January 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC updates for 3.3
by Mark Brown 12 Jan '12

12 Jan '12

The following changes since commit d6b24507972341e5babdca8e25448325c9137232: Merge branch 'for-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc into topic/asoc (2012-01-11 15:30:53 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 The MFD patch is an incremental patch on top of the WM8994 changes being carried by ASoC this merge window. Axel Lin (2): ASoC: Fix build dependency for SND_ATMEL_SOC_SSC ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC Joachim Eastwood (1): ASoC: check for substream not channels_min in pcm engines Mark Brown (3): ASoC: Fix idma build after update for channel count check mfd: Put WM8994 into cache only mode when suspending ASoC: Dynamically allocate the rtd device for a non-empty release() drivers/mfd/wm8994-core.c | 2 + include/sound/soc.h | 6 ++-- sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel-pcm.c | 5 +-- sound/soc/blackfin/bf5xx-ac97-pcm.c | 5 +-- sound/soc/blackfin/bf5xx-i2s-pcm.c | 5 +-- sound/soc/blackfin/bf5xx-tdm-pcm.c | 5 +-- sound/soc/davinci/davinci-pcm.c | 5 +-- sound/soc/ep93xx/ep93xx-pcm.c | 5 +-- sound/soc/jz4740/jz4740-pcm.c | 5 +-- sound/soc/kirkwood/kirkwood-dma.c | 5 +-- sound/soc/mid-x86/sst_platform.c | 5 +-- sound/soc/omap/omap-pcm.c | 5 +-- sound/soc/samsung/dma.c | 5 +-- sound/soc/samsung/idma.c | 4 +- sound/soc/soc-core.c | 44 +++++++++++++++++++--------------- sound/soc/soc-dapm.c | 3 +- sound/soc/tegra/tegra_pcm.c | 5 +-- 18 files changed, 58 insertions(+), 63 deletions(-)

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[alsa-devel] [PATCH] ASoC: Use regmap update bits operation for drivers using regmap
by Mark Brown 12 Jan '12

12 Jan '12

If a driver is using regmap directly ensure that we're coherent with non-ASoC register updates by using the regmap API directly to do our read/modify/write cycles. This will bypass the ASoC cache but drivers using regmap directly should not be using the ASoC cache. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 25 +++++++++++++++---------- sound/soc/soc-dapm.c | 27 +++++++++++++++++---------- sound/soc/soc-io.c | 1 + 4 files changed, 34 insertions(+), 20 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0992dff..fde163b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -559,6 +559,7 @@ struct snd_soc_codec { unsigned int ac97_created:1; /* Codec has been created by SoC */ unsigned int sysfs_registered:1; /* codec has been sysfs registered */ unsigned int cache_init:1; /* codec cache has been initialized */ + unsigned int using_regmap:1; /* using regmap access */ u32 cache_only; /* Suppress writes to hardware */ u32 cache_sync; /* Cache needs to be synced to hardware */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3986520..07ab062e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1865,23 +1865,28 @@ EXPORT_SYMBOL_GPL(snd_soc_bulk_write_raw); int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, unsigned int mask, unsigned int value) { - int change; + bool change; unsigned int old, new; int ret; - ret = snd_soc_read(codec, reg); - if (ret < 0) - return ret; - - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = snd_soc_write(codec, reg, new); + if (codec->using_regmap) { + ret = regmap_update_bits_check(codec->control_data, reg, + mask, value, &change); + } else { + ret = snd_soc_read(codec, reg); if (ret < 0) return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) + ret = snd_soc_write(codec, reg, new); } + if (ret < 0) + return ret; + return change; } EXPORT_SYMBOL_GPL(snd_soc_update_bits); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3ad1f59..3b8eaa8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -197,21 +197,28 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { - int change; + bool change; unsigned int old, new; int ret; - ret = soc_widget_read(w, reg); - if (ret < 0) - return ret; - - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = soc_widget_write(w, reg, new); + if (w->codec && w->codec->using_regmap) { + ret = regmap_update_bits_check(w->codec->control_data, + reg, mask, value, &change); + if (ret != 0) + return ret; + } else { + ret = soc_widget_read(w, reg); if (ret < 0) return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) { + ret = soc_widget_write(w, reg, new); + if (ret < 0) + return ret; + } } return change; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index c8610cb..39ba507 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -140,6 +140,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ + codec->using_regmap = true; break; default: -- 1.7.7.3

