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Re: [alsa-devel] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Dan Carpenter 01 Apr '15

01 Apr '15

Hello Jyri Sarha, The patch a75a053f1eef: "ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used" from Mar 20, 2015, leads to the following Smatch warning: sound/soc/davinci/davinci-mcasp.c:1152 davinci_mcasp_startup() error: buffer overflow 'mcasp->ruledata' 2 <= 2 sound/soc/davinci/davinci-mcasp.c 1138 1139 /* 1140 * Limit the maximum allowed channels for the first stream: 1141 * number of serializers for the direction * tdm slots per serializer 1142 */ 1143 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 1144 dir = TX_MODE; 1145 else 1146 dir = RX_MODE; dir is either 1 or 2. 1147 1148 for (i = 0; i < mcasp->num_serializer; i++) { 1149 if (mcasp->serial_dir[i] == dir) 1150 max_channels++; 1151 } 1152 mcasp->ruledata[dir].serializers = max_channels; ^^^^^^^^^^^^^ ->ruledata[] has only two elements so ->ruledata[2] is beyond the end of the array. 1153 max_channels *= mcasp->tdm_slots; regards, dan carpenter

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[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: No not use IEC958_AES1_PRO_MODE_NOTID
by Jyri Sarha 01 Apr '15

01 Apr '15

No IEC958_AES?_PRO_* macros should be used in HDMI consumer audio mode and IEC958_AES1_PRO_MODE_NOTID should be applied to byte 1 when applicable. However IEC958_AES1_PRO_MODE_NOTID is defined as 0 so this fix does not affect the functionality in any way. Reported-by: Russell King <linux(a)arm.linux.org.uk> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- When I refactored the OMAP HDMI audio functionality I intentionally wanted to keep all the HW bits identical to the earlier implementation so that I could compare the register values of the old and the new implementations one to one. One of these days I should go trough and check all the bits now that new implementation has settled in. Best regards, Jyri sound/soc/omap/omap-hdmi-audio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index f7eb42a..4775da4 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -142,8 +142,6 @@ static int hdmi_dai_hw_params(struct snd_pcm_substream *substream, iec->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE; - iec->status[0] |= IEC958_AES1_PRO_MODE_NOTID; - iec->status[1] = IEC958_AES1_CON_GENERAL; iec->status[2] |= IEC958_AES2_CON_SOURCE_UNSPEC; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/8] ASoC: wm8971: Use system_power_efficient_wq instead of custom workqueue
by Lars-Peter Clausen 01 Apr '15

01 Apr '15

The delayed work used by the wm8971 driver to manage the caps charging doesn't have any special requirements that would justify using a custom workqueue, just use the generic system_power_efficient_wq instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8971.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 39ddb9b..44baacd 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -31,8 +31,6 @@ #define WM8971_REG_COUNT 43 -static struct workqueue_struct *wm8971_workq = NULL; - /* codec private data */ struct wm8971_priv { unsigned int sysclk; @@ -636,7 +634,8 @@ static int wm8971_resume(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_ON; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); } @@ -649,9 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) u16 reg; INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); - wm8971_workq = create_workqueue("wm8971"); - if (wm8971_workq == NULL) - return -ENOMEM; wm8971_reset(codec); @@ -659,7 +655,8 @@ static int wm8971_probe(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ @@ -681,8 +678,6 @@ static int wm8971_remove(struct snd_soc_codec *codec) { wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm8971_workq) - destroy_workqueue(wm8971_workq); return 0; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: atmel-pcm-pdc: merge atmel-pcm back in
by Alexandre Belloni 01 Apr '15

01 Apr '15

atmel-pcm.c was split into two files to create a generic framework for both PDC and DMA. atmel-pcm-dma.c is using the generic dmaengine framework since 95e0e07e710e (ASoC: atmel-pcm: use generic dmaengine framework). Merge atmel-pcm.c in atmel-pcm-pdc.c as this is now the only user. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/Makefile | 2 - sound/soc/atmel/atmel-pcm-pdc.c | 79 ++++++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.c | 121 ---------------------------------------- sound/soc/atmel/atmel-pcm.h | 5 -- 4 files changed, 79 insertions(+), 128 deletions(-) delete mode 100644 sound/soc/atmel/atmel-pcm.c diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 466a821da98c..b327e5cc8de3 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -1,10 +1,8 @@ # AT91 Platform Support -snd-soc-atmel-pcm-objs := atmel-pcm.o snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o -obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index a366b3503c28..da861b44413f 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -47,6 +47,85 @@ #include "atmel-pcm.h" +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = ATMEL_SSC_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", + (void *)buf->area, (void *)(long)buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + /*--------------------------------------------------------------------------*\ * Hardware definition \*--------------------------------------------------------------------------*/ diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c deleted file mode 100644 index 8ae3fa5ac60a..000000000000 --- a/sound/soc/atmel/atmel-pcm.c +++ /dev/null @@ -1,121 +0,0 @@ -/* - * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. - * - * Copyright (C) 2005 SAN People - * Copyright (C) 2008 Atmel - * - * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> - * - * Based on at91-pcm. by: - * Frank Mandarino <fmandarino(a)endrelia.com> - * Copyright 2006 Endrelia Technologies Inc. - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -#include <linux/module.h> -#include <linux/dma-mapping.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include "atmel-pcm.h" - -static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = ATMEL_SSC_DMABUF_SIZE; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", - (void *)buf->area, (void *)(long)buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -int atmel_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} -EXPORT_SYMBOL_GPL(atmel_pcm_mmap); - -int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} -EXPORT_SYMBOL_GPL(atmel_pcm_new); - -void atmel_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} -EXPORT_SYMBOL_GPL(atmel_pcm_free); - diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 12ae814eff21..6eaf081cad50 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -83,11 +83,6 @@ struct atmel_pcm_dma_params { #define ssc_readx(base, reg) (__raw_readl((base) + (reg))) #define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) -int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd); -void atmel_pcm_free(struct snd_pcm *pcm); -int atmel_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma); - #if defined(CONFIG_SND_ATMEL_SOC_PDC) || \ defined(CONFIG_SND_ATMEL_SOC_PDC_MODULE) int atmel_pcm_pdc_platform_register(struct device *dev); -- 2.1.0

