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

[alsa-devel] [PATCH -next] ASoC: ux500: Fix sparse non static symbol warning
by Wei Yongjun 08 Jan '14

08 Jan '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warning: sound/soc/ux500/ux500_msp_i2s.c:649:5: warning: symbol 'ux500_msp_i2s_of_init_msp' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/ux500/ux500_msp_i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 7f2a4ac..959d7b4 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -646,9 +646,9 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) } -int ux500_msp_i2s_of_init_msp(struct platform_device *pdev, - struct ux500_msp *msp, - struct msp_i2s_platform_data **platform_data) +static int ux500_msp_i2s_of_init_msp(struct platform_device *pdev, + struct ux500_msp *msp, + struct msp_i2s_platform_data **platform_data) { struct msp_i2s_platform_data *pdata;

5 5

[alsa-devel] [PATCH] Patched to allow fine-grained control of PLL and clock dividers
by Daniel Matuschek 08 Jan '14

08 Jan '14

Changes: 1. allow the driver to accept S32_LE PCM streams 2. fine-grained control of the PLL generation WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using the "pllid" argument of the "set_pll" function is is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. When pllid=0, the behaviour of the driver does not change. --- sound/soc/codecs/wm8804.c | 25 +++++++++++++++++-------- sound/soc/codecs/wm8804.h | 7 +++++++ 2 files changed, 24 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..f3f73b2 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -2,6 +2,7 @@ * wm8804.c -- WM8804 S/PDIF transceiver driver * * Copyright 2010-11 Wolfson Microelectronics plc + * patched by Daniel Matuschek <info(a)crazyaudio.com> to allow fine-grained + * control of PLL and dividers * * Author: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> * @@ -277,6 +278,7 @@ static int wm8804_hw_params(struct snd_pcm_substream *substream, blen = 0x1; break; case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: blen = 0x2; break; default: @@ -318,7 +320,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -332,9 +334,14 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, tmp = target * post_table[i].div; if (tmp >= 90000000 && tmp <= 100000000) { pll_div->freqmode = post_table[i].freqmode; - pll_div->mclkdiv = post_table[i].mclkdiv; - target *= post_table[i].div; - break; + if ((mclk_div == WM8804_MCLKDIV_DONTCARE) || + ((post_table[i].mclkdiv == 1) && + (mclk_div == WM8804_MCLKDIV_1)) || + ((post_table[i].mclkdiv == 0) && + (mclk_div == WM8804_MCLKDIV_0))) { + pll_div->mclkdiv = post_table[i].mclkdiv; + target *= post_table[i].div; + break; + } } } @@ -388,7 +395,7 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, int ret; struct pll_div pll_div; - ret = pll_factors(&pll_div, freq_out, freq_in); + ret = pll_factors(&pll_div, freq_out, freq_in, pll_id); if (ret) return ret; @@ -641,7 +648,8 @@ static const struct snd_soc_dai_ops wm8804_dai_ops = { }; #define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE) #define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ @@ -674,7 +682,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .suspend = wm8804_suspend, .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, - .idle_bias_off = true, + .idle_bias_off = false, .controls = wm8804_snd_controls, .num_controls = ARRAY_SIZE(wm8804_snd_controls), @@ -760,6 +768,7 @@ static int wm8804_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8804, &wm8804_dai, 1); + return ret; } diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..0365177 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -58,4 +58,11 @@ #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLKDIV_DONTCARE 0 +#define WM8804_MCLKDIV_0 1 +#define WM8804_MCLKDIV_1 2 +#define WM8804_PLL_MCLKDIV_DONTCARE WM8804_MCLKDIV_DONTCARE +#define WM8804_PLL_MCLKDIV_0 WM8804_MCLKDIV_0 +#define WM8804_PLL_MCLKDIV_1 WM8804_MCLKDIV_1 + #endif /* _WM8804_H */ -- 1.7.9.5

3 2

[alsa-devel] [PATCH 1/2] ASoC: utils: Add internal call to determine if DAI is dummy.
by Liam Girdwood 08 Jan '14

08 Jan '14

Provide a quick way to tell if a DAI is a dummy DAI or a regular DAI. This is for internal DAPM usage only and is used to determe whether to insert a DAI link connection into the DAPM graph. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-utils.c | 7 +++++++ 2 files changed, 9 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 243d3b6..71f27c4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -123,6 +123,8 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); +int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); + struct snd_soc_dai_ops { /* * DAI clocking configuration, all optional. diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 6ebdfd9..7f22ca3 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -119,6 +119,13 @@ static struct snd_soc_dai_driver dummy_dai = { }, }; +int snd_soc_dai_is_dummy(struct snd_soc_dai *dai) +{ + if (dai->driver == &dummy_dai) + return 1; + return 0; +} + static int snd_soc_dummy_probe(struct platform_device *pdev) { int ret; -- 1.8.3.2

