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

[alsa-devel] Get sound card available channels
by Dan Hordern 25 Sep '14

25 Sep '14

Hi, We have a need for our process to poll/monitor the sound cards. This process needs to provide a unique id for the sound card (e.g. some concat of name+hardware address) which we can get using snd_ctl_t handle. However, we also need to provide the number or rx/tx channels each sound card provides. Doing this using the pcm hw_params after snd_pcm_open won't work as the device may be in use. Is there a way for this process to get the number of rx/tx channels without needing to 'open' the device, e.g. using the snd_ctl_t handle? One observation I have made is that there is a proc file ( cat /proc/asound/card1/stream0) that seems to only be populated by the USB soundcards I have and not the Intel builtin. This has all the information I require but if it isn't a standard file we can't rely on it. Thanks

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Re: [alsa-devel] mackie onyx 400F works
by Takashi Sakamoto 24 Sep '14

24 Sep '14

Hi, (C.C.ed to alsa-devel and ffado-devel, fot information to the others) On 23 Sep 2014 00:42, José Razafindranaly wrote: > Hi, > > I own a mackie Onyx 400F that works very well with your driver (and which was not supported by ffado). > So I'm very happy with it now: thank you so much! > It works for playing and recording sound. > My computer runs linux Mint Petra, with a 3.10.0-20 lowlatency kernel > > Few monthes ago, when I first installed your driver I had some minor issues: > > - with the alsa-loop-jack bridge of "cadence" (weird sound after several minutes of video playback or gaming for example), but It may not be your driver's fault. It worked again after stopping and restarting jack. > - some sampling frequencies didn't work with the jack server. > > I recently reinstalled the driver, and these issues seem fixed. > I still have to do further recording tests with my new setup. > I'm ready to do more state of the art bug reporting if you want me to. Just keep in mind that I'm quite a rookie regarding command lines. > > > In conclusion, I'm really impressed by your work and I think you can write that the onyx 400F is reported to work with your driver. > > Kind regards. Thanks for your report. It's pretty nice to hear that your 400F works with my drivers. Here I have a question. Have you evet tried with FFADO to drive your 400F? If so and FFADO doesn't drive, I want to clear the reason. I would like you to get teminal dumps of "ffado-debug ffado-test Discover -v 8" and send it to me. I reccomend to redirect stdout to file by using ">", like: $ ffado-debug ffado-test Discover -v 8 > dump_file_name Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

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[alsa-devel] [PATCH] ASoC: samsung: Remove goni or aquila with the WM8994
by Paul Bolle 24 Sep '14

24 Sep '14

On Thu, 2014-09-04 at 18:02 +0200, Arnd Bergmann wrote: > On Thursday 04 September 2014, Paul Bolle wrote: > > References to MACH_GONI and MACH_AQUILA can still be seen in v3.17-rc3 > > and next-20140903. Note that as consequence SND_SOC_GONI_AQUILA_WM8994 > > can not be set anymore. > > > [...] > > I think it would be nice if you could submit a patch to remove the > drivers from ASoC, then we can see if anybody complains. Same thing for v3.17-rc5 and next-20140918. Let's see if we can remove the goni or aquila with wm8994 driver. Done on top of next-20140918. Untested. --------->8--------- From: Paul Bolle <pebolle(a)tiscali.nl> 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> --- sound/soc/samsung/Kconfig | 11 -- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/goni_wm8994.c | 304 ---------------------------------------- 3 files changed, 317 deletions(-) delete mode 100644 sound/soc/samsung/goni_wm8994.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 55a38697443d..9d9b149ad977 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 91505ddaaf95..c30293a67176 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 @@ -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-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(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 3b527dcfc0aa..000000000000 --- a/sound/soc/samsung/goni_wm8994.c +++ /dev/null @@ -1,304 +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; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 24000000; - int ret = 0; - - /* set the cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* 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 codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* 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", - .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", - .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

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[alsa-devel] [PATCH] ASoC: samsung-i2s: Maintain CDCLK settings across i2s_{shutdown/startup}
by Sylwester Nawrocki 24 Sep '14