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[alsa-devel] [PATCH] ASoC: rsnd: add Synchronous SRC mode
by Kuninori Morimoto 01 Apr '15

01 Apr '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas R-Car sound SRC (= Sampling Rate Converter) has Asynchronous/Synchronous SRC mode. Asynchronous mode is already supported via DPCM. This patch adds Synchronous mode on it. The condition of enabling Synchronous mode are - SoC is clock master - Sound uses SRC - Sound doesn't use DVC - Sound card uses DPCM (= rsrc-card card) amixer set "SRC Out Rate" on aplay xxx.wav & amixer set "SRC Out Rate" 48000 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 126 ++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 121 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index a0a2bda..3beb32e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -22,13 +22,15 @@ struct rsnd_src { struct rsnd_src_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; + struct rsnd_kctrl_cfg_s sen; /* sync convert enable */ + struct rsnd_kctrl_cfg_s sync; /* sync convert */ u32 convert_rate; /* sampling rate convert */ int err; }; #define RSND_SRC_NAME_SIZE 16 -#define rsnd_src_convert_rate(s) ((s)->convert_rate) +#define rsnd_enable_sync_convert(src) ((src)->sen.val) #define rsnd_src_of_node(priv) \ of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,src") @@ -233,6 +235,30 @@ int rsnd_src_ssi_irq_disable(struct rsnd_mod *ssi_mod) return 0; } +static u32 rsnd_src_convert_rate(struct rsnd_src *src) +{ + struct rsnd_mod *mod = &src->mod; + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 convert_rate; + + if (!runtime) + return 0; + + if (!rsnd_enable_sync_convert(src)) + return src->convert_rate; + + convert_rate = src->sync.val; + + if (!convert_rate) + convert_rate = src->convert_rate; + + if (!convert_rate) + convert_rate = runtime->rate; + + return convert_rate; +} + unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime) @@ -333,6 +359,9 @@ static int rsnd_src_init(struct rsnd_mod *mod, src->err = 0; + /* reset sync convert_rate */ + src->sync.val = 0; + /* * Initialize the operation of the SRC internal circuits * see rsnd_src_start() @@ -356,6 +385,9 @@ static int rsnd_src_quit(struct rsnd_mod *mod, src->convert_rate = 0; + /* reset sync convert_rate */ + src->sync.val = 0; + return 0; } @@ -672,6 +704,7 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod) struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_src *src = rsnd_mod_to_src(mod); u32 convert_rate = rsnd_src_convert_rate(src); + u32 cr, route; uint ratio; int ret; @@ -692,13 +725,21 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod) if (ret < 0) return ret; - rsnd_mod_write(mod, SRC_SRCCR, 0x00011110); - + cr = 0x00011110; + route = 0x0; if (convert_rate) { - /* Gen1/Gen2 are not compatible */ - rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1); + route = 0x1; + + if (rsnd_enable_sync_convert(src)) { + cr |= 0x1; + route |= rsnd_io_is_play(io) ? + (0x1 << 24) : (0x1 << 25); + } } + rsnd_mod_write(mod, SRC_SRCCR, cr); + rsnd_mod_write(mod, SRC_ROUTE_MODE0, route); + switch (rsnd_mod_id(mod)) { case 5: case 6: @@ -811,6 +852,80 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod, return ret; } +static void rsnd_src_reconvert_update(struct rsnd_mod *mod) +{ + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + u32 convert_rate = rsnd_src_convert_rate(src); + u32 fsrate; + + if (!runtime) + return; + + if (!convert_rate) + convert_rate = runtime->rate; + + fsrate = 0x0400000 / convert_rate * runtime->rate; + + /* update IFS */ + rsnd_mod_write(mod, SRC_IFSVR, fsrate); +} + +static int rsnd_src_pcm_new(struct rsnd_mod *mod, + struct snd_soc_pcm_runtime *rtd) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + int ret; + + /* + * enable SRC sync convert if possible + */ + + /* + * Gen1 is not supported + */ + if (rsnd_is_gen1(priv)) + return 0; + + /* + * SRC sync convert needs clock master + */ + if (!rsnd_rdai_is_clk_master(rdai)) + return 0; + + /* + * We can't use SRC sync convert + * if it has DVC + */ + if (rsnd_io_to_mod_dvc(io)) + return 0; + + /* + * enable sync convert + */ + ret = rsnd_kctrl_new_s(mod, rtd, + rsnd_io_is_play(io) ? + "SRC Out Rate Switch" : + "SRC In Rate Switch", + rsnd_src_reconvert_update, + &src->sen, 1); + if (ret < 0) + return ret; + + ret = rsnd_kctrl_new_s(mod, rtd, + rsnd_io_is_play(io) ? + "SRC Out Rate" : + "SRC In Rate", + rsnd_src_reconvert_update, + &src->sync, 192000); + + return ret; +} + static struct rsnd_mod_ops rsnd_src_gen2_ops = { .name = SRC_NAME, .dma_req = rsnd_src_dma_req, @@ -821,6 +936,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = { .start = rsnd_src_start_gen2, .stop = rsnd_src_stop_gen2, .hw_params = rsnd_src_hw_params, + .pcm_new = rsnd_src_pcm_new, }; struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id) -- 1.9.1

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[alsa-devel] [PATCH 00/25] treewide: Use bool function return values of true/false not 1/0
by Joe Perches 01 Apr '15