3 4

[alsa-devel] [PATCH] alsa-utils: fix pcm_read() return value
by Olivier Langlois 08 Jan '14

08 Jan '14

Because of the way the pcm_read() functions are currently used, returning rcount or result is equivalent but I feel it is more accurate to return 'result'. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- aplay/aplay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index e0631c4..69e8bda 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2039,7 +2039,7 @@ static ssize_t pcm_read(u_char *data, size_t rcount) data += r * bits_per_frame / 8; } } - return rcount; + return result; } static ssize_t pcm_readv(u_char **data, unsigned int channels, size_t rcount) @@ -2084,7 +2084,7 @@ static ssize_t pcm_readv(u_char **data, unsigned int channels, size_t rcount) count -= r; } } - return rcount; + return result; } /* -- 1.8.5.2

2 1

[alsa-devel] [PATCH] ALSA: hda - Split Thinkpad ACPI-related code
by Takashi Iwai 08 Jan '14

08 Jan '14

Both patch_realtek.c and patch_conexant.c contain the fairy same code snippet for supporting Thinkpad ACPI LED controls. Split them into thinkpad_helper.c and include it from both places. Although this isn't the best approach from the code size POV, the probability for coexistence of both Realtek and Conexant codecs on a single machine is pretty low, thus it'll end up with less memory footprint than splitting to yet another module. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 100 ++-------------------------------------- sound/pci/hda/patch_realtek.c | 93 ++----------------------------------- sound/pci/hda/thinkpad_helper.c | 99 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 105 insertions(+), 187 deletions(-) create mode 100644 sound/pci/hda/thinkpad_helper.c diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6df4c35b48c5..4e0ec146553d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3240,102 +3240,8 @@ enum { CXT_FIXUP_THINKPAD_ACPI, }; -#if IS_ENABLED(CONFIG_THINKPAD_ACPI) - -#include <linux/thinkpad_acpi.h> -#include <linux/acpi.h> - -static int (*led_set_func)(int, bool); - -static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, - void **rv) -{ - bool *found = context; - *found = true; - return AE_OK; -} - -static bool is_thinkpad(struct hda_codec *codec) -{ - bool found = false; - if (codec->subsystem_id >> 16 != 0x17aa) - return false; - if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) - return true; - found = false; - return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; -} - -static void update_tpacpi_mute_led(void *private_data, int enabled) -{ - struct hda_codec *codec = private_data; - struct conexant_spec *spec = codec->spec; - - if (spec->dynamic_eapd) - cx_auto_vmaster_hook(private_data, enabled); - - if (led_set_func) - led_set_func(TPACPI_LED_MUTE, !enabled); -} - -static void update_tpacpi_micmute_led(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) -{ - if (!ucontrol || !led_set_func) - return; - if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { - /* TODO: How do I verify if it's a mono or stereo here? */ - bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; - led_set_func(TPACPI_LED_MICMUTE, !val); - } -} - -static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct conexant_spec *spec = codec->spec; - - bool removefunc = false; - - if (action == HDA_FIXUP_ACT_PROBE) { - if (!is_thinkpad(codec)) - return; - if (!led_set_func) - led_set_func = symbol_request(tpacpi_led_set); - if (!led_set_func) { - snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); - return; - } - - removefunc = true; - if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { - spec->gen.vmaster_mute.hook = update_tpacpi_mute_led; - removefunc = false; - } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->gen.num_adc_nids > 1) - snd_printdd("Skipping micmute LED control due to several ADCs"); - else { - spec->gen.cap_sync_hook = update_tpacpi_micmute_led; - removefunc = false; - } - } - } - - if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { - symbol_put(tpacpi_led_set); - led_set_func = NULL; - } -} - -#else - -static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ -} - -#endif +/* for hda_fixup_thinkpad_acpi() */ +#include "thinkpad_helper.c" static void cxt_fixup_stereo_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -3492,7 +3398,7 @@ static const struct hda_fixup cxt_fixups[] = { }, [CXT_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, - .v.func = cxt_fixup_thinkpad_acpi, + .v.func = hda_fixup_thinkpad_acpi, }, }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 74b496b0d179..d35cf3dc6711 