sleep(hsw); } snd_soc_suspend(pdata->soc_card->dev); snd_soc_poweroff(pdata->soc_card->dev); - /* enter D3 state and stall */ - sst_hsw_dsp_runtime_suspend(hsw); - - /* preserve persistent memory */ - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - - if (!pcm_data->substream) - continue; - - dev_dbg(dev, "saving context pcm %d\n", i); - err = sst_module_runtime_save(pcm_data->runtime, - &pcm_data->context); - if (err < 0) - dev_err(dev, "failed to save context for PCM %d\n", i); - } - - /* put the DSP to sleep */ - sst_hsw_dsp_runtime_sleep(hsw); pdata->pm_state = HSW_PM_STATE_D3; return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH] linux-firmware: update firmware for Intel Broadwell SST DSP
by Jie Yang 05 Feb '15

05 Feb '15

Update firmware intel/IntcSST2.bin from Version 8.4.1.68 to 8.4.1.77 Fix for freeing capture stream just after resetting it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- WHENCE | 1 + intel/IntcSST2.bin | Bin 260256 -> 260320 bytes 2 files changed, 1 insertion(+) diff --git a/WHENCE b/WHENCE index 7f2d6bf..6cb0778 100644 --- a/WHENCE +++ b/WHENCE @@ -2646,6 +2646,7 @@ Licence: Redistributable. See LICENCE.it913x for details Driver: snd_soc_sst_acpi File: intel/IntcSST2.bin +Version: 8.4.1.77 License: Redistributable. 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1 0

[alsa-devel] [PATCH v2] ASoC: Intel: add a status for runtime suspend/resume
by Jie Yang 05 Feb '15

05 Feb '15

For runtime suspend/resume, it is some different with suspend/resume, e.g. codec power supply won't be switch off, codec jack detection still working(to wake up system from Jack event), won't call call snd_soc_suspend/resume, etc. So here, we add a platform PM status, HSW_PM_STATE_RTD3, to make the status more clear, when in idle, it will enter this status, to transfer from HSW_PM_STATE_RTD3 to HSW_PM_STATE_D3, we will do those extra jobs, and vice versa for resuming. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 70 ++++++++++++++++++++------------------- 1 file changed, 36 insertions(+), 34 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index ad7f4a5..edf8eaf 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -119,8 +119,9 @@ struct hsw_pcm_data { }; enum hsw_pm_state { - HSW_PM_STATE_D3 = 0, - HSW_PM_STATE_D0 = 1, + HSW_PM_STATE_D0 = 0, + HSW_PM_STATE_RTD3 = 1, + HSW_PM_STATE_D3 = 2, }; /* private data for the driver */ @@ -1035,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev) struct hsw_priv_data *pdata = dev_get_drvdata(dev); struct sst_hsw *hsw = pdata->hsw; - if (pdata->pm_state == HSW_PM_STATE_D3) + if (pdata->pm_state >= HSW_PM_STATE_RTD3) return 0; sst_hsw_dsp_runtime_suspend(hsw); sst_hsw_dsp_runtime_sleep(hsw); - pdata->pm_state = HSW_PM_STATE_D3; + pdata->pm_state = HSW_PM_STATE_RTD3; return 0; } @@ -1051,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev) struct sst_hsw *hsw = pdata->hsw; int ret; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_RTD3) return 0; ret = sst_hsw_dsp_load(hsw); @@ -1091,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev) struct hsw_pcm_data *pcm_data; int i, err; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_D3) return; err = sst_hsw_dsp_load(hsw); @@ -1139,41 +1140,42 @@ static int hsw_pcm_prepare(struct device *dev) if (pdata->pm_state == HSW_PM_STATE_D3) return 0; - /* suspend all active streams */ - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + else if (pdata->pm_state == HSW_PM_STATE_D0){ + /* suspend all active streams */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; + dev_dbg(dev, "suspending pcm %d\n", i); + snd_pcm_suspend_all(pcm_data->hsw_pcm); + + /* We need to wait until the DSP FW stops the streams */ + msleep(2); + } - if (!pcm_data->substream) - continue; - dev_dbg(dev, "suspending pcm %d\n", i); - snd_pcm_suspend_all(pcm_data->hsw_pcm); + /* preserve persistent memory */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; - /* We need to wait until the DSP FW stops the streams */ - msleep(2); + dev_dbg(dev, "saving context pcm %d\n", i); + err = sst_module_runtime_save(pcm_data->runtime, + &pcm_data->context); + if (err < 0) + dev_err(dev, "failed to save context for PCM %d\n", i); + } + /* enter D3 state and stall */ + sst_hsw_dsp_runtime_suspend(hsw); + /* put the DSP to sleep */ + sst_hsw_dsp_runtime_sleep(hsw); } snd_soc_suspend(pdata->soc_card->dev); snd_soc_poweroff(pdata->soc_card->dev); - /* enter D3 state and stall */ - sst_hsw_dsp_runtime_suspend(hsw); - - /* preserve persistent memory */ - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - - if (!pcm_data->substream) - continue; - - dev_dbg(dev, "saving context pcm %d\n", i); - err = sst_module_runtime_save(pcm_data->runtime, - &pcm_data->context); - if (err < 0) - dev_err(dev, "failed to save context for PCM %d\n", i); - } - - /* put the DSP to sleep */ - sst_hsw_dsp_runtime_sleep(hsw); pdata->pm_state = HSW_PM_STATE_D3; return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: Add a helper to add a new attribute group to card
by Takashi Iwai 05 Feb '15

05 Feb '15

For assigning sysfs entries for a card device from the driver, introduce a new helper function, snd_card_add_dev_attr(). In this way, we can avoid the possible race between the device registration and the sysfs addition / removal. The driver can pass a new attribute group to add freely. This has to be called before snd_card_register(). Currently, up to two extra groups can be added. More than that, it'll return an error. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 3 +++ sound/core/init.c | 31 +++++++++++++++++++++++++------ 2 files changed, 28 insertions(+), 6 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 58882bfacdd7..da5748289968 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -132,6 +132,7 @@ struct snd_card { struct completion *release_completion; struct device *dev; /* device assigned to this card */ struct device card_dev; /* cardX object for sysfs */ + const struct attribute_group *dev_groups[4]; /* assigned sysfs attr */ bool registered; /* card_dev is registered? */ #ifdef CONFIG_PM @@ -262,6 +263,8 @@ void snd_card_set_id(struct snd_card *card, const char *id); int snd_card_register(struct snd_card *card); int snd_card_info_init(void); int snd_card_info_done(void); +int snd_card_add_dev_attr(struct snd_card *card, + const struct attribute_group *group); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); diff --git a/sound/core/init.c b/sound/core/init.c index 96194599e82e..35419054821c 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -181,7 +181,7 @@ void snd_device_initialize(struct device *dev, struct snd_card *card) EXPORT_SYMBOL_GPL(snd_device_initialize); static int snd_card_do_free(struct snd_card *card); -static const struct attribute_group *card_dev_attr_groups[]; +static const struct attribute_group card_dev_attr_group; static void release_card_device(struct device *dev) { @@ -269,7 +269,8 @@ int snd_card_new(struct device *parent, int idx, const char *xid, card->card_dev.parent = parent; card->card_dev.class = sound_class; card->card_dev.release = release_card_device; - card->card_dev.groups = card_dev_attr_groups; + card->card_dev.groups = card->dev_groups; + card->dev_groups[0] = &card_dev_attr_group; err = kobject_set_name(&card->card_dev.kobj, "card%d", idx); if (err < 0) goto __error; @@ -700,14 +701,32 @@ static struct attribute *card_dev_attrs[] = { NULL }; -static struct attribute_group card_dev_attr_group = { +static const struct attribute_group card_dev_attr_group = { .attrs = card_dev_attrs, }; -static const struct attribute_group *card_dev_attr_groups[] = { - &card_dev_attr_group, - NULL +/** + * snd_card_add_dev_attr - Append a new sysfs attribute group to card + * @card: card instance + * @group: attribute group to append + */ +int snd_card_add_dev_attr(struct snd_card *card, + const struct attribute_group *group) +{ + int i; + + /* loop for (arraysize-1) here to keep NULL at the last entry */ + for (i = 0; i < ARRAY_SIZE(card->dev_groups) - 1; i++) { + if (!card->dev_groups[i]) { + card->dev_groups[i] = group; + return 0; + } + } + + dev_err(card->dev, "Too many groups assigned\n"); + return -ENOSPC; }; +EXPORT_SYMBOL_GPL(snd_card_add_dev_attr); /** * snd_card_register - register the soundcard -- 2.2.2

1 1

[alsa-devel] ASoC updates for v3.20
by Mark Brown 05 Feb '15

05 Feb '15

The following changes since commit e36f014edff70fc02b3d3d79cead1d58f289332e: Linux 3.19-rc7 (2015-02-01 20:07:21 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.20-2 for you to fetch changes up to 16ca41c6cc904d7eb9e92afc5f576e375c314f90: Merge remote-tracking branches 'asoc/topic/wm8960', 'asoc/topic/wm8988' and 'asoc/topic/xtfpga' into asoc-next (2015-02-04 20:57:26 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.20 More updates for v3.20: - Lots of refactoring from Lars-Peter Clausen, moving drivers to more data driven initialization and rationalizing a lot of DAPM usage. - Much improved handling of CDCLK clocks on Samsung I2S controllers. - Lots of driver specific cleanups and feature improvements. - CODEC support for TI PCM514x and TLV320AIC3104 devices. - Board support for Tegra systems with Realtek RT5677. ---------------------------------------------------------------- Alexander Morozov (1): ASoC: wm8904: add dt ids table Alexandre Belloni (2): ASoC: atmel: sam9g20_wm8731: remove useless include Asoc: sam9x5_wm8731: depend on ARCH_AT91 instead of SOC_AT91SAM9X5 Anatol Pomozov (3): ASoC: tegra: Add platform driver for rt5677 audio codec ASoC: ts3a227e: Remap keys to match Android headset specification ASoC: ts3a227e: Add dts property that allows to specify micbias voltage Andrew Jackson (9): ASoC: dwc: Remove unnecessary debug messages and tests ASoC: dwc: Reorder code in preparation for DT support ASoC: dwc: Switch to managed clock resource ASoC: dwc: Read I2S block configuration from registers ASoC: dwc: Prepare clock before use ASoC: dwc: Register components with managed interface ASoC: dwc: Add documentation for I2S DT ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Remove unnecessary conditional compilation Arnd Bergmann (3): ASoC: davinci: fix DM365_EVM codec selection ASoC: rt5677: fix SPI dependency ASoC: pxa: make TTC DKB tristate Bard Liao (6): ASoC: rt5670: set platform data by dmi ASoC: rt5670: Add runtime PM support ASoC: rt5670: Keep sysclk on if JD func is used ASoC: rt5670: redefine ASRC control registers 0x84 and 0x85 ASoC: rt286: Fix capture volume setting issue ASoC: rt5645: Add rt5650 codec support Bo Shen (7): ASoC: atmel_ssc_dai: fix start event for I2S mode ASoC: atmel_ssc_dai: remove clock pin comments ASoC: wm8904: add new compatible string ASoC: atmel_ssc_dai: fix the setting for DSP mode ASoC: atmel_ssc_dai: refactor the startup and shutdown ASoC: atmel_ssc_dai: only clean PDC when using PDC ASoC: atmel-pcm-dma: won't check direction when configure dma Charles Keepax (2): ASoC: arizona: Add delay for output disable ASoC: arizona: Coalesce output power up delays Dan Carpenter (1): ASoC: Intel: remove an unused struct member Eric Nelson (1): ASoC: sgtl5000: add delay before first I2C access Fabio Estevam (10): ASoC: fsl_spdif: Use dev_name() for registering the irq ASoC: fsl_asrc: Use dev_name() for registering the irq ASoC: mxs-saif: Do not store the irq number in the private structure ASoC: fsl_ssi: Make error message concise ASoC: fsl_asrc: Make error message concise ASoC: fsl_sai: Make error message concise ASoC: fsl_esai: Make error message concise ASoC: fsl_spdif: Make error message concise ASoC: fsl_ssi: Change irq type to 'int' ASoC: fsl: imx-spdif: Set the card owner field Fang, Yang A (1): ASoC: Intel: Add support rt5645 in sst driver Filip Brozovic (1): ASoC: sgtl5000: Use shift mask when setting codec mode Jarkko Nikula (2): ASoC: pcm: Fix vague codec and cpu DAI prepare error messages ASoC: rt5640: Add RT5642 ACPI ID for Intel Baytrail Jianqun Xu (1): ASoC: rockchip: i2s: set TDL and RDL to 16 samples Jie Yang (8): ASoC: Intel: Store the entry_point read from FW file ASoC: Intel: Always enable DRAM block for FW dump ASoC: Intel: Add stream direction for pcm-module map ASoC: Intel: Split hsw_pcm_data for playback and capture ASoC: Intel: initial stream_hw_id to invalid value ASoC: Intel: Used lock version to update shim registers MAINTAINERS: ASoC: add maintainer for Intel BDW/HSW ASoC driver ASoC: Intel: initial scalar variable ba Jyri Sarha (3): ASoC: simple-card: Enable and disable DAI clocks as needed ASoC: tlv320aic3x: Add support for tlv320aic3104 ASoC: tlv320aic3x: Fix bad comment before intercon_extra_3104 definition Krzysztof Kozlowski (8): ASoC: alc5632: Constify struct regmap_config and snd_soc_codec_driver ASoC: cs35l32: Constify struct regmap_config and snd_soc_codec_driver ASoC: cs42l52: Constify struct regmap_config and snd_soc_codec_driver ASoC: cs42l56: Constify struct regmap_config and snd_soc_codec_driver ASoC: cs42l73: Constify struct regmap_config and snd_soc_codec_driver ASoC: wm8804: Constify struct regmap_config and snd_soc_codec_driver ASoC: wm8988: Constify struct regmap_config and snd_soc_codec_driver ASoC: wm8804: Constify struct regmap_config and snd_soc_codec_driver Kuninori Morimoto (19): ASoC: rsnd: add recovery support for under/over flow error on SRC ASoC: rsnd: remove unsed .rate ASoC: rsnd: tidyup rsnd_io_to_xxx() macro declaration position ASoC: rsnd: rsnd_mod controls clock ASoC: rsnd: rename rsnd_dai_get() to rsnd_rdai_get() ASoC: rsnd: rename rsnd_dai_is_clk_master() to rsnd_rdai_is_clk_master() ASoC: rsnd: remove unused rsnd_dai_id() ASoC: rsnd: remove rsnd_dai_get_platform_info() macro ASoC: rsnd: add rsnd_io_to_rdai() ASoC: rsnd: replace rsnd_dai_is_play() to rsnd_io_is_play() ASoC: rsnd: remove unsed rdai parameter from each function ASoC: rsnd: replace rdai to priv in each function ASoC: rsnd: save priv in struct rsnd_dai ASoC: rsnd: error meesage indicates its port ASoC: rsnd: ignore DIRQ when DMA transfer ASoC: rsnd: tidyup DMA start/stop timing on SSI ASoC: rsnd: tidyup DVC/SRC/SSI order ASoC: rsnd: call missing snd_ctl_remove() ASoC: rsnd: call missing snd_soc_unregiter_component/platform() Lad, Prabhakar (1): ASoC: ts3a227e: fix sparse warning Lars-Peter Clausen (109): ASoC: wm8750: Make VREF a VMID instead of OUTPUT widget ASoC: wm8750: Cleanup manual bias level transitions ASoC: dapm: Don't mark MICBIAS widgets as auto non-connected ASoC: dapm: Simplify fully route card handling ASoC: dmaengine: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: au1x: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: nuc900: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: rcar: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: sh: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: txx9: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: broadwell: Drop unnecessary snd_soc_dapm_enable() calls ASoC: byt-rt5640: Fix snd_soc_dapm_ignore_suspend() calls ASoC: byt-rt5640: Register microphone routes with the card DAPM context ASoC: intel: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: sam9g20_wm8731: Use static DAI format setup ASoC: db1200: Use static DAI format setup ASoC: eureka-tlv320: Use static DAI format setup ASoC: mx27vis-aci32x4: Use static DAI format setup ASoC: wm1133-ev1: Use static DAI format setup ASoC: mxs-sgtl5000: Use static DAI format setup ASoC: ams-delta: Use static DAI format setup ASoC: raumfeld: Use static DAI format setup ASoC: zylonite: Use static DAI format setup ASoC: goni_wm8994: Use static DAI format setup ASoC: h1940_uda1380: Use static DAI format setup ASoC: jive_wm8750: Use static DAI format setup ASoC: neo1973_wm8753: Use static DAI format setup ASoC: rx1950_uda1380: Use static DAI format setup ASoC: s3c24xx_simtec: Use static DAI format setup ASoC: s3c24xx_uda134x: Use static DAI format setup ASoC: smartq_wm8987: Use static DAI format setup ASoC: smdk_wm8580: Use static DAI format setup ASoC: smdk_wm8570pcm: Use static DAI format setup ASoC: smdk_wm8994pcm: Use static DAI format setup ASoC: migor: Use static DAI format setup ASoC: Add helper function for changing the DAI link format ASoC: omap-twl4030: Use snd_soc_runtime_set_dai_fmt() ASoC: mop500_ab8500: Use snd_soc_runtime_set_dai_fmt() ASoC: Export snd_soc_runtime_set_dai_fmt() ASoC: ttc-dkb: Remove unnecessary snd_soc_dapm_disable_pin() calls ASoC: corgi: Automatically disconnect non-connected pins ASoC: e740: Automatically disconnect non-connected pins ASoC: e750: Automatically disconnect non-connected pins ASoC: hx4700: Automatically disconnect non-connected pins ASoC: magician: Automatically disconnect non-connected pins ASoC: palm27x: Fix microphone route ASoC: palm27x: Automatically disconnect non-connected pins ASoC: spitz: Automatically disconnect non-connected pins ASoC: ad193x: Don't power down DAC in CODEC probe ASoC: mc13783: Update set_tdm_slot() semantics ASoC: fsl: Update set_tdm_slot() semantics ASoC: fsl: Remove fsl_asoc_xlate_tdm_slot_mask() ASoC: Update snd_soc_dai_set_tdm_slot() documentation ASoC: 88pm860x: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: ak4671: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: da732x: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: sgtl5000: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: sn95031: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: arizona: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm2000: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm5102: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm5102: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm5110: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8350: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8400: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8731: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8770: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8900: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8903: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8904: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8955: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8961: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8961: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8962: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8988: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8990: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8991: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8993: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8995: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8995: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8996: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8997: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm9081: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm9090: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm9713: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm_adsp: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm_hubs: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: mioa701_wm9713: Fix speaker event ASoC: pcm3008: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320aic31xx: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320aic3x: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320dac33: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: twl4030: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: twl6040: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: alc5623: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: alc5632: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt286: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5631: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5640: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5645: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5651: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5670: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5677: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: Remove codec field from snd_soc_dapm_widget ASoC: jive: Automatically disconnect non-connected pins ASoC: smdk_wm8580: Pass card DAPM context to snd_soc_dapm_disable_pin() ASoC: samsung: Replace depends on REGMAP_I2C with depends on I2C ASoC: Add support for allocating AC'97 device before registering it ASoC: wm97xx: Reset AC'97 device before registering it Manuel Lauss (1): ASoC: wm8731: init mutex in i2c init path Mark Brown (27): ASoC: dapm: Don't use async I/O Merge branch 'topic/rt5670' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge remote-tracking branch 'asoc/fix/fsl-ssi' into asoc-fsl Merge branch 'topic/rt5677' of git://git.kernel.org/.../broonie/sound into asoc-w-codec Merge branch 'topic/dai' of git://git.kernel.org/.../broonie/sound into asoc-samsung Merge branch 'fix/atmel' of git://git.kernel.org/.../broonie/sound into asoc-atmel Merge branch 'topic/rt286' of git://git.kernel.org/.../broonie/sound into asoc-w-codec Merge branch 'fix/rt5677' of git://git.kernel.org/.../broonie/sound into asoc-w-codec Merge branch 'topic/arizona' of git://git.kernel.org/.../broonie/sound into asoc-w-codec Merge remote-tracking branches 'asoc/fix/ac97', 'asoc/fix/atmel', 'asoc/fix/intel', 'asoc/fix/rt286', 'asoc/fix/rt5640', 'asoc/fix/sgtl5000', 'asoc/fix/sta32x', 'asoc/fix/tlv320aic3x' and 'asoc/fix/wm8731' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dmaengine' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branch 'asoc/topic/w-codec' into asoc-next Merge remote-tracking branches 'asoc/topic/ad193x', 'asoc/topic/alc5632', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next Merge remote-tracking branches 'asoc/topic/bt-sco', 'asoc/topic/cs35l32', 'asoc/topic/cs42l52' and 'asoc/topic/cs42l56' into asoc-next Merge remote-tracking branches 'asoc/topic/cs42l73', 'asoc/topic/dai' and 'asoc/topic/davinci' into asoc-next Merge remote-tracking branches 'asoc/topic/doc', 'asoc/topic/dwc', 'asoc/topic/fsi', 'asoc/topic/fsl' and 'asoc/topic/fsl-asrc' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/imx', 'asoc/topic/intel', 'asoc/topic/mxs-saif' and 'asoc/topic/nuc900' into asoc-next Merge remote-tracking branches 'asoc/topic/omap', 'asoc/topic/pxa' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/rockchip', 'asoc/topic/rt5645' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/rx51', 'asoc/topic/samsung', 'asoc/topic/sh', 'asoc/topic/simple' and 'asoc/topic/sta32x' into asoc-next Merge remote-tracking branches 'asoc/topic/tdm-slot', 'asoc/topic/tegra', 'asoc/topic/tlv320aic3x', 'asoc/topic/ts3a227e' and 'asoc/topic/ts3a277e' into asoc-next Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/wm8750', 'asoc/topic/wm8804' and 'asoc/topic/wm8904' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8960', 'asoc/topic/wm8988' and 'asoc/topic/xtfpga' into asoc-next Markus Elfring (3): ASoC: Intel: Delete an unnecessary check before the function call "sst_dma_free" ASoC: fsi: Deletion of unnecessary checks before the function call "clk_enable" ASoC: Intel: Delete an unnecessary check before the function call "release_firmware" Masanari Iida (2): ASoC: Fix typo in bt-sco.c ASoC: Fix warning with make xmldocs caused by soc-devres.c Max Filippov (1): ASoC: add xtensa xtfpga I2S interface and platform Mengdong Lin (2): ASoC: rt5670: add API to select ASRC clock source ASoC: Intel: Select RT5672 ASRC clock source on Cherrytrail and Braswell Oder Chiou (6): ASoC: rt5677: Revise the filter powers ASoC: rt5677: Add the ASRC support ASoC: rt5677: Adjust the routing of "PLL1" ASoC: rt5677: Add the MICBIAS VDD setting in the platform data ASoC: rt5677: Use the regmap functions instead of the snd_soc functions ASoC: rt5677: Add the slot_width "25" support in the TDM mode Peter Rosin (10): ASoC: pcm512x: Also support PCM514x devices ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS ASoC: pcm512x: Fixup warning splat ASoC: pcm512x: Use the correct range constraints for S24_LE ASoC: atmel_ssc_dai: Support SND_SOC_DAIFMT_CBM_CFS on I2S Peter Ujfalusi (7): ASoC: davinci-mcasp: Support for combined tx/rx interrupt line ASoC: rx51: Correct DAPM route for capture ASoC: tlv320aic3x: Fix data delay configuration ASoC: davinci-evm: Do not include edma headers ASoC: davinci-evm: Do not include davinci-pcm.h and davinci-i2s.h ASoC: davicni-mcasp: Mark the common irq line as shared ASoC: davinci-mcasp: Fix DIT only McASP instance support Sylwester Nawrocki (13): ASoC: samsung: i2s: Remove unused gpios field from struct i2s ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clk provider DT binding documentation ASoC: samsung: i2s: Add clock provider for the I2S internal clocks Takashi Iwai (2): ASoC: intel: Remove superfluous backslash in Kconfig ASoC: Add sysfs entries via static attribute groups Thomas Niederprüm (9): ASoC: sta32x: correct bit shift value for IDE register ASoC: sta32x: Convert to direct regmap API usage. ASoC: sta32x: make sta32x a gpio consumer for the reset GPIO ASoC: sta32x: use DECLARE_TLV_DB_RANGE macro. ASoC: sta32x: move code to calculate mclk divider and extrapolation ratio to sta32x_hw_params() ASoC: sta32x: add device tree binding. ASoC: sta32x: use dev_dbg() for debug output. ASoC: sta32x: minor Kconfig update. ASoC: sta32x: change dai name to be in line with the sta350 driver. Vaishali Thakkar (1): ASoC: wm5102: Use put_unaligned_be16 Wei Yongjun (1): ASoC: Intel: fix sparse non static symbol warnings Wolfram Sang (4): ASoC: intel: sst: drop owner assignment from platform_drivers ASoC: omap: drop owner assignment from platform_drivers ASoC: pxa: drop owner assignment from platform_drivers ASoC: samsung: drop owner assignment from platform_drivers Zidan Wang (1): ASoC: wm8960: Let wm8960 codec driver manage its own MCLK kbuild test robot (2): ASoC: Intel: fix platform_no_drv_owner.cocci warnings ASoC: Intel: fix platform_no_drv_owner.cocci warnings .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + .../devicetree/bindings/sound/designware-i2s.txt | 31 + .../bindings/sound/nvidia,tegra-audio-rt5677.txt | 67 ++ .../devicetree/bindings/sound/pcm512x.txt | 28 +- .../devicetree/bindings/sound/samsung-i2s.txt | 22 + .../devicetree/bindings/sound/simple-card.txt | 5 + .../devicetree/bindings/sound/st,sta32x.txt | 92 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 10 +- .../devicetree/bindings/sound/ts3a227e.txt | 5 + Documentation/devicetree/bindings/sound/wm8904.txt | 2 +- MAINTAINERS | 11 + include/dt-bindings/sound/samsung-i2s.h | 8 + include/sound/pcm.h | 12 + include/sound/rcar_snd.h | 1 + include/sound/rt5677.h | 3 + include/sound/simple_card.h | 1 + include/sound/soc-dapm.h | 4 +- include/sound/soc.h | 4 + include/sound/sta32x.h | 18 +- sound/core/pcm_lib.c | 85 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel-pcm-dma.c | 12 +- sound/soc/atmel/atmel_ssc_dai.c | 132 +-- sound/soc/atmel/sam9g20_wm8731.c | 31 +- sound/soc/au1x/db1200.c | 19 +- sound/soc/au1x/dbdma2.c | 6 - sound/soc/au1x/dma.c | 6 - sound/soc/codecs/88pm860x-codec.c | 4 +- sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/ad193x.c | 4 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 8 +- sound/soc/codecs/alc5632.c | 12 +- sound/soc/codecs/arizona.c | 78 +- sound/soc/codecs/arizona.h | 5 + sound/soc/codecs/bt-sco.c | 2 +- sound/soc/codecs/cs35l32.c | 4 +- sound/soc/codecs/cs42l52.c | 4 +- sound/soc/codecs/cs42l56.c | 4 +- sound/soc/codecs/cs42l73.c | 4 +- sound/soc/codecs/da732x.c | 4 +- sound/soc/codecs/mc13783.c | 10 +- sound/soc/codecs/pcm3008.c | 4 +- sound/soc/codecs/pcm512x-i2c.c | 4 + sound/soc/codecs/pcm512x-spi.c | 4 + sound/soc/codecs/pcm512x.c | 932 ++++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- sound/soc/codecs/rt286.c | 51 +- sound/soc/codecs/rt5631.c | 28 +- sound/soc/codecs/rt5640.c | 12 +- sound/soc/codecs/rt5645.c | 174 +++- sound/soc/codecs/rt5645.h | 15 + sound/soc/codecs/rt5651.c | 18 +- sound/soc/codecs/rt5670.c | 137 ++- sound/soc/codecs/rt5670.h | 80 +- sound/soc/codecs/rt5677.c | 243 +++++- sound/soc/codecs/sgtl5000.c | 27 +- sound/soc/codecs/sn95031.c | 33 +- sound/soc/codecs/sta32x.c | 534 +++++++----- sound/soc/codecs/sta32x.h | 2 +- sound/soc/codecs/tlv320aic31xx.c | 9 +- sound/soc/codecs/tlv320aic3x.c | 349 +++++--- sound/soc/codecs/tlv320dac33.c | 9 +- sound/soc/codecs/ts3a227e.c | 35 +- sound/soc/codecs/twl4030.c | 55 +- sound/soc/codecs/twl6040.c | 4 +- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm5100.c | 5 +- sound/soc/codecs/wm5102.c | 23 +- sound/soc/codecs/wm5110.c | 20 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8731.c | 5 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8770.c | 8 +- sound/soc/codecs/wm8804.c | 4 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 35 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8958-dsp2.c | 2 +- sound/soc/codecs/wm8960.c | 51 +- sound/soc/codecs/wm8961.c | 4 +- sound/soc/codecs/wm8962.c | 6 +- sound/soc/codecs/wm8988.c | 6 +- sound/soc/codecs/wm8990.c | 9 +- sound/soc/codecs/wm8991.c | 9 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 23 +- sound/soc/codecs/wm8995.c | 20 +- sound/soc/codecs/wm8996.c | 8 +- sound/soc/codecs/wm8997.c | 11 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9705.c | 16 +- sound/soc/codecs/wm9712.c | 12 +- sound/soc/codecs/wm9713.c | 14 +- sound/soc/codecs/wm_adsp.c | 6 +- sound/soc/codecs/wm_hubs.c | 10 +- sound/soc/davinci/Kconfig | 3 +- sound/soc/davinci/davinci-evm.c | 6 - sound/soc/davinci/davinci-mcasp.c | 103 ++- sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 311 +++++-- sound/soc/fsl/eukrea-tlv320.c | 23 +- sound/soc/fsl/fsl_asrc.c | 5 +- sound/soc/fsl/fsl_asrc.h | 3 - sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 17 +- sound/soc/fsl/fsl_ssi.c | 10 +- sound/soc/fsl/fsl_utils.c | 27 - sound/soc/fsl/fsl_utils.h | 3 - sound/soc/fsl/imx-mc13783.c | 5 +- sound/soc/fsl/imx-spdif.c | 1 + sound/soc/fsl/imx-ssi.c | 5 +- sound/soc/fsl/mx27vis-aic32x4.c | 12 +- sound/soc/fsl/wm1133-ev1.c | 16 +- sound/soc/generic/simple-card.c | 34 + sound/soc/intel/Kconfig | 4 +- sound/soc/intel/broadwell.c | 10 - sound/soc/intel/byt-rt5640.c | 12 +- sound/soc/intel/bytcr_dpcm_rt5640.c | 1 - sound/soc/intel/cht_bsw_rt5672.c | 15 +- sound/soc/intel/sst-baytrail-pcm.c | 6 - sound/soc/intel/sst-dsp.c | 3 +- sound/soc/intel/sst-firmware.c | 3 + sound/soc/intel/sst-haswell-dsp.c | 17 +- sound/soc/intel/sst-haswell-ipc.c | 9 +- sound/soc/intel/sst-haswell-pcm.c | 105 ++- sound/soc/intel/sst-mfld-platform-pcm.c | 7 - sound/soc/intel/sst/sst_acpi.c | 7 +- sound/soc/intel/sst/sst_loader.c | 3 +- sound/soc/mxs/mxs-saif.c | 10 +- sound/soc/mxs/mxs-saif.h | 1 - sound/soc/mxs/mxs-sgtl5000.c | 27 +- sound/soc/nuc900/nuc900-pcm.c | 6 - sound/soc/omap/ams-delta.c | 18 +- sound/soc/omap/omap-hdmi-audio.c | 1 - sound/soc/omap/omap-twl4030.c | 20 +- sound/soc/omap/rx51.c | 8 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/corgi.c | 16 +- sound/soc/pxa/e740_wm9705.c | 20 +- sound/soc/pxa/e750_wm9705.c | 20 +- sound/soc/pxa/hx4700.c | 8 +- sound/soc/pxa/magician.c | 21 +- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/pxa/palm27x.c | 15 +- sound/soc/pxa/raumfeld.c | 35 +- sound/soc/pxa/spitz.c | 23 +- sound/soc/pxa/ttc-dkb.c | 4 - sound/soc/pxa/zylonite.c | 12 +- sound/soc/rockchip/rockchip_i2s.c | 4 + sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/arndale_rt5631.c | 1 - sound/soc/samsung/goni_wm8994.c | 23 +- sound/soc/samsung/h1940_uda1380.c | 15 +- sound/soc/samsung/i2s.c | 362 ++++---- sound/soc/samsung/jive_wm8750.c | 34 +- sound/soc/samsung/neo1973_wm8753.c | 25 +- sound/soc/samsung/odroidx2_max98090.c | 6 +- sound/soc/samsung/rx1950_uda1380.c | 15 +- sound/soc/samsung/s3c24xx_simtec.c | 20 +- sound/soc/samsung/s3c24xx_uda134x.c | 12 +- sound/soc/samsung/smartq_wm8987.c | 16 +- sound/soc/samsung/smdk_wm8580.c | 26 +- sound/soc/samsung/smdk_wm8580pcm.c | 19 +- sound/soc/samsung/smdk_wm8994pcm.c | 16 +- sound/soc/sh/dma-sh7760.c | 6 - sound/soc/sh/fsi.c | 15 +- sound/soc/sh/migor.c | 12 +- sound/soc/sh/rcar/adg.c | 14 +- sound/soc/sh/rcar/core.c | 111 ++- sound/soc/sh/rcar/dvc.c | 63 +- sound/soc/sh/rcar/gen.c | 15 + sound/soc/sh/rcar/rsnd.h | 81 +- sound/soc/sh/rcar/src.c | 269 ++++-- sound/soc/sh/rcar/ssi.c | 100 +-- sound/soc/sh/siu_pcm.c | 1 - sound/soc/soc-ac97.c | 36 +- sound/soc/soc-core.c | 207 +++-- sound/soc/soc-dapm.c | 125 +-- sound/soc/soc-devres.c | 2 +- sound/soc/soc-generic-dmaengine-pcm.c | 15 +- sound/soc/soc-pcm.c | 7 +- sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5677.c | 347 ++++++++ sound/soc/txx9/txx9aclc.c | 6 - sound/soc/ux500/mop500_ab8500.c | 16 +- sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 675 +++++++++++++++ 196 files changed, 5566 insertions(+), 2227 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt create mode 100644 Documentation/devicetree/bindings/sound/st,sta32x.txt create mode 100644 include/dt-bindings/sound/samsung-i2s.h create mode 100644 sound/soc/tegra/tegra_rt5677.c create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c