01 Apr '15

Joe Perches (25): arm: Use bool function return values of true/false not 1/0 arm64: Use bool function return values of true/false not 1/0 hexagon: Use bool function return values of true/false not 1/0 ia64: Use bool function return values of true/false not 1/0 mips: Use bool function return values of true/false not 1/0 powerpc: Use bool function return values of true/false not 1/0 s390: Use bool function return values of true/false not 1/0 sparc: Use bool function return values of true/false not 1/0 tile: Use bool function return values of true/false not 1/0 unicore32: Use bool function return values of true/false not 1/0 x86: Use bool function return values of true/false not 1/0 virtio_console: Use bool function return values of true/false not 1/0 csiostor: Use bool function return values of true/false not 1/0 dcache: Use bool function return values of true/false not 1/0 nfsd: nfs4state: Use bool function return values of true/false not 1/0 include/linux: Use bool function return values of true/false not 1/0 sound: Use bool function return values of true/false not 1/0 rcu: tree_plugin: Use bool function return values of true/false not 1/0 sched: Use bool function return values of true/false not 1/0 ftrace: Use bool function return values of true/false not 1/0 slub: Use bool function return values of true/false not 1/0 bridge: Use bool function return values of true/false not 1/0 netfilter: Use bool function return values of true/false not 1/0 security: Use bool function return values of true/false not 1/0 sound: wm5100-tables: Use bool function return values of true/false not 1/0 arch/arm/include/asm/dma-mapping.h | 8 ++-- arch/arm/include/asm/kvm_emulate.h | 2 +- arch/arm/mach-omap2/powerdomain.c | 14 +++--- arch/arm64/include/asm/dma-mapping.h | 2 +- arch/hexagon/include/asm/dma-mapping.h | 2 +- arch/ia64/include/asm/dma-mapping.h | 2 +- arch/mips/include/asm/dma-mapping.h | 2 +- arch/powerpc/include/asm/dcr-native.h | 2 +- arch/powerpc/include/asm/dma-mapping.h | 4 +- arch/powerpc/include/asm/kvm_book3s_64.h | 4 +- arch/powerpc/sysdev/dcr.c | 2 +- arch/s390/include/asm/dma-mapping.h | 2 +- arch/sparc/mm/init_64.c | 8 ++-- arch/tile/include/asm/dma-mapping.h | 2 +- arch/unicore32/include/asm/dma-mapping.h | 2 +- arch/x86/include/asm/archrandom.h | 2 +- arch/x86/include/asm/dma-mapping.h | 2 +- arch/x86/include/asm/kvm_para.h | 2 +- arch/x86/kvm/cpuid.h | 2 +- arch/x86/kvm/vmx.c | 72 ++++++++++++++-------------- drivers/char/virtio_console.c | 2 +- drivers/scsi/csiostor/csio_scsi.c | 4 +- fs/dcache.c | 12 ++--- fs/nfsd/nfs4state.c | 2 +- include/linux/blkdev.h | 2 +- include/linux/ide.h | 2 +- include/linux/kgdb.h | 2 +- include/linux/mfd/db8500-prcmu.h | 2 +- include/linux/mm.h | 2 +- include/linux/power_supply.h | 8 ++-- include/linux/ssb/ssb_driver_extif.h | 2 +- include/linux/ssb/ssb_driver_gige.h | 16 +++---- include/sound/soc.h | 4 +- kernel/rcu/tree_plugin.h | 4 +- kernel/sched/auto_group.h | 2 +- kernel/sched/completion.c | 16 ++++--- kernel/trace/ftrace.c | 10 ++-- mm/slub.c | 12 ++--- net/bridge/br_private.h | 2 +- net/ipv4/netfilter/ipt_ah.c | 2 +- net/netfilter/ipset/ip_set_hash_ip.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipmark.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipport.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipportip.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipportnet.c | 8 ++-- net/netfilter/ipset/ip_set_hash_net.c | 8 ++-- net/netfilter/ipset/ip_set_hash_netiface.c | 8 ++-- net/netfilter/ipset/ip_set_hash_netport.c | 8 ++-- net/netfilter/ipset/ip_set_hash_netportnet.c | 8 ++-- net/netfilter/xt_connlimit.c | 2 +- net/netfilter/xt_hashlimit.c | 2 +- net/netfilter/xt_ipcomp.c | 2 +- security/apparmor/file.c | 8 ++-- security/apparmor/policy.c | 10 ++-- sound/soc/codecs/wm5100-tables.c | 12 ++--- 55 files changed, 178 insertions(+), 176 deletions(-) -- 2.1.2

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[alsa-devel] [PATCH v3 0/5] ALSA: hda - add hda controller to hda core
by Vinod Koul 01 Apr '15

01 Apr '15

This incorporates the comments from Takashi on 1st patch and uses the change from Takashi's unstable tree for controller and then adds the DSP loader code on top. This is based on sound/topic/hda-core Comments welcome Jeeja KP (1): ALSA: hda: move common hda controller register defines up Vinod Koul (4): ALSA: hda: add HDA_MAX_CODECS ALSA: hda - Add the controller helper codes to hda-core module ALSA: hda - move dsp lock to hdac ALSA: hda - add the DSP loader code include/sound/hda_registers.h | 177 +++++++++++++ include/sound/hdaudio.h | 185 ++++++++++++- sound/hda/Makefile | 3 +- sound/hda/hdac_bus.c | 13 +- sound/hda/hdac_controller.c | 401 ++++++++++++++++++++++++++++ sound/hda/hdac_stream.c | 565 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.c | 17 +- sound/pci/hda/hda_controller.h | 171 +----------- 8 files changed, 1347 insertions(+), 185 deletions(-) create mode 100644 include/sound/hda_registers.h create mode 100644 sound/hda/hdac_controller.c create mode 100644 sound/hda/hdac_stream.c -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: rcar: mark device data as constant
by Uwe Kleine-König 01 Apr '15