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[alsa-devel] [PATCH 1/2] ASoC: Add WM8962 DAC and ADC L/R swap support
by Mark Brown 12 Jan '12

12 Jan '12

Simple switches since there's no per-channel control. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 11125c9..e35237a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2223,6 +2223,8 @@ SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8962_LEFT_DAC_VOLUME, WM8962_RIGHT_DAC_VOLUME, 1, 127, 0, digital_tlv), SOC_SINGLE("DAC High Performance Switch", WM8962_ADC_DAC_CONTROL_2, 0, 1, 0), +SOC_SINGLE("DAC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 5, 1, 0), +SOC_SINGLE("ADC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 8, 1, 0), SOC_SINGLE("ADC High Performance Switch", WM8962_ADDITIONAL_CONTROL_1, 5, 1, 0), -- 1.7.7.3

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[alsa-devel] [PATCH 4/4] snd-usb-6fire: add analog input volume control
by Torsten Schenk 12 Jan '12

12 Jan '12

- Add a stereo volume control for analog input channel pair 1/2. - Add missing end-of-list entry at end of list for array mute_elements and vol_elements Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-12 00:19:40.680844042 +0100 +++ b/sound/usb/6fire/control.c 2012-01-12 00:23:21.013835655 +0100 @@ -76,6 +76,17 @@ comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); } +static void usb6fire_control_input_vol_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + int i; + + if (comm_rt) + for (i = 0; i < 2; i++) + comm_rt->write8(comm_rt, 0x12, 0x1c + i, + rt->input_vol[i] & 0x3f); +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -257,6 +268,48 @@ return 0; } +static int usb6fire_control_input_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = -15; + uinfo->value.integer.max = 15; + return 0; +} + +static int usb6fire_control_input_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + int changed = 0; + + if (rt->input_vol[0] != ucontrol->value.integer.value[0]) { + rt->input_vol[0] = ucontrol->value.integer.value[0]; + changed = 1; + } + if (rt->input_vol[1] != ucontrol->value.integer.value[1]) { + rt->input_vol[1] = ucontrol->value.integer.value[1]; + changed = 1; + } + + if (changed) + usb6fire_control_input_vol_update(rt); + + return changed; +} + +static int usb6fire_control_input_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = rt->input_vol[0]; + ucontrol->value.integer.value[1] = rt->input_vol[1]; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -378,7 +431,8 @@ .get = usb6fire_control_output_vol_get, .put = usb6fire_control_output_vol_put, .private_value = 2 - } + }, + {} }; static struct __devinitdata snd_kcontrol_new mute_elements[] = { @@ -411,7 +465,8 @@ .get = usb6fire_control_output_mute_get, .put = usb6fire_control_output_mute_put, .private_value = 2 - } + }, + {} }; static struct __devinitdata snd_kcontrol_new elements[] = { @@ -442,6 +497,15 @@ .get = usb6fire_control_digital_thru_get, .put = usb6fire_control_digital_thru_put }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 1/2 Capture Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_input_vol_info, + .get = usb6fire_control_input_vol_get, + .put = usb6fire_control_input_vol_put + }, {} }; @@ -505,6 +569,7 @@ usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); usb6fire_control_output_mute_update(rt); + usb6fire_control_input_vol_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-11 19:54:50.651190007 +0100 +++ b/sound/usb/6fire/control.h 2012-01-11 23:44:43.212950616 +0100 @@ -45,6 +45,7 @@ bool usb_streaming; u8 output_vol[6]; u8 output_mute; + s8 input_vol[2]; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