2 4

[alsa-devel] [PATCH_V4 0/3] sound:soc: jz4740: Add DT, dynamic sampling
by Zubair Lutfullah Kakakhel 05 Feb '15

05 Feb '15

Hi, Here are a few simple patches for the jz4740. First adds dynamic sampling support to jz4740-i2s. Then two to add a simple binding and DT support. Then a patch to set the clock rate for i2s. These are in preparation for jz4780 and ci20 later on. Patches are based on 3.19-rc7. Quite disjoint and stay within jz4740 so should apply easily on other trees. If you would like to have them rebased to a different tree, please tell. Thank-you V4 changes: Added Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Rebased on 3.19-rc7 Removed patch 4 which was redundant V3 changes: Added dma binding to example but forgot to put it in required properties. V2 changes: Removed a redundant #define Added #ifdef CONFIG_OF Added dma in binding document Removed a redundant call to clk_prepare_enable Zubair Lutfullah Kakakhel (3): sound: soc: jz4740: Add dynamic sampling rate support to jz4740-i2s dt: sound: jz4740: Add binding documentation for jz4740-i2s sound: soc: jz4740: Add DT support to jz4740-i2s driver .../bindings/sound/ingenic,jz4740-i2s.txt | 23 ++++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.c | 21 +++++++++++++++++++- 2 files changed, 43 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: Intel: add a status for runtime suspend/resume
by Jie Yang 05 Feb '15

05 Feb '15

For runtime suspend/resume, it is some different with suspend/resume, e.g. codec power supply won't be switch off, codec jack detection still working(to wake up system from Jack event), won't call call snd_soc_suspend/resume, etc. So here, we add a platform PM status, HSW_PM_STATE_RTD3, to make the status clearer, when in idle, it will enter this status, to transfer from HSW_PM_STATE_RTD3 to HSW_PM_STATE_D3, we will do those extra jobs, and vice versa for resuming. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index ad7f4a5..06b149e 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -119,8 +119,9 @@ struct hsw_pcm_data { }; enum hsw_pm_state { - HSW_PM_STATE_D3 = 0, - HSW_PM_STATE_D0 = 1, + HSW_PM_STATE_D0 = 0, + HSW_PM_STATE_RTD3 = 1, + HSW_PM_STATE_D3 = 2, }; /* private data for the driver */ @@ -1035,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev) struct hsw_priv_data *pdata = dev_get_drvdata(dev); struct sst_hsw *hsw = pdata->hsw; - if (pdata->pm_state == HSW_PM_STATE_D3) + if (pdata->pm_state >= HSW_PM_STATE_RTD3) return 0; sst_hsw_dsp_runtime_suspend(hsw); sst_hsw_dsp_runtime_sleep(hsw); - pdata->pm_state = HSW_PM_STATE_D3; + pdata->pm_state = HSW_PM_STATE_RTD3; return 0; } @@ -1051,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev) struct sst_hsw *hsw = pdata->hsw; int ret; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_RTD3) return 0; ret = sst_hsw_dsp_load(hsw); @@ -1091,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev) struct hsw_pcm_data *pcm_data; int i, err; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_D3) return; err = sst_hsw_dsp_load(hsw); @@ -1139,6 +1140,9 @@ static int hsw_pcm_prepare(struct device *dev) if (pdata->pm_state == HSW_PM_STATE_D3) return 0; + else if (pdata->pm_state == HSW_PM_STATE_RTD3) + goto suspend; + /* suspend all active streams */ for (i = 0; i < ARRAY_SIZE(mod_map); i++) { pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; @@ -1152,9 +1156,6 @@ static int hsw_pcm_prepare(struct device *dev) msleep(2); } - snd_soc_suspend(pdata->soc_card->dev); - snd_soc_poweroff(pdata->soc_card->dev); - /* enter D3 state and stall */ sst_hsw_dsp_runtime_suspend(hsw); @@ -1174,6 +1175,11 @@ static int hsw_pcm_prepare(struct device *dev) /* put the DSP to sleep */ sst_hsw_dsp_runtime_sleep(hsw); + +suspend: + snd_soc_suspend(pdata->soc_card->dev); + snd_soc_poweroff(pdata->soc_card->dev); + pdata->pm_state = HSW_PM_STATE_D3; return 0; -- 1.9.1

3 2

[alsa-devel] [PATCH] sound: soc: codecs/ts3a227e: fix sparse warning
by Lad Prabhakar 04 Feb '15

04 Feb '15

From: "Lad, Prabhakar" <prabhakar.csengg(a)gmail.com> this patch fixes following sparse warning: ts3a227e.c:222:5: warning: symbol 'ts3a227e_enable_jack_detect' was not declared. Should it be static? Signed-off-by: Lad, Prabhakar <prabhakar.csengg(a)gmail.com> --- Found this issue on linux-next (gcc version 4.8.2, sparse version 0.4.5-rc1)and applies on top linux-next. sound/soc/codecs/ts3a227e.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 5ecfa31..9fd80ac 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -20,6 +20,8 @@ #include <sound/jack.h> #include <sound/soc.h> +#include "ts3a227e.h" + struct ts3a227e { struct regmap *regmap; struct snd_soc_jack *jack; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add support for tlv320aic3104
by Jyri Sarha 04 Feb '15

04 Feb '15

Disables GPIO support and LINE2 input and renames Mic3 input to Mic2, if tlv320aic3104 mode is seleced. Devicetree binding document is updated accordingly. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- .../devicetree/bindings/sound/tlv320aic3x.txt | 10 +- sound/soc/codecs/tlv320aic3x.c | 344 ++++++++++++++------ 2 files changed, 252 insertions(+), 102 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 5e6040c..47a213c 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -9,6 +9,7 @@ Required properties: "ti,tlv320aic33" - TLV320AIC33 "ti,tlv320aic3007" - TLV320AIC3007 "ti,tlv320aic3106" - TLV320AIC3106 + "ti,tlv320aic3104" - TLV320AIC3104 - reg - <int> - I2C slave address @@ -18,6 +19,7 @@ Optional properties: - gpio-reset - gpio pin number used for codec reset - ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + - Not supported on tlv320aic3104 - ai3x-micbias-vg - MicBias Voltage required. 1 - MICBIAS output is powered to 2.0V, 2 - MICBIAS output is powered to 2.5V, @@ -36,7 +38,13 @@ CODEC output pins: * HPLCOM * HPRCOM -CODEC input pins: +CODEC input pins for TLV320AIC3104: + * MIC2L + * MIC2R + * LINE1L + * LINE1R + +CODEC input pins for other compatible codecs: * MIC3L * MIC3R * LINE1L diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b7ebce0..49ba3c7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -87,6 +87,7 @@ struct aic3x_priv { #define AIC3X_MODEL_3X 0 #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 +#define AIC3X_MODEL_3104 3 u16 model; /* Selects the micbias voltage */ @@ -316,52 +317,37 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * only for swapped L-to-R and R-to-L routes. See below stereo controls * for direct L-to-L and R-to-R routes. */ - SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume", - LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left Line Mixer PGAR Bypass Volume", PGAR_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left Line Mixer DACR1 Playback Volume", DACR1_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume", - LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right Line Mixer PGAL Bypass Volume", PGAL_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right Line Mixer DACL1 Playback Volume", DACL1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume", - LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HP Mixer PGAR Bypass Volume", PGAR_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HP Mixer DACR1 Playback Volume", DACR1_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume", - LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HP Mixer PGAL Bypass Volume", PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HP Mixer DACL1 Playback Volume", DACL1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume", - LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HPCOM Mixer PGAR Bypass Volume", PGAR_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HPCOM Mixer DACR1 Playback Volume", DACR1_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume", - LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HPCOM Mixer PGAL Bypass Volume", PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HPCOM Mixer DACL1 Playback Volume", DACL1_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), /* Stereo output controls for direct L-to-L and R-to-R routes */ - SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume", - LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("Line PGA Bypass Volume", PGAL_2_LLOPM_VOL, PGAR_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), @@ -369,9 +355,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", - LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("HP PGA Bypass Volume", PGAL_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), @@ -379,9 +362,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume", - LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("HPCOM PGA Bypass Volume", PGAL_2_HPLCOM_VOL, PGAR_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), @@ -424,6 +404,45 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), }; +/* For other than tlv320aic3104 */ +static const struct snd_kcontrol_new aic3x_extra_snd_controls[] = { + /* + * Output controls that map to output mixer switches. Note these are + * only for swapped L-to-R and R-to-L routes. See below stereo controls + * for direct L-to-L and R-to-R routes. + */ + SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume", + LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume", + LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume", + LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume", + LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume", + LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume", + LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), + + /* Stereo output controls for direct L-to-L and R-to-R routes */ + SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume", + LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", + LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume", + LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), +}; + static const struct snd_kcontrol_new aic3x_mono_controls[] = { SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, @@ -464,22 +483,24 @@ SOC_DAPM_ENUM("Route", aic3x_right_hpcom_enum); /* Left Line Mixer */ static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_LLOPM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), }; /* Right Line Mixer */ static const struct snd_kcontrol_new aic3x_right_line_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_RLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), }; /* Mono Mixer */ @@ -494,42 +515,46 @@ static const struct snd_kcontrol_new aic3x_mono_mixer_controls[] = { /* Left HP Mixer */ static const struct snd_kcontrol_new aic3x_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLOUT_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0), }; /* Right HP Mixer */ static const struct snd_kcontrol_new aic3x_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPROUT_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), }; /* Left HPCOM Mixer */ static const struct snd_kcontrol_new aic3x_left_hpcom_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLCOM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0), }; /* Right HPCOM Mixer */ static const struct snd_kcontrol_new aic3x_right_hpcom_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPRCOM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), }; /* Left PGA Mixer */ @@ -550,6 +575,22 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), }; +/* Left PGA Mixer for tlv320aic3104 */ +static const struct snd_kcontrol_new aic3104_left_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1), +}; + +/* Right PGA Mixer for tlv320aic3104 */ +static const struct snd_kcontrol_new aic3104_right_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), +}; + /* Left Line1 Mux */ static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls = SOC_DAPM_ENUM("Route", aic3x_line1l_2_l_enum); @@ -593,26 +634,56 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), - SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_left_pga_mixer_controls[0], - ARRAY_SIZE(aic3x_left_pga_mixer_controls)), SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1l_mux_controls), SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1r_mux_controls), - SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, - &aic3x_left_line2_mux_controls), /* Inputs to Right ADC */ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", LINE1R_2_RADC_CTRL, 2, 0), - SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_right_pga_mixer_controls[0], - ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1l_mux_controls), SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1r_mux_controls), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("LLOUT"), + SND_SOC_DAPM_OUTPUT("RLOUT"), + SND_SOC_DAPM_OUTPUT("HPLOUT"), + SND_SOC_DAPM_OUTPUT("HPROUT"), + SND_SOC_DAPM_OUTPUT("HPLCOM"), + SND_SOC_DAPM_OUTPUT("HPRCOM"), + + SND_SOC_DAPM_INPUT("LINE1L"), + SND_SOC_DAPM_INPUT("LINE1R"), + + /* + * Virtual output pin to detection block inside codec. This can be + * used to keep codec bias on if gpio or detection features are needed. + * Force pin on or construct a path with an input jack and mic bias + * widgets. + */ + SND_SOC_DAPM_OUTPUT("Detection"), +}; + +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_widget aic3x_extra_dapm_widgets[] = { + /* Inputs to Left ADC */ + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_left_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line2_mux_controls), + + /* Inputs to Right ADC */ + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), @@ -637,11 +708,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), - /* Mic Bias */ - SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, - mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - /* Output mixers */ SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_line_mixer_controls[0], @@ -662,27 +728,46 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_right_hpcom_mixer_controls[0], ARRAY_SIZE(aic3x_right_hpcom_mixer_controls)), - SND_SOC_DAPM_OUTPUT("LLOUT"), - SND_SOC_DAPM_OUTPUT("RLOUT"), - SND_SOC_DAPM_OUTPUT("HPLOUT"), - SND_SOC_DAPM_OUTPUT("HPROUT"), - SND_SOC_DAPM_OUTPUT("HPLCOM"), - SND_SOC_DAPM_OUTPUT("HPRCOM"), - SND_SOC_DAPM_INPUT("MIC3L"), SND_SOC_DAPM_INPUT("MIC3R"), - SND_SOC_DAPM_INPUT("LINE1L"), - SND_SOC_DAPM_INPUT("LINE1R"), SND_SOC_DAPM_INPUT("LINE2L"), SND_SOC_DAPM_INPUT("LINE2R"), +}; - /* - * Virtual output pin to detection block inside codec. This can be - * used to keep codec bias on if gpio or detection features are needed. - * Force pin on or construct a path with an input jack and mic bias - * widgets. - */ - SND_SOC_DAPM_OUTPUT("Detection"), +/* For tlv320aic3104 */ +static const struct snd_soc_dapm_widget aic3104_extra_dapm_widgets[] = { + /* Inputs to Left ADC */ + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3104_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3104_left_pga_mixer_controls)), + + /* Inputs to Right ADC */ + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3104_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3104_right_pga_mixer_controls)), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_line_mixer_controls[0], + ARRAY_SIZE(aic3x_left_line_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_line_mixer_controls[0], + ARRAY_SIZE(aic3x_right_line_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hp_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hp_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Left HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hpcom_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hpcom_mixer_controls) - 2), + + SND_SOC_DAPM_INPUT("MIC2L"), + SND_SOC_DAPM_INPUT("MIC2R"), }; static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = { @@ -712,17 +797,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Line1R Mux", "single-ended", "LINE1R"}, {"Left Line1R Mux", "differential", "LINE1R"}, - {"Left Line2L Mux", "single-ended", "LINE2L"}, - {"Left Line2L Mux", "differential", "LINE2L"}, - {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, - {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, - {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, - {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Left ADC", NULL, "Left PGA Mixer"}, - {"Left ADC", NULL, "GPIO1 dmic modclk"}, /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, @@ -730,25 +808,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line1L Mux", "single-ended", "LINE1L"}, {"Right Line1L Mux", "differential", "LINE1L"}, - {"Right Line2R Mux", "single-ended", "LINE2R"}, - {"Right Line2R Mux", "differential", "LINE2R"}, - {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"}, {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"}, - {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, - {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, - {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, - {"Right ADC", NULL, "GPIO1 dmic modclk"}, - - /* - * Logical path between digital mic enable and GPIO1 modulator clock - * output function - */ - {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, - {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, - {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, /* Left DAC Output */ {"Left DAC Mux", "DAC_L1", "Left DAC"}, @@ -761,10 +824,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right DAC Mux", "DAC_R3", "Right DAC"}, /* Left Line Output */ - {"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left Line Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left Line Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -773,10 +834,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"LLOUT", NULL, "Left Line Out"}, /* Right Line Output */ - {"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right Line Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right Line Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -785,10 +844,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"RLOUT", NULL, "Right Line Out"}, /* Left HP Output */ - {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left HP Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left HP Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -797,10 +854,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPLOUT", NULL, "Left HP Out"}, /* Right HP Output */ - {"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right HP Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right HP Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -809,10 +864,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPROUT", NULL, "Right HP Out"}, /* Left HPCOM Output */ - {"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -823,10 +876,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPLCOM", NULL, "Left HP Com"}, /* Right HPCOM Output */ - {"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -839,6 +890,72 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPRCOM", NULL, "Right HP Com"}, }; +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_route intercon_extra[] = { + /* Left Input */ + {"Left Line2L Mux", "single-ended", "LINE2L"}, + {"Left Line2L Mux", "differential", "LINE2L"}, + + {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, + {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Left ADC", NULL, "GPIO1 dmic modclk"}, + + /* Right Input */ + {"Right Line2R Mux", "single-ended", "LINE2R"}, + {"Right Line2R Mux", "differential", "LINE2R"}, + + {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, + {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Right ADC", NULL, "GPIO1 dmic modclk"}, + + /* + * Logical path between digital mic enable and GPIO1 modulator clock + * output function + */ + {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, + + /* Left Line Output */ + {"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right Line Output */ + {"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Left HP Output */ + {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right HP Output */ + {"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Left HPCOM Output */ + {"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right HPCOM Output */ + {"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, +}; + +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_route intercon_extra_3104[] = { + /* Left Input */ + {"Left PGA Mixer", "Mic2L Switch", "MIC2L"}, + {"Left PGA Mixer", "Mic2R Switch", "MIC2R"}, + + /* Right Input */ + {"Right PGA Mixer", "Mic2L Switch", "MIC2L"}, + {"Right PGA Mixer", "Mic2R Switch", "MIC2R"}, +}; + static const struct snd_soc_dapm_route intercon_mono[] = { /* Mono Output */ {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, @@ -867,17 +984,31 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets, + ARRAY_SIZE(aic3x_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra, + ARRAY_SIZE(intercon_extra)); snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets, ARRAY_SIZE(aic3x_dapm_mono_widgets)); snd_soc_dapm_add_routes(dapm, intercon_mono, ARRAY_SIZE(intercon_mono)); break; case AIC3X_MODEL_3007: + snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets, + ARRAY_SIZE(aic3x_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra, + ARRAY_SIZE(intercon_extra)); snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_3007, ARRAY_SIZE(intercon_3007)); break; + case AIC3X_MODEL_3104: + snd_soc_dapm_new_controls(dapm, aic3104_extra_dapm_widgets, + ARRAY_SIZE(aic3104_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra_3104, + ARRAY_SIZE(intercon_extra_3104)); + break; } return 0; @@ -1438,23 +1569,33 @@ static int aic3x_probe(struct snd_soc_codec *codec) aic3x_init(codec); if (aic3x->setup) { - /* setup GPIO functions */ - snd_soc_write(codec, AIC3X_GPIO1_REG, - (aic3x->setup->gpio_func[0] & 0xf) << 4); - snd_soc_write(codec, AIC3X_GPIO2_REG, - (aic3x->setup->gpio_func[1] & 0xf) << 4); + if (aic3x->model != AIC3X_MODEL_3104) { + /* setup GPIO functions */ + snd_soc_write(codec, AIC3X_GPIO1_REG, + (aic3x->setup->gpio_func[0] & 0xf) << 4); + snd_soc_write(codec, AIC3X_GPIO2_REG, + (aic3x->setup->gpio_func[1] & 0xf) << 4); + } else { + dev_warn(codec->dev, "GPIO functionality is not supported on tlv320aic3104\n"); + } } switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + snd_soc_add_codec_controls(codec, aic3x_extra_snd_controls, + ARRAY_SIZE(aic3x_extra_snd_controls)); snd_soc_add_codec_controls(codec, aic3x_mono_controls, ARRAY_SIZE(aic3x_mono_controls)); break; case AIC3X_MODEL_3007: + snd_soc_add_codec_controls(codec, aic3x_extra_snd_controls, + ARRAY_SIZE(aic3x_extra_snd_controls)); snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); break; + case AIC3X_MODEL_3104: + break; } /* set mic bias voltage */ @@ -1522,6 +1663,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, { "tlv320aic3106", AIC3X_MODEL_3X }, + { "tlv320aic3104", AIC3X_MODEL_3104 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); -- 1.7.9.5