01 Apr '15

A driver's device data should and can be const. This is a follow-up on commit 33187fb4a203 (ASoC: rsnd: constify of_device_id array) which marked the of_device_id as const. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- This is also patch 4 of my series[1] which degraded to this hunk when rebasing to next because all the other changes were already addressed by Fabian. Best regards Uwe [1] http://thread.gmane.org/gmane.linux.alsa.devel/136241/ sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 3d8e4d60c83d..9f48d75fa992 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -99,11 +99,11 @@ #define RSND_RATES SNDRV_PCM_RATE_8000_96000 #define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) -static struct rsnd_of_data rsnd_of_data_gen1 = { +static const struct rsnd_of_data rsnd_of_data_gen1 = { .flags = RSND_GEN1, }; -static struct rsnd_of_data rsnd_of_data_gen2 = { +static const struct rsnd_of_data rsnd_of_data_gen2 = { .flags = RSND_GEN2, }; -- 2.1.4 -- Pengutronix e.K. | Uwe Kleine-König | Industrial Linux Solutions | http://www.pengutronix.de/ |

2 1

[alsa-devel] [PATCH v2 0/4] ASoC: constification and random cleanups
by Uwe Kleine-König 01 Apr '15

01 Apr '15

Hello, this is the series sent on Sunday rebased on next. There was a conflict with commit c660c0a80586 (ASoC: fsi: constify of_device_id array) which did a part of my 2nd patch but already conflicted with the first one. Mark said he already took patch 3, but that should not be a problem. Patch 4 applied just fine to next, so this is just a resend. Best regards Uwe Uwe Kleine-König (4): ASoC: fsi: reorder code to make a forward declaration superfluous ASoC: fsi: mark several data structures as const ASoC: fsi: fix license specification ASoC: mark of_device_id and device data as constant sound/soc/sh/fsi.c | 71 ++++++++++++++++++++++++------------------------ sound/soc/sh/rcar/core.c | 4 +-- 2 files changed, 37 insertions(+), 38 deletions(-) -- 2.1.4

2 10

[alsa-devel] [PATCH 0/5] Intel SST driver restructure
by Jie Yang 01 Apr '15

01 Apr '15

In order to prepare for the upstreaming of Skylake drivers and to allow reuse of code between different Intel audio DSP products, this patch series restructures the Intel directory as follows: 1. Create "common" directory. Move common files into this directory. 2. Create "boards" directory. Move all machine drivers into this directory. 3. Create "haswell" directory. Move all remaining hsw/bdw code into this directory. 4. Create "baytrail" directory. Move remaining baytrail driver to this directory. 5. Create "atom" directory. Move all MCG platform drivers(including sst/ folder) to this directory. Todo next step: 1. Change and move atom related codes(in atom/sst/) to corresponding and try share as many as possible codes(e.g. sst_acpi.c). 2. Share more common codes about IPC for different platforms, e.g. The methods for queueing and sending/reciving Tx/Rx messages, The methods for mailbox IO, The IPC error return codes, etc. Jie Yang (5): ASoC: Intel: create common folder and move common files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create atom folder and move atom platform files in sound/soc/intel/Makefile | 42 +++------------------- sound/soc/intel/atom/Makefile | 7 ++++ sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform.h | 0 sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 0 sound/soc/intel/{ => atom}/sst/sst.h | 0 sound/soc/intel/{ => atom}/sst/sst_acpi.c | 0 sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 0 sound/soc/intel/{ => atom}/sst/sst_ipc.c | 0 sound/soc/intel/{ => atom}/sst/sst_loader.c | 0 sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 0 sound/soc/intel/{ => atom}/sst/sst_stream.c | 0 sound/soc/intel/baytrail/Makefile | 4 +++ sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 +-- sound/soc/intel/boards/Makefile | 15 ++++++++ sound/soc/intel/{ => boards}/broadwell.c | 6 ++-- sound/soc/intel/{ => boards}/byt-max98090.c | 2 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 +-- .../{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 0 sound/soc/intel/{ => boards}/haswell.c | 6 ++-- sound/soc/intel/{ => boards}/mfld_machine.c | 0 sound/soc/intel/common/Makefile | 6 ++++ sound/soc/intel/{ => common}/sst-acpi.c | 0 sound/soc/intel/{ => common}/sst-dsp-priv.h | 0 sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 0 sound/soc/intel/{ => common}/sst-firmware.c | 0 sound/soc/intel/haswell/Makefile | 4 +++ sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 6 ++-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 4 +-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 0 sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 6 ++-- 43 files changed, 64 insertions(+), 60 deletions(-) create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (100%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (100%) rename sound/soc/intel/{ => atom}/sst/sst.h (100%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (100%) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (98%) rename sound/soc/intel/{ => boards}/byt-max98090.c (99%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (100%) rename sound/soc/intel/{ => boards}/haswell.c (98%) rename sound/soc/intel/{ => boards}/mfld_machine.c (100%) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (100%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (100%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (100%) rename sound/soc/intel/{ => common}/sst-firmware.c (100%) create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (100%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (99%) -- 1.9.1

6 16

[alsa-devel] Rane SL2/3/4 driver?
by Be 01 Apr '15

01 Apr '15

Hi, I recently published the first version of Crossfade GNU/Linux, a distro for DJing from USB drives on any PC (see http://mixxx.org/forums/viewtopic.php?f=1&t=7031 for details). I just tried it on my friend's Macbook which he uses with a Rane SL4 USB sound card. I realized there is no Linux driver for the current line of sound cards sold for digital vinyl emulation with Serato DJ. These are the Rane SL2, SL3, and SL4 (http://serato.com/hardware/all/mixers-interfaces ; the Denon DS1 was also recently announced but not available yet: http://serato.com/latest/blog/17973/announcing-denon-ds1 ). They range from 4 in/4 out to 10 in/10 out and all have phono preamps built in. There appears to be support for the old and discontinued Rane SL1, which used USB 1.1, but the SL2/3/4 use USB 2.0. Has anyone thought about writing a driver for these?