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[alsa-devel] [PATCH 2/4] snd-usb-6fire: add individual volume control for analog channels
by Torsten Schenk 12 Jan '12

12 Jan '12

Add a stereo volume control for every analog output pair 1/2, 3/4, 5/6. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-09 22:33:01.512182294 +0100 +++ b/sound/usb/6fire/control.c 2012-01-12 00:10:48.981232196 +0100 @@ -7,6 +7,10 @@ * Created: Jan 01, 2011 * Copyright: (C) Torsten Schenk * + * Thanks to: + * - Holger Ruckdeschel: he found out how to control individual channel + * volumes and introduced mute switch + * * 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 @@ -24,26 +28,6 @@ static char *line_phono_texts[2] = { "Line", "Phono" }; /* - * calculated with $value\[i\] = 128 \cdot sqrt[3]{\frac{i}{128}}$ - * this is done because the linear values cause rapid degredation - * of volume in the uppermost region. - */ -static const u8 log_volume_table[128] = { - 0x00, 0x19, 0x20, 0x24, 0x28, 0x2b, 0x2e, 0x30, 0x32, 0x34, - 0x36, 0x38, 0x3a, 0x3b, 0x3d, 0x3e, 0x40, 0x41, 0x42, 0x43, - 0x44, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4c, 0x4d, 0x4e, - 0x4e, 0x4f, 0x50, 0x51, 0x52, 0x53, 0x53, 0x54, 0x55, 0x56, - 0x56, 0x57, 0x58, 0x58, 0x59, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c, - 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x62, 0x62, - 0x63, 0x63, 0x64, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x68, - 0x68, 0x69, 0x69, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c, - 0x6d, 0x6d, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71, - 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75, - 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79, - 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c, - 0x7d, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f }; - -/* * data that needs to be sent to device. sets up card internal stuff. * values dumped from windows driver and filtered by trial'n'error. */ @@ -58,7 +42,7 @@ { 0x22, 0x03, 0x00 }, { 0x20, 0x03, 0x08 }, { 0x22, 0x04, 0x00 }, { 0x20, 0x04, 0x08 }, { 0x22, 0x05, 0x01 }, { 0x20, 0x05, 0x08 }, { 0x22, 0x04, 0x01 }, { 0x12, 0x04, 0x00 }, { 0x12, 0x05, 0x00 }, - { 0x12, 0x0d, 0x78 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, + { 0x12, 0x0d, 0x38 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, { 0x12, 0x23, 0x00 }, { 0x12, 0x06, 0x02 }, { 0x12, 0x03, 0x00 }, { 0x12, 0x02, 0x00 }, { 0x22, 0x03, 0x01 }, { 0 } /* TERMINATING ENTRY */ @@ -73,16 +57,15 @@ DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; -static void usb6fire_control_master_vol_update(struct control_runtime *rt) +static void usb6fire_control_output_vol_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; - if (comm_rt) { - /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 0x7f - - log_volume_table[rt->master_vol]); - /* unmute */ - comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); - } + int i; + + if (comm_rt) + for (i = 0; i < 6; i++) + comm_rt->write8(comm_rt, 0x12, 0x0f + i, + 180 - rt->output_vol[i]); } static void usb6fire_control_line_phono_update(struct control_runtime *rt) @@ -141,7 +124,6 @@ return ret; /* disable digital inputs and outputs */ - /* TODO: use spdif_x to enable/disable digital channels */ ret = comm_rt->write16(comm_rt, 0x02, 0x03, 0x00, 0x00); if (ret < 0) return ret; @@ -164,34 +146,61 @@ return -EINVAL; } -static int usb6fire_control_master_vol_info(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 127; + uinfo->value.integer.max = 180; return 0; } -static int usb6fire_control_master_vol_put(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; int changed = 0; - if (rt->master_vol != ucontrol->value.integer.value[0]) { - rt->master_vol = ucontrol->value.integer.value[0]; - usb6fire_control_master_vol_update(rt); + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return 0; + } + + if (rt->output_vol[ch0] != ucontrol->value.integer.value[0]) { + rt->output_vol[ch0] = ucontrol->value.integer.value[0]; changed = 1; } + if (rt->output_vol[ch1] != ucontrol->value.integer.value[1]) { + rt->output_vol[ch1] = ucontrol->value.integer.value[1]; + changed = 1; + } + + if (changed) + usb6fire_control_output_vol_update(rt); + return changed; } -static int usb6fire_control_master_vol_get(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = rt->master_vol; + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return 0; + } + + ucontrol->value.integer.value[0] = rt->output_vol[ch0]; + ucontrol->value.integer.value[1] = rt->output_vol[ch1]; + return 0; } @@ -286,18 +295,42 @@ return 0; } -static struct __devinitdata snd_kcontrol_new elements[] = { +static struct __devinitdata snd_kcontrol_new vol_elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 1/2 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 0 + }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Volume", + .name = "Output 3/4 Playback Volume", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = usb6fire_control_master_vol_info, - .get = usb6fire_control_master_vol_get, - .put = usb6fire_control_master_vol_put + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 1 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 5/6 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 2 + } +}; + +static struct __devinitdata snd_kcontrol_new elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line/Phono Capture Route", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -326,6 +359,39 @@ {} }; +static int usb6fire_control_add_virtual( + struct control_runtime *rt, + struct snd_card *card, + char *name, + struct snd_kcontrol_new *elems) +{ + int ret; + int i; + struct snd_kcontrol *vmaster = snd_ctl_make_virtual_master(name, NULL); + struct snd_kcontrol *control; + + if (!vmaster) + return -ENOMEM; + ret = snd_ctl_add(card, vmaster); + if (ret < 0) + return ret; + + i = 0; + while (elems[i].name) { + control = snd_ctl_new1(&elems[i], rt); + if (!control) + return -ENOMEM; + ret = snd_ctl_add(card, control); + if (ret < 0) + return ret; + ret = snd_ctl_add_slave(vmaster, control); + if (ret < 0) + return ret; + i++; + } + return 0; +} + int __devinit usb6fire_control_init(struct sfire_chip *chip) { int i; @@ -351,9 +417,17 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); - usb6fire_control_master_vol_update(rt); + usb6fire_control_output_vol_update(rt); usb6fire_control_streaming_update(rt); + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Volume", vol_elements); + if (ret) { + kfree(rt); + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + return ret; + } + i = 0; while (elements[i].name) { ret = snd_ctl_add(chip->card, snd_ctl_new1(&elements[i], rt)); diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 22:32:56.088002206 +0100 +++ b/sound/usb/6fire/control.h 2012-01-09 22:36:07.698544501 +0100 @@ -43,7 +43,7 @@ bool line_phono_switch; bool digital_thru_switch; bool usb_streaming; - u8 master_vol; + u8 output_vol[6]; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

