March 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 6/6] ESI W192M : Add sampling rate control of the ADC/DAC
by klem.dev＠gmail.com 14 Mar '15

14 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add sampling rate control for ADC/DAC for ESI W192M. Allow to switch between 48K/96K/192K sampling rate. All DAC need to be mute when changing samplerate. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 674e2a1..d3f0e24 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -30,12 +30,18 @@ #include <linux/init.h> #include <sound/core.h> #include <sound/tlv.h> +#include <linux/slab.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +struct wtm_spec { + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus @@ -69,15 +75,65 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned short int *change_mask) +{ + unsigned char new, old; + int id, idx, change; + + /*stac9460 1*/ + for (id = 0; id < 7; id++) { + if (*change_mask & (0x01 << id)) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } + + /*stac9460 2*/ + for (id = 0; id < 3; id++) { + if (*change_mask & (0x01 << (id + 7))) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_2_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char val; int idx, id; + mutex_lock(&spec->mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; id = 0; @@ -90,6 +146,8 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, else val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; + + mutex_unlock(&spec->mute_mutex); return 0; } @@ -397,6 +455,44 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned short int changed; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + changed = 0xFFFF; + stac9460_dac_mute_all(ice, 0, &changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + stac9460_2_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, &changed); + mutex_unlock(&spec->mute_mutex); +} + + /*Limits value in dB for fader*/ static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -496,21 +592,32 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + /*init mutex for dac mute conflict*/ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } -- 2.1.4

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[alsa-devel] [PATCH 4/6] ESI W192M : Add TLV support for control value in dB scale for ESI W192M
by klem.dev＠gmail.com 14 Mar '15

14 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add TLV support to control volume using dB scale for input and ouput on ESI W192M. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 59483b4..c7ffafa 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <sound/tlv.h> #include "ice1712.h" #include "envy24ht.h" @@ -380,17 +381,25 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/*Limits value in dB for fader*/ +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ static struct snd_kcontrol_new stac9640_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Switch", .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +428,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +448,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; -- 2.1.4

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[alsa-devel] [PATCH 6/6] ESI W192M : Add sampling rate control of the ADC/DAC
by klem.dev＠gmail.com 14 Mar '15

14 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add sampling rate control for ADC/DAC for ESI W192M. Allow to switch between 48K/96K/192K sampling rate. All DAC need to be mute when changing samplerate. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 497ff1a..4140200 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -30,12 +30,18 @@ #include <linux/init.h> #include <sound/core.h> #include <sound/tlv.h> +#include <linux/slab.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +struct wtm_spec { + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus @@ -69,15 +75,68 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned short int *change_mask) +{ + unsigned char new, old; + int id, idx, change; + + /*stac9460 1*/ + for (id = 0; id < 7; id++) { + if (*change_mask & (0x01 << id)) { + if(id == 0){ + idx = STAC946X_MASTER_VOLUME; + }else{ + idx = STAC946X_LF_VOLUME - 1 + id; + } + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change){ + stac9460_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + }else{ + *change_mask = *change_mask & ~(0x01 << id); + } + } + } + + /*stac9460 2*/ + for (id = 0; id < 3; id++) { + if (*change_mask & (0x01 << (idx + 7))) { + if(id == 0){ + idx = STAC946X_MASTER_VOLUME; + }else{ + idx = STAC946X_LF_VOLUME - 1 + id; + } + + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change){ + stac9460_2_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + }else{ + *change_mask = *change_mask & ~(0x01 << id); + } + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char val; int idx, id; + mutex_lock(&spec->mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; id = 0; @@ -90,6 +149,8 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, else val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; + + mutex_unlock(&spec->mute_mutex); return 0; } @@ -397,6 +458,44 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned short int changed; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + changed = 0xFFFF; + stac9460_dac_mute_all(ice, 0, &changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + stac9460_2_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, &changed); + mutex_unlock(&spec->mute_mutex); +} + + /*Limits value in dB for fader*/ static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -496,21 +595,32 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + /*init mutex for dac mute conflict*/ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } -- 2.1.4

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[alsa-devel] [PATCH 3/6] ESI W192M : Enable midi i/o port for envy24 chip
by klem.dev＠gmail.com 14 Mar '15