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3781,95 +3781,8 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, } } -#if IS_ENABLED(CONFIG_THINKPAD_ACPI) - -#include <linux/thinkpad_acpi.h> -#include <linux/acpi.h> - -static int (*led_set_func)(int, bool); - -static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, - void **rv) -{ - bool *found = context; - *found = true; - return AE_OK; -} - -static bool is_thinkpad(struct hda_codec *codec) -{ - bool found = false; - if (codec->subsystem_id >> 16 != 0x17aa) - return false; - if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) - return true; - found = false; - return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; -} - -static void update_tpacpi_mute_led(void *private_data, int enabled) -{ - if (led_set_func) - led_set_func(TPACPI_LED_MUTE, !enabled); -} - -static void update_tpacpi_micmute_led(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) -{ - if (!ucontrol || !led_set_func) - return; - if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { - /* TODO: How do I verify if it's a mono or stereo here? */ - bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; - led_set_func(TPACPI_LED_MICMUTE, !val); - } -} - -static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct alc_spec *spec = codec->spec; - bool removefunc = false; - - if (action == HDA_FIXUP_ACT_PROBE) { - if (!is_thinkpad(codec)) - return; - if (!led_set_func) - led_set_func = symbol_request(tpacpi_led_set); - if (!led_set_func) { - snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); - return; - } - - removefunc = true; - if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { - spec->gen.vmaster_mute.hook = update_tpacpi_mute_led; - removefunc = false; - } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->gen.num_adc_nids > 1) - snd_printdd("Skipping micmute LED control due to several ADCs"); - else { - spec->gen.cap_sync_hook = update_tpacpi_micmute_led; - removefunc = false; - } - } - } - - if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { - symbol_put(tpacpi_led_set); - led_set_func = NULL; - } -} - -#else - -static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ -} - -#endif +/* for hda_fixup_thinkpad_acpi() */ +#include "thinkpad_helper.c" enum { ALC269_FIXUP_SONY_VAIO, @@ -4273,7 +4186,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC269_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, - .v.func = alc_fixup_thinkpad_acpi, + .v.func = hda_fixup_thinkpad_acpi, }, [ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c new file mode 100644 index 000000000000..8492b8a3ce47 --- /dev/null +++ b/sound/pci/hda/thinkpad_helper.c @@ -0,0 +1,99 @@ +/* Helper functions for Thinkpad LED control; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) + +#include <linux/acpi.h> +#include <linux/thinkpad_acpi.h> + +static int (*led_set_func)(int, bool); + +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} + +static void update_tpacpi_mute_led(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct hda_gen_spec *spec = codec->spec; + + if (spec->vmaster_mute.hook) + spec->vmaster_mute.hook(private_data, enabled); + + if (led_set_func) + led_set_func(TPACPI_LED_MUTE, !enabled); +} + +static void update_tpacpi_micmute_led(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + if (!ucontrol || !led_set_func) + return; + if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { + /* TODO: How do I verify if it's a mono or stereo here? */ + bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; + led_set_func(TPACPI_LED_MICMUTE, !val); + } +} + +static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct hda_gen_spec *spec = codec->spec; + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; + if (!led_set_func) + led_set_func = symbol_request(tpacpi_led_set); + if (!led_set_func) { + snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); + return; + } + + removefunc = true; + if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + spec->vmaster_mute.hook = update_tpacpi_mute_led; + removefunc = false; + } + if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { + if (spec->num_adc_nids > 1) + snd_printdd("Skipping micmute LED control due to several ADCs"); + else { + spec->cap_sync_hook = update_tpacpi_micmute_led; + removefunc = false; + } + } + } + + if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(tpacpi_led_set); + led_set_func = NULL; + } +} + +#else /* CONFIG_THINKPAD_ACPI */ + +static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif /* CONFIG_THINKPAD_ACPI */ -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant
by Lars-Peter Clausen 08 Jan '14