4 10

[alsa-devel] [PATCH] ASoC: rt286: add irq enable/disable APIs and export
by Jie Yang 04 Feb '15

04 Feb '15

Some platforms, e.g. WSB, don't need irq handler and jack detection when system is in Suspend, for power save reason. Here add irq enable/disable APIs and export them, also disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reviewed-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 24 ++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 2 ++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8104d22..66afb9b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -400,6 +400,30 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(rt286_mic_detect); +int rt286_enable_irq(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + enable_irq(rt286->i2c->irq); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_enable_irq); + +int rt286_disable_irq(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + disable_irq(rt286->i2c->irq); + + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_disable_irq); + static int is_mclk_mode(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index b539b73..fb18331 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -193,6 +193,8 @@ enum { }; int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int rt286_enable_irq(struct snd_soc_codec *codec); +int rt286_disable_irq(struct snd_soc_codec *codec); #endif /* __RT286_H__ */ -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: Intel: initial scalar variable ba
by Jie Yang 04 Feb '15

04 Feb '15

Reported by Coverity: CID 1267985 CID 1267986 Fix these two Defects: Uninitialized scalar variable. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index a2ae2c5..5f71ef6 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -791,6 +791,7 @@ int sst_module_alloc_blocks(struct sst_module *module) struct sst_block_allocator ba; int ret; + memset(&ba, 0, sizeof(ba)); ba.size = module->size; ba.type = module->type; ba.offset = module->offset; @@ -864,6 +865,7 @@ int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime, if (module->persistent_size == 0) return 0; + memset(&ba, 0, sizeof(ba)); ba.size = module->persistent_size; ba.type = SST_MEM_DRAM; -- 1.9.1

2 1

[alsa-devel] [RFC PATCH] ASoC: wm8731: let codec to manage clock by itself
by Bo Shen 04 Feb '15

04 Feb '15

Let the wm8731 codec to manage clock by itself. As all at91 related boards have been switch to CCF framework. So, on the at91 related boards which use wm8731 will let it manage the clock, or else the board use wm8731 is broken. However, at the same time the wm8731 codec is used on other boards, I am sure this change will broken some boards. For example: poodle and corgi based on PXA SoC (in default configuration file, no one use it). DB1200 board which is a mips based board. So I have no idea how to fix them. So, my suggestion is to add CCF check based on the following patch? Any idea or suggestions? Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8731.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b9211b4..83f75d66 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -13,6 +13,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -45,6 +46,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; @@ -389,6 +391,8 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8731_SYSCLK_XTAL: case WM8731_SYSCLK_MCLK: + if (clk_set_rate(wm8731->mclk, freq)) + return -EINVAL; wm8731->sysclk_type = clk_id; break; default: @@ -490,6 +494,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + clk_prepare_enable(wm8731->mclk); break; case SND_SOC_BIAS_PREPARE: break; @@ -508,6 +513,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -666,6 +672,13 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + dev_err(&spi->dev, "Failed to get MCLK\n"); + return ret; + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); @@ -717,6 +730,13 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + dev_err(&i2c->dev, "Failed to get MCLK\n"); + return ret; + } + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); -- 2.3.0.rc0

5 10

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix bad comment before intercon_extra_3104 definition
by Jyri Sarha 04 Feb '15

04 Feb '15

The intercon_extra_3104 is obviously for tlv320aic3104. Reported-by: Benoît Thébaudeau <benoit.thebaudeau.dev(a)gmail.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cb92cdb..ed35e8f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -945,7 +945,7 @@ static const struct snd_soc_dapm_route intercon_extra[] = { {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, }; -/* For other than tlv320aic3104 */ +/* For tlv320aic3104 */ static const struct snd_soc_dapm_route intercon_extra_3104[] = { /* Left Input */ {"Left PGA Mixer", "Mic2L Switch", "MIC2L"}, -- 1.9.1

2 1

[alsa-devel] Single vs. multiple seq client ports
by Tim E. Real 04 Feb '15

04 Feb '15

Hi. Our app currently uses a single alsa midi seq client port to address all external midi ports. I have an opportunity now to change this to allow as many client ports as desired, for example one client port for each external port found. If I proceed, it is possible the user may have many client ports if desired. What are the advantages and disadvantages if any, in terms of speed and resources, of having a single client port versus multiple client ports? Thanks. Tim.

2 1

[alsa-devel] [PATCH V3 00/15] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 03 Feb '15

03 Feb '15

This series is an attempt to resolve the CDCLK clock gating issue on Odroid X2/U3 as reported by Daniel Drake [1], by exposing the CDCLK gate clock (and the two other clocks) through clk API. The upside is we can switch Odroid X2/U3 to the simple-card, once the CDCLK clock is taken care of by the clk core and DT. Changes since v2: - skipped the first, already merged patch, - modified description of the patch moving clk_prepare_enable() from DAI to the platform device probe(), - the last patch marked for stable. The patch series has been created on top of branch: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/samsung Changes since the first version: - removed check for the i2s_opclk1 mux input clock while creating the mux and div clocks, - the DT binding documentation changes reworked (addressing review comments), - added include/dt-bindings/sound/samsung-i2s.h header file defining the clk indices, it's been put into a separate patch together with the I2S DT binding documentation updates to make merging of the ASoC and the dts patches separately easier, - a patch fixing compatible strings of I2S1, I2S2 in exynos4.dtsi is included in this series. This whole series may need more testing on other SoCs, so far I only tested it on Odroid Exynos4412 X2, with the I2S working in slave mode. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081753.… Sylwester Nawrocki (15): ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clk provider DT binding documentation ASoC: samsung: i2s: Add clock provider for the I2S internal clocks ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update ARM: dts: Switch Odroid X2/U2 to simple-audio-card ARM: dts: Fix I2S1, I2S2 compatible for exynos4 SoCs .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++ arch/arm/boot/dts/exynos4.dtsi | 13 +- arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 27 +- arch/arm/boot/dts/exynos4412-odroidu3.dts | 8 +- arch/arm/boot/dts/exynos4412-odroidx2.dts | 8 +- include/dt-bindings/sound/samsung-i2s.h | 8 + sound/soc/samsung/i2s.c | 361 ++++++++++++-------- sound/soc/samsung/odroidx2_max98090.c | 6 +- 8 files changed, 295 insertions(+), 158 deletions(-) create mode 100644 include/dt-bindings/sound/samsung-i2s.h -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: sgtl5000: add delay before first I2C access
by Eric Nelson 03 Feb '15

03 Feb '15

To quote from section 1.3.1 of the data sheet: The SGTL5000 has an internal reset that is deasserted 8 SYS_MCLK cycles after all power rails have been brought up. After this time, communication can start ... 1.0μs represents 8 SYS_MCLK cycles at the minimum 8.0 MHz SYS_MCLK. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 29cf7ce..7d65a5b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1462,6 +1462,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (ret) return ret; + /* Need 8 clocks before I2C accesses */ + usleep_range(1, 2); + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); if (ret) -- 1.9.1

4 7

[alsa-devel] [PATCH] add Steinberg UR44 USB Audio Interface support to quirks-table
by Cyrus Harmon 03 Feb '15

03 Feb '15

Signed-off-by: Cyrus Harmon <cyrus(a)bobobeach.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..987ca01 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -436,6 +436,36 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR44", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- 1.8.3.2

5 32

[alsa-devel] [PATCH 2/2] ALSA: line6: fixup of line6_start_timer argument type
by Nicholas Mc Guire 03 Feb '15

03 Feb '15

line6_start_timer passes an unsigned int as argument to be used in mod_timer which is then used by mod_timer as unsigned long, this just fixes up the argument type. This change helps make static code checkers happy. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- The change of argument type was checked against all call-sites - in all cases it is passing in a constant (either VARIAX_STARTUP_DELAY1 or POD_STARTUP_DELAY) which should be fine with the argument type change. Patch was compile tested for x86_64_defconfig + CONFIG_SOUND=y, CONFIG_SND_USB_POD=m (implies CONFIG_SND_USB_LINE6=y) Patch is against 3.0.19-rc6 (localversion = -next-20150202) sound/usb/line6/driver.c | 2 +- sound/usb/line6/driver.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 3cf12cc..e2a2603 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -162,7 +162,7 @@ static int line6_send_raw_message_async_part(struct message *msg, /* Setup and start timer. */ -void line6_start_timer(struct timer_list *timer, unsigned int msecs, +void line6_start_timer(struct timer_list *timer, unsigned long msecs, void (*function)(unsigned long), unsigned long data) { setup_timer(timer, function, data); diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index fa877a3..2276b78 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -159,7 +159,7 @@ extern int line6_send_sysex_message(struct usb_line6 *line6, const char *buffer, int size); extern ssize_t line6_set_raw(struct device *dev, struct device_attribute *attr, const char *buf, size_t count); -extern void line6_start_timer(struct timer_list *timer, unsigned int msecs, +extern void line6_start_timer(struct timer_list *timer, unsigned long msecs, void (*function)(unsigned long), unsigned long data); extern int line6_version_request_async(struct usb_line6 *line6); -- 1.7.10.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: line6: use msecs_to_jiffies for conversion
by Nicholas Mc Guire 03 Feb '15

03 Feb '15

This is only an API consolidation and should make things more readable it replaces var * HZ / 1000 by msecs_to_jiffies(var). Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Converting milliseconds to jiffies by val * HZ / 1000 is technically not wrong but msecs_to_jiffies(val) is the cleaner solution and handles all corner cases correctly. This is a minor API cleanup only. Patch was only compile tested for x86_64_defconfig + CONFIG_SOUND=y, CONFIG_SND_USB_POD=m (implies CONFIG_SND_USB_LINE6=y) Patch is against 3.0.19-rc6 (localversion = -next-20150202) sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 2328ec9..3cf12cc 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -166,7 +166,7 @@ void line6_start_timer(struct timer_list *timer, unsigned int msecs, void (*function)(unsigned long), unsigned long data) { setup_timer(timer, function, data); - mod_timer(timer, jiffies + msecs * HZ / 1000); + mod_timer(timer, jiffies + msecs_to_jiffies(msecs)); } EXPORT_SYMBOL_GPL(line6_start_timer); -- 1.7.10.4

2 1

[alsa-devel] [PATCH v4 0/3] Sound support for Exynos4412 Trats2 board
by Inha Song 03 Feb '15

03 Feb '15

This patch-set adds basic sound support for the Trats2 boards. It just support primary I2s and external speaker playback. Changes in v4: - Move to clock out parent settings in exynos4412-trats2.dts - Remove unnecessary lines in Trats2 machine driver. - Change name of dai stream to more cleary. - Add MFD_WM8994 to select in machine driver kconfig. Changes in v3: - Set clock out parent to XUSBXTI in pmu_system_controller DT node. Changes in v2: - Remove unnecessary lines in Trats2 machine driver. - move clock enable/disable to .startup / .shutdown. - Fix code style issues. - Remove mclk DT property in Trats2 sound and add MCLK1 property in WM1811 node. (MCLK1 is optional property in WM8994 node - refer: Document/devicetree/bindings/sound/wm8994.txt) - Use mclk that defined in WM1811 DT node. - Add "samsung,audio-codec" property to required properties. Inha Song (3): ASoC: samsung: Add machine driver for Trats2 ASoC: samsung: Document Trats2 audio subsystem bindings ARM: dts: Add sound nodes for exynos4412-trats2 .../bindings/sound/samsung,trats2-wm1811.txt | 25 +++ arch/arm/boot/dts/exynos4412-trats2.dts | 42 ++++ sound/soc/samsung/Kconfig | 9 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/trats2_wm1811.c | 211 +++++++++++++++++++++ 5 files changed, 289 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,trats2-wm1811.txt create mode 100644 sound/soc/samsung/trats2_wm1811.c -- 2.0.0.390.gcb682f8

4 11

[alsa-devel] [PATCH v2] ASoC: Add sysfs entries via static attribute groups
by Takashi Iwai 02 Feb '15

02 Feb '15

Instead of calling device_create_file() manually, assign the static attribute group entries at the device registration. This simplifies the error handling and avoids the possible races. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Enable attributes conditionally via is_visible callback include/sound/soc-dapm.h | 2 +- sound/soc/soc-core.c | 58 ++++++++++++++++++++++++++++-------------------- sound/soc/soc-dapm.c | 14 ++++-------- 3 files changed, 39 insertions(+), 35 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 89823cfe6f04..e8e0220404d4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -405,7 +405,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_update *update); /* dapm sys fs - used by the core */ -int snd_soc_dapm_sys_add(struct device *dev); +extern struct attribute *soc_dapm_dev_attrs[]; void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2c62620abca6..4bd4d77d4ba8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -191,6 +191,39 @@ static ssize_t pmdown_time_set(struct device *dev, static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); +static struct attribute *soc_dev_attrs[] = { + &dev_attr_codec_reg.attr, + &dev_attr_pmdown_time.attr, + NULL +}; + +static umode_t soc_dev_attr_is_visible(struct kobject *kobj, + struct attribute *attr, int idx) +{ + struct device *dev = kobj_to_dev(kobj); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + + if (attr == &dev_attr_pmdown_time.attr) + return attr->mode; /* always visible */ + return rtd->codec ? attr->mode : 0; /* enabled only with codec */ +} + +static const struct attribute_group soc_dapm_dev_group = { + .attrs = soc_dapm_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group soc_dev_roup = { + .attrs = soc_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group *soc_dev_attr_groups[] = { + &soc_dapm_dev_group, + &soc_dev_roup, + NULL +}; + #ifdef CONFIG_DEBUG_FS static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) @@ -949,8 +982,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(rtd->dev, &dev_attr_pmdown_time); - device_remove_file(rtd->dev, &dev_attr_codec_reg); device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1120,6 +1151,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; + rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); @@ -1136,23 +1168,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return ret; } rtd->dev_registered = 1; - - if (rtd->codec) { - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", - ret); - - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", - ret); - } - return 0; } @@ -1308,11 +1323,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } #endif - ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); - if (ret < 0) - dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", - ret); - if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ ret = soc_new_compress(rtd, num); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..6c5cf7d1ff54 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2127,15 +2127,10 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -int snd_soc_dapm_sys_add(struct device *dev) -{ - return device_create_file(dev, &dev_attr_dapm_widget); -} - -static void snd_soc_dapm_sys_remove(struct device *dev) -{ - device_remove_file(dev, &dev_attr_dapm_widget); -} +struct attribute *soc_dapm_dev_attrs[] = { + &dev_attr_dapm_widget.attr, + NULL +}; static void dapm_free_path(struct snd_soc_dapm_path *path) { @@ -3903,7 +3898,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) */ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_sys_remove(dapm->dev); dapm_debugfs_cleanup(dapm); dapm_free_widgets(dapm); list_del(&dapm->list); -- 2.2.2

2 1

[alsa-devel] [PATCH] ASoC: rx51: Correct DAPM route for capture
by Peter Ujfalusi 02 Feb '15

02 Feb '15

"Mic Bias" is DAPM_SUPPLY so it has to be connected in the route accordingly. Fixes audio capture on the board. Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Tested-by: Pavel Machek <pavel(a)ucw.cz> --- sound/soc/omap/rx51.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 04896d6252a2..7f299357c2d2 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -250,14 +250,14 @@ static const struct snd_soc_dapm_route audio_map[] = { {"FM Transmitter", NULL, "LLOUT"}, {"FM Transmitter", NULL, "RLOUT"}, - {"DMic Rate 64", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "DMic"}, + {"DMic Rate 64", NULL, "DMic"}, + {"DMic", NULL, "Mic Bias"}, {"b LINE2R", NULL, "MONO_LOUT"}, {"Earphone", NULL, "b HPLOUT"}, - {"LINE1L", NULL, "b Mic Bias"}, - {"b Mic Bias", NULL, "HS Mic"} + {"LINE1L", NULL, "HS Mic"}, + {"HS Mic", NULL, "b Mic Bias"}, }; static const char * const spk_function[] = {"Off", "On"}; -- 2.2.2

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Re: [alsa-devel] Digital Input on ALC3661 (from laptop HDMI IN) not working under Linux
by Oliver Freyermuth 02 Feb '15

