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

Re: [alsa-devel] [PATCH] hda-beep: Update authors dead email address
by Takashi Iwai 13 Jun '15

13 Jun '15

At Fri, 12 Jun 2015 21:47:46 -0700, Matt Ranostay wrote: > > Signed-off-by: Matt Ranostay <mranostay(a)gmail.com> Applied, thanks. Takashi > --- > sound/pci/hda/hda_beep.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c > index 3364dc0..c397e7d 100644 > --- a/sound/pci/hda/hda_beep.c > +++ b/sound/pci/hda/hda_beep.c > @@ -1,7 +1,7 @@ > /* > * Digital Beep Input Interface for HD-audio codec > * > - * Author: Matthew Ranostay <mranostay(a)embeddedalley.com> > + * Author: Matt Ranostay <mranostay(a)gmail.com> > * Copyright (c) 2008 Embedded Alley Solutions Inc > * > * This driver is free software; you can redistribute it and/or modify > -- > 1.9.1 >

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[alsa-devel] [PATCH] hda-beep: Update authors dead email address
by Matt Ranostay 13 Jun '15

13 Jun '15

Signed-off-by: Matt Ranostay <mranostay(a)gmail.com> --- sound/pci/hda/hda_beep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 3364dc0..c397e7d 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -1,7 +1,7 @@ /* * Digital Beep Input Interface for HD-audio codec * - * Author: Matthew Ranostay <mranostay(a)embeddedalley.com> + * Author: Matt Ranostay <mranostay(a)gmail.com> * Copyright (c) 2008 Embedded Alley Solutions Inc * * This driver is free software; you can redistribute it and/or modify -- 1.9.1

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[alsa-devel] [PATCH 1/1 linux-next] ALSA: gus: use swap() in snd_ics_put_double()
by Fabian Frederick 12 Jun '15

12 Jun '15

Use kernel.h macro definition. Thanks to Julia Lawall for Coccinelle scripting support. Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/isa/gus/gus_mixer.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/isa/gus/gus_mixer.c b/sound/isa/gus/gus_mixer.c index 0dd4341..3b5d9a7 100644 --- a/sound/isa/gus/gus_mixer.c +++ b/sound/isa/gus/gus_mixer.c @@ -109,7 +109,7 @@ static int snd_ics_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem unsigned long flags; int addr = kcontrol->private_value & 0xff; int change; - unsigned char val1, val2, oval1, oval2, tmp; + unsigned char val1, val2, oval1, oval2; val1 = ucontrol->value.integer.value[0] & 127; val2 = ucontrol->value.integer.value[1] & 127; @@ -120,11 +120,8 @@ static int snd_ics_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem gus->gf1.ics_regs[addr][0] = val1; gus->gf1.ics_regs[addr][1] = val2; if (gus->ics_flag && gus->ics_flipped && - (addr == SNDRV_ICS_GF1_DEV || addr == SNDRV_ICS_MASTER_DEV)) { - tmp = val1; - val1 = val2; - val2 = tmp; - } + (addr == SNDRV_ICS_GF1_DEV || addr == SNDRV_ICS_MASTER_DEV)) + swap(val1, val2); addr <<= 3; outb(addr | 0, GUSP(gus, MIXCNTRLPORT)); outb(1, GUSP(gus, MIXDATAPORT)); -- 2.4.2

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[alsa-devel] [PATCH] ALSA: aoa: convert bus code to use dev_groups
by Quentin Lambert 12 Jun '15

12 Jun '15

You should use dev_groups instead of the dev_attrs field of struct bus_type. This converts the MDIO bus code to use the correct field. These modifications were made using Coccinelle. Signed-off-by: Quentin Lambert <lambert.quentin(a)gmail.com> --- sound/aoa/soundbus/core.c | 3 ++- sound/aoa/soundbus/soundbus.h | 2 +- sound/aoa/soundbus/sysfs.c | 13 ++++++++----- 3 files changed, 11 insertions(+), 7 deletions(-) --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -150,6 +150,7 @@ static int soundbus_device_resume(struct #endif /* CONFIG_PM */ +ATTRIBUTE_GROUPS(soundbus_dev); static struct bus_type soundbus_bus_type = { .name = "aoa-soundbus", .probe = soundbus_probe, @@ -160,7 +161,7 @@ static struct bus_type soundbus_bus_type .suspend = soundbus_device_suspend, .resume = soundbus_device_resume, #endif - .dev_attrs = soundbus_dev_attrs, + .dev_groups = soundbus_dev_groups, }; int soundbus_add_one(struct soundbus_dev *dev) --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -199,6 +199,6 @@ struct soundbus_driver { extern int soundbus_register_driver(struct soundbus_driver *drv); extern void soundbus_unregister_driver(struct soundbus_driver *drv); -extern struct device_attribute soundbus_dev_attrs[]; +extern struct attribute *soundbus_dev_attrs[]; #endif /* __SOUNDBUS_H */ --- a/sound/aoa/soundbus/sysfs.c +++ b/sound/aoa/soundbus/sysfs.c @@ -30,13 +30,16 @@ static ssize_t modalias_show(struct devi return length; } +static DEVICE_ATTR_RO(modalias); soundbus_config_of_attr (name, "%s\n"); +static DEVICE_ATTR_RO(name); soundbus_config_of_attr (type, "%s\n"); +static DEVICE_ATTR_RO(type); -struct device_attribute soundbus_dev_attrs[] = { - __ATTR_RO(name), - __ATTR_RO(type), - __ATTR_RO(modalias), - __ATTR_NULL +struct attribute *soundbus_dev_attrs[] = { + &dev_attr_name.attr, + &dev_attr_type.attr, + &dev_attr_modalias.attr, + NULL, };