08 Jan '14

The driver defines ADAU1701_SEROCTL_WORD_LEN_16 as 0x10 while it should be b10, so 0x2. This patch fixes it. Reported-by: Magnus Reftel <magnus.reftel(a)lockless.no> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/adau1701.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index ebff112..adee866 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -71,7 +71,7 @@ #define ADAU1701_SEROCTL_WORD_LEN_24 0x0000 #define ADAU1701_SEROCTL_WORD_LEN_20 0x0001 -#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010 +#define ADAU1701_SEROCTL_WORD_LEN_16 0x0002 #define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003 #define ADAU1701_AUXNPOW_VBPD 0x40 -- 1.8.0

1 0

[alsa-devel] [PATCH] ALSA: compress: add num_sample_rates in snd_codec_desc
by Vinod Koul 08 Jan '14

08 Jan '14

this gives ability to convey the valid values of supported rates in sample_rates array Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/compress_params.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index b62b24b..165e705 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -326,6 +326,7 @@ union snd_codec_options { /** struct snd_codec_desc - description of codec capabilities * @max_ch: Maximum number of audio channels * @sample_rates: Sampling rates in Hz, use values like 48000 for this + * @num_sample_rates: Number of valid values in sample_rates array * @bit_rate: Indexed array containing supported bit rates * @num_bitrates: Number of valid values in bit_rate array * @rate_control: value is specified by SND_RATECONTROLMODE defines. @@ -348,6 +349,7 @@ union snd_codec_options { struct snd_codec_desc { __u32 max_ch; __u32 sample_rates[MAX_NUM_SAMPLE_RATES]; + __u32 num_sample_rates; __u32 bit_rate[MAX_NUM_BITRATES]; __u32 num_bitrates; __u32 rate_control; -- 1.7.0.4

3 9

[alsa-devel] [PATCH v2] ASoC: soc-core: add snd_soc_of_parse_master_clkdir() for dt
by Xiubo Li 08 Jan '14

08 Jan '14

This patch adds snd_soc_of_parse_master_clkdir() and supports below style on dt: master-clkdir-out; If this property is absent 0 will be returned, 1 otherwise. 0 : SND_SOC_CLOCK_IN 1 : SND_SOC_CLOCK_OUT Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 11 +++++++++++ 2 files changed, 12 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5a049d9..690f4f6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1170,6 +1170,7 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); +unsigned int snd_soc_of_parse_master_clkdir(struct device_node *np); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix); int snd_soc_of_get_dai_name(struct device_node *of_node, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index be88df5..f155207 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4472,6 +4472,17 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +unsigned int snd_soc_of_parse_master_clkdir(struct device_node *np) +{ + /* + * check "master-clkdir-out" + * false means SND_SOC_CLOCK_IN + * true means SND_SOC_CLOCK_OUT + */ + return !!of_property_read_bool(np, "master-clkdir-out"); +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_master_clkdir); + unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix) { -- 1.8.4

3 5

[alsa-devel] exceprionally large value
by Johan Esvelt 07 Jan '14

07 Jan '14

Hi, My PC crashed and these are the last 3 lines in /var/log/messages: -------------------------8<---------------------- Jan 7 19:11:48 charles pulseaudio[2646]: alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 416448 bytes (2360 ms). Jan 7 19:11:48 charles pulseaudio[2646]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_emu10k1'. Please report this issue to the ALSA developers. Jan 7 19:11:48 charles pulseaudio[2646]: alJan 7 19:20:03 charles kernel: imklog 5.8.10, log source = /proc/kmsg started. -------------------------------8<----------------- I am running Centos 6.5: Linux version 2.6.32-358.18.1.el6.x86_64 (mockbuild(a)c6b10.bsys.dev.centos.org) (gcc version 4.4.7 20120313 (Red Hat 4.4.7-3) (GCC) ) #1 SMP Wed Aug 28 17:19:38 UTC 2013 What can I do to prevent this from happening again? Best regards, Johan Esvelt

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[alsa-devel] [PATCH] ASoC: Docs: Update the Overview document
by Liam Girdwood 07 Jan '14

07 Jan '14

Update the ASoC overview to bring it up to date with the current code base and include multi-component. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Documentation/sound/alsa/soc/overview.txt | 27 ++++++++++++++++++--------- 1 file changed, 18 insertions(+), 9 deletions(-) diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt index 138ac88..ff88f52 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -49,18 +49,23 @@ features :- * Machine specific controls: Allow machines to add controls to the sound card (e.g. volume control for speaker amplifier). -To achieve all this, ASoC basically splits an embedded audio system into 3 -components :- +To achieve all this, ASoC basically splits an embedded audio system into +multiple re-usable component drivers :- - * Codec driver: The codec driver is platform independent and contains audio - controls, audio interface capabilities, codec DAPM definition and codec IO - functions. + * Codec class drivers: The codec class driver is platform independent and + contains audio controls, audio interface capabilities, codec DAPM + definition and codec IO functions. This class extends to BT, FM and MODEM + ICs if required. Codec class drivers should be generic code that can run + on any architecture and machine. - * Platform driver: The platform driver contains the audio DMA engine and audio - interface drivers (e.g. I2S, AC97, PCM) for that platform. + * Platform class drivers: The platform class driver includes the audio DMA + engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM) + and any audio DSP drivers for that platform. - * Machine driver: The machine driver handles any machine specific controls and - audio events (e.g. turning on an amp at start of playback). + * Machine class driver: The machine driver class acts as the glue that + decribes and binds the other component drivers together to form an ALSA + "sound card device". It handles any machine specific controls and + machine level audio events (e.g. turning on an amp at start of playback). Documentation @@ -84,3 +89,7 @@ machine.txt: Machine driver internals. pop_clicks.txt: How to minimise audio artifacts. clocking.txt: ASoC clocking for best power performance. + +jack.txt: ASoC jack detection. + +DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples. -- 1.8.3.2

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