02 Feb '15

Am 02.02.2015 um 08:05 schrieb Raymond Yau: > >> >> >> I have an Alienware 17 laptop (sometimes called M17RX5) which has a >> > Realtek ALC3661. >> >>The problem I describe here should also be valid for the Alienware 18 >> > (all the late 2013 models, which are still the most current). At least >> > some scattered forum reports suggest so. >> > >> >>The card works almost perfect in Linux, but one functionality is >> > missing: The laptop features an HDMI input, and the Realtek card is >> > supposed to take care of the incoming digital audio. >> > This works fine in Windows, but in Linux, I only get the video and the >> > audio stays missing. >> > >> >> Here you can find my alsa-info: >> >>http://www.alsa-project.org/db/?f=0a0502e49212a0436a30fde9d4ee56163bbc2ebc > <http://www.alsa-project.org/db/?f=0a0502e49212a0436a30fde9d4ee56163bbc2ebc> >> > >> > The only digial input is node 0x0a >> > >> > Try hda-jack-retask to change node 0x1f with pin default [Jack] > digital In >> >> Thanks a lot for your suggestion! >> I tried, 0x0a stays as "not present" but I can select it in alsa-mixer >> as expected. Sadly, I can only record silence. >> Also setting Jack Detection to "Present" in the advanced overrides of >> hdajackretask did not help. > > http://git.alsa-project.org/?p=alsa-tools.git;a=blob;f=hdajackretask/README… > > You have to select "show unconnected pin" and "advanced override" > > Change the pin default of node 0x1f to 0x3c61130 [Jack] SPDIF In > > Set as boot default and reboot > > Digital capture device 1 will be created after you fix the pin default > of node 0x1f > > Turn on iec958 capture switch and try > > arecord -D IEC958:CARD=PCH -f dat -v test.wav Thanks a lot for your continued help! Here's what I did - the sad news, I still record only silence: 1. use hdajackretask as you explained, apply as boot default. 2. I got the following files: /etc/modprobe.d/hda-jack-retask.conf (below the automatic comment): options snd-hda-intel patch=hda-jack-retask.fw,hda-jack-retask.fw,hda-jack-retask.fw,hda-jack-retask.fw /lib64/firmware/hda-jack-retask.fw: [codec] 0x10ec0668 0x102805aa 0 [pincfg] 0x12 0x99a30140 0x14 0x90170110 0x15 0x0321101f 0x16 0x03211020 0x18 0x40030008 0x19 0x03a11030 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x41000001 0x1e 0x411111f0 0x1f 0x03c61130 It contains exactly the line 0x1f 0x03c61130, so I hope I clicked correctly in the GUI. 3. After reboot, I see in dmesg (I assume this means it worked): [ 6.021374] snd_hda_intel 0000:00:1b.0: Applying patch firmware 'hda-jack-retask.fw' [ 6.021468] snd_hda_intel 0000:01:00.1: Disabling MSI [ 6.021471] snd_hda_intel 0000:01:00.1: Applying patch firmware 'hda-jack-retask.fw' [ 6.022749] snd_hda_intel 0000:00:1b.0: irq 29 for MSI/MSI-X 4. I activate S/PDIF capture in alsamixer (IEC958 is not shown there, but S/PDIF appeared new after the change). 5. arecord -L tells me at the end: iec958:CARD=PCH,DEV=0 HDA Intel PCH, ALC3661 Digital IEC958 (S/PDIF) Digital Audio Output 6. I run, as suggested: arecord -D iec958:CARD=PCH -f dat -v test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hooks PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 Slave: Hardware PCM card 1 'HDA Intel PCH' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 But I get only silence, I tried to first attach the HDMI + switch to the video and start recording when everything is connected (i.e. when I would hear audio in Windows). I also tried to just record with the HDMI-cable connected and audio coming in (but not switched to video - in this case, I would hear nothing in Windows, but the audio should still arrive). Both cases, I record just silence. Any further ideas? Thanks a lot for your assistance! > >> > >> > Node 0x1f [Pin Complex] wcaps 0x400681: Stereo Digital >> > Pincap 0x00000020: IN >> > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear >> > Conn = 1/8, Color = Black >> > DefAssociation = 0xf, Sequence = 0x0 >> > Misc = NO_PRESENCE >> > Pin-ctls: 0x20: IN >> > Unsolicited: tag=00, enabled=0 >> > Power states: D0 D1 D2 D3 EPSS >> > >> >>In Windows, I observe the HDMI-IN audio routing to the speakers is only >> > done if the Realtek Audio Service is running. In Linux, I observe >> > keypress-events when I either switch manually to the HDMI-in signal > and also >> > if a new signal is detected (something (the UEFI?) injects a keypress >> > then). >> > I assume the service in Windows catches these events and changes >> > "something" in the Realtek Audio Settings. I see no special recording >> > device in Windows, nor any explicit HDMI-input setting, nor any related >> > setting in the driver. Dell links a standard Realtek driver in their >> > support section. >> >

1 0

[alsa-devel] [PATCH 0/2] ASoC: davinci-mcasp: Fixes for 3.20
by Peter Ujfalusi 02 Feb '15

02 Feb '15

Hi, While doing the conversion of davinci DAI drivers to use edma-pcm I have noticed couple of issues with the driver regarding to old daVinci device support which resulted in the following two patch. Some SoC uses shared IRQ line for all McASP instances There are SoCs where a given McASP instance only supports DIT mode. Currently the arch code feeds dummy resource in place of RX (which is not available). With the second patch this is not going to be needed anymore. If it is not too late for 3.20.. regards, Peter --- Peter Ujfalusi (2): ASoC: davicni-mcasp: Mark the common irq line as shared ASoC: davinci-mcasp: Fix DIT only McASP instance support sound/soc/davinci/davinci-mcasp.c | 75 +++++++++++++++++++++------------------ 1 file changed, 41 insertions(+), 34 deletions(-) -- 2.2.2

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[alsa-devel] [PATCH 0/2] ASoC: davinci-evm: Remove not needed header files
by Peter Ujfalusi 02 Feb '15

02 Feb '15

Hi, Trivial changes to remove headers should not be included in the machine driver. The first patch was sent as part of the edma header cleanup series in November: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084734.h… http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084735.h… The series missed 3.19 and it seams to not going to make it to 3.20 either.. With these header files gone from the davinci-evm I will be able to send my davinci audio conversion from davinci-pcm to edma-pcm for 3.21. Mark: is it possible to take these for 3.20 still? Regards, Peter --- Peter Ujfalusi (2): ASoC: davinci-evm: Do not include edma headers ASoC: davinci-evm: Do not include davinci-pcm.h and davinci-i2s.h sound/soc/davinci/davinci-evm.c | 6 ------ 1 file changed, 6 deletions(-) -- 2.2.2

2 4

[alsa-devel] [PATCH v4 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 02 Feb '15

02 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 67 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 36 +++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 58 ++++++++- sound/pci/hda/hda_controller.c | 43 +++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 478 insertions(+), 59 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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[alsa-devel] [PATCH 00/14] Refactoring ALSA core device management
by Takashi Iwai 02 Feb '15

02 Feb '15

Hi, this is a series of relatively small patches for refactoring the ALSA core device management. I experimented a similar thing some time ago but didn't merge it in the end due to too intrusive changes. At this time, the approach is rather minimalistic: only the device registration part is replaced by embedding the struct device into each ALSA device object. To the outside of ALSA core, there shouldn't be many visible changes. The only possible breakage would be an access to some renamed / dropped fields by a downstream driver. But it must be easy to correct. Some patches in the series are merely cleanup patches that can be applied individually, but included just for completeness. The whole patches are found in test/snd-device branch of sound git tree, too. Takashi === Takashi Iwai (14): ALSA: Allow to pass the device object to snd_register_device*() ALSA: control: Provide a helper to look for the preferred subdevice ALSA: Add a helper to initialize device ALSA: control: Embed struct device ALSA: hwdep: Embed struct device ALSA: pcm: Embed struct device ALSA: rawmidi: Embed struct device ALSA: rawmidi: Use rawmidi device file for kernel messages ALSA: timer: Propagate the error at initialization ALSA: timer: Handle the device directly ALSA: seq: Handle the device directly ALSA: compress: Embed struct device ALSA: Simplify snd_device_register() variants ALSA: Drop snd_get_device() helper include/sound/compress_driver.h | 4 +- include/sound/control.h | 11 +++- include/sound/core.h | 41 ++---------- include/sound/hwdep.h | 3 +- include/sound/pcm.h | 2 +- include/sound/rawmidi.h | 4 +- sound/aoa/soundbus/i2sbus/pcm.c | 5 +- sound/core/compress_offload.c | 26 ++++++-- sound/core/control.c | 66 +++++++++++-------- sound/core/hwdep.c | 88 ++++++++++--------------- sound/core/init.c | 19 ++++++ sound/core/pcm.c | 70 ++++++-------------- sound/core/rawmidi.c | 47 +++++++------- sound/core/seq/seq_clientmgr.c | 14 +++- sound/core/sound.c | 112 ++++++++++---------------------- sound/core/timer.c | 42 ++++++++---- sound/pci/hda/hda_controller.c | 3 +- sound/pci/hda/hda_hwdep.c | 7 +- sound/soc/intel/sst-mfld-platform-pcm.c | 1 - 19 files changed, 265 insertions(+), 300 deletions(-) -- 2.2.2

3 20

[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: Fix data delay configuration
by Peter Ujfalusi 02 Feb '15

02 Feb '15

The CTRLC register were not receiving the correct delay configuration, which will corrupt DSP_A audio mode. Fixes: 368494093354 ("ASoC: tlv320aic3x: Add TDM support") Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Tested-by: Pavel Machek <pavel(a)ucw.cz> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Changes since v1: Fixes tag and Tested-by from Pavel and Jarkko added to commit message. Regards, Peter sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a78ad361f494..07603d142923 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, delay += aic3x->tdm_delay; /* Configure data delay */ - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } -- 2.2.2

2 2

[alsa-devel] [PATCH] ASoC: atmel-pcm-dma: won't check direction when configure dma
by Bo Shen 02 Feb '15

02 Feb '15

As DMA framework request DMA using direction only in prep_slave function, (The At91 xdma driver has adapted to this request). So won't check direction when do DMA configuration. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/atmel/atmel-pcm-dma.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 33fb3bb..b8e7bad 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -105,13 +105,11 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, return ret; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config->dst_addr = ssc->phybase + SSC_THR; - slave_config->dst_maxburst = 1; - } else { - slave_config->src_addr = ssc->phybase + SSC_RHR; - slave_config->src_maxburst = 1; - } + slave_config->dst_addr = ssc->phybase + SSC_THR; + slave_config->dst_maxburst = 1; + + slave_config->src_addr = ssc->phybase + SSC_RHR; + slave_config->src_maxburst = 1; prtd->dma_intr_handler = atmel_pcm_dma_irq; -- 2.3.0.rc0

2 1

[alsa-devel] [PATCH] MAINTAINERS: ASoC: add maintainer for Intel BDW/HSW ASoC driver
by Jie Yang 02 Feb '15

02 Feb '15

Adding myself as the Intel BDW/HSW ASoC driver maintainer. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Acked-by: Liam Girdwood <lgirdwood(a)gmail.com> --- MAINTAINERS | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index dc1e9a8..78a0913 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4943,6 +4943,16 @@ F: Documentation/input/multi-touch-protocol.txt F: drivers/input/input-mt.c K: \b(ABS|SYN)_MT_ +INTEL ASoC BDW/HSW DRIVERS +M: Jie Yang <yang.jie(a)linux.intel.com> +L: alsa-devel(a)alsa-project.org +S: Supported +F: sound/soc/intel/sst-haswell* +F: sound/soc/intel/sst-dsp* +F: sound/soc/intel/sst-firmware.c +F: sound/soc/intel/broadwell.c +F: sound/soc/intel/haswell.c + INTEL C600 SERIES SAS CONTROLLER DRIVER M: Intel SCU Linux support <intel-linux-scu(a)intel.com> M: Artur Paszkiewicz <artur.paszkiewicz(a)intel.com> -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] Remove unused functions
by Chris Rorvick 01 Feb '15

01 Feb '15

Found a couple more things to rip out. Regards, Chris Chris Rorvick (2): ALSA: line6: Remove unused line6_midibuf_status() ALSA: line6: Remove unused line6_midibuf_skip_message() sound/usb/line6/midibuf.c | 18 ------------------ 1 file changed, 18 deletions(-) -- 2.1.0

2 4

[alsa-devel] [PATCH 0/3] snd-bcd2000: add audio support and control interface
by Mario Kicherer 01 Feb '15

01 Feb '15

The current version of the driver only provides a MIDI interface to read controller status and switch LEDs. These patches add audio playback and capture support as well as a control interface to switch between Phono A and Mic input.

2 7

[alsa-devel] [PATCH alsa-lib] conf: add support for USB AudioSport Quattro (tm)
by Alexander E. Patrakov 01 Feb '15

01 Feb '15

This patch resulted from discussion with redlion_ on #alsa IRC channel on Freenode. 4-channel playback now works. 4-channel capture works too, but not simultaneously with playback (hardware limitation). Alsa-info before the fix: http://www.alsa-project.org/db/?f=a3673622074b88a1abf4ccc6e7f37d0b5b72f34a Signed-off-by: Alexander E. Patrakov <patrakov(a)gmail.com> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 05e5d09..4cbf85e 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -30,6 +30,7 @@ USB-Audio.pcm.surround40_type { "OmniStudio" two_stereo_devices "Quattro" two_stereo_devices "SB Audigy 2 NX" six_channels + "USB AudioSport Quattro (tm)" two_stereo_devices } # If a device does not use the first PCM device for digital data, the device -- 2.2.1

2 1

[alsa-devel] [PATCH v2 12/12] dt-bindings: Add documentation for Rockchip dw-hdmi-audio
by Yakir Yang 31 Jan '15

31 Jan '15

Required properties: - compatible: platform specific - cpu-of-node: the device node of cpu_dai Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Changes in v2: - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt .../bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt diff --git a/Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt b/Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt new file mode 100644 index 0000000..5b86eed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt @@ -0,0 +1,12 @@ +Rockchip hdmi audio bindings + +Required properties: +- compatible: platform specific +- cpu-of-node: the device node of cpu_dai + +Example: + +sound { + compatible = "rockchip,rk3288-hdmi-audio"; + cpu-of-node = <&i2s>; +}; -- 2.1.2

3 2

[alsa-devel] [PATCH v2 10/12] ASoC: dw-hdmi-audio: add codec driver for dw hdmi audio
by Yakir Yang 31 Jan '15

31 Jan '15

codec driver creat an standard alsa device, than config audio and report jack status through some callback interfaces that dw_hdmi driver support. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Changes in v2: - Update dw_hdmi audio control interfaces, and adjust jack report process sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 314 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 17 +++ 4 files changed, 337 insertions(+) create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8349f98..b34dd12 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_DW_HDMI_AUDIO select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 @@ -459,6 +460,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_DW_HDMI_AUDIO + tristate + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bbdfd1e..0ebb664 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-dw-hdmi-audio-objs := dw-hdmi-audio.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o @@ -249,6 +250,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_DW_HDMI_AUDIO) += snd-soc-dw-hdmi-audio.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o diff --git a/sound/soc/codecs/dw-hdmi-audio.c b/sound/soc/codecs/dw-hdmi-audio.c new file mode 100644 index 0000000..ed046ce --- /dev/null +++ b/sound/soc/codecs/dw-hdmi-audio.c @@ -0,0 +1,314 @@ +/* + * dw-hdmi-codec.c + * + * DesignerWare ALSA SoC DAI driver for DW HDMI audio. + * Copyright (c) 2014, CORPORATION. All rights reserved. + * Authors: Yakir Yang <ykk(a)rock-chips.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>.* + * + */ +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/moduleparam.h> + +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> + +#include <drm/bridge/dw_hdmi.h> +#include "dw-hdmi-audio.h" + +struct snd_dw_hdmi { + struct device *dev; + struct dw_hdmi_audio_data data; + + u8 jack_status; + struct snd_soc_jack *jack; + struct delayed_work jack_work; +}; + +/** + * dw_hdmi_jack_detect - Enable hdmi detection via the HDMI IRQ + * + * @codec: HDMI codec + * @jack: jack to report detection events on + * + * Enable HDMI detection via IRQ on the HDMI. + * + * If no jack is supplied detection will be disabled. + */ +int dw_hdmi_jack_detect(struct snd_soc_codec *codec_dai, + struct snd_soc_jack *jack) +{ + struct snd_dw_hdmi *hdmi = snd_soc_codec_get_drvdata(codec_dai); + + hdmi->jack = jack; + hdmi->jack_status = 0; + + snd_soc_jack_report(hdmi->jack, 0, SND_JACK_LINEOUT); + + queue_delayed_work(system_power_efficient_wq, &hdmi->jack_work, + msecs_to_jiffies(50)); + + return 0; +} +EXPORT_SYMBOL_GPL(dw_hdmi_jack_detect); + +static void dw_hdmi_jack_work(struct work_struct *work) +{ + struct snd_dw_hdmi *hdmi = container_of(work, struct snd_dw_hdmi, + jack_work.work); + u8 jack_status, status; + + status = hdmi->data.get_connect_status(hdmi->data.dw); + jack_status = status ? SND_JACK_LINEOUT : 0; + if (jack_status != hdmi->jack_status) { + snd_soc_jack_report(hdmi->jack, jack_status, + SND_JACK_LINEOUT); + hdmi->jack_status = jack_status; + + dev_dbg(hdmi->dev, "jack report [%d]\n", hdmi->jack_status); + } +} + +static irqreturn_t snd_dw_hdmi_irq(int irq, void *dev_id) +{ + struct snd_dw_hdmi *hdmi = dev_id; + + queue_delayed_work(system_power_efficient_wq, &hdmi->jack_work, + msecs_to_jiffies(50)); + + return IRQ_HANDLED; +} + +static int dw_hdmi_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_dw_hdmi *hdmi = snd_soc_dai_get_drvdata(codec_dai); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hdmi_audio_fmt hdmi_fmt; + unsigned int fmt, rate, chan, width; + + fmt = rtd->dai_link->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK; + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + hdmi_fmt.dai_fmt = AUDIO_DAIFMT_IIS; + break; + case SND_SOC_DAIFMT_LEFT_J: + hdmi_fmt.dai_fmt = AUDIO_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + hdmi_fmt.dai_fmt = AUDIO_DAIFMT_RIGHT_J; + break; + default: + dev_err(codec_dai->dev, "DAI format unsupported"); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: dai_fmt = %d.\n", fmt); + + width = params_width(params); + switch (width) { + case 16: + case 24: + hdmi_fmt.word_length = width; + break; + default: + dev_err(codec_dai->dev, "width[%d] not support!\n", width); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: word_length = %d.\n", width); + + chan = params_channels(params); + switch (chan) { + case 2: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_2; + break; + case 4: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_4; + break; + case 6: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_6; + break; + case 8: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_8; + break; + default: + dev_err(codec_dai->dev, "channel[%d] not support!\n", chan); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: chan_num = %d.\n", chan); + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + case 176400: + case 192000: + hdmi_fmt.sample_rate = rate; + break; + default: + dev_err(codec_dai->dev, "rate[%d] not support!\n", rate); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: sample_rate = %d.\n", rate); + + hdmi->data.config_audio(hdmi->data.dw, &hdmi_fmt); + + return 0; +} + +static int dw_hdmi_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_dw_hdmi *hdmi = snd_soc_dai_get_drvdata(codec_dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + hdmi->data.enable_clk(hdmi->data.dw); + dev_dbg(codec_dai->dev, "[codec_dai]: trigger enable.\n"); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + hdmi->data.disable_clk(hdmi->data.dw); + dev_dbg(codec_dai->dev, "[codec_dai]: trigger disable.\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget dw_hdmi_widgets[] = { + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route dw_hdmi_routes[] = { + { "TX", NULL, "Playback" }, +}; + +static const struct snd_soc_dai_ops dw_hdmi_dai_ops = { + .hw_params = dw_hdmi_dai_hw_params, + .trigger = dw_hdmi_dai_trigger, +}; + +static struct snd_soc_dai_driver dw_hdmi_audio_dai = { + .name = "dw-hdmi-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &dw_hdmi_dai_ops, +}; + +static const struct snd_soc_codec_driver dw_hdmi_audio = { + .dapm_widgets = dw_hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(dw_hdmi_widgets), + .dapm_routes = dw_hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(dw_hdmi_routes), +}; + +static int dw_hdmi_audio_probe(struct platform_device *pdev) +{ + struct dw_hdmi_audio_data *data = pdev->dev.platform_data; + struct snd_dw_hdmi *hdmi; + int ret; + + hdmi = devm_kzalloc(&pdev->dev, sizeof(*hdmi), GFP_KERNEL); + if (!hdmi) + return -ENOMEM; + + hdmi->data = *data; + hdmi->dev = &pdev->dev; + platform_set_drvdata(pdev, hdmi); + + INIT_DELAYED_WORK(&hdmi->jack_work, dw_hdmi_jack_work); + + ret = devm_request_irq(&pdev->dev, hdmi->data.irq, snd_dw_hdmi_irq, + IRQF_SHARED, "dw-hdmi-audio", hdmi); + if (ret) { + dev_err(&pdev->dev, "request irq failed (%d)\n", ret); + goto free_hdmi_data; + } + + ret = snd_soc_register_codec(&pdev->dev, &dw_hdmi_audio, + &dw_hdmi_audio_dai, 1); + if (ret) { + dev_err(&pdev->dev, "register codec failed (%d)\n", ret); + goto free_irq; + } + + dev_info(&pdev->dev, "hdmi audio init success.\n"); + + return 0; + +free_irq: + devm_free_irq(&pdev->dev, hdmi->data.irq, hdmi); +free_hdmi_data: + devm_kfree(&pdev->dev, hdmi); + + return ret; +} + +static int dw_hdmi_audio_remove(struct platform_device *pdev) +{ + struct snd_dw_hdmi *hdmi = platform_get_drvdata(pdev); + + snd_soc_unregister_codec(&pdev->dev); + devm_free_irq(&pdev->dev, hdmi->data.irq, hdmi); + devm_kfree(&pdev->dev, hdmi); + + return 0; +} + +static const struct of_device_id dw_hdmi_audio_ids[] = { + { .compatible = "dw-hdmi-audio", }, + { } +}; + +static struct platform_driver dw_hdmi_audio_driver = { + .driver = { + .name = "dw-hdmi-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(dw_hdmi_audio_ids), + }, + .probe = dw_hdmi_audio_probe, + .remove = dw_hdmi_audio_remove, +}; +module_platform_driver(dw_hdmi_audio_driver); + +MODULE_AUTHOR("Yakir Yang <ykk(a)rock-chips.com>"); +MODULE_DESCRIPTION("DW HDMI Audio ASoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" "dw-hdmi-audio"); +MODULE_DEVICE_TABLE(of, dw_hdmi_audio_ids); diff --git a/sound/soc/codecs/dw-hdmi-audio.h b/sound/soc/codecs/dw-hdmi-audio.h new file mode 100644 index 0000000..2a7b5b4 --- /dev/null +++ b/sound/soc/codecs/dw-hdmi-audio.h @@ -0,0 +1,17 @@ +/* + * dw-hdmi-audio.h -- DW HDMI ALSA SoC Audio driver + * + * Copyright 2011-2012 DesignerWare Products + * + * 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 _DW_HDMI_AUDIO_H +#define _DW_HDMI_AUDIO_H + +int dw_hdmi_jack_detect(struct snd_soc_codec *codec_dai, + struct snd_soc_jack *jack); + +#endif -- 2.1.2

2 1

[alsa-devel] [PATCH 1/5] sync asound.h with kernel version
by Pierre-Louis Bossart 31 Jan '15