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[alsa-devel] [PATCH 1/1 linux-next] sound: oss/sb_audio: use swap() in sb_audio_close()
by Fabian Frederick 12 Jun '15

12 Jun '15

Use kernel.h macro definition. Thanks to Julia Lawall for Coccinelle scripting support. Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/oss/sb_audio.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index 048439a..dc91072 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -102,12 +102,8 @@ void sb_audio_close(int dev) if(devc->duplex && !devc->fullduplex && (devc->opened & OPEN_READ) && (devc->opened & OPEN_WRITE)) - { - struct dma_buffparms *dmap_temp; - dmap_temp = audio_devs[dev]->dmap_out; - audio_devs[dev]->dmap_out = audio_devs[dev]->dmap_in; - audio_devs[dev]->dmap_in = dmap_temp; - } + swap(audio_devs[dev]->dmap_out, audio_devs[dev]->dmap_in); + audio_devs[dev]->dmap_out->dma = devc->dma8; audio_devs[dev]->dmap_in->dma = ( devc->duplex ) ? devc->dma16 : devc->dma8; -- 2.4.2

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[alsa-devel] [PATCH] ASoC: wm8523: Set bclk ratio for master mode
by Julian Scheel 12 Jun '15

12 Jun '15

When running in master mode the bclk divider must be configured to generate a sane bitclock. Pick the smallest fs multiplicator, which can hold all transmitted bits. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/soc/codecs/wm8523.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index ec1f574..9c60058 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -113,6 +113,15 @@ static struct { { 7, 1152 }, }; +static struct { + int value; + int ratio; +} bclk_ratios[WM8523_NUM_RATES] = { + { 2, 32 }, + { 3, 64 }, + { 4, 128 }, +}; + static int wm8523_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -162,6 +171,23 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, aifctrl2 &= ~WM8523_SR_MASK; aifctrl2 |= lrclk_ratios[i].value; + if (aifctrl1 & WM8523_AIF_MSTR) { + /* Find a fs->bclk ratio */ + for (i = 0; i < ARRAY_SIZE(bclk_ratios); i++) + if (params_width(params) * 2 <= bclk_ratios[i].ratio) + break; + + if (i == ARRAY_SIZE(bclk_ratios)) { + dev_err(codec->dev, + "No matching BCLK/fs ratio for word length %d\n", + params_width(params)); + return -EINVAL; + } + + aifctrl2 &= ~WM8523_BCLKDIV_MASK; + aifctrl2 |= bclk_ratios[i].value << WM8523_BCLKDIV_SHIFT; + } + aifctrl1 &= ~WM8523_WL_MASK; switch (params_width(params)) { case 16: -- 2.4.0

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[alsa-devel] [PATCH - seq 1/1] ALSA seq: expose the card number of ALSA seq clients
by Adam Goode 12 Jun '15