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[alsa-devel] [PATCH v2] ASoC: Route Mic Bias in Visstrim_M10 board.
by Javier Martin 12 Jan '12

12 Jan '12

Visstrim_M10 board uses an external microphone that can be enabled/disabled by the user --- Changes since v1: - Change "audio_map" to "aic32x4_dapm_routes". - Use snd_soc_card fields to register controls, routes and widgets. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/mx27vis-aic32x4.c | 24 ++++++++++++++++++++++++ 1 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 054110b..30adbe1 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -74,6 +74,24 @@ static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { .hw_params = mx27vis_aic32x4_hw_params, }; +static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = { + SOC_DAPM_PIN_SWITCH("External Mic"), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_MIC("External Mic", NULL), +}; + +static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { + {"Mic Bias", NULL, "External Mic"}, + {"IN1_R", NULL, "Mic Bias"}, + {"IN2_R", NULL, "Mic Bias"}, + {"IN3_R", NULL, "Mic Bias"}, + {"IN1_L", NULL, "Mic Bias"}, + {"IN2_L", NULL, "Mic Bias"}, + {"IN3_L", NULL, "Mic Bias"}, +}; + static struct snd_soc_dai_link mx27vis_aic32x4_dai = { .name = "tlv320aic32x4", .stream_name = "TLV320AIC32X4", @@ -88,6 +106,12 @@ static struct snd_soc_card mx27vis_aic32x4 = { .name = "visstrim_m10-audio", .dai_link = &mx27vis_aic32x4_dai, .num_links = 1, + .controls = mx27vis_aic32x4_controls, + .num_controls = ARRAY_SIZE(mx27vis_aic32x4_controls), + .dapm_widgets = aic32x4_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), + .dapm_routes = aic32x4_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; static struct platform_device *mx27vis_aic32x4_snd_device; -- 1.7.0.4