14 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Enable midi i/o port of envy24 chip as their are available on ESI W192M soundcard. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6e1026e..59483b4 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,7 +485,8 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_SYSCONF] = 0x67, /*SYSCONF: clock 192KHz, mpu401, + 4ADC, 8DAC */ [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ -- 2.1.4

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[alsa-devel] [PATCH 2/6] ESI W192M : Update eeprom structure to C99 standard
by klem.dev＠gmail.com 14 Mar '15

14 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Update eeprom structure to C99 standard to be compliant with change in alsa. It's just a notation change, no configuration change. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index f65ac19..6e1026e 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,19 +485,19 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ + [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, }; -- 2.1.4

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[alsa-devel] [PATCH 1/6] ESI W192M : Correct copy/paste from prodigy driver
by klem.dev＠gmail.com 14 Mar '15

14 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Correct copy/paste name from prodigy driver, no behaviour change, only name. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..f65ac19 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -463,7 +463,7 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, (unsigned short)-1 }; @@ -475,7 +475,7 @@ static int wtm_init(struct snd_ice1712 *ice) ice->force_rdma1 = 1; /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); -- 2.1.4

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[alsa-devel] [PATCH RFC (do not merge)] ASoC: davinci-mcasp: Set rule constraint if implicit bclk divider is used
by Jyri Sarha 13 Mar '15

13 Mar '15

Set a rule constraint to allow only sample-rate and sample-format combinations that can be played/captured with reasonable accuracy. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- In theory this patch does exactly what it is supposed to. It only allows a sample-rate and sample-format combination if the rate can be produced with reasonable accuracy. Unfortunately the alsa-lib and alsa-tools are not able use this information too well. If the requested sample-rate and sample-format is not available the aplay/arecord fails, even if plughw is selected, with: pcm_params.c:170: snd1_pcm_hw_param_get_min: Assertion `!snd_interval_empty(i)' failed. I think we are better of without this patch until the behavior of ALSA userspace improved. sound/soc/davinci/davinci-mcasp.c | 94 ++++++++++++++++++++++++++++++++++----- 1 file changed, 84 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155..4ff5820 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -855,6 +855,29 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, return 0; } +static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, + unsigned int bclk_freq, + int *error_ppm) +{ + int div = mcasp->sysclk_freq / bclk_freq; + int rem = mcasp->sysclk_freq % bclk_freq; + + if (rem != 0) { + if (div == 0 || + ((mcasp->sysclk_freq / div) - bclk_freq) > + (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + div++; + rem = rem - bclk_freq; + } + } + if (error_ppm) + *error_ppm = + (div*1000000 + 1000*((rem*1000)/(int)bclk_freq))/div + - 1000000; + + return div; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -872,16 +895,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, * the machine driver, we need to calculate the ratio. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - unsigned int bclk_freq = snd_soc_params_to_bclk(params); - unsigned int div = mcasp->sysclk_freq / bclk_freq; - if (mcasp->sysclk_freq % bclk_freq != 0) { - if (((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) - div++; - dev_warn(mcasp->dev, - "Inaccurate BCLK: %u Hz / %u != %u Hz\n", - mcasp->sysclk_freq, div, bclk_freq); - } + int ppm; + uint bclk_freq = snd_soc_params_to_bclk(params); + int div = davinci_mcasp_calc_clk_div(mcasp, bclk_freq, &ppm); + + if (ppm) + dev_info(mcasp->dev, "BCLK rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -973,6 +994,46 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static const unsigned int davinci_mcasp_dai_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, +}; + +#define DAVINCI_MAX_RATE_ERROR_PPM 1000 + +static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp *mcasp = rule->private; + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int frame_size; + int i, count = 0; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + uint bclk_freq = frame_size*davinci_mcasp_dai_rates[i]; + int ppm; + + davinci_mcasp_calc_clk_div(mcasp, bclk_freq, &ppm); + + dev_dbg(mcasp->dev, + "%u bit frames @ %u Hz => %d PPM error\n", + frame_size, davinci_mcasp_dai_rates[i], ppm); + + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + + } + dev_dbg(mcasp->dev, "%d frequencies for %u fsize\n", + count, frame_size); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { @@ -1012,6 +1073,19 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, max_channels); + + /* + * If we rely on implicit BCLK divider setting we should + * set constraints based on what we can provide. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) + return snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + mcasp, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + return 0; } -- 1.9.1