1 1

[alsa-devel] [RESEND PATCH] MAINTAINERS: change the Atmel audio alsa driver entry
by Nicolas Ferre 31 Mar '15

31 Mar '15

I take over the the maintainship of Atmel alsa drivers from Voice. Thanks for your work! Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> Acked-by: Bo Shen <voice.shen(a)atmel.com> Cc: Mark Brown <broonie(a)kernel.org> --- Hi Mark, Here is the MAINTAINERS' change that I sent to the wrong address. Can you please take it with your tree? Thanks, bye. MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 6cb40f8736ee..a915bb124fe7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1762,7 +1762,7 @@ S: Supported F: drivers/tty/serial/atmel_serial.c ATMEL Audio ALSA driver -M: Bo Shen <voice.shen(a)atmel.com> +M: Nicolas Ferre <nicolas.ferre(a)atmel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/atmel -- 2.1.3

2 1

[alsa-devel] [PATCH 1/4] ASoC: fsi: reorder code to make a forward declaration superfluous
by Uwe Kleine-König 31 Mar '15

31 Mar '15

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/sh/fsi.c | 69 +++++++++++++++++++++++++++--------------------------- 1 file changed, 34 insertions(+), 35 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b87b22e88e43..dab6aa661a22 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1876,7 +1876,40 @@ static void fsi_handler_init(struct fsi_priv *fsi, } } -static struct of_device_id fsi_of_match[]; +static struct fsi_core fsi1_core = { + .ver = 1, + + /* Interrupt */ + .int_st = INT_ST, + .iemsk = IEMSK, + .imsk = IMSK, +}; + +static struct fsi_core fsi2_core = { + .ver = 2, + + /* Interrupt */ + .int_st = CPU_INT_ST, + .iemsk = CPU_IEMSK, + .imsk = CPU_IMSK, + .a_mclk = A_MST_CTLR, + .b_mclk = B_MST_CTLR, +}; + +static struct of_device_id fsi_of_match[] = { + { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, + { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, + {}, +}; +MODULE_DEVICE_TABLE(of, fsi_of_match); + +static struct platform_device_id fsi_id_table[] = { + { "sh_fsi", (kernel_ulong_t)&fsi1_core }, + { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, + {}, +}; +MODULE_DEVICE_TABLE(platform, fsi_id_table); + static int fsi_probe(struct platform_device *pdev) { struct fsi_master *master; @@ -2072,40 +2105,6 @@ static struct dev_pm_ops fsi_pm_ops = { .resume = fsi_resume, }; -static struct fsi_core fsi1_core = { - .ver = 1, - - /* Interrupt */ - .int_st = INT_ST, - .iemsk = IEMSK, - .imsk = IMSK, -}; - -static struct fsi_core fsi2_core = { - .ver = 2, - - /* Interrupt */ - .int_st = CPU_INT_ST, - .iemsk = CPU_IEMSK, - .imsk = CPU_IMSK, - .a_mclk = A_MST_CTLR, - .b_mclk = B_MST_CTLR, -}; - -static struct of_device_id fsi_of_match[] = { - { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, - { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, - {}, -}; -MODULE_DEVICE_TABLE(of, fsi_of_match); - -static struct platform_device_id fsi_id_table[] = { - { "sh_fsi", (kernel_ulong_t)&fsi1_core }, - { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, - {}, -}; -MODULE_DEVICE_TABLE(platform, fsi_id_table); - static struct platform_driver fsi_driver = { .driver = { .name = "fsi-pcm-audio", -- 2.1.4

2 6

[alsa-devel] [PATCH] ASoC: pcm512x: Do not redefine DIV_ROUND_CLOSEST_ULL
by Fabio Estevam 31 Mar '15

31 Mar '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> DIV_ROUND_CLOSEST_ULL is already defined at <linux/kernel.h>, so use the definition from this header file rather than re-defining it locally. This fixes the following build warning: sound/soc/codecs/pcm512x.c:34:0: warning: "DIV_ROUND_CLOSEST_ULL" redefined [enabled by default] In file included from include/linux/list.h:8:0, from include/linux/module.h:9, from sound/soc/codecs/pcm512x.c:19: include/linux/kernel.h:110:0: note: this is the location of the previous definition Reported-by: Olof's autobuilder <build(a)lixom.net> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/pcm512x.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5a30fdd..e12764d 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -18,6 +18,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/clk.h> +#include <linux/kernel.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -31,8 +32,6 @@ #define DIV_ROUND_DOWN_ULL(ll, d) \ ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) -#define DIV_ROUND_CLOSEST_ULL(ll, d) \ - ({ unsigned long long _tmp = (ll)+(d)/2; do_div(_tmp, d); _tmp; }) #define PCM512x_NUM_SUPPLIES 3 static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { -- 1.9.1