31 Jan '15

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/asound.h | 37 ++++++++++++++++++++++++++++++++----- 1 file changed, 32 insertions(+), 5 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 1f23cd6..3d46e9a 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -140,7 +140,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -267,9 +267,17 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ -#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ +#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ + +#define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ + + typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ #define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */ @@ -407,6 +415,22 @@ struct snd_pcm_channel_info { unsigned int step; /* samples distance in bits */ }; +enum { + /* + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + + /* timestamp definitions */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +}; + struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ @@ -418,9 +442,11 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - __u32 reserved_alignment; /* must be filled with zero */ - struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */ + struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */ + __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -533,6 +559,7 @@ enum { #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) #define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) +#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) #define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40) #define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41) -- 1.9.1

1 4

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix data delay configuration
by Peter Ujfalusi 30 Jan '15

30 Jan '15

Fix the issue introduced by: 368494093354 ASoC: tlv320aic3x: Add TDM support The CTRLC register were not receiving the correct delay configuration, which will corrupt DSP_A audio mode. Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, most of my boards are using DSP_B or I2S mode... This is why this is slipped. Peter sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a78ad361f494..07603d142923 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, delay += aic3x->tdm_delay; /* Configure data delay */ - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } -- 2.2.2

4 3

[alsa-devel] [PATCH] ASoC: Add sysfs entries via static attribute groups
by Takashi Iwai 30 Jan '15

30 Jan '15

Instead of calling device_create_file() manually, assign the static attribute group entries at the device registration. This simplifies the code and avoids the possible races. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-core.c | 40 ++++++++++++++++------------------------ sound/soc/soc-dapm.c | 16 +++++++--------- 3 files changed, 24 insertions(+), 34 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 89823cfe6f04..9fecd6752fcb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -405,7 +405,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_update *update); /* dapm sys fs - used by the core */ -int snd_soc_dapm_sys_add(struct device *dev); +extern const struct attribute_group soc_dapm_dev_attr_group; void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2c62620abca6..386365f4a628 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -949,8 +949,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(rtd->dev, &dev_attr_pmdown_time); - device_remove_file(rtd->dev, &dev_attr_codec_reg); device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1103,6 +1101,21 @@ err_probe: return ret; } +static struct attribute *soc_dev_attrs[] = { + &dev_attr_codec_reg.attr, + &dev_attr_pmdown_time.attr, + NULL +}; + +static const struct attribute_group soc_dev_attr_group = { + .attrs = soc_dev_attrs, +}; + +static const struct attribute_group *soc_dev_attr_groups[] = { + &soc_dev_attr_group, + &soc_dapm_dev_attr_group, +}; + static void rtd_release(struct device *dev) { kfree(dev); @@ -1120,6 +1133,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; + rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); @@ -1136,23 +1150,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return ret; } rtd->dev_registered = 1; - - if (rtd->codec) { - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", - ret); - - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", - ret); - } - return 0; } @@ -1308,11 +1305,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } #endif - ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); - if (ret < 0) - dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", - ret); - if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ ret = soc_new_compress(rtd, num); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..a997bb8cae93 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2127,15 +2127,14 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -int snd_soc_dapm_sys_add(struct device *dev) -{ - return device_create_file(dev, &dev_attr_dapm_widget); -} +static struct attribute *soc_dapm_dev_attrs[] = { + &dev_attr_dapm_widget.attr, + NULL +}; -static void snd_soc_dapm_sys_remove(struct device *dev) -{ - device_remove_file(dev, &dev_attr_dapm_widget); -} +const struct attribute_group soc_dapm_dev_attr_group = { + .attrs = soc_dapm_dev_attrs, +}; static void dapm_free_path(struct snd_soc_dapm_path *path) { @@ -3903,7 +3902,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) */ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_sys_remove(dapm->dev); dapm_debugfs_cleanup(dapm); dapm_free_widgets(dapm); list_del(&dapm->list); -- 2.2.2

2 2

[alsa-devel] [PATCH] ASoC: rt286: Add rt288 codec support
by Bard Liao 30 Jan '15

30 Jan '15

This patch adds support for rt288 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 81bdd276..2eafbcf 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -34,6 +34,7 @@ #include "rt286.h" #define RT286_VENDOR_ID 0x10ec0286 +#define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { struct regmap *regmap; @@ -1210,7 +1211,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_read(rt286->regmap, RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID) { + if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %x is not rt286\n", ret); return -ENODEV; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt286: set idle_bias_off to false
by Bard Liao 30 Jan '15

30 Jan '15

We do the power down process at bias level STANDBY. And do nothing at bias level OFF. As a result, the register setting is the same between bias level STANDBY and OFF. So we don't need to set idle_bias_off true. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 847cc4b..81bdd276 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1144,7 +1144,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt286 = { .suspend = rt286_suspend, .resume = rt286_resume, .set_bias_level = rt286_set_bias_level, - .idle_bias_off = true, + .idle_bias_off = false, .controls = rt286_snd_controls, .num_controls = ARRAY_SIZE(rt286_snd_controls), .dapm_widgets = rt286_dapm_widgets, -- 1.8.1.1.439.g50a6b54

3 7

[alsa-devel] [PATCH] ASoC: sgtl5000: Use shift mask when setting codec mode
by Filip Brozovic 30 Jan '15

30 Jan '15

Shift the I2S mode value by the necessary amount before writing the registers. This makes Right Justified and PCM mode work in addition to the default Left Justified mode. Signed-off-by: Filip Brozovic <fbrozovic(a)gmail.com> --- sound/soc/codecs/sgtl5000.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 29cf7ce..7665016 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -483,21 +483,21 @@ static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* setting i2s data format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_DSP_B: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; case SND_SOC_DAIFMT_I2S: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_RIGHT_J: - i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_MODE_RJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRPOL; break; case SND_SOC_DAIFMT_LEFT_J: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; default: -- 2.1.4

1 0

[alsa-devel] [PATCH v2 11/12] ASoC: rockchip-hdmi-audio: add sound driver for hdmi audio
by Yakir Yang 30 Jan '15

30 Jan '15

Add a sound driver that combines rockchip-i2s cpu_dai and dw-hdmi-codec as codec_dai to provide hdmi audio output on rk3288 platforms. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Changes in v2: - give "codec-name" & "codec-dai-name" an const name sound/soc/rockchip/Kconfig | 9 ++ sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 196 +++++++++++++++++++++++++++++++ 3 files changed, 207 insertions(+) create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index e181826..ed2b7f0 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -14,3 +14,12 @@ config SND_SOC_ROCKCHIP_I2S Say Y or M if you want to add support for I2S driver for Rockchip I2S device. The device supports upto maximum of 8 channels each for play and record. + +config SND_SOC_ROCKCHIP_HDMI_AUDIO + tristate "ASoC support for Rockchip HDMI audio" + depends on SND_SOC_ROCKCHIP + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_DW_HDMI_AUDIO + help + Say Y or M here if you want to add support for SoC audio on Rockchip + HDMI, such as rk3288 hdmi. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index b921909..b9185b3 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,4 +1,6 @@ # ROCKCHIP Platform Support snd-soc-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-hdmi-audio-objs := rockchip_hdmi_audio.o obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_HDMI_AUDIO) += snd-soc-rockchip-hdmi-audio.o diff --git a/sound/soc/rockchip/rockchip_hdmi_audio.c b/sound/soc/rockchip/rockchip_hdmi_audio.c new file mode 100644 index 0000000..0c9f497 --- /dev/null +++ b/sound/soc/rockchip/rockchip_hdmi_audio.c @@ -0,0 +1,196 @@ +/* + * rockchip-hdmi-card.c + * + * ROCKCHIP ALSA SoC DAI driver for HDMI audio on rockchip processors. + * Copyright (c) 2014, ROCKCHIP CORPORATION. All rights reserved. + * Authors: Yakir Yang <ykk(a)rock-chips.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>.* + * + */ +#include <linux/module.h> +#include <linux/device.h> +#include <linux/platform_device.h> + +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/jack.h> +#include <sound/core.h> +#include <sound/pcm_params.h> + +#include "rockchip_i2s.h" +#include "../codecs/dw-hdmi-audio.h" + +#define DRV_NAME "rockchip-hdmi-audio" + +struct hdmi_audio_private { + struct snd_soc_jack hdmi_jack; +}; + +static int rockchip_hdmi_audio_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int dai_fmt = rtd->dai_link->dai_fmt; + int mclk, ret; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt); + if (ret < 0) { + dev_err(cpu_dai->dev, "failed to set cpu_dai fmt.\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(cpu_dai->dev, "failed to set cpu_dai sysclk.\n"); + return ret; + } + + return 0; +} + +static int hdmi_audio_dai_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + struct hdmi_audio_private *drv = snd_soc_card_get_drvdata(card); + + /* Enable headphone jack detection */ + snd_soc_jack_new(codec, "HDMI Jack", SND_JACK_LINEOUT, + &drv->hdmi_jack); + + return dw_hdmi_jack_detect(codec, &drv->hdmi_jack); +} + +static struct snd_soc_ops hdmi_audio_dai_ops = { + .hw_params = rockchip_hdmi_audio_hw_params, +}; + +static struct snd_soc_dai_link hdmi_audio_dai = { + .name = "RockchipHDMI", + .stream_name = "RockchipHDMI", + .codec_name = "dw-hdmi-audio", + .codec_dai_name = "dw-hdmi-hifi", + .init = hdmi_audio_dai_init, + .ops = &hdmi_audio_dai_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card rockchip_hdmi_audio_card = { + .name = "RockchipHDMI", + .owner = THIS_MODULE, + .dai_link = &hdmi_audio_dai, + .num_links = 1, +}; + +static int rockchip_hdmi_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_hdmi_audio_card; + struct device_node *np = pdev->dev.of_node; + struct hdmi_audio_private *drv; + int ret; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (!drv) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, drv); + + hdmi_audio_dai.cpu_of_node = of_parse_phandle(np, "i2s-controller", 0); + if (!hdmi_audio_dai.cpu_of_node) { + dev_err(&pdev->dev, "Property 'i2s-controller' missing !\n"); + goto free_priv_data; + } + + hdmi_audio_dai.platform_of_node = hdmi_audio_dai.cpu_of_node; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "register card failed (%d)\n", ret); + card->dev = NULL; + goto free_cpu_of_node; + } + + dev_info(&pdev->dev, "hdmi audio init success.\n"); + + return 0; + +free_cpu_of_node: + hdmi_audio_dai.cpu_of_node = NULL; + hdmi_audio_dai.platform_of_node = NULL; +free_priv_data: + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + devm_kfree(&pdev->dev, drv); + card->dev = NULL; + + return ret; +} + +static int rockchip_hdmi_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; + + return 0; +} + +static const struct of_device_id rockchip_hdmi_audio_of_match[] = { + { .compatible = "rockchip,rk3288-hdmi-audio", }, + {}, +}; + +static struct platform_driver rockchip_hdmi_audio_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = rockchip_hdmi_audio_of_match, + }, + .probe = rockchip_hdmi_audio_probe, + .remove = rockchip_hdmi_audio_remove, +}; +module_platform_driver(rockchip_hdmi_audio_driver); + +MODULE_AUTHOR("Yakir Yang <ykk(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip HDMI Audio ASoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_hdmi_audio_of_match); -- 2.1.2

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Re: [alsa-devel] [PATCH 0/2 v5] dmaengine: rcar-audmapp: independent from SH_DMAE_BASE
by Kuninori Morimoto 30 Jan '15

30 Jan '15

Hi Laurent, Vinod, and Arnd # I added Linux ALSA ML and Mark > > >> This time I added SoC/platform side setting patches too (= 3) - 6)). > > >> SoC/platform side setting needs many entries for this rcar-audmapp, > > >> because it has many combinations. > > >> But, I believe this is very normal DMAEngine style, not special. > > > > > > Vinod commented on 22/12/2014 (Message-ID: > > > <20141222151447.GL16827(a)intel.com>) > > > > > > "I think this makes sense. Going thru the driver, it was clear that we > > > were not really gaining anything for using dmaengine API here. So agree > > > that lets use dmanegine for 1st dmac thru dmaengine library and then > > > configure this in your sound driver.." > > > > > > My understanding is that a solution specific to the sound driver was > > > preferred, instead of a generic DMA engine driver. Have I missed something > > > ? > > > > Grr... my understanding was that > > "1st DMAC will use dmaengine library (= sound framework has sound-dma-xxx > > functions) 2nd DMAC will use normal dmaengine API" > > > > But, I need to fixup sound driver ? > > - 1st DMAC = normal DMAEngine API > > - 2nd DMAC = part of sound driver > > > > Sorry, for discuss it again here, but I want flexible switching > > for 1st/2nd DMAC (because of 1st/2nd DMAC path combination). > > So, using same DMAEngine interface from sound driver is easy to > > implement/understanding. > > - 1st DMAC = normal DMAEngine API > > - 2nd DMAC = normal DMAEngine API > > The first DMA engine (the one handling transfer from/to memory) is a general- > purpose DMA engine. It should be handled by a driver that implement the DMA > engine API, no doubt about that. > > The second "DMA engine" is dedicated to the sound IP cores and is far from > being general-purpose, given that it only supports peripheral-to-peripheral > transfers, without even interrupts to report transfer completion. I'm not even > sure we can call it a DMA engine as there's no Direct Memory Access involved > here :-) The hardware looks more like a crossbar switch with programmable > routing of audio channels. That's why Vinod and I were wondering whether it > really makes sense to implement it using the DMA engine API, given the > resulting complexity of the DT bindings (the sound DT nodes look a bit scary), > or if it could be simpler to implement it as part of the Renesas sound > drivers. If you are caring about naming (= DMA), it is "Audio *DMAC* peri peri". I wonder dma_transfer_direction has DMA_DEV_TO_DEV (this driver is not using it though...) it is for peripheral-to-peripheral ? And API point of view, 2nd DMAC doesn't need new DMAEngine API. From DRY (= Don't Repeat Yourself) point of view, I don't want to re-create "similar but different" implementation for naming issue. From DT bindings complexity point of view, which is complex ? DMAC driver side ? DT node side ? Indeed sound driver needs many node, but is is regular arrangement, not complex, and, it needs many node for 1st DMAC too. I don't understand why 1st is OK, 2nd is not OK ? From DMAC driver side complexity point of view, 1st DMAC has same complexity (= it accepts many node from many drivers) ? If I need to move 2nd DMAC from DMAEngine to sound driver side, please explain it to Mark Brown (= ALSA SoC maintainer) Best regards --- Kuninori Morimoto Best regards --- Kuninori Morimoto

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[alsa-devel] [RFC PATCH] ASoC: atmel_ssc_dai: Allow more rates
by Peter Rosin 29 Jan '15

29 Jan '15

From: Peter Rosin <peda(a)axentia.se> When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to within 500ppm of the SSC master clock, possibly divided by 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 98 +++++++++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 95 insertions(+), 4 deletions(-) This patch depends on the patch I sent earlier that enables the _CBS_CFS mode on the atmel SSC dai (1). I'm unsure about the rate rules for reception on the TK pin when outputting LRCLK and the rate rules for transmitting on the RK pin when inputting LRCLK. The datasheet for "my" chip (SAMA5D31) doesn't say anything about what rates are allowed when using the RK pin for transmitting or the TK pin for receiving. Another question I have is if these rate rules are the same across all chips supported by this driver, or if there is need for some amount of special casing? And what special casing if that is the case? The reason I'm digging around in this area is that I need support for inputting a BCK of 16MHz on the TK pin during playback, i.e. with the codec mastering BCK. And the only way I can get that is to use _CBM_CFS. My SSC master clock is 66MHz, which means that the max supported rate for _CBM_CFM is 11MHz, i.e. not enough. This patch unlocks my use case, but I would like to correctly stop rates that the SSC does not support in other use cases as well. I simply do not have the hardware to fully test this myself. An lastly, is it reasonable to deduct 500ppm to allow for some clock skew? FYI, the 500ppm was pulled out of my hat... Cheers, Peter (1) ASoC: atmel_ssc_dai: Support SND_SOC_DAIFMT_CBM_CFS on I2S https://lkml.org/lkml/2015/1/29/373 diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f55f3aab8bdd..11eab65c86c8 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,76 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro is output on RK pin. + * What rules are there when outputting on the TK pin? + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro is input on TK pin. + * What rules are there when inputting from the RK pin? + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + /* Take away 500ppm, just to be on the safe side. */ + t.max -= t.max / 2000; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,10 +270,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); + if (!ssc_p->initialized) + clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk) * 2; + if (!ssc_p->initialized) + clk_disable(ssc_p->ssc->clk); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dir = 0; dir_mask = SSC_DIR_MASK_PLAYBACK; @@ -212,6 +289,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -787,8 +875,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -808,12 +894,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); -- 1.7.10.4

1 0

[alsa-devel] simple-audio-card vs. platform question
by Ricard Wanderlof 29 Jan '15

29 Jan '15

I'm trying to set up a simple device tree framework so I can do some development on an I2S driver. I'm trying to use simple-audio-card, but I keep running into the problem that once soc_bind_dai_link() gets called, dai_link->platform is set to NULL (because it is not initialized by simple-card.c when there is a DT node), while dai_link->platform_of_node has been set to dai_link->cpu_of_node in asoc_simple_card_parse_of(). This results in the following code in soc-core:soc_bind_dai_link() to set platform to NULL, /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; /* find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { if (dai_link->platform_of_node) { if (platform->dev->of_node != dai_link->platform_of_node) continue; } else { if (strcmp(platform->component.name, platform_name)) continue; } rtd->platform = platform; } if (!rtd->platform) { dev_err(card->dev, "ASoC: platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } with -EPROBE_DEFER as a result. I can't figure out either how to set up a platform (=DMA) driver in this environment. Perhaps the case of having a CPU DAI, DMA driver and codec DAI is not supported by simple-audio-card? On other hand, the way it looks there is no way the resulting setup can get through soc_bind_dai_link() unless there is a platform and associated driver. This indicates to me that I'm doing something seriously wrong in my setup, but having looked at various DT configurations in arch/arm/boot/dts in this case, as well as the documentation in Documentation/devicetree/bindings/sound/simple-card.txt I can't figure how to set it up any other way. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH] ASoC: pxa: make TTC DKB tristate
by Arnd Bergmann 29 Jan '15

29 Jan '15

In a rare combination of Kconfig settings, the 88pm860x-codec module may be selected as a loadable module, while it's also being used by the ttb-dkb code that is built-in, resulting in a link error: sound/built-in.o: In function `ttc_pm860x_init': :(.text+0x3e888): undefined reference to `pm860x_hs_jack_detect' :(.text+0x3e898): undefined reference to `pm860x_mic_jack_detect' Changing ttb-tkb to a tristate option tells Kconfig that 88pm86x actually needs to be built-in if ttc-dkb is also built-in. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 2434b6d61675..39cea80846c3 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -140,7 +140,7 @@ config SND_PXA910_SOC Marvell PXA910 reference platform. config SND_SOC_TTC_DKB - bool "SoC Audio support for TTC DKB" + tristate "SoC Audio support for TTC DKB" depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y select PXA_SSP select SND_PXA_SOC_SSP

1 0

[alsa-devel] [PATCH] ASoC: Samsung: add missing I2C/SPI dependencies
by Arnd Bergmann 29 Jan '15

29 Jan '15

The SND_SOC_ARNDALE_RT5631_ALC5631 selects the rt5631 codec that requires I2C to work, so we get a build error if I2C is disabled: codecs/rt5631.c:1737:1: warning: data definition has no type or storage class 31_i2c_driver); codecs/rt5631.c:1737:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] codecs/rt5631.c:1737:1: warning: parameter names (without types) in function declaration codecs/rt5631.c:1726:26: warning: 'rt5631_i2c_driver' defined but not used [-Wunused-variable] The same thing can happen on odroid-x2, littlemill, bells and speyside, so this patch adds all the dependencies. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index e817a2f43ea8..dd781d9a1c66 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -185,7 +185,7 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -200,7 +200,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -217,7 +217,7 @@ config SND_SOC_LOWLAND config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 @@ -234,7 +234,7 @@ config SND_SOC_SNOW config SND_SOC_ODROIDX2 tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S help @@ -242,6 +242,6 @@ config SND_SOC_ODROIDX2 config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631

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[alsa-devel] [PATCH 0/2] ASoC: pcm512x: Fixups for the Clock master modes series
by Peter Rosin 29 Jan '15

29 Jan '15

From: Peter Rosin <peda(a)axentia.se> These fixups can either be squashed in with 5/7 from the Clock master mode series, or they can be added on top of the current topic/pcm512x. Sorry for the trouble. I can squash them, and resend the Clock master modes series if that helps. Let me know how you'd like it. Cheers, Peter Peter Rosin (2): ASoC: pcm512x: Fixup warning splat ASoC: pcm512x: Use the correct range constraints for S24_LE sound/soc/codecs/pcm512x.c | 41 +++++++++++++++++++++-------------------- 1 file changed, 21 insertions(+), 20 deletions(-) -- 1.7.10.4

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[alsa-devel] [RFC] ASoC: rt5677: fix SPI dependency
by Arnd Bergmann 29 Jan '15

29 Jan '15

The rt5677 codec has gained code that requires SPI to work correctly, but there is no provision in Kconfig to prevent the driver from being used when SPI is disabled or a loadable module, resulting in this build error: sound/built-in.o: In function `rt5677_spi_write': :(.text+0xa7ba0): undefined reference to `spi_sync' sound/built-in.o: In function `rt5677_spi_driver_init': :(.init.text+0x253c): undefined reference to `spi_register_driver' ERROR: "spi_sync" [sound/soc/codecs/snd-soc-rt5677-spi.ko] undefined! ERROR: "spi_register_driver" [sound/soc/codecs/snd-soc-rt5677-spi.ko] undefined! This makes the SPI portion of the driver depend on the SPI subsystem, and disables the function that uses SPI for firmware download if SPI is disabled. The latter may not be the correct solution, but I could not come up with a better one. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: af48f1d08a54741 ("ASoC: rt5677: Support DSP function for VAD application") diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8349f982a586..ef2c70e77d91 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -525,7 +525,7 @@ config SND_SOC_RT5677 config SND_SOC_RT5677_SPI tristate - default SND_SOC_RT5677 + default SND_SOC_RT5677 && SPI #Freescale sgtl5000 codec config SND_SOC_SGTL5000 diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 26fc538f03b1..5d0bb8748dd1 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -702,6 +702,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) static bool activity; int ret; + if (!IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI)) + return -ENXIO; + if (on && !activity) { activity = true;

2 1

[alsa-devel] [PATCH] ASoC: davinci: fix DM365_EVM codec selection
by Arnd Bergmann 29 Jan '15

29 Jan '15

An earlier bug fix of mine made the SND_DM365_VOICE_CODEC symbol tristate to avoid creating an undefined reference from the davinci-vcif.c driver to the davinci_soc_platform_register function that may be in a module. However, this may now lead to a different error on randconfig kernels: "warning: SND_DM365_VOICE_CODEC creates inconsistent choice state" This happens because we now have a choice statement with one bool and one tristate option, and the latter might not support being set to 'y' because of dependencies. This new change turns the other option into 'tristate' as well, which avoids the problem. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 19926c6de0c3 ("ASoC: davinci: vcif must be a module if SND_DAVINCI_SOC is") diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 8e948c63f3d9..2b81ca418d2a 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -58,13 +58,12 @@ choice depends on MACH_DAVINCI_DM365_EVM config SND_DM365_AIC3X_CODEC - bool "Audio Codec - AIC3101" + tristate "Audio Codec - AIC3101" help Say Y if you want to add support for AIC3101 audio codec config SND_DM365_VOICE_CODEC tristate "Voice Codec - CQ93VC" - depends on SND_DAVINCI_SOC select MFD_DAVINCI_VOICECODEC select SND_DAVINCI_SOC_VCIF select SND_SOC_CQ0093VC

2 1

[alsa-devel] ESI Juli@ crash with external clock switch - patch
by Pavel Hofman 28 Jan '15

28 Jan '15

Hi, I have investigated a crash/kernel thread lockup when Juli@ is switched to external SPDIF clock and the incoming SPDIF stream changes samplerate. The problem appears to occur in the timed thread in charge of reading incoming samplerate and acting upon its change. The problem disappears with the following patch: diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index c7f5633..68bb326 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -154,7 +154,7 @@ void snd_ak4114_reinit(struct ak4114 *chip) { chip->init = 1; mb(); - flush_delayed_work(&chip->work); + //flush_delayed_work(&chip->work); ak4114_init_regs(chip); /* bring up statistics / event queing */ chip->init = 0; I am afraid I do not know enough about kernel workqueues to determine whether this "fix" is OK. Interestingly, the almost identical driver ak4113.c for a very similar card Infrasonic Quartet (ice1724/quartet.c) does not suffer from this problem (tested now). Thanks a lot for your opinion. Best regards, Pavel Hofman.