24 Sep '14

Currently configuration of the CDCLK pad is being overwritten in the i2s_shutdown() callback in order to gate the SoC output clock. However if an ASoC machine driver doesn't restore that clock settings each time after opening the sound device this results in the CDCLK pin being permanently configured into input mode. I.e. the output clock will always stay disabled. Fix that by saving the CDCLKCON bit state in i2s_shutdown() and and restoring it in the i2s_startup() callback. Signed-off-by: Chen Zhen <zhen1.chen(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2ac76fa..3cb3e95 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -68,6 +68,8 @@ struct i2s_dai { #define DAI_OPENED (1 << 0) /* Dai is opened */ #define DAI_MANAGER (1 << 1) /* Dai is the manager */ unsigned mode; + /* CDCLK pin direction: 0 - input, 1 - output */ + unsigned int cdclk_out:1; /* Driver for this DAI */ struct snd_soc_dai_driver i2s_dai_drv; /* DMA parameters */ @@ -737,6 +739,9 @@ static int i2s_startup(struct snd_pcm_substream *substream, spin_unlock_irqrestore(&lock, flags); + if (!is_opened(other) && i2s->cdclk_out) + i2s_set_sysclk(dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); return 0; } @@ -752,9 +757,13 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, i2s->mode &= ~DAI_OPENED; i2s->mode &= ~DAI_MANAGER; - if (is_opened(other)) + if (is_opened(other)) { other->mode |= DAI_MANAGER; - + } else { + u32 mod = readl(i2s->addr + I2SMOD); + i2s->cdclk_out = !(mod & MOD_CDCLKCON); + other->cdclk_out = i2s->cdclk_out; + } /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; i2s->bfs = 0; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: soc-compress: fix double unlock of fe card mutex
by Qiao Zhou 24 Sep '14

24 Sep '14

Fix double unlock of fe card mutex introduced by patch 8f70e515a8bb "ASoC: soc-pcm: fix dpcm_path_get error handling" The first unlock is at line 106, and the unlock is at line 149. we should remove the first unlock. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-compress.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3092b58..cecfab3 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -102,13 +102,11 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].runtime = fe_substream->runtime; ret = dpcm_path_get(fe, stream, &list); - if (ret < 0) { - mutex_unlock(&fe->card->mutex); + if (ret < 0) goto fe_err; - } else if (ret == 0) { + else if (ret == 0) dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - } /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); -- 1.7.0.4

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[alsa-devel] [PATCH 1/2] ALSA: hda - Move the function "check_amp_caps" to hda_codec.c
by David Henningsson 24 Sep '14

24 Sep '14

The next patch will use it, so make it visible across modules. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 20 ++++++++++++++++++++ sound/pci/hda/hda_generic.c | 17 ----------------- sound/pci/hda/hda_local.h | 8 ++++++++ 3 files changed, 28 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0aa2e1e..15e0089 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2002,6 +2002,26 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) EXPORT_SYMBOL_GPL(query_amp_caps); /** + * snd_hda_check_amp_caps - query AMP capabilities + * @codec: the HD-audio codec + * @nid: the NID to query + * @dir: either #HDA_INPUT or #HDA_OUTPUT + * + * Check whether the widget has the given amp capability for the direction. + */ +bool snd_hda_check_amp_caps(struct hda_codec *codec, hda_nid_t nid, + int dir, unsigned int bits) +{ + if (!nid) + return false; + if (get_wcaps(codec, nid) & (1 << (dir + 1))) + if (query_amp_caps(codec, nid, dir) & bits) + return true; + return false; +} +EXPORT_SYMBOL_GPL(snd_hda_check_amp_caps); + +/** * snd_hda_override_amp_caps - Override the AMP capabilities * @codec: the CODEC to clean up * @nid: the NID to clean up diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 32a85f9..64220c0 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -519,18 +519,6 @@ static unsigned int amp_val_replace_channels(unsigned int val, unsigned int chs) return val; } -/* check whether the widget has the given amp capability for the direction */ -static bool check_amp_caps(struct hda_codec *codec, hda_nid_t nid, - int dir, unsigned int bits) -{ - if (!nid) - return false; - if (get_wcaps(codec, nid) & (1 << (dir + 1))) - if (query_amp_caps(codec, nid, dir) & bits) - return true; - return false; -} - static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, hda_nid_t nid2, int dir) { @@ -540,11 +528,6 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, query_amp_caps(codec, nid2, dir)); } -#define nid_has_mute(codec, nid, dir) \ - check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) -#define nid_has_volume(codec, nid, dir) \ - check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS) - /* look for a widget suitable for assigning a mute switch in the path */ static hda_nid_t look_for_out_mute_nid(struct hda_codec *codec, struct nid_path *path) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 8a018d4..7eb44e7 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -603,6 +603,14 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, unsigned int caps); +bool snd_hda_check_amp_caps(struct hda_codec *codec, hda_nid_t nid, + int dir, unsigned int bits); + +#define nid_has_mute(codec, nid, dir) \ + snd_hda_check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) +#define nid_has_volume(codec, nid, dir) \ + snd_hda_check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS) + /* flags for hda_nid_item */ #define HDA_NID_ITEM_AMP (1<<0) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5677: Add a configuration option for LDO2_POW pin
by Anatol Pomozov 24 Sep '14