12 Jun '15

Expose the card number of seq clients. This allows interested userspace programs to discover the hardware device that backs a particular seq client. Before this change, the only way to get information about the hardware for a client was by using brittle heuristics. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 5a5fa49..c2a659be 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -25,7 +25,7 @@ #include <sound/asound.h> /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -357,7 +357,8 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card_number; /* RO: card number, or -1 if no card. Added in protocol version 1.0.2 */ + char reserved[60]; /* for future use */ }; diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index edbdab8..fcae784 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -216,7 +216,7 @@ int __init client_init_data(void) } -static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) +static struct snd_seq_client *seq_create_client1(int client_index, int poolsize, int card_number) { unsigned long flags; int c; @@ -232,6 +232,7 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) return NULL; } client->type = NO_CLIENT; + client->card_number = card_number; snd_use_lock_init(&client->use_lock); rwlock_init(&client->ports_lock); mutex_init(&client->ports_mutex); @@ -327,7 +328,7 @@ static int snd_seq_open(struct inode *inode, struct file *file) if (mutex_lock_interruptible(&register_mutex)) return -ERESTARTSYS; - client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS); + client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS, -1); if (client == NULL) { mutex_unlock(&register_mutex); return -ENOMEM; /* failure code */ @@ -1194,6 +1195,7 @@ static void get_client_info(struct snd_seq_client *cptr, info->event_lost = cptr->event_lost; memcpy(info->event_filter, cptr->event_filter, 32); info->num_ports = cptr->num_ports; + info->card_number = cptr->card_number; memset(info->reserved, 0, sizeof(info->reserved)); } @@ -2239,6 +2241,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, { struct snd_seq_client *client; va_list args; + int card_number = -1; if (snd_BUG_ON(in_interrupt())) return -EBUSY; @@ -2252,6 +2255,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, return -ERESTARTSYS; if (card) { + card_number = card->number; client_index += SNDRV_SEQ_GLOBAL_CLIENTS + card->number * SNDRV_SEQ_CLIENTS_PER_CARD; if (client_index >= SNDRV_SEQ_DYNAMIC_CLIENTS_BEGIN) @@ -2259,7 +2263,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, } /* empty write queue as default */ - client = seq_create_client1(client_index, 0); + client = seq_create_client1(client_index, 0, card_number); if (client == NULL) { mutex_unlock(&register_mutex); return -EBUSY; /* failure code */ @@ -2533,9 +2537,13 @@ void snd_seq_info_clients_read(struct snd_info_entry *entry, continue; } - snd_iprintf(buffer, "Client %3d : \"%s\" [%s]\n", + snd_iprintf(buffer, "Client %3d : \"%s\" [%s]", c, client->name, client->type == USER_CLIENT ? "User" : "Kernel"); + if (client->card_number != -1) { + snd_iprintf(buffer, ", card %d", client->card_number); + } + snd_iprintf(buffer, "\n"); snd_seq_info_dump_ports(buffer, client); if (snd_seq_write_pool_allocated(client)) { snd_iprintf(buffer, " Output pool :\n"); diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index 20f0a72..627be9d 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -50,6 +50,7 @@ struct snd_seq_client { accept_output: 1; char name[64]; /* client name */ int number; /* client number */ + int card_number; /* card number, or -1 if no card */ unsigned int filter; /* filter flags */ DECLARE_BITMAP(event_filter, 256); snd_use_lock_t use_lock; -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] [PATCH 1/2] ALSA: hda - Continue probing even if i915 binding fails
by Takashi Iwai 12 Jun '15

12 Jun '15

Currently snd-hda-intel driver aborts the probing of Intel HD-audio controller with i915 power well management when binding with i915 driver via hda_i915_init() fails. This is no big problem for Haswell and Broadwell where the HD-audio controllers are dedicated to HDMI/DP, thus i915 link is mandatory. However, Skylake, Baytrail and Braswell have only one controller and both HDMI/DP and analog codecs share the same bus. Thus, even if HDMI/DP isn't usable, we should keep the controller working for other codecs. For fixing this, this patch simply allows continuing the probing even if hda_i915_init() call fails. This may leave stale sound components for HDMI/DP devices that are unbound with graphics. We could abort the probing selectively, but from the code simplicity POV, it's better to continue in all cases. Reported-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is for 4.1. Appying to 4.2 may result in trivial conflicts. sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fea198c58196..8a0af6770e1d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1855,7 +1855,7 @@ static int azx_probe_continue(struct azx *chip) #ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(hda); if (err < 0) - goto out_free; + goto skip_i915; err = hda_display_power(hda, true); if (err < 0) { dev_err(chip->card->dev, @@ -1865,6 +1865,7 @@ static int azx_probe_continue(struct azx *chip) #endif } + skip_i915: err = azx_first_init(chip); if (err < 0) goto out_free; -- 2.4.3

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[alsa-devel] [PATCH 4/4] ASoC: wm8960: the enum of "DAC Polarity" should be wm8960_enum[1]
by Zidan Wang 12 Jun '15

12 Jun '15

the enum of "DAC Polarity" should be wm8960_enum[1]. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 62246f1..d482c0f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -254,7 +254,7 @@ SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0), SOC_ENUM("ADC Polarity", wm8960_enum[0]), SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0), -SOC_ENUM("DAC Polarity", wm8960_enum[2]), +SOC_ENUM("DAC Polarity", wm8960_enum[1]), SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, wm8960_get_deemph, wm8960_put_deemph), -- 1.9.1

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[alsa-devel] [PATCH 3/4] ASoC: wm8960: add kcontrol to select ADC data output
by Zidan Wang 12 Jun '15

12 Jun '15

add kcontrol to select ADC data output. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index c1be15c..62246f1 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -141,6 +141,12 @@ static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; +static const char *wm8960_adc_data_sel[] = { + "ldata = left ADC; rdata = right ADC", + "ldata = left ADC; rdata = left ADC", + "ldata = right ADC; rdata = right ADC", + "ldata = right ADC; rdata = left ADC", +}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -149,6 +155,7 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff), SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), + SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 4, wm8960_adc_data_sel), }; static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -279,6 +286,8 @@ SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume", WM8960_BYPASS2, 4, 7, 1, bypass_tlv), SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), + +SOC_ENUM("ADC Data Output Select", wm8960_enum[6]), }; static const struct snd_kcontrol_new wm8960_lin_boost[] = { -- 1.9.1

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