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
by Mark Brown 31 Mar '15

31 Mar '15

On Mon, Mar 30, 2015 at 08:16:24PM -0700, Scott Branden wrote: The CC list for this patch is pretty wide - please look at who you're sending this to and try to send to only relevant people (for example I'm not sure the Raspberry Pi people need to review this). People working upstream get a lot of mail so it's helpful to avoid filling their inboxes with random irrelevant stuff. > sound/soc/bcm/Kconfig | 11 + > sound/soc/bcm/Makefile | 5 +- > sound/soc/bcm/cygnus-pcm.c | 918 +++++++++++++++++++++++++ > sound/soc/bcm/cygnus-pcm.h | 45 ++ > sound/soc/bcm/cygnus-ssp.c | 1613 ++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/bcm/cygnus-ssp.h | 84 +++ > 6 files changed, 2675 insertions(+), 1 deletion(-) Send the DMA and DAI drivers as separate patches please, it makes review easier. > +config SND_SOC_CYGNUS > + tristate "SoC platform audio for Broadcom Cygnus chips" > + depends on ARCH_BCM_CYGNUS || COMPILE_TEST > + default ARCH_BCM_CYGNUS Remove the default setting here - we don't do this for other drivers, we shouldn't do it here. > +/* > + * PERIOD_BYTES_MIN is the number of bytes to at which the interrupt will tick. > + * This number should be a multiple of 256 > + */ > +#define PERIOD_BYTES_MIN 0x100 This sounds like it's a setting rather than actually the minimum? > +static const struct snd_pcm_hardware cygnus_pcm_hw = { > + .info = SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_INTERLEAVED, The DMA controller is integrated into the IP? > +static int enable_count; This looks bogus - why is this a global variable not part of the device struct and if it does need to be global why does it need no locking? > + if (aio->portnum == 0) > + *p_rbuf = RINGBUF_REG_PLAYBACK(0); > + else if (aio->portnum == 1) > + *p_rbuf = RINGBUF_REG_PLAYBACK(2); > + else if (aio->portnum == 2) > + *p_rbuf = RINGBUF_REG_PLAYBACK(4); > + else if (aio->portnum == 3) > + *p_rbuf = RINGBUF_REG_PLAYBACK(6); /*SPDIF */ > + else > + status = -EINVAL; This looks like a switch statement, there are many places in the code where you're writing switch statements as chains of ifs. > +static void ringbuf_inc(void __iomem *audio_io, bool is_playback, > + const struct ringbuf_regs *p_rbuf) > +{ > + u32 regval, endval, active_ptr; > + > + if (is_playback) > + active_ptr = p_rbuf->wraddr; > + else > + active_ptr = p_rbuf->rdaddr; > + > + endval = readl(audio_io + p_rbuf->endaddr); > + regval = readl(audio_io + active_ptr); > + regval = regval + p_rbuf->period_bytes; > + if (regval > endval) > + regval -= p_rbuf->buf_size; > + > + writel(regval, audio_io + active_ptr); > +} So it looks like we're getting an interrupt per period and we have to manually advance to the next one? > +static irqreturn_t cygnus_dma_irq(int irq, void *data) > +{ > + u32 r5_status; > + struct cygnus_audio *cygaud; > + > + cygaud = (struct cygnus_audio *)data; If you need to cast away from void something is very wrong. > + /* > + * R5 status bits Description > + * 0 ESR0 (playback FIFO interrupt) > + * 1 ESR1 (playback rbuf interrupt) > + * 2 ESR2 (capture rbuf interrupt) > + * 3 ESR3 (Freemark play. interrupt) > + * 4 ESR4 (Fullmark capt. interrupt) > + */ > + r5_status = readl(cygaud->audio + INTH_R5F_STATUS_OFFSET); > + > + /* If playback interrupt happened */ > + if (ANY_PLAYBACK_IRQ & r5_status) > + handle_playback_irq(cygaud); > + > + /* If capture interrupt happened */ > + if (ANY_CAPTURE_IRQ & r5_status) > + handle_capture_irq(cygaud); > + > + /* > + * clear r5 interrupts after servicing them to avoid overwriting > + * esr_status > + */ > + writel(r5_status, cygaud->audio + INTH_R5F_CLEAR_OFFSET); This feels racy especially given that we seem to need every interrupt delivering. What if another period happens during the servicing? I don't understand what "overwriting esr_status" means here. > + return IRQ_HANDLED; If neither playback nor capture interrupts were flagged we should return IRQ_NONE. > +/* > + * This code is identical to what is done by the framework, when we do not > + * supply a 'copy' function. Having our own copy hook in place allows for > + * us to easily add some diagnotics when needed. > + */ > +int cygnus_pcm_copy(struct snd_pcm_substream *substream, int channel, > + snd_pcm_uframes_t pos, > + void __user *buf, snd_pcm_uframes_t count) This is obviously not suitable for mainline - "let's just cut'n'paste the shared code in case we want to add diagnostics in future" does not scale, you can always add local diagnostics in the core code. > +}; > +/* Blank line between these two please. Missing blanks between bits of code seem to be a common issue in this driver. > +static int audio_ssp_in_enable(struct cygnus_aio_port *aio, bool enable) > +{ > + u32 value; > + > + if (enable) { > + } else { Why not just write two functions? > +static int audio_ssp_source_bitres(struct cygnus_aio_port *aio, unsigned bits) > +{ > + u32 mask = 0x1f; > + u32 value = 0; > + > + if ((bits == 8) || (bits == 16) || (bits == 32)) { > + value = readl(aio->audio + aio->regs.bf_sourcech_cfg); > + value &= ~(mask << BF_SRC_CFGX_BIT_RES); > + > + /* 32 bit mode is coded as 0 */ > + if ((bits == 8) || (bits == 16)) > + value |= (bits << BF_SRC_CFGX_BIT_RES); > + > + writel(value, aio->audio + aio->regs.bf_sourcech_cfg); > + return 0; > + } > + > + pr_err("Bit resolution not supported %d\n", bits); > + return -EINVAL; dev_err() (this applies throughout the driver). It's not clear why this is a function and not just written as a single statement in the one place that it's used (there are multiple calls but they're all together and could just be inlined). > + if (!aio->bitrate) { > + pr_err("%s Use .set_clkdiv() to set bitrate\n", __func__); > + return -EINVAL; > + } This seems bogus - why are we enforcing the use of set_clkdiv()? Can't we figure out something esneible? > + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { > + /* Set the SSP up as slave */ > + case SND_SOC_DAIFMT_CBM_CFM: The comments here look odd due to the indentation and really aren't adding much anyway. > + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > + case SND_SOC_DAIFMT_RIGHT_J: > + case SND_SOC_DAIFMT_LEFT_J: > + return -EINVAL; These are just eaten by the default case. > + case SND_SOC_DAIFMT_DSP_A: > + case SND_SOC_DAIFMT_DSP_B: > + ssp_newcfg |= BIT(I2S_OUT_CFGX_TDM_MODE); > + > + /* DSP_A = data after FS, DSP_B = data during FS */ > + if (SND_SOC_DAIFMT_DSP_A) > + ssp_newcfg |= BIT(I2S_OUT_CFGX_DATA_ALIGNMENT); That if statement isn't doing what was intended... > + } else { > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + audio_ssp_in_enable(aio, 1); > + break; > + > + case SNDRV_PCM_TRIGGER_STOP: > + audio_ssp_in_enable(aio, 0); > + break; > + } We can just ignore other triggers? It's not clear why this is different to the playback case. > +int cygnus_ssp_get_mode(struct snd_soc_dai *cpu_dai) > +{ > + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai); > + > + return aio->mode; > +} > +EXPORT_SYMBOL(cygnus_ssp_get_mode); What is this for, it's setting off alarm bells? Note also that ASoC is _GPL() only. > +static const struct snd_kcontrol_new pll_tweak_controls[] = { > + SOC_SINGLE_EXT("PLL Tweak", 0, 0, PLL_NDIV_FRACT_MAX, 0, > + pll_tweak_get, pll_tweak_put), > +}; Whatever this control is doing it should be a separate patch (as I think we discussed in person a ELC?), it's clearly something that's unusual and is likely to block the rest of the code as a result. At a high level this needs at least some documentation. > +int cygnus_ssp_add_pll_tweak_controls(struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + > + return snd_soc_add_dai_controls(cpu_dai, > + pll_tweak_controls, > + ARRAY_SIZE(pll_tweak_controls)); > +} > +EXPORT_SYMBOL(cygnus_ssp_add_pll_tweak_controls); Again, why is this being exported and why is it not _GPL? If the driver is adding controls I'd expect it to just add its controls itself. > +static int cygnus_ssp_probe(struct platform_device *pdev) > +{ > + struct device *dev = &pdev->dev; > + struct device_node *child_node; > + struct resource *res = pdev->resource; > + struct cygnus_audio *cygaud; > + int err = -EINVAL; > + int node_count; > + int active_port_count; > + > + if (!of_match_device(cygnus_ssp_of_match, dev)) { > + dev_err(dev, "Failed to find ssp controller\n"); > + return -ENODEV; > + } This error message is misleading - you mean to say that the driver got loaded for a device it doesn't understand. > + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + cygaud->audio = devm_ioremap_resource(dev, res); > + if (IS_ERR(cygaud->audio)) { > + dev_err(dev, "audio_io ioremap failed\n"); > + return PTR_ERR(cygaud->audio); devm_ioremap_resource() will complain for you, and in general you should be printing error codes. > + node_count = 0; This doesn't seem to be needed? > + node_count = of_get_child_count(pdev->dev.of_node); > + if ((node_count < 1) || (node_count > CYGNUS_MAX_PORTS)) { > + dev_err(dev, "Incorrct number of child nodes\n"); > + return -EINVAL; Spelling mistake there and it would be helpful to the user to tell them what we parsed.