24 Sep '14

Some boards have this pin statically tied and do not require any configuration, some other boards allow to enable chip using GPIO. Add an option that tells which GPIO is used to power the audio codec. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 41 ++++++++++++++++ sound/soc/codecs/rt5677.c | 54 ++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 1 + 3 files changed, 96 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5677.txt diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt new file mode 100644 index 0000000..98509fb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -0,0 +1,41 @@ +RT5677 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5677". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. + +Pins on the device (for linking into audio routes): + + * IN1P + * IN1N + * IN2P + * IN2N + * MICBIAS1 + * DMIC1 + * DMIC2 + * DMIC3 + * DMIC4 + * LOUT1 + * LOUT2 + * LOUT3 + +Example: + +rt5677 { + compatible = "realtek,rt5677"; + reg = <0x2c>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + realtek,pow-ldo2-gpio = + <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 9847473..83785db 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/of_gpio.h> #include <linux/regmap.h> #include <linux/i2c.h> #include <linux/platform_device.h> @@ -3319,6 +3320,8 @@ static int rt5677_remove(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); return 0; } @@ -3330,6 +3333,8 @@ static int rt5677_suspend(struct snd_soc_codec *codec) regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); return 0; } @@ -3338,6 +3343,10 @@ static int rt5677_resume(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + if (gpio_is_valid(rt5677->pow_ldo2)) { + gpio_set_value_cansleep(rt5677->pow_ldo2, 1); + msleep(10); + } regcache_cache_only(rt5677->regmap, false); regcache_sync(rt5677->regmap); @@ -3495,6 +3504,24 @@ static const struct i2c_device_id rt5677_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); +static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) +{ + rt5677->pow_ldo2 = of_get_named_gpio(np, + "realtek,pow-ldo2-gpio", 0); + + /* + * POW_LDO2 is optional (it may be statically tied on the board). + * -ENOENT means that the property doesn't exist, i.e. there is no + * GPIO, so is not an error. Any other error code means the property + * exists, but could not be parsed. + */ + if (!gpio_is_valid(rt5677->pow_ldo2) && + (rt5677->pow_ldo2 != -ENOENT)) + return rt5677->pow_ldo2; + + return 0; +} + static int rt5677_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3513,6 +3540,33 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (pdata) rt5677->pdata = *pdata; + if (i2c->dev.of_node) { + ret = rt5677_parse_dt(rt5677, i2c->dev.of_node); + if (ret) { + dev_err(&i2c->dev, "Failed to parse device tree: %d\n", + ret); + return ret; + } + } else { + rt5677->pow_ldo2 = -EINVAL; + } + + if (gpio_is_valid(rt5677->pow_ldo2)) { + ret = devm_gpio_request_one(&i2c->dev, rt5677->pow_ldo2, + GPIOF_OUT_INIT_HIGH, + "RT5677 POW_LDO2"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request POW_LDO2 %d: %d\n", + rt5677->pow_ldo2, ret); + return ret; + } + /* Wait a while until I2C bus becomes available. The datasheet + * does not specify the exact we should wait but startup + * sequence mentiones at least a few milliseconds. + */ + msleep(10); + } + rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap); if (IS_ERR(rt5677->regmap)) { ret = PTR_ERR(rt5677->regmap); diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index b61b72c..222a4137 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1550,6 +1550,7 @@ struct rt5677_priv { int pll_src; int pll_in; int pll_out; + int pow_ldo2; /* POW_LDO2 pin */ #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif -- 2.1.0.rc2.206.gedb03e5