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[alsa-devel] Adding new (old) SNDRV_SEQ_PORT_TYPE_MIDI_* defines
by Adam Goode 13 Mar '15

13 Mar '15

Hi, I found the /* TODO: read port flags from descriptors */ in sound/usb/midi.c and was thinking about what to do to implement this. The first thing to do would be to ensure that all the flags from the element descriptors in the USB MIDI standard are available for returning from snd_seq_port_info_get_type. USB MIDI (see page 23): http://www.usb.org/developers/docs/devclass_docs/midi10.pdf I propose adding these flags to ALSA: #define SNDRV_SEQ_PORT_TYPE_MIDI_CLOCK (1<<7) /* MIDI CLOCK compatible device */ #define SNDRV_SEQ_PORT_TYPE_MIDI_TIME_CODE (1<<8) /* MTC compatible device */ #define SNDRV_SEQ_PORT_TYPE_MIDI_MACHINE_CONTROL (1<<9) /* MMC compatible device */ #define SNDRV_SEQ_PORT_TYPE_MIDI_EFX (1<<21) /* Audio effects processor device */ #define SNDRV_SEQ_PORT_TYPE_MIDI_PATCH_BAY (1<<22) /* MIDI patcher or router device */ #define SNDRV_SEQ_PORT_TYPE_MIDI_DLS1 (1<<23) /* DownLoadable Sounds Standard Level 1 compatible device */ #define SNDRV_SEQ_PORT_TYPE_MIDI_DLS2 (1<<24) /* DownLoadable Sounds Standard Level 2 compatible device */ Thoughts? I can send a couple patches for the kernel and alsa-lib. I don't think this is a breaking change. Thanks, Adam

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[alsa-devel] struct timespec as part of kernel/userspace ABI
by Tim Cussins 13 Mar '15

13 Mar '15

Hi all, Like the other ALSA ioctls, the proposed start_at ioctl currently uses struct timespec. A reviewer (of the start_at patchset) has mentioned that communicating timespecs across the kernel/userspace boundary requires special 32bit/64bit compatibility handling (pcm_compat.c). I didn't appreciate this :( Should I follow the same pcm_compat.c pattern for dealing with timespecs for the proposed start_at ioctl? Cheers, Tim

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[alsa-devel] [PATCH] ASoC: rt5645: Remove adc stereo2 filter
by Bard Liao 13 Mar '15

13 Mar '15

Remove adc stereo2 filter since it is not in rt5645/rt5650 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- "ADC STO2 ASRC" is not defined in the driver already. So it is not seen in this patch. --- sound/soc/codecs/rt5645.c | 3 --- sound/soc/codecs/rt5645.h | 2 -- 2 files changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index af468ae..e1433f9 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1536,8 +1536,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("adc stereo1 filter", 1, RT5645_PWR_DIG2, RT5645_PWR_ADC_S1F_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY_S("adc stereo2 filter", 1, RT5645_PWR_DIG2, - RT5645_PWR_ADC_S2F_BIT, 0, NULL, 0), SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, rt5645_sto1_adc_l_mix, ARRAY_SIZE(rt5645_sto1_adc_l_mix), NULL, 0), @@ -1727,7 +1725,6 @@ static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { 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 }, { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc }, { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc }, { "dac mono left filter", NULL, "DAC MONO L ASRC", is_using_asrc }, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index dbfd98c..db78e94 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -804,8 +804,6 @@ #define RT5645_PWR_DAC_MF_L_BIT 10 #define RT5645_PWR_DAC_MF_R (0x1 << 9) #define RT5645_PWR_DAC_MF_R_BIT 9 -#define RT5645_PWR_ADC_S2F (0x1 << 8) -#define RT5645_PWR_ADC_S2F_BIT 8 #define RT5645_PWR_PDM1 (0x1 << 7) #define RT5645_PWR_PDM1_BIT 7 #define RT5645_PWR_PDM2 (0x1 << 6) -- 1.8.1.1.439.g50a6b54

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