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[alsa-devel] [PATCH 3/4] snd-usb-6fire: add mute controls for analog outputs
by Torsten Schenk 12 Jan '12

12 Jan '12

Add an individual mute control for every analog output channel. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-12 00:10:48.981232196 +0100 +++ b/sound/usb/6fire/control.c 2012-01-12 00:19:40.680844042 +0100 @@ -68,6 +68,14 @@ 180 - rt->output_vol[i]); } +static void usb6fire_control_output_mute_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + + if (comm_rt) + comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -204,6 +212,51 @@ return 0; } +static int usb6fire_control_output_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + u8 old = rt->output_mute; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + rt->output_mute &= ~((1 << ch0) | (1 << ch1)); + rt->output_mute |= ucontrol->value.integer.value[0] << ch0; + rt->output_mute |= ucontrol->value.integer.value[1] << ch1; + + if (rt->output_mute != old) + usb6fire_control_output_mute_update(rt); + + return rt->output_mute != old; +} + +static int usb6fire_control_output_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + ucontrol->value.integer.value[0] = 1 & (rt->output_mute >> ch0); + ucontrol->value.integer.value[1] = 1 & (rt->output_mute >> ch1); + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -328,6 +381,39 @@ } }; +static struct __devinitdata snd_kcontrol_new mute_elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 1/2 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 3/4 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + .private_value = 1 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 5/6 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + .private_value = 2 + } +}; + static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -418,13 +504,21 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); + usb6fire_control_output_mute_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, "Master Playback Volume", vol_elements); if (ret) { + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); kfree(rt); + return ret; + } + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Switch", mute_elements); + if (ret) { snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + kfree(rt); return ret; } diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 22:36:07.698544501 +0100 +++ b/sound/usb/6fire/control.h 2012-01-11 19:54:50.651190007 +0100 @@ -44,6 +44,7 @@ bool digital_thru_switch; bool usb_streaming; u8 output_vol[6]; + u8 output_mute; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

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[alsa-devel] [PATCH 1/4] snd-usb-6fire: remove driver version information
by Torsten Schenk 12 Jan '12

12 Jan '12

Remove unused driver version information from the individual files. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c --- a/sound/usb/6fire/chip.c 2012-01-09 21:55:55.863579333 +0100 +++ b/sound/usb/6fire/chip.c 2012-01-09 22:33:49.857786986 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify @@ -29,7 +28,7 @@ #include <sound/initval.h> MODULE_AUTHOR("Torsten Schenk <torsten.schenk(a)zoho.com>"); -MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.0"); +MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver"); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); diff -Nur a/sound/usb/6fire/chip.h b/sound/usb/6fire/chip.h --- a/sound/usb/6fire/chip.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/chip.h 2012-01-09 22:33:38.469409108 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c --- a/sound/usb/6fire/comm.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/comm.c 2012-01-09 22:33:11.620517902 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h --- a/sound/usb/6fire/comm.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/comm.h 2012-01-09 22:33:16.900693132 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/common.h b/sound/usb/6fire/common.h --- a/sound/usb/6fire/common.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/common.h 2012-01-09 22:33:24.636949983 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/control.c 2012-01-09 22:33:01.512182294 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/control.h 2012-01-09 22:32:56.088002206 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c --- a/sound/usb/6fire/firmware.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/firmware.c 2012-01-09 22:33:30.933158868 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c --- a/sound/usb/6fire/midi.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/midi.c 2012-01-09 22:35:07.356357430 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h --- a/sound/usb/6fire/midi.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/midi.h 2012-01-09 22:33:58.866085888 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c --- a/sound/usb/6fire/pcm.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/pcm.c 2012-01-09 22:34:04.546274285 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h --- a/sound/usb/6fire/pcm.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/pcm.h 2012-01-09 22:34:07.798382278 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify