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[alsa-devel] [PATCH] ASoC: rt286: Fix sync function
by Bard Liao 24 Sep '14

24 Sep '14

We try to write index registers into cache when we write an index register, but we change the reg value before updating the cache. As a result, the cache is never be updated. This patch will fix this issue and also correct some incorrect default value in the cache. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 7dfca82..8ad4b24 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = { { 0x04, 0xaf01 }, { 0x08, 0x000d }, { 0x09, 0xd810 }, - { 0x0a, 0x0060 }, + { 0x0a, 0x0120 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, @@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = { { 0x33, 0x0208 }, { 0x49, 0x0004 }, { 0x4f, 0x50e9 }, - { 0x50, 0x2c00 }, + { 0x50, 0x2000 }, { 0x63, 0x2902 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, @@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = { { 0x02170700, 0x00000000 }, { 0x02270100, 0x00000000 }, { 0x02370100, 0x00000000 }, - { 0x02040000, 0x00004002 }, { 0x01870700, 0x00000020 }, { 0x00830000, 0x000000c3 }, { 0x00930000, 0x000000c3 }, @@ -192,7 +191,6 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) /*handle index registers*/ if (reg <= 0xff) { rt286_hw_write(client, RT286_COEF_INDEX, reg); - reg = RT286_PROC_COEF; for (i = 0; i < INDEX_CACHE_SIZE; i++) { if (reg == rt286->index_cache[i].reg) { rt286->index_cache[i].def = value; @@ -200,6 +198,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) } } + reg = RT286_PROC_COEF; } data[0] = (reg >> 24) & 0xff; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/5] ASoC: max98090: Move interrupt request from codec probe to i2c probe
by Jarkko Nikula 24 Sep '14

24 Sep '14

Keep MAX98090 interrupt requested after i2c device probing as long as the driver is loaded. This fixes the issue where subsequent codec probe max98090_probe() call fails in interrupt request since interrupt wasn't freed over codec remove-reprobe cycle. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/codecs/max98090.c | 29 ++++++++++++++++------------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f1543653a699..fe77df6a76c2 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2159,12 +2159,16 @@ static void max98090_jack_work(struct work_struct *work) static irqreturn_t max98090_interrupt(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_priv *max98090 = data; + struct snd_soc_codec *codec = max98090->codec; int ret; unsigned int mask; unsigned int active; + /* Treat interrupt before codec is initialized as spurious */ + if (codec == NULL) + return IRQ_NONE; + dev_dbg(codec->dev, "***** max98090_interrupt *****\n"); ret = regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); @@ -2367,17 +2371,6 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_write(codec, M98090_REG_JACK_DETECT, M98090_JDETEN_MASK | M98090_JDEB_25MS); - /* Register for interrupts */ - dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - - ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, - max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, - "max98090_interrupt", codec); - if (ret < 0) { - dev_err(codec->dev, "request_irq failed: %d\n", - ret); - } - /* * Clear any old interrupts. * An old interrupt ocurring prior to installing the ISR @@ -2417,6 +2410,7 @@ static int max98090_remove(struct snd_soc_codec *codec) cancel_delayed_work_sync(&max98090->pll_det_enable_work); cancel_work_sync(&max98090->pll_det_disable_work); cancel_work_sync(&max98090->pll_work); + max98090->codec = NULL; return 0; } @@ -2478,6 +2472,15 @@ static int max98090_i2c_probe(struct i2c_client *i2c, goto err_enable; } + ret = devm_request_threaded_irq(&i2c->dev, max98090->irq, NULL, + max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "max98090_interrupt", max98090); + if (ret < 0) { + dev_err(&i2c->dev, "request_irq failed: %d\n", + ret); + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98090, max98090_dai, ARRAY_SIZE(max98090_dai)); -- 2.1.0

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Re: [alsa-devel] [PATCH] soc-core: Fix volsw_range funcs so SOC_DOUBLE_R_RANGE_TLV works.
by Mark Brown 24 Sep '14

24 Sep '14

On Fri, Sep 19, 2014 at 12:50:31PM +0100, Howard Mitchell wrote: > This fixes a bug when using the SOC_DOUBLE_R_RANGE_TLV macro in > the invert mode. In the non-invert case, e.g. Applied, thanks.

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