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[alsa-devel] [PATCH v2 0/7] ASoC: pcm512x: Clock master modes
by Peter Rosin 28 Jan '15

28 Jan '15

From: Peter Rosin <peda(a)axentia.se> Hi! [ Note that the dt change is in patch 5/7, for those only interested in that particular bit. ] This series implements BCLK master modes for the pcm512x driver. It has only been tested with the pcm5142 chip, but they are from the same family and should be compatible. I have mainly used the spec for the newer pcm5242 chip (also from the same family) as it fills in a lot of blanks in the pcm512x/pcm514x specs. The code has also seen most of its testing in a 3.10 environment, so there might be some forward-porting warts. But it is able to play sound in 3.18 as well, and most of the changes have little to do with anything but the clocking in the chip itself. Changes since v1: - Use snd_soc_params_to_frame_size and snd_soc_params_to_bclk instead of implementing home-grown versions based on the physical width. The I2S-communication will no longer have padding bits for S24_LE. Cheers, Peter Peter Rosin (7): ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Change register default to match actual content after reset ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS .../devicetree/bindings/sound/pcm512x.txt | 25 +- include/sound/pcm.h | 12 + sound/core/pcm_lib.c | 85 ++ sound/soc/codecs/pcm512x.c | 933 +++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- 5 files changed, 1139 insertions(+), 25 deletions(-) -- 1.7.10.4

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[alsa-devel] [PATCH v3 1/2] ASoC: codecs: wm8904: add dt ids table
by Bo Shen 28 Jan '15

28 Jan '15

From: Alexander Morozov <linux(a)meltdown.ru> The WM8904 and WM8918 has the same data type, while the WM8912 has different data type. So, use the data in dt ids table to distinguish them. Signed-off-by: Alexander Morozov <linux(a)meltdown.ru> [voice.shen(a)atmel.com: add code to distinguish device type] Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v3: - Get the device type from match data in probe function. Changes in v2: - Add driver data for distinguish the device capability. sound/soc/codecs/wm8904.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..a80bc52 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2098,6 +2098,24 @@ static const struct regmap_config wm8904_regmap = { .num_reg_defaults = ARRAY_SIZE(wm8904_reg_defaults), }; +#ifdef CONFIG_OF +static enum wm8904_type wm8904_data = WM8904; +static enum wm8904_type wm8912_data = WM8912; + +static const struct of_device_id wm8904_of_match[] = { + { + .compatible = "wlf,wm8904", + .data = &wm8904_data, + }, { + .compatible = "wlf,wm8912", + .data = &wm8912_data, + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, wm8904_of_match); +#endif + static int wm8904_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2125,7 +2143,17 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, return ret; } - wm8904->devtype = id->driver_data; + if (i2c->dev.of_node) { + const struct of_device_id *match; + + match = of_match_node(wm8904_of_match, i2c->dev.of_node); + if (match == NULL) + return -EINVAL; + wm8904->devtype = *((enum wm8904_type *)match->data); + } else { + wm8904->devtype = id->driver_data; + } + i2c_set_clientdata(i2c, wm8904); wm8904->pdata = i2c->dev.platform_data; @@ -2259,6 +2287,7 @@ static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(wm8904_of_match), }, .probe = wm8904_i2c_probe, .remove = wm8904_i2c_remove, -- 2.3.0.rc0

2 2

[alsa-devel] [PATCH v3 00/10] Improvements to Tegra-based Chromebook support
by Tomeu Vizoso 28 Jan '15

28 Jan '15

v3: * Added bindings for the LTN140AT29 panel * Removed the delay in pwrseq, as what was actually needed was to add a dependency on the power supplies of the host * Uses the pinmux for the Blaze as generated by tegra-pinmux-scripts * Uses the pinmux for the Big as in the last patch from Simon Glass Hello, this series adds support for the Tegra-based HP Chromebook 14 (aka nyan blaze), which is very similar to the Acer Chromebook 13 (aka nyan big). Because they both include tegra124-nyan.dtsi, some improvements to Blaze support have also benefitted the Big. I have tested that USB2, the panels, HDMI, the trackpad, Wifi and sound work on both. The DT for the Big includes the pinmux configuration as generated by tegra-pinmux-scripts with Simon's patch at: https://patchwork.ozlabs.org/patch/417779/ These patches are based on top of linux-next 20150128. http://cgit.collabora.com/git/user/tomeu/linux.git/log/?h=nyan-v3 Regards, Tomeu Stéphane Marchesin (1): drm/panel: add support for Samsung LTN140AT29 panel Tomeu Vizoso (9): ARM: tegra: Set the sound card model that alsaucm expects ARM: tegra: Move out nyan-generic parts out from the nyan-big DT ARM: tegra: Add DTS for the nyan-blaze board ARM: tegra: Add node for trackpad in Nyan boards ASoC: tegra: Add a control for the headphone switch ASoC: tegra: add sink for the internal mic to tegra_max98090 ARM: tegra: Use pwrseq-simple for the wifi in Nyan ARM: tegra: Use the generated pinmux data ARM: tegra: Set spi-max-frequency property to flash node .../bindings/panel/samsung,ltn140at29-301.txt | 7 + .../bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + arch/arm/boot/dts/Makefile | 1 + arch/arm/boot/dts/tegra124-nyan-big.dts | 2112 +++++++++++--------- arch/arm/boot/dts/tegra124-nyan-blaze.dts | 1325 ++++++++++++ arch/arm/boot/dts/tegra124-nyan.dtsi | 692 +++++++ drivers/gpu/drm/panel/panel-simple.c | 26 + sound/soc/tegra/tegra_max98090.c | 3 + 8 files changed, 3206 insertions(+), 961 deletions(-) create mode 100644 Documentation/devicetree/bindings/panel/samsung,ltn140at29-301.txt create mode 100644 arch/arm/boot/dts/tegra124-nyan-blaze.dts create mode 100644 arch/arm/boot/dts/tegra124-nyan.dtsi -- 1.9.3

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[alsa-devel] [PATCH PING] ASoC: dwc: Remove unnecessary conditional compilation
by Andrew Jackson 28 Jan '15

28 Jan '15

From: Andrew Jackson <Andrew.Jackson(a)arm.com> of_match_ptr is already conditionally compiled based on CONFIG_OF so further conditional compilation is not required. Remove conditional compilation surrounding of_match_ptr. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 99cf64b..a3e97b4 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -624,9 +624,7 @@ static struct platform_driver dw_i2s_driver = { .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", -#ifdef CONFIG_OF .of_match_table = of_match_ptr(dw_i2s_of_match), -#endif }, }; -- 1.7.1

2 1

[alsa-devel] [RFC] XEN support in aserver
by Vitaly V. Ch 28 Jan '15

28 Jan '15

Hi all! What do you think about introducing XEN support into aserver? It will allow to share sound between domains with few little changes. \\wbr Vitaly Ch

3 3

[alsa-devel] [PATCH 0/7] Yet another line6 cleanups
by Takashi Iwai 28 Jan '15

28 Jan '15

OK, I did enough for this month. The only real fix is the first patch for big-endianess. The rest are rather cleanups and beautifying the code. As usual, the latest code is found in test/line6 branch. thanks, Takashi === Takashi Iwai (7): ALSA: line6: Fix volume calculation for big-endian ALSA: line6: Tidy up and typo fixes in comments ALSA: line6: Remove revision.h ALSA: line6: Move the contents of usbdefs.h into driver.h ALSA: line6: Remove struct usb_line6_podhd ALSA: line6: Remove invalid capability bits for PODxt Live Variax ALSA: line6: Remove snd_line6_ prefix of pcm property fields sound/usb/line6/capture.c | 5 ++- sound/usb/line6/driver.c | 4 +-- sound/usb/line6/driver.h | 89 ++++++++++++++++++++-------------------------- sound/usb/line6/midi.c | 1 - sound/usb/line6/midi.h | 32 +++++------------ sound/usb/line6/pcm.h | 57 +++++++++-------------------- sound/usb/line6/playback.c | 24 +++++++------ sound/usb/line6/pod.c | 39 ++++++-------------- sound/usb/line6/podhd.c | 16 +++------ sound/usb/line6/revision.h | 4 --- sound/usb/line6/toneport.c | 31 +++++----------- sound/usb/line6/usbdefs.h | 27 -------------- sound/usb/line6/variax.c | 25 ++++--------- 13 files changed, 109 insertions(+), 245 deletions(-) delete mode 100644 sound/usb/line6/revision.h delete mode 100644 sound/usb/line6/usbdefs.h -- 2.2.2

1 8

[alsa-devel] [PATCH 0/4] Misc cleanup for 3.20 kernel
by Takashi Iwai 28 Jan '15

28 Jan '15

Hi, here is a few trivial cleanup patches, targeted for 3.20. All these shouldn't result in any functional change. Takashi === Takashi Iwai (4): ALSA: Include linux/io.h instead of asm/io.h ALSA: Include linux/uaccess.h and linux/bitopts.h instead of asm/* ALSA: au88x0: Remove superfluous ifdef __KERNEL__ ALSA: usx2y: Move UAPI definition into include/uapi/sound/usb_stream.h include/sound/emu10k1.h | 2 +- include/sound/gus.h | 2 +- include/sound/sb.h | 2 +- include/uapi/sound/usb_stream.h | 76 ++++++++++++++++++++++++++++++++++ sound/aoa/soundbus/i2sbus/control.c | 2 +- sound/aoa/soundbus/i2sbus/pcm.c | 2 +- sound/core/memory.c | 4 +- sound/core/pcm_memory.c | 2 +- sound/core/pcm_native.c | 2 +- sound/drivers/mpu401/mpu401_uart.c | 2 +- sound/drivers/opl3/opl3_lib.c | 2 +- sound/drivers/opl4/opl4_lib.c | 2 +- sound/drivers/opl4/opl4_synth.c | 2 +- sound/drivers/pcsp/pcsp.c | 2 +- sound/drivers/pcsp/pcsp_input.c | 2 +- sound/drivers/pcsp/pcsp_lib.c | 2 +- sound/drivers/serial-u16550.c | 3 +- sound/drivers/vx/vx_core.c | 2 +- sound/i2c/other/ak4xxx-adda.c | 2 +- sound/isa/ad1816a/ad1816a_lib.c | 2 +- sound/isa/azt2320.c | 2 +- sound/isa/cs423x/cs4236_lib.c | 2 +- sound/isa/es1688/es1688_lib.c | 2 +- sound/isa/es18xx.c | 2 +- sound/isa/opl3sa2.c | 3 +- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- sound/isa/sb/emu8000.c | 4 +- sound/isa/sb/emu8000_patch.c | 2 +- sound/isa/sb/sb16_main.c | 2 +- sound/isa/sb/sb8_main.c | 2 +- sound/isa/sb/sb8_midi.c | 2 +- sound/isa/sb/sb_common.c | 2 +- sound/isa/sb/sb_mixer.c | 2 +- sound/isa/wavefront/wavefront_fx.c | 2 +- sound/isa/wavefront/wavefront_midi.c | 2 +- sound/isa/wavefront/wavefront_synth.c | 2 +- sound/isa/wss/wss_lib.c | 2 +- sound/parisc/harmony.c | 2 +- sound/pci/ad1889.c | 3 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/als300.c | 3 +- sound/pci/als4000.c | 2 +- sound/pci/asihpi/hpioctl.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0.h | 4 +- sound/pci/aw2/aw2-saa7146.c | 2 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 2 +- sound/pci/ca0106/ca0106_mixer.c | 2 +- sound/pci/ca0106/ca0106_proc.c | 2 +- sound/pci/cmipci.c | 2 +- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx_lib.c | 3 +- sound/pci/cs46xx/dsp_spos.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/es1938.c | 3 +- sound/pci/es1968.c | 2 +- sound/pci/ice1712/ak4xxx.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 3 +- sound/pci/maestro3.c | 2 +- sound/pci/mixart/mixart_core.c | 2 +- sound/pci/mixart/mixart_hwdep.c | 2 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/oxygen_io.c | 2 +- sound/pci/pcxhr/pcxhr_core.c | 2 +- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- sound/pci/riptide/riptide.c | 2 +- sound/pci/rme32.c | 3 +- sound/pci/rme96.c | 3 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 2 +- sound/pci/sonicvibes.c | 3 +- sound/pci/trident/trident_main.c | 3 +- sound/pci/trident/trident_memory.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/vx222/vx222_ops.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/ppc/awacs.c | 2 +- sound/ppc/beep.c | 2 +- sound/ppc/burgundy.c | 2 +- sound/ppc/pmac.c | 2 +- sound/ppc/tumbler.c | 2 +- sound/sh/aica.c | 2 +- sound/sparc/amd7930.c | 2 +- sound/synth/emux/emux_hwdep.c | 2 +- sound/synth/emux/emux_oss.c | 2 +- sound/synth/emux/soundfont.c | 2 +- sound/usb/usx2y/usb_stream.h | 78 ++--------------------------------- 108 files changed, 188 insertions(+), 195 deletions(-) create mode 100644 include/uapi/sound/usb_stream.h -- 2.2.2

1 4

[alsa-devel] sigmadsp: adau1701: Version mismatch while loading firmware in 3.18.3
by Pascal Huerst 28 Jan '15

28 Jan '15

Hello Lars-Peter, i followed the instructions on: http://wiki.analog.com/resources/tools-software/linux-software/sigmadsp_gen… and built the firmware from my xml-file. But when the driver loads, I always get: adau1701 2-0034: Failed to load firmware: Invalid version 2. Supported firmware versions: 1 adau1701 2-0034: Failed to load firmware adau1701 2-0034: ASoC: can't set adau1701 hw params: -22 I have been using that .xml/.bin file in 3.14.13 for some time, but now in 3.18.3 I always get the version mismatch from above. What am I doing wrong? regards pascal

2 2

[alsa-devel] [PATCH] sound: sscape: add missing include of asm/io.h
by Arnd Bergmann 28 Jan '15

28 Jan '15

The soundscape driver uses the ISA inb/outb functions declared in asm/io.h, so it needs to include this header to avoid a build error: sscape.c: In function 'sscape_write_unsafe': sscape.c:203:2: error: implicit declaration of function 'outb' [-Werror=implicit-function-declaration] Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 018ab140c2be..7b248cdf06e2 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -23,6 +23,7 @@ #include <linux/init.h> #include <linux/err.h> +#include <linux/io.h> #include <linux/isa.h> #include <linux/delay.h> #include <linux/firmware.h>

2 3

[alsa-devel] [PATCH] ASoC: rx51: do not fail if could not get jack detection gpio
by Pali Rohár 28 Jan '15

28 Jan '15

qemu does not emulate it Signed-off-by: Pali Rohár <pali.rohar(a)gmail.com> --- sound/soc/omap/rx51.c | 45 +++++++++++++++++++++++++-------------------- 1 file changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 04896d6..a689021 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -310,25 +310,27 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; } - /* AV jack detection */ - err = snd_soc_jack_new(codec, "AV Jack", - SND_JACK_HEADSET | SND_JACK_VIDEOOUT, - &rx51_av_jack); - if (err) { - dev_err(card->dev, "Failed to add AV Jack\n"); - return err; - } + if (!IS_ERR(pdata->jack_detection_gpio)) { + /* AV jack detection */ + err = snd_soc_jack_new(codec, "AV Jack", + SND_JACK_HEADSET | SND_JACK_VIDEOOUT, + &rx51_av_jack); + if (err) { + dev_err(card->dev, "Failed to add AV Jack\n"); + return err; + } - /* prepare gpio for snd_soc_jack_add_gpios */ - rx51_av_jack_gpios[0].gpio = desc_to_gpio(pdata->jack_detection_gpio); - devm_gpiod_put(card->dev, pdata->jack_detection_gpio); + /* prepare gpio for snd_soc_jack_add_gpios */ + rx51_av_jack_gpios[0].gpio = desc_to_gpio(pdata->jack_detection_gpio); + devm_gpiod_put(card->dev, pdata->jack_detection_gpio); - err = snd_soc_jack_add_gpios(&rx51_av_jack, - ARRAY_SIZE(rx51_av_jack_gpios), - rx51_av_jack_gpios); - if (err) { - dev_err(card->dev, "Failed to add GPIOs\n"); - return err; + err = snd_soc_jack_add_gpios(&rx51_av_jack, + ARRAY_SIZE(rx51_av_jack_gpios), + rx51_av_jack_gpios); + if (err) { + dev_err(card->dev, "Failed to add GPIOs\n"); + return err; + } } return err; @@ -336,8 +338,11 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) static int rx51_card_remove(struct snd_soc_card *card) { - snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), - rx51_av_jack_gpios); + struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card); + + if (!IS_ERR(pdata->jack_detection_gpio)) + snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), + rx51_av_jack_gpios); return 0; } @@ -470,8 +475,8 @@ static int rx51_soc_probe(struct platform_device *pdev) pdata->jack_detection_gpio = devm_gpiod_get(card->dev, "jack-detection"); if (IS_ERR(pdata->jack_detection_gpio)) { + /* Do not fail, qemu does not emulate jack detection gpio */ dev_err(card->dev, "could not get jack detection gpio\n"); - return PTR_ERR(pdata->jack_detection_gpio); } pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch"); -- 1.7.9.5

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[alsa-devel] [PATCH 00/16] Yet more cleanups of line6 drivers
by Takashi Iwai 28 Jan '15

28 Jan '15

Hi, here is a series of cleanup patches for line6 stuff. They are applied on top of my previous patches. The latest patchset is found in test/line6 branch of sound.git tree. Takashi Takashi Iwai (16): ALSA: line6: Minor refactoring ALSA: line6: Fix memory leak at probe error path ALSA: line6: Remove unused line6_nop_read() ALSA: line6: Reduce superfluous spinlock in midi.c ALSA: line6: Fix missing error handling in line6_pcm_acquire() ALSA: line6: Use logical OR ALSA: line6: Fix the error recovery in line6_pcm_acquire() ALSA: line6: Drop superfluous spinlock for trigger ALSA: line6: Use incremental loop ALSA: line6: Drop voodoo workarounds ALSA: line6: Rearrange PCM structure ALSA: line6: Consolidate URB unlink and sync helpers ALSA: line6: Use dev_err() ALSA: line6: Consolidate PCM stream buffer allocation and free ALSA: line6: Do clipping in volume / monitor manipulations ALSA: line6: Skip volume manipulation during silence copying sound/usb/line6/capture.c | 122 ++++++-------------------- sound/usb/line6/capture.h | 5 -- sound/usb/line6/driver.c | 157 ++++++++++++++++----------------- sound/usb/line6/driver.h | 2 - sound/usb/line6/midi.c | 18 ++-- sound/usb/line6/midi.h | 7 +- sound/usb/line6/pcm.c | 210 +++++++++++++++++++++++++++------------------ sound/usb/line6/pcm.h | 163 ++++++++++------------------------- sound/usb/line6/playback.c | 159 ++++++++++------------------------ sound/usb/line6/playback.h | 5 -- 10 files changed, 329 insertions(+), 519 deletions(-) -- 2.2.2

2 26

[alsa-devel] [patch] ASoC: Intel: remove an unused struct member
by Dan Carpenter 28 Jan '15

28 Jan '15

We never set the ->scratch pointer, so let's delete it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index eeef9a5..78daa46 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -277,7 +277,6 @@ struct sst_hsw { /* FW config */ struct sst_hsw_ipc_fw_ready fw_ready; struct sst_hsw_ipc_fw_version version; - struct sst_module *scratch; bool fw_done; struct sst_fw *sst_fw; @@ -2135,7 +2134,6 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, hsw->dx_context, hsw->dx_context_paddr); sst_dsp_free(hsw->dsp); - kfree(hsw->scratch); kthread_stop(hsw->tx_thread); kfree(hsw->msg); }

2 1

[alsa-devel] [PATCH 0/5] Reduce races in line6 drivers
by Takashi Iwai 28 Jan '15

28 Jan '15

Hi, another day, another patchset. This time it mostly covers the races in PCM stream handling. This should work for most cases, but there are slightly uncovered conditions that is fairly difficult to fix. Maybe moving to the non-atomic PCM ops is one way, but it's too far ahead, ther are much more things to clean up before that. Takashi === Takashi Iwai (5): ALSA: line6: Fix racy loopback handling ALSA: line6: Clear prev_fbuf and prev_fsize properly ALSA: line6: Reorganize PCM stream handling ALSA: line6: Make common PCM pointer callback ALSA: line6: Handle error from line6_pcm_acquire() sound/usb/line6/capture.c | 113 +++------------ sound/usb/line6/capture.h | 1 - sound/usb/line6/pcm.c | 351 ++++++++++++++++++++++++++------------------- sound/usb/line6/pcm.h | 145 ++++++------------- sound/usb/line6/playback.c | 139 ++++-------------- sound/usb/line6/playback.h | 1 - sound/usb/line6/toneport.c | 17 ++- 7 files changed, 302 insertions(+), 465 deletions(-) -- 2.2.2

2 7

[alsa-devel] [PATCH 0/7 RESEND] ASoC: pcm512x: Clock master modes
by Peter Rosin 27 Jan '15

27 Jan '15

From: Peter Rosin <peda(a)axentia.se> Hi! [ Note that the dt change is in patch 5/7, for those only interested in that particular bit. ] This series implements BCLK master modes for the pcm512x driver. It has only been tested with the pcm5142 chip, but they are from the same family and should be compatible. I have mainly used the spec for the newer pcm5242 chip (also from the same family) as it fills in a lot of blanks in the pcm512x/pcm514x specs. The code has also seen most of its testing in a 3.10 environment, so there might be some forward-porting warts. But it is able to play sound in 3.18 as well, and most of the changes have little to do with anything but the clocking in the chip itself. This is technically not a clean resend, since I'm now sending the whole series to all parties, hopefully using the correct email addesses this time. I have also augmented the commit message of patch 1/7 according to the review comments by Lars-Peter Clausen. Cheers, Peter Peter Rosin (7): ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Change register default to match actual content after reset ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS .../devicetree/bindings/sound/pcm512x.txt | 25 +- include/sound/pcm.h | 12 + sound/core/pcm_lib.c | 85 ++ sound/soc/codecs/pcm512x.c | 955 +++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- 5 files changed, 1161 insertions(+), 25 deletions(-) -- 1.7.10.4

1 7

[alsa-devel] [PATCH] ASoC: rt5645: Add rt5650 codec support
by Bard Liao 27 Jan '15