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[alsa-devel] [PATCH] ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
by Liam Girdwood 11 Jan '12

11 Jan '12

The DL1 PDM interface adds a little gain depending on the output device. Add a method to retrieve the gain value for machine driver usage. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/codecs/twl6040.c | 23 +++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 1 + 2 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4ae6b82..3b00ee7 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -33,6 +33,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> @@ -1042,6 +1043,28 @@ int twl6040_get_hs_step_size(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(twl6040_get_hs_step_size); +int twl6040_get_dl1_gain(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + + if (snd_soc_dapm_get_pin_status(dapm, "EP")) + return -1; /* -1dB */ + + if (snd_soc_dapm_get_pin_status(dapm, "HSOR") || + snd_soc_dapm_get_pin_status(dapm, "HSOL")) { + + u8 val = snd_soc_read(codec, TWL6040_REG_HSLCTL); + if (val & TWL6040_HSDACMODE) + /* HSDACL in LP mode */ + return -8; /* -8dB */ + else + /* HSDACL in HP mode */ + return -1; /* -1dB */ + } + return 0; /* 0dB */ +} +EXPORT_SYMBOL_GPL(twl6040_get_dl1_gain); + static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* Capture gains */ SOC_DOUBLE_TLV("Capture Preamplifier Volume", diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index 0028699..b580c9d 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -39,5 +39,6 @@ void twl6040_hs_jack_detect(struct snd_soc_codec *codec, int twl6040_get_clk_id(struct snd_soc_codec *codec); int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim); int twl6040_get_hs_step_size(struct snd_soc_codec *codec); +int twl6040_get_dl1_gain(struct snd_soc_codec *codec); #endif /* End of __TWL6040_H__ */ -- 1.7.5.4

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[alsa-devel] [FIXED PATCH - ice1724 1/1] ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
by pavel.hofman＠ivitera.com 11 Jan '12

11 Jan '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Add the capture pcm only if there is at least one ADC configured in the SYSCONF register. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h index a0c5e00..4ca33a8 100644 --- a/sound/pci/ice1712/envy24ht.h +++ b/sound/pci/ice1712/envy24ht.h @@ -66,6 +66,7 @@ enum { #define VT1724_CFG_CLOCK384 0x40 /* 16.9344Mhz, 44.1kHz*384 */ #define VT1724_CFG_MPU401 0x20 /* MPU401 UARTs */ #define VT1724_CFG_ADC_MASK 0x0c /* one, two or one and S/PDIF, stereo ADCs */ +#define VT1724_CFG_ADC_NONE 0x0c /* no ADCs */ #define VT1724_CFG_DAC_MASK 0x03 /* one, two, three, four stereo DACs */ #define VT1724_REG_AC97_CFG 0x05 /* byte */ diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 9e18d3a..e797823 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1117,14 +1117,21 @@ static struct snd_pcm_ops snd_vt1724_capture_pro_ops = { static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device) { struct snd_pcm *pcm; - int err; + int capt, err; - err = snd_pcm_new(ice->card, "ICE1724", device, 1, 1, &pcm); + if ((ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_ADC_MASK) == + VT1724_CFG_ADC_NONE) + capt = 0; + else + capt = 1; + err = snd_pcm_new(ice->card, "ICE1724", device, 1, capt, &pcm); if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_vt1724_playback_pro_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_vt1724_capture_pro_ops); + if (capt) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_vt1724_capture_pro_ops); pcm->private_data = ice; pcm->info_flags = 0; -- 1.7.0.4

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