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Re: [alsa-devel] [pulseaudio-discuss] Can't get Pulseadio to playback stream on Raspbian with HifiBerry DAC+
by Raymond Yau 31 Mar '15

31 Mar '15

2015-3-30 上午12:22 於 "David Martinez" <david(a)sojournmusic.net> 寫道： > > Label says PiFi DAC+. v2.0 > > I'll need to look up how to apply the patch. I'm not familiar with how to do it. This look like driver bug since capabilities should be pvolume instead of volume if your sound card are half duplex Simple mixer control 'PCM',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 215 Front Left: 172 [80%] Front Right: 172 [80%] Simple mixer control 'Playback',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 1 Front Left: 0 [0%] [-6.00dB] Front Right: 0 [0%] [-6.00dB] Simple mixer control 'Playback Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 1 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] > >> >> > >> > I'll check HiFiBerry's site and see if I can get it. >> > >> >> https://www.hifiberry.com/dacplus/ >> >> What is the label on the black DAC chip ? >> >> Aplay/Arecord output >> !!-------------------- >> >> APLAY >> >> **** List of PLAYBACK Hardware Devices **** >> card 0: sndrpihifiberry [snd_rpi_hifiberry_dacplus], device 0: HiFiBerry DAC HiFi pcm512x-hifi-0 [] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> ARECORD >> >> **** List of CAPTURE Hardware Devices **** >> >> Seem half duplex >> >>

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[alsa-devel] [RFC v2][PATCH 00/11] digi00x: new driver for Digidesign 002/003 family
by Takashi Sakamoto 31 Mar '15

31 Mar '15

This patchset is revised version of my previous set. [alsa-devel] [RFC][PATCH 00/11] digi00x: new driver for Digidesign 002/003 family http://mailman.alsa-project.org/pipermail/alsa-devel/2015-March/089141.html Improvements from the previous set: * improve double-oh-three (DOT) application * extend hwdep interface for asynchronous messaging * limit sampling rate for PCM substream when clock source is external or any PCM substreams are running * support s16 samples for PCM playback In this time, I test PCM functionality with my test programs. As a result, I can confirm that S/PDIF and ADAT output from my Digi 002 Rack includes expected PCM samples as my programs generates. I think we can judge the implementation of DOT is proper for Digi 002 rack. Currently, several issues remains: * The port for MIDI machine control message is not tested yet, because 002/003 console model are required. * When allocates 2 or more channel numbers for the device, after 15 to 20 seconds from playbacking, any PCM samples causes noisy sound. Then, all of LED on the front panel light. The streaming still continues correctly. * The actual effects of external clock source is not clear. When set the clock source is somewhat external, even if stopping the clock source, the device continues to sound PCM samples against my expectation. * The meaning of asynchronous messages is unknown. This patchset adds a functionality to receive it in userspace. You can test it with updated libhinawa sample script. https://github.com/takaswie/libhinawa Well, I'll stop my work for this driver in this developing period (for Linux 4.0), because this work is not in my original plan. I hope any users test this driver and report the result till Linux 4.2 merge window. If not, this driver will not be merged into upstream, naturally. I prepare for DKMS package to test this patchset. Please use it if needed. https://github.com/takaswie/snd-firewire-improve Takashi Sakamoto (11): ALSA: digi00x: add skeleton for Digi 002/003 device driver ALSA: digi00x: add streaming functionality ALSA: digi00x: add proc node for clock status ALSA: digi00x: add PCM functionality ALSA: digi00x: add MIDI functionality ALSA: digi00x: add hwdep interface ALSA: digi00x: support unknown asynchronous message ALSA: digi00x: support MIDI ports for device control ALSA: firewire-lib: allows to implement external MIDI callback function ALSA: digi00x: improve MIDI capture/playback ALSA: digi00x: apply double-oh-three algorithm to multiplex PCM samples include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 8 + sound/firewire/Kconfig | 10 + sound/firewire/Makefile | 1 + sound/firewire/amdtp.c | 30 ++- sound/firewire/amdtp.h | 5 + sound/firewire/digi00x/Makefile | 3 + sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++++++ sound/firewire/digi00x/digi00x-midi.c | 199 +++++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 344 +++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 86 +++++++ sound/firewire/digi00x/digi00x-protocol.c | 406 ++++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-stream.c | 393 +++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x.c | 179 +++++++++++++ sound/firewire/digi00x/digi00x.h | 167 ++++++++++++ 15 files changed, 2026 insertions(+), 8 deletions(-) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-protocol.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h -- 2.1.0