27 Jan '15

This patch adds support for rt5650 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 158 +++++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/rt5645.h | 15 +++++ 2 files changed, 156 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 6e9cd8e..306db11 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -31,6 +31,7 @@ #include "rt5645.h" #define RT5645_DEVICE_ID 0x6308 +#define RT5650_DEVICE_ID 0x6419 #define RT5645_PR_RANGE_BASE (0xff + 1) #define RT5645_PR_SPACING 0x100 @@ -59,6 +60,10 @@ static const struct reg_default init_list[] = { }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) +static const struct reg_default rt5650_init_list[] = { + {0xf6, 0x0100}, +}; + static const struct reg_default rt5645_reg[] = { { 0x00, 0x0000 }, { 0x01, 0xc8c8 }, @@ -86,6 +91,7 @@ static const struct reg_default rt5645_reg[] = { { 0x2a, 0x5656 }, { 0x2b, 0x5454 }, { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, { 0x2f, 0x1002 }, { 0x31, 0x5000 }, { 0x32, 0x0000 }, @@ -192,6 +198,8 @@ static const struct reg_default rt5645_reg[] = { { 0xdb, 0x0003 }, { 0xdc, 0x0049 }, { 0xdd, 0x001b }, + { 0xdf, 0x0008 }, + { 0xe0, 0x4000 }, { 0xe6, 0x8000 }, { 0xe7, 0x0200 }, { 0xec, 0xb300 }, @@ -241,6 +249,7 @@ static bool rt5645_volatile_register(struct device *dev, unsigned int reg) case RT5645_IRQ_CTRL3: case RT5645_INT_IRQ_ST: case RT5645_IL_CMD: + case RT5650_4BTN_IL_CMD1: case RT5645_VENDOR_ID: case RT5645_VENDOR_ID1: case RT5645_VENDOR_ID2: @@ -286,6 +295,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_STO_DAC_MIXER: case RT5645_MONO_DAC_MIXER: case RT5645_DIG_MIXER: + case RT5650_A_DAC_SOUR: case RT5645_DIG_INF1_DATA: case RT5645_PDM_OUT_CTRL: case RT5645_REC_L1_MIXER: @@ -376,6 +386,8 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_IL_CMD: case RT5645_IL_CMD2: case RT5645_IL_CMD3: + case RT5650_4BTN_IL_CMD1: + case RT5650_4BTN_IL_CMD2: case RT5645_DRC1_HL_CTRL1: case RT5645_DRC2_HL_CTRL1: case RT5645_ADC_MONO_HP_CTRL1: @@ -1005,6 +1017,44 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5645_if1_adc_in_mux = SOC_DAPM_ENUM("IF1 ADC IN source", rt5645_if1_adc_in_enum); +/* MX-2d [3] [2] */ +static const char * const rt5650_a_dac1_src[] = { + "DAC1", "Stereo DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_L_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_l_mux = + SOC_DAPM_ENUM("A DAC1 L source", rt5650_a_dac1_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_R_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_r_mux = + SOC_DAPM_ENUM("A DAC1 R source", rt5650_a_dac1_r_enum); + +/* MX-2d [1] [0] */ +static const char * const rt5650_a_dac2_src[] = { + "Stereo DAC Mixer", "Mono DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_L_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_l_mux = + SOC_DAPM_ENUM("A DAC2 L source", rt5650_a_dac2_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_R_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_r_mux = + SOC_DAPM_ENUM("A DAC2 R source", rt5650_a_dac2_r_enum); + /* MX-2F [13:12] */ static const char * const rt5645_if2_adc_in_src[] = { "IF_ADC1", "IF_ADC2", "VAD_ADC" @@ -1149,11 +1199,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: hp_amp_power(codec, 1); /* headphone unmute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, RT5645_CP_FQ1_MASK | - RT5645_CP_FQ2_MASK | RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1173,12 +1228,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMD: /* headphone mute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, - RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | - RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1572,6 +1631,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPOR"), }; +static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { + SND_SOC_DAPM_MUX("A DAC1 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_l_mux), + SND_SOC_DAPM_MUX("A DAC1 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_r_mux), + SND_SOC_DAPM_MUX("A DAC2 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_l_mux), + SND_SOC_DAPM_MUX("A DAC2 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_r_mux), +}; + static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, @@ -1777,13 +1847,9 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, - { "DAC L1", NULL, "Stereo DAC MIXL" }, { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R1", NULL, "Stereo DAC MIXR" }, { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC L2", NULL, "Mono DAC MIXL" }, { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R2", NULL, "Mono DAC MIXR" }, { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, { "SPK MIXL", "BST1 Switch", "BST1" }, @@ -1872,6 +1938,30 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "SPOR", NULL, "SPK amp" }, }; +static const struct snd_soc_dapm_route rt5650_specific_dapm_routes[] = { + { "A DAC1 L Mux", "DAC1", "DAC1 MIXL"}, + { "A DAC1 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC1 R Mux", "DAC1", "DAC1 MIXR"}, + { "A DAC1 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + + { "A DAC2 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC2 L Mux", "Mono DAC Mixer", "Mono DAC MIXL"}, + { "A DAC2 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + { "A DAC2 R Mux", "Mono DAC Mixer", "Mono DAC MIXR"}, + + { "DAC L1", NULL, "A DAC1 L Mux" }, + { "DAC R1", NULL, "A DAC1 R Mux" }, + { "DAC L2", NULL, "A DAC2 L Mux" }, + { "DAC R2", NULL, "A DAC2 R Mux" }, +}; + +static const struct snd_soc_dapm_route rt5645_specific_dapm_routes[] = { + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC R2", NULL, "Mono DAC MIXR" }, +}; + static int rt5645_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2287,6 +2377,22 @@ static int rt5645_probe(struct snd_soc_codec *codec) rt5645->codec = codec; + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5645: + snd_soc_dapm_add_routes(&codec->dapm, + rt5645_specific_dapm_routes, + ARRAY_SIZE(rt5645_specific_dapm_routes)); + break; + case CODEC_TYPE_RT5650: + snd_soc_dapm_new_controls(&codec->dapm, + rt5650_specific_dapm_widgets, + ARRAY_SIZE(rt5650_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5650_specific_dapm_routes, + ARRAY_SIZE(rt5650_specific_dapm_routes)); + break; + } + rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); @@ -2418,6 +2524,7 @@ static const struct regmap_config rt5645_regmap = { static const struct i2c_device_id rt5645_i2c_id[] = { { "rt5645", 0 }, + { "rt5650", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); @@ -2450,9 +2557,18 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } regmap_read(rt5645->regmap, RT5645_VENDOR_ID2, &val); - if (val != RT5645_DEVICE_ID) { + + switch (val) { + case RT5645_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5645; + break; + case RT5650_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5650; + break; + default: dev_err(&i2c->dev, - "Device with ID register %x is not rt5645\n", val); + "Device with ID register %x is not rt5645 or rt5650\n", + val); return -ENODEV; } @@ -2463,6 +2579,14 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + ret = regmap_register_patch(rt5645->regmap, rt5650_init_list, + ARRAY_SIZE(rt5650_init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Apply rt5650 patch failed: %d\n", + ret); + } + if (rt5645->pdata.in2_diff) regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, RT5645_IN_DF2, RT5645_IN_DF2); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index a815e36..7454231 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -47,6 +47,7 @@ #define RT5645_STO_DAC_MIXER 0x2a #define RT5645_MONO_DAC_MIXER 0x2b #define RT5645_DIG_MIXER 0x2c +#define RT5650_A_DAC_SOUR 0x2d #define RT5645_DIG_INF1_DATA 0x2f /* Mixer - PDM */ #define RT5645_PDM_OUT_CTRL 0x31 @@ -150,6 +151,8 @@ #define RT5645_IL_CMD 0xdb #define RT5645_IL_CMD2 0xdc #define RT5645_IL_CMD3 0xdd +#define RT5650_4BTN_IL_CMD1 0xdf +#define RT5650_4BTN_IL_CMD2 0xe0 #define RT5645_DRC1_HL_CTRL1 0xe7 #define RT5645_DRC2_HL_CTRL1 0xe9 #define RT5645_MUTI_DRC_CTRL1 0xea @@ -472,6 +475,12 @@ #define RT5645_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) #define RT5645_DAC_L2_DAC_R_VOL_SFT 4 +/* Analog DAC1/2 Input Source Control (0x2d) */ +#define RT5650_A_DAC1_L_IN_SFT 3 +#define RT5650_A_DAC1_R_IN_SFT 2 +#define RT5650_A_DAC2_L_IN_SFT 1 +#define RT5650_A_DAC2_R_IN_SFT 0 + /* Digital Interface Data Control (0x2f) */ #define RT5645_IF1_ADC2_IN_SEL (0x1 << 15) #define RT5645_IF1_ADC2_IN_SFT 15 @@ -2175,6 +2184,11 @@ enum { RT5645_DMIC_DATA_GPIO11, }; +enum { + CODEC_TYPE_RT5645, + CODEC_TYPE_RT5650, +}; + struct rt5645_priv { struct snd_soc_codec *codec; struct rt5645_platform_data pdata; @@ -2184,6 +2198,7 @@ struct rt5645_priv { struct snd_soc_jack *mic_jack; struct delayed_work jack_detect_work; + int codec_type; int sysclk; int sysclk_src; int lrck[RT5645_AIFS]; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: Intel: Used lock version to update shim registers
by Jie Yang 27 Jan '15

27 Jan '15

We need hold lock each time updating shirm registers, otherwise, we may set unexpected values to them when they are set in different thread at different time sequence. The notification work will be scheduled in global work queue, which won't hold this sst->spinlock itself, so here we need change to use the lock version to update shim registers. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index eeef9a5..e18cb07 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -653,11 +653,11 @@ static void hsw_notification_work(struct work_struct *work) } /* tell DSP that notification has been handled */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD, + sst_dsp_shim_update_bits(hsw->dsp, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); /* unmask busy interrupt */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); + sst_dsp_shim_update_bits(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); } static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: ts3a227e: Add dts property that allows to specify micbias voltage
by Anatol Pomozov 27 Jan '15

27 Jan '15

The voltage controls key press threshold. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/ts3a227e.txt | 5 ++++ sound/soc/codecs/ts3a227e.c | 27 ++++++++++++++++++++++ 2 files changed, 32 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt index e8bf23e..a836881 100644 --- a/Documentation/devicetree/bindings/sound/ts3a227e.txt +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -13,6 +13,11 @@ Required properties: - interrupt-parent: The parent interrupt controller - interrupts: Interrupt number for /INT pin from the 227e +Optional properies: + - ti,micbias: Intended MICBIAS voltage (datasheet section 9.6.7). + Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage + 2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V + Default value is "1" (2.2V). Examples: diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 54d431c..5ecfa31 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -79,6 +79,10 @@ static const int ts3a227e_buttons[] = { /* TS3A227E_REG_SETTING_2 0x05 */ #define KP_ENABLE 0x04 +/* TS3A227E_REG_SETTING_3 0x06 */ +#define MICBIAS_SETTING_SFT (3) +#define MICBIAS_SETTING_MASK (0x7 << MICBIAS_SETTING_SFT) + /* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ #define TYPE_3_POLE 0x01 #define TYPE_4_POLE_OMTP 0x02 @@ -248,6 +252,21 @@ static const struct regmap_config ts3a227e_regmap_config = { .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), }; +static int ts3a227e_parse_dt(struct ts3a227e *ts3a227e, struct device_node *np) +{ + u32 micbias; + int err; + + err = of_property_read_u32(np, "ti,micbias", &micbias); + if (!err) { + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_3, + MICBIAS_SETTING_MASK, + (micbias & 0x07) << MICBIAS_SETTING_SFT); + } + + return 0; +} + static int ts3a227e_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -266,6 +285,14 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, if (IS_ERR(ts3a227e->regmap)) return PTR_ERR(ts3a227e->regmap); + if (dev->of_node) { + ret = ts3a227e_parse_dt(ts3a227e, dev->of_node); + if (ret) { + dev_err(dev, "Failed to parse device tree: %d\n", ret); + return ret; + } + } + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "TS3A227E", ts3a227e); -- 2.2.0.rc0.207.ga3a616c

2 1

[alsa-devel] [PATCH] ASoC: ts3a227e: Remap keys to match Android headset specification
by Anatol Pomozov 27 Jan '15

27 Jan '15

ts3a227e datasheet says typical key resistance is key1 50 Om key2 135 Om key3 240 Om key4 470 Om The android headset specification expect buttons impedance: A (MEDIA) 0-70 Om D (VOICECOMMAND) 110-180 Om B (VOLUMEUP) 210-290 Om C (VOLUMEDOWN) 360-680 Om Thus key mapping should be key1 - MEDIA key2 - VOICECOMMAND key3 - VOLUMEUP key3 - VOLUMEDOWN Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/ts3a227e.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 9f2dced..54d431c 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -221,9 +221,9 @@ int ts3a227e_enable_jack_detect(struct snd_soc_component *component, struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); ts3a227e->jack = jack; ts3a227e_jack_report(ts3a227e); -- 2.2.0.rc0.207.ga3a616c

3 3

[alsa-devel] [PATCH] ASoC: wm8731: init mutex in i2c init path
by Manuel Lauss 27 Jan '15

27 Jan '15

The I2C init path forgot to init the mutex, leading to an oops when controls are accessed. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Fixes an oops on bootup when alsactl tries to restore codec parameters on my MIPS DB1300 board. sound/soc/codecs/wm8731.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b9211b4..b115ed8 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -717,6 +717,8 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + mutex_init(&wm8731->lock); + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); -- 2.2.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: atmel_ssc_dai: fix start event for I2S mode
by Bo Shen 27 Jan '15

27 Jan '15

According to the I2S specification information as following: - WS = 0, channel 1 (left) - WS = 1, channel 2 (right) So, the start event should be TF/RF falling edge. Reported-by: Songjun Wu <songjun.wu(a)atmel.com> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: None sound/soc/atmel/atmel_ssc_dai.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 99ff35e..e691aab 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -348,7 +348,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; - int start_event; int ret; int fslen, fslen_ext; @@ -457,19 +456,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * The SSC transmit clock is obtained from the BCLK signal on * on the TK line, and the SSC receive clock is * generated from the transmit clock. - * - * For single channel data, one sample is transferred - * on the falling edge of the LRC clock. - * For two channel data, one sample is - * transferred on both edges of the LRC clock. */ - start_event = ((channels == 1) - ? SSC_START_FALLING_RF - : SSC_START_EDGE_RF); - rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) - | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? @@ -478,14 +468,14 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(RFMR_FSLEN, 0) - | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) | SSC_BF(RFMR_DATLEN, (bits - 1)); tcmr = SSC_BF(TCMR_PERIOD, 0) | SSC_BF(TCMR_STTDLY, START_DELAY) - | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? @@ -495,7 +485,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(TFMR_FSLEN, 0) - | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) | SSC_BF(TFMR_DATLEN, (bits - 1)); -- 2.3.0.rc0

2 3

[alsa-devel] add Steinberg UR44 USB Audio Interface support to quirks-table
by Vlastimil Wagner 27 Jan '15

27 Jan '15

hi In what state is the driver for Steinberg UR44 thanks Bukowski

2 1

[alsa-devel] [PATCH 00/10] ASoC: sta32x: Add device tree binding
by Thomas Niederprüm 27 Jan '15

27 Jan '15

This patchset is the result of comparing the sta32x driver with the sta350 driver with the goal to make the sta32x driver device tree capable. In the merging process the coding style of the sta350 driver was adopted since it seemed more up to date. I did my best to break up the changes in logical chunks. Additionally some small fixes were added. The comparison with the sta350 lead to the following changes: - Switch to direct usage of the regmap API - Use the kernel gpio framework to control the reset pin of the sta32x - adopt calculation of mlck divider and extrapolation ratio from sta350 - add device tree bindings - Change the Codecs DAI name to be coherent with the sta350 driver Moreover some minor fixes were added: - Add the status register - Fix bit shift value for the IDE bit in the CONFF register - Add description for the driver and I2C dependency in Kconfig Thomas Niederprüm (10): ASoC: sta32x: Convert to direct regmap API usage. ASoC: sta32x: make sta32x a gpio consumer for the reset GPIO ASoC: sta32x: use DECLARE_TLV_DB_RANGE macro. ASoC: sta32x: add status register. ASoC: sta32x: move code to calculate mclk divider and extrapolation ratio to sta32x_hw_params() ASoC: sta32x: add device tree binding. ASoC: sta32x: use dev_dbg() for debug output. ASoC: sta32x: minor Kconfig update. ASoC: sta32x: correct bit shift value for IDE register. ASoC: sta32x: change dai name to be in line with the sta350 driver. .../devicetree/bindings/sound/st,sta32x.txt | 92 ++++ include/sound/sta32x.h | 18 +- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/sta32x.c | 537 +++++++++++++-------- sound/soc/codecs/sta32x.h | 4 +- 5 files changed, 450 insertions(+), 205 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/st,sta32x.txt -- 2.1.1

4 17

[alsa-devel] [PATCH_V3 0/4] sound:soc: jz4740: DT, dynamic sampling, enable clocks
by Zubair Lutfullah Kakakhel 27 Jan '15

27 Jan '15

Hi, Here are a few simple patches for the jz4740. First adds dynamic sampling support to jz4740-i2s. Then two to add a simple binding and DT support. Then a patch to set the clock rate for i2s. These are in preparation for jz4780 and ci20 later on. Patches are based on 3.19-rc6. Quite disjoint and stay within jz4740 so should apply easily on other trees. If you would like to have them rebased to a different tree, please tell. Thank-you V3 changes: Added dma binding to example but forgot to put it in required properties. V2 changes: Removed a redundant #define Added #ifdef CONFIG_OF Added dma in binding document Removed a redundant call to clk_prepare_enable Zubair Lutfullah Kakakhel (4): sound: soc: jz4740: Add dynamic sampling rate support to jz4740-i2s dt: sound: jz4740: Add binding documentation for jz4740-i2s sound: soc: jz4740: Add DT support to jz4740-i2s driver sound: jz4740: set i2s clk rate to 12MHz .../bindings/sound/ingenic,jz4740-i2s.txt | 23 ++++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.c | 22 ++++++++++++++++++++- 2 files changed, 44 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt -- 1.9.1

2 6

Re: [alsa-devel] [PATCH v2 00/16] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 27 Jan '15

27 Jan '15

Hello Tobias, On 24/01/15 02:53, Tobias Jakobi wrote: > Hello, > > I've tested the series on my X2 and so far I haven't encountered any > obvious issues with it. > > I have a small question though. With the move to simple-audio-card the > old driver (selected by SND_SOC_ODROIDX2) is probably going away after > some time. > Currently SND_SOC_ODROIDX2 also selects SND_SOC_MAX98090 and > SND_SAMSUNG_I2S, which I believe (correct me if I'm wrong) are necessary > to use the sound subsystem even with simple-audio-card (since the > max98090 codec has to be built at least). I've found no way of selecting > these two manually, so at the moment I need both CONFIG_SND_SIMPLE_CARD > and SND_SOC_ODROIDX2 enabled to use sound. Is this intended behaviour? Thanks for the feedback. It seems I finally forgot to update Kconfig, presumably we should make SND_SOC_ODROIDX2 selecting SND_SIMPLE_CARD, i.e. diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 58c797a..7fb728c 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ODROIDX2 depends on SND_SOC_SAMSUNG select SND_SOC_MAX98090 select SND_SAMSUNG_I2S + select SND_SIMPLE_CARD help Say Y here to enable audio support for the Odroidx2. Making SND_SOC_SAMSUNG selecting SND_SIMPLE_CARD is probably not a good idea and still we would be missing the codec part. I'm going to post a patch as above. > And while I'm at it. I'm booting the board with an upstream u-boot > version which sets the MPLL to 800MHz, which AFAIK is the clocking rate > that makes less 'trouble' (but at the expense of performance). > I was wondering if anyone can comment on whether the recent rework has > any influence on the behaviour when booting with the vendor u-boot > (which sets the MPLL to 880MHz). IIRC then clock rounding issues arose > there. > > It would be interesting to know if this series gets us closer to drive > the MPLL with the recommended (?) higher clocking rate. No, (unfortunately) this series has nothing to do with MPLL. The whole clock tree for sound is configured by the kernel, including the EPLL, which provides reference clock for the audio subsystem. MPLL and EPLL are normally used in different clock domains. So there should be no any clock rounding issues related to bootloader as far as audio is concerned. Please let me know if you find any related problems. -- Regards, Sylwester

1 0

[alsa-devel] [PATCH v2 1/3] ASoC: codecs: wm8904: add dt ids table
by Bo Shen 27 Jan '15

27 Jan '15

From: Alexander Morozov <linux(a)meltdown.ru> Signed-off-by: Alexander Morozov <linux(a)meltdown.ru> [Add driver data to distinguish device type] Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: - Add driver data for distinguish the device capability. sound/soc/codecs/wm8904.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..6e3f175 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2255,10 +2255,32 @@ static const struct i2c_device_id wm8904_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); +#ifdef CONFIG_OF +static enum wm8904_type wm8904_data = WM8904; +static enum wm8904_type wm8912_data = WM8912; + +static const struct of_device_id wm8904_of_match[] = { + { + .compatible = "wlf,wm8904", + .data = &wm8904_data, + }, { + .compatible = "wlf,wm8912", + .data = &wm8912_data, + }, { + .compatible = "wlf,wm8918", + .data = &wm8904_data, + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, wm8904_of_match); +#endif + static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(wm8904_of_match), }, .probe = wm8904_i2c_probe, .remove = wm8904_i2c_remove, -- 2.1.0.24.g4109c28

4 9

[alsa-devel] Regression in kernel 3.19-rc5 for usb2 scarlett 18i20
by Guido Aulisi 26 Jan '15

26 Jan '15

Hi, I noticed a regression in 3.19-rc5 kernel (as shipped by fedora rawhide) with usb2 focusrite scarlett 18i20. I get a lot of xruns with jack, even at 1024 frame buffer and 2 periods. In 3.17.8 it works quite well without xruns at the same settings. I use a fedora 20 (64bit) laptop with 8GB ram, I only installed the rawhide kernel to try the new included scarlett mixer, no other software components changed. I can try to do some other test, but compiling the kernel on this laptop is time consuming. Thanks Guido

3 5

[alsa-devel] [PATCH 0/7] ASoC: pcm512x: Clock master modes
by Peter Rosin 26 Jan '15

26 Jan '15

From: Peter Rosin <peda(a)axentia.se> Hi! This series implements BCLK master modes for the pcm512x driver. It has only been tested with the pcm5142 chip, but they are from the same family and should be compatible. I have mainly used the spec for the newer pcm5242 chip (also from the same family) as it fills in a lot of blanks in the pcm512x/pcm514x specs. The code has also seen most of its testing in a 3.10 environment, so there might be some forward-porting warts. But it is able to play sound in 3.18 as well, and most of the changes have little to do with anything but the clocking in the chip itself. Cheers, Peter Peter Rosin (7): ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Change register default to match actual content after reset ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS .../devicetree/bindings/sound/pcm512x.txt | 25 +- include/sound/pcm.h | 12 + sound/core/pcm_lib.c | 85 ++ sound/soc/codecs/pcm512x.c | 955 +++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- 5 files changed, 1161 insertions(+), 25 deletions(-) -- 1.7.10.4

2 8

[alsa-devel] [PATCH] ASoC: rt5640: Add RT5642 ACPI ID for Intel Baytrail
by Jarkko Nikula 26 Jan '15

26 Jan '15

Asus T100TAF uses ACPI ID "10EC5642" for its audio codec. I suppose it is updated ACPI ID for the RT5642 codec since some earlier platforms are using "10EC5640" with the RT5642 too. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> --- ACPI ID found from the DSDT table of Asus T100TAF. This makes the rt5640 codec driver probing but the audio itself is not working due a bit different ACPI data and perhaps HW differences too compared to for instance Asus T100TA. --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b386958de042..178e55d4d481 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2125,6 +2125,7 @@ MODULE_DEVICE_TABLE(of, rt5640_of_match); static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { "10EC5640", 0 }, + { "10EC5642", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: rt286: Fix capture volume setting issue
by Bard Liao 26 Jan '15

26 Jan '15

The purpose of rt286_adc_event is to mute/numnte the ADC mixer. However, it will also set the capture volume to default value. As a result, "ADC0 Capture Volume" is not working if it is set before capture start. This patch remove rt286_adc_event and add "ADC0 Capture Switch" to mute/unmute ADC mixer. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 38 ++++++-------------------------------- 1 file changed, 6 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1d1c7f8..847cc4b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -417,6 +417,8 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); static const struct snd_kcontrol_new rt286_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R("ADC0 Capture Switch", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 7, 1, 1), SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, @@ -538,32 +540,6 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } -static int rt286_adc_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - unsigned int nid; - - nid = (w->reg >> 20) & 0xff; - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7000); - break; - case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7080); - break; - default: - return 0; - } - - return 0; -} - static int rt286_vref_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -667,12 +643,10 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), /* ADC Mux */ - SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, - &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, - &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux), + SND_SOC_DAPM_MUX("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux), /* Audio Interface */ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 0/2] ASoC: Fix wm97xx touchscreen driver
by Lars-Peter Clausen 26 Jan '15

26 Jan '15

The wm97xx touchscreen driver will bind itself to the snd_ac97 device registered by CODEC drivers and expects that the device has already been reset by the CODEC driver. Previous to commit 6794f709b712 ("ASoC: ac97: Drop delayed device registration") the snd_ac97 device was only registered after the CODEC driver probe function had finished running, but starting with the commit it is registered within snd_soc_new_ac97_codec(). This breaks the touchscreen driver as the reset is no longer performed before the touchscreen driver probe function runs. This patch series introduces a new function snd_soc_alloc_ac97_codec() which allocates the snd_ac97 device, but does not yet register it yet. This allows the CODEC drivers to perform the reset before the device is registered. The series is meant to perform a minimum amount of changes while fixing the issue to prevent introducing other regressions. I have a few more patches that consolidate the reset handling in the drivers and puts it into the core in snd_soc_new_ac97_codec() which makes the split in the drivers unnecessary, but that's something for next and not for stable. Manuel can you give this another round of testing? Thanks, - Lars Lars-Peter Clausen (2): ASoC: Add support for allocating AC'97 device before registering it ASoC: wm97xx: Reset AC'97 device before registering it include/sound/soc.h | 1 + sound/soc/codecs/wm9705.c | 16 ++++++++++------ sound/soc/codecs/wm9712.c | 12 ++++++++---- sound/soc/codecs/wm9713.c | 12 ++++++++---- sound/soc/soc-ac97.c | 34 +++++++++++++++++++++++++++++----- 5 files changed, 56 insertions(+), 19 deletions(-) -- 1.8.0

4 6

[alsa-devel] broadwell-audio is unable to initialize on certain laptops
by Tom Li 26 Jan '15

26 Jan '15

Hello everyone. I'm here for reporting an issue. I bought a new ThinkPad Helix 2, but I found that the built-in sound output device is not recognize, only external HDMI sound output is possible. # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 When I check dmesg output, I found some error message that indicated that broadwell-audio is unable to initialize. # dmesg | egrep "sound|audio" [ 1.827797] input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input8 [ 1.827935] input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input9 [ 1.828071] input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input10 [ 1.952913] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 1.952924] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.062812] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.062837] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.350951] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.350987] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.375744] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.375759] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.386019] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.386037] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.706751] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.706777] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.042190] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.042202] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.048200] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.048212] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.048382] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.048391] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.199788] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.199812] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.208241] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.208259] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.216577] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.216600] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.270629] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.270659] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.324400] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.324419] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.604723] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.604755] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.604800] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.604817] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.673159] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.673190] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.727312] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.727352] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.727387] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.727401] platform broadwell-audio: Driver broadwell-audio requests probe deferral I tried both 3.18.3 and 3.19.0-rc6, the results were same. It seems that we need to add some quirks for this hardware. I also found a similar bug on Launchpad [0]. I filed a kernel bug [1] and I hope someone could fix it. [0]: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1413446 [1]: https://bugzilla.kernel.org/show_bug.cgi?id=92061 Thanks.

1 0

[alsa-devel] [alsa] Crackling after 6.45 hours
by a.renault＠kaptalia.com 26 Jan '15

26 Jan '15

Hi everyone, I currently facing the following issue : My sound crackling after 6.45 hours of music without reinitialization. I am using sdl-mixer, sdl and alsa. I deployed a bug on libsdl at the address : https://bugzilla.libsdl.org/show_bug.cgi?id=2794 And after some works we discover that the snd_pcm_writei answer "-1073741824". The error code "-1073741824" is not one of -EINTR, -EPIPE, -ESTRPIPE, -EAGAIN. So "snd_pcm_recover" does not handle anything. If you are brave enought please take a look at the link, Any idea would be greatly appreciated. Best regards, -- Adrien

4 6

[alsa-devel] [PATCH_V2 0/4] sound:soc: jz4740: DT, dynamic sampling, enable clocks
by Zubair Lutfullah Kakakhel 26 Jan '15

26 Jan '15

Hi, Here are a few simple patches for the jz4740. First adds dynamic sampling support to jz4740-i2s. Then two to add a simple binding and DT support. Then a patch to set the clock rate for i2s. These are in preparation for jz4780 and ci20 later on. Patches are based on 3.19-rc6. Quite disjoint and stay within jz4740 so should apply easily on other trees. If you would like to have them rebased to a different tree, please tell. Thank-you V2 changes: Removed a redundant #define Added #ifdef CONFIG_OF Added dma in binding document Removed a redundant call to clk_prepare_enable Zubair Lutfullah Kakakhel (4): sound: soc: jz4740: Add dynamic sampling rate support to jz4740-i2s dt: sound: jz4740: Add binding documentation for jz4740-i2s sound: soc: jz4740: Add DT support to jz4740-i2s driver sound: jz4740: set i2s clk rate to 12MHz .../bindings/sound/ingenic,jz4740-i2s.txt | 21 +++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.c | 22 +++++++++++++++++++++- 2 files changed, 42 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt -- 1.9.1

2 5

[alsa-devel] [PATCH 0/6] sequencer fixes
by Clemens Ladisch 26 Jan '15

26 Jan '15

One bug fix, and various unimportant cleanups for -next. include/sound/seq_kernel.h | 9 +-------- sound/core/seq/seq_clientmgr.c | 3 +-- sound/core/seq/seq_dummy.c | 31 ------------------------------- sound/core/seq/seq_ports.c | 9 +++------ sound/core/seq/seq_ports.h | 1 - 5 files changed, 5 insertions(+), 48 deletions(-)

2 8

[alsa-devel] [PATCH 0/3] Another round of line6 cleanup
by Takashi Iwai 26 Jan '15

26 Jan '15

Hi, here is yet another series of cleanup patches for line6 drivers. This time, the extra allocation for struct usb_line6 was removed by using snd_card_new()'s allocation feature. In addition to that, a few trivial cleanups. The test/line6 branch of sound git tree was updated with these patches, too. Takashi === Takashi Iwai (3): ALSA: line6: Drop interface argument from private_init and disconnect callbacks ALSA: line6: Let snd_card_new() allocate private data ALSA: line6: Minor tidy up in line6_probe() sound/usb/line6/driver.c | 52 ++++++++++++++++++++-------------------------- sound/usb/line6/driver.h | 8 ++++--- sound/usb/line6/pod.c | 21 +++++++------------ sound/usb/line6/podhd.c | 13 ++++-------- sound/usb/line6/toneport.c | 21 +++++++------------ sound/usb/line6/variax.c | 19 ++++++----------- 6 files changed, 54 insertions(+), 80 deletions(-) -- 2.2.2

1 3

[alsa-devel] ASoC fixes for v3.19
by Mark Brown 26 Jan '15

26 Jan '15

The following changes since commit 26bc420b59a38e4e6685a73345a0def461136dce: Linux 3.19-rc6 (2015-01-25 20:04:41 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19-rc6 for you to fetch changes up to 88343ee38d7ae55e39cda74c5d137cb2c454e66c: Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/simple', 'asoc/fix/ts3a227e', 'asoc/fix/wm8904' and 'asoc/fix/wm8960' into asoc-linus (2015-01-26 11:29:58 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.19 There's a lot more fixes here than I'd like since I've been lax in sending things throughout the release cycle but there's only two in generic code and they've had quite some time to cook in -next so hopefully on balance are OK. The two generic fixes are a fix for crashes on capture DAIs in the compress code and a fix for error handling on probe failures which would have been harmless in the past but now oopses with the new dynamic probe code. ---------------------------------------------------------------- Andrew Jackson (1): ASoC: adi: Add missing return statement. Aurelien BOUIN (1): ASoC: fsl_esai: Fix incorrect xDC field width of xCCR registers Bard Liao (1): ASoC: rt286: set the same format for dac and adc Bo Shen (1): ASoC: wm8904: fix runtime warning Cheng-Yi Chiang (1): ASoC: ts3a227e: Check and report jack status at probe Fabio Estevam (2): ASoC: fsl_ssi: Fix irq error check ASoC: fsl: imx-wm8962: Set the card owner field Geert Uytterhoeven (1): ASoC: simple-card: Fix crash in asoc_simple_card_unref() Jianqun Xu (1): ASoC: rockchip: i2s: applys rate symmetry for CPU DAI Jie Yang (2): ASoC: Intel: Don't change offset of block allocator during fixed allocate ASoC: Intel: Add NULL checks for the stream pointer Mark Brown (4): Merge remote-tracking branch 'asoc/fix/fsl-esai' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm512x' into asoc-linus Merge remote-tracking branches 'asoc/fix/adi', 'asoc/fix/compress', 'asoc/fix/fsl-ssi', 'asoc/fix/imx', 'asoc/fix/intel', 'asoc/fix/omap', 'asoc/fix/rockchip' and 'asoc/fix/rt286' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/simple', 'asoc/fix/ts3a227e', 'asoc/fix/wm8904' and 'asoc/fix/wm8960' into asoc-linus Oder Chiou (1): ASoC: rt5677: Modify the behavior that updates the PLL parameter. Peter Rosin (1): ASoC: pcm512x: Fix DSP program selection Peter Ujfalusi (1): ASoC: omap-mcbsp: Correct CBM_CFS dai format configuration Qais Yousef (1): ASoC: soc-compress.c: fix NULL dereference Zidan Wang (1): ASoC: wm8960: Fix capture sample rate from 11250 to 11025 sound/soc/adi/axi-i2s.c | 2 ++ sound/soc/codecs/pcm512x.c | 2 +- sound/soc/codecs/rt286.c | 6 ++---- sound/soc/codecs/rt5677.c | 18 ++++++++++++++---- sound/soc/codecs/ts3a227e.c | 6 ++++++ sound/soc/codecs/wm8904.c | 23 +++++++++++++++-------- sound/soc/codecs/wm8960.c | 2 +- sound/soc/fsl/fsl_esai.h | 2 +- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/imx-wm8962.c | 1 + sound/soc/generic/simple-card.c | 7 +++---- sound/soc/intel/sst-firmware.c | 13 +++++++------ sound/soc/intel/sst-haswell-ipc.c | 30 ++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 1 + sound/soc/soc-compress.c | 9 ++++++--- 16 files changed, 93 insertions(+), 35 deletions(-)

2 1

[alsa-devel] [PATCH 0/4] sound:soc: jz4740: DT, dynamic sampling, enable clocks
by Zubair Lutfullah Kakakhel 26 Jan '15

26 Jan '15

Hi, Here are a few simple patches for the jz4740. First adds dynamic sampling support to jz4740-i2s. Then two to add a simple binding and DT support. Then a patch to enable the codec clock. These are in preparation for jz4780 and ci20 later on. These are based on 3.19-rc6. If you would like to have them rebased to a different tree, please tell. Thank-you Zubair Lutfullah Kakakhel (4): sound: soc: jz4740: Add dynamic sampling rate support to jz4740-i2s dt: sound: jz4740: Add binding documentation for jz4740-i2s sound: soc: jz4740: Add DT support to jz4740-i2s driver sound: jz4740: Enable codec clock during dai_probe .../bindings/sound/ingenic,jz4740-i2s.txt | 18 ++++++++++++++++ sound/soc/jz4740/jz4740-i2s.c | 24 +++++++++++++++++++++- 2 files changed, 41 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt -- 1.9.1

3 12

[alsa-devel] [PATCH - UCM 1/1] ucm: add binary configure file parse
by han.lu＠intel.com 26 Jan '15

26 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> with cset command, UCM set kcontrol parameters directly: cset "name='<KCONTROL_NAME>' 1<,2,3,...>" This patch enables UCM to set kcontrol with parameters from configure file: cset-bin-file "name='<KCONTROL_NAME>' <path/to/file>" where "cset-bin-file" is a newly added keyword alongside of "cset", to indicate cset with binary data in file. The binary data in file is parameter for audio DSPs, and it's just passed by UCM/ALSA as raw data. The data type of parameter elements must be byte, and the count must matches driver definition. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 37ae4c8..182f174 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -34,6 +34,7 @@ #include <ctype.h> #include <stdarg.h> #include <pthread.h> +#include <sys/stat.h> /* * misc @@ -160,11 +161,65 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int binary_file_parse(snd_ctl_elem_value_t *dst, + snd_ctl_elem_info_t *info, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + char *res; + snd_ctl_elem_type_t type; + unsigned int idx, count; + + type = snd_ctl_elem_info_get_type(info); + if (type != SND_CTL_ELEM_TYPE_BYTES) { + uc_error("only support byte type!"); + err = -EINVAL; + return err; + } + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + count = snd_ctl_elem_info_get_count(info); + if (sz != count || sz > sizeof(dst->value.bytes)) { + uc_error("invalid parameter size %d!", sz); + err = -EINVAL; + goto __fail; + } + res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -errno; + goto __fail_read; + } + for (idx = 0; idx < sz; idx++) + snd_ctl_elem_value_set_byte(dst, idx, *(res + idx)); + __fail_read: + free(res); + __fail: + close(fd); + return err; +} + extern int __snd_ctl_ascii_elem_id_parse(snd_ctl_elem_id_t *dst, const char *str, const char **ret_ptr); -static int execute_cset(snd_ctl_t *ctl, const char *cset) +static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) { const char *pos; int err; @@ -194,7 +249,10 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset) err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); if (err < 0) goto __fail; err = snd_ctl_elem_write(ctl, value); @@ -239,6 +297,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail_nomem; break; case SEQUENCE_ELEMENT_TYPE_CSET: + case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: if (cdev == NULL) { const char *cdev1 = NULL, *cdev2 = NULL; err = get_value3(&cdev1, "PlaybackCTL", @@ -274,7 +333,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail; } } - err = execute_cset(ctl, s->data.cset); + err = execute_cset(ctl, s->data.cset, s->type); if (err < 0) { uc_error("unable to execute cset '%s'\n", s->data.cset); goto __fail; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index d7517f6..9e1cb41 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -306,6 +306,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-bin-file") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-bin-file requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 87f14a2..c1655c7 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -47,6 +47,7 @@ #define SEQUENCE_ELEMENT_TYPE_CSET 2 #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 +#define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 struct ucm_value { struct list_head list; -- 2.1.0

2 1

[alsa-devel] [PATCH 0/3] ARM: edma: Correct header file usage
by Peter Ujfalusi 26 Jan '15

26 Jan '15

Hi, The linux/platform_data/edma.h file was used for API definition as well, which is not correct since the header should only contain platform data related structures, defines, etc. Mark: I think this series can go via arm since it is just changing include stuff under ASoC. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-evm: Do not include edma headers ARM: edma: Rename header file for dmaengine filter function definition ARM: edma: Split up header file to platform_data and API file arch/arm/common/edma.c | 3 +- arch/arm/mach-davinci/devices.c | 1 + arch/arm/mach-davinci/include/mach/da8xx.h | 1 + drivers/dma/edma.c | 2 +- drivers/mmc/host/davinci_mmc.c | 3 +- drivers/spi/spi-davinci.c | 2 +- include/linux/edma-dmaengine.h | 29 ++++++ include/linux/edma.h | 152 ++++++++++++++++++++++++++--- include/linux/platform_data/edma.h | 148 ++-------------------------- sound/soc/davinci/davinci-evm.c | 3 - sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/davinci/edma-pcm.c | 2 +- 12 files changed, 182 insertions(+), 165 deletions(-) create mode 100644 include/linux/edma-dmaengine.h -- 2.1.3

7 19

[alsa-devel] [PATCH 0/3] Add master mode, tmd and right-j mode support
by Zidan Wang 23 Jan '15

23 Jan '15

Add sai master mode support. Add tdm slots support. Add Right-J mode support. Zidan Wang (3): SoC: fsl_sai: add sai master mode support ASoC: fsl_sai: Add support for tdm slots operation ASoC: fsl_sai: Add support for Right-J mode sound/soc/fsl/fsl_sai.c | 164 ++++++++++++++++++++++++++++++++++++++++++++---- sound/soc/fsl/fsl_sai.h | 13 +++- 2 files changed, 164 insertions(+), 13 deletions(-) -- 1.9.1

4 18

[alsa-devel] [PATCH v3 0/4] Sound support for Exynos4412 Trats2 board
by Inha Song 23 Jan '15

23 Jan '15

This patch-set adds basic sound support for the Trats2 boards. It just support primary I2s and external speaker playback. Changes in v3: - Set clock out parent to XUSBXTI in pmu_system_controller DT node. Changes in v2: - Remove unnecessary lines in Trats2 machine driver - move clock enable/disable to .startup / .shutdown - Fix code style issues - Remove mclk DT property in Trats2 sound and add MCLK1 property in WM1811 node. (MCLK1 is optional property in WM8994 node - refer: Document/devicetree/bindings/sound/wm8994.txt) - Use mclk that defined in WM1811 DT node. - Add "samsung,audio-codec" property to required properties. Inha Song (4): ASoC: samsung: Add machine driver for Trats2 ASoC: samsung: Document Trats2 audio subsystem bindings ARM: dts: set clock out parent to XUSBXTI ARM: dts: Add sound nodes for exynos4412-trats2 .../bindings/sound/samsung,trats2-wm1811.txt | 25 +++ arch/arm/boot/dts/exynos4.dtsi | 2 + arch/arm/boot/dts/exynos4412-trats2.dts | 37 ++++ sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/trats2_wm1811.c | 218 +++++++++++++++++++++ 6 files changed, 292 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,trats2-wm1811.txt create mode 100644 sound/soc/samsung/trats2_wm1811.c -- 2.0.0.390.gcb682f8

3 10

[alsa-devel] alsa-lib support for compress offload
by Qais Yousef 23 Jan '15

23 Jan '15

Hi, As far as I understand I need to use something similar to tinycompress to use alsa devices that supports compress offload. When I asked on gstreamer list [1] about alsasink support for compress offload they said yes. But I can see that gstalsasink.c [2] only uses the standard alsa-lib api which AFAICT doesn't support compress offload. Did I misread the code and alsa-lib actually works with compress offload? Gstreamer refers to the feature "passthrough" and associate it with SPDIF[3], are they taking advantage of some other alsa feature that looks like compress offload? Thanks, Qais [1] http://gstreamer-devel.966125.n4.nabble.com/ALSA-Compress-Offload-support-t… [2] http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/alsa/gstals… [3] http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/alsa/gstals…

6 16

[alsa-devel] [PATCH 1/1] ASoC: tas5086: Reinit register values in probe function
by pascal.huerst＠gmail.com 22 Jan '15

22 Jan '15

From: Pascal Huerst <pascal.huerst(a)gmail.com> If the machine driver has been un/reloaded, the register values of the tas5086 codec have to be reinitialized, in order to work properly. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 249ef5c..cd19171 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -851,10 +851,16 @@ static int tas5086_probe(struct snd_soc_codec *codec) } tas5086_reset(priv); + regcache_mark_dirty(priv->regmap); + ret = tas5086_init(codec->dev, priv); if (ret < 0) goto exit_disable_regulators; + ret = regcache_sync(priv->regmap); + if (ret < 0) + goto exit_disable_regulators; + /* set master volume to 0 dB */ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); if (ret < 0) -- 1.9.3

1 0

[alsa-devel] [PATCH - UCM 1/1] ucm: add binary configure file parse
by han.lu＠intel.com 22 Jan '15

22 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> with cset command, UCM set kcontrol parameters directly: cset "name='<KCONTROL_NAME>' 1<,2,3,...>" This patch enables UCM to set kcontrol with parameters from configure file: cset-bin-file "name='<KCONTROL_NAME>' <path/to/file>" where "cset-bin-file" is a newly added keyword alongside of "cset", to indicate cset with binary data in file. The binary data in file is parameter for audio DSPs, and it's just passed by UCM/ALSA as raw data. The data type of parameter elements must be byte, and the count must matches driver definition. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 37ae4c8..4c1e4b2 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -160,11 +160,62 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int binary_file_parse(snd_ctl_elem_value_t *dst, + snd_ctl_elem_info_t *info, + const char *filepath) +{ + int err, fd; + struct stat st; + size_t sz; + char *res; + snd_ctl_elem_type_t type; + unsigned int idx, count; + + type = snd_ctl_elem_info_get_type(info); + if (type != SND_CTL_ELEM_TYPE_BYTES) { + uc_error("only support byte type!"); + err = -EINVAL; + return err; + } + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + count = snd_ctl_elem_info_get_count(info); + if (sz != count || sz > sizeof(dst->value.bytes)) { + uc_error("invalid parameter size %d!", sz); + err = -EINVAL; + goto __fail; + } + res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + if (read(fd, res, sz) != sz) { + err = -errno; + goto __fail_read; + } + for (idx = 0; idx < sz; idx++) + snd_ctl_elem_value_set_byte(dst, idx, *(res + idx)); + __fail_read: + free(res); + __fail: + close(fd); + return err; +} + extern int __snd_ctl_ascii_elem_id_parse(snd_ctl_elem_id_t *dst, const char *str, const char **ret_ptr); -static int execute_cset(snd_ctl_t *ctl, const char *cset) +static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) { const char *pos; int err; @@ -194,7 +245,10 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset) err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); if (err < 0) goto __fail; err = snd_ctl_elem_write(ctl, value); @@ -239,6 +293,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail_nomem; break; case SEQUENCE_ELEMENT_TYPE_CSET: + case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: if (cdev == NULL) { const char *cdev1 = NULL, *cdev2 = NULL; err = get_value3(&cdev1, "PlaybackCTL", @@ -274,7 +329,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail; } } - err = execute_cset(ctl, s->data.cset); + err = execute_cset(ctl, s->data.cset, s->type); if (err < 0) { uc_error("unable to execute cset '%s'\n", s->data.cset); goto __fail; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index d7517f6..9e1cb41 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -306,6 +306,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-bin-file") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-bin-file requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 87f14a2..c1655c7 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -47,6 +47,7 @@ #define SEQUENCE_ELEMENT_TYPE_CSET 2 #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 +#define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 struct ucm_value { struct list_head list; -- 2.1.0

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