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Re: [alsa-devel] [PATCH 00/25] treewide: Use bool function return values of true/false not 1/0
by Joe Perches 31 Mar '15

31 Mar '15

On Mon, 2015-03-30 at 17:07 -0700, Casey Schaufler wrote: > On 3/30/2015 4:45 PM, Joe Perches wrote: > > Joe Perches (25): > > arm: Use bool function return values of true/false not 1/0 [etc...] > Why, and why these in particular? bool functions are probably better returning bool values instead of 1 and 0. Especially when the functions intermix returning returning 1/0 and true/false. (there are only a couple of those though) These are all the remaining instances in the kernel tree.

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Re: [alsa-devel] Realtek ALC662 issues - Intel DG41CN
by Raymond Yau 31 Mar '15

31 Mar '15

> > > Well, I was just trying different modes. The last one was ASUS-MODE1. Before this I tried the "laptop" mode and the "generic" one, both of which did not work. > > What do you suggest? You don't need any model since the pin defaults of your Intel DG41CN are correct unless you want to use CD audio in connector !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x01014010 0x15 0x411111f0 0x16 0x411111f0 0x18 0x01a19840 0x19 0x02a19c50 0x1a 0x0181304f 0x1b 0x02214c20 0x1c 0x593301f0 0x1d 0x4004c601 0x1e 0x01441130 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN http://www.intel.com/support/motherboards/desktop/dg41cn/sb/CS-031343.htm You should check the front audio panel of your computer chassis if jack state of headphone is alway plugged http://www.intel.com/support/motherboards/desktop/sb/cs-015851.htm http://fedoraproject.org/wiki/How_to_debug_PulseAudio_problems You have to provide pulseaudio verbose log Default setting support multi channels but NO headphone playback volume control http://www.intel.com/support/motherboards/desktop/sb/CS-034198.htm You can disable multi channels by using early patching and hint multi_io=0 to get back the headphone playback volume control - multi_io (bool): try probing multi-I/O config (e.g. shared line-in/surround, mic/clfe jacks) https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… >> >> > I have recently installed Fedora 21 Mate on my desktop which runs on >> Intel DG41CN motherboard with Realtek ALC662 Audio. Prior to this I >> was running Linux Mint. It was working all fine then, I may have fixed >> it when I had installed Mint two years ago but I do not remember what >> was done then. I also run Windows 7 on the same machine and there is >> no problem with the audio. >> >> > So the problem I have is that there is no sound from my speakers. A >> few times, there is a little "choppy" sound but thats it. When I put >> the headphones on, the sound is there. >> >> > I must also mention that I also installed Ubuntu Mate on it and faced >> the same problem. >> >> > I have run the Alsa diagnostic tool and the output is uploaded here: >> http://www.alsa-project.org/db/?f=1f6dd98e26f68e890258fa2fe5eb08ee343c5c13 >> >> >I will really appreciate any help. I am stuck bad here.. >> >> Any reason to use asus-mode1 ? >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> Seem asus-mode1 is a broken pin fixup since node 0x21 is not any pin complex >> >> Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono >> >> + [ALC662_FIXUP_ASUS_MODE1] = { >> + .type = ALC_FIXUP_PINS, >> + .v.pins = (const struct alc_pincfg[]) { >> + { 0x14, 0x99130110 }, /* speaker */ >> + { 0x18, 0x01a19c20 }, /* mic */ >> + { 0x19, 0x99a3092f }, /* int-mic */ >> + { 0x21, 0x0121401f }, /* HP out */ >> + { } >> + }, >> + .chained = true, >> + .chain_id = ALC662_FIXUP_SKU_IGNORE >> + }, >> >> Line out is muted when headphone jack is plugged >> >> control.30 { >> iface CARD >> name 'Front Headphone Jack' >> value true >> comment { >> access read >> type BOOLEAN >> count 1 >> } >> } >> >> Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Control: name="Speaker Playback Switch", index=0, device=0 >> ControlAmp: chs=3, dir=Out, idx=0, ofs=0 >> Control: name="Speaker Phantom Jack", index=0, device=0 >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0001003c: IN OUT HP EAPD Detect >> EAPD 0x2: EAPD >> Pin Default 0x01014010: [Jack] Line Out at Ext Rear >> Conn = 1/8, Color = Green >> DefAssociation = 0x1, Sequence = 0x0 >> Pin-ctls: 0x00: >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0c > >

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Re: [alsa-devel] Realtek ALC662 issues - Intel DG41CN
by Raymond Yau 30 Mar '15

30 Mar '15

> I have recently installed Fedora 21 Mate on my desktop which runs on Intel DG41CN motherboard with Realtek ALC662 Audio. Prior to this I was running Linux Mint. It was working all fine then, I may have fixed it when I had installed Mint two years ago but I do not remember what was done then. I also run Windows 7 on the same machine and there is no problem with the audio. > So the problem I have is that there is no sound from my speakers. A few times, there is a little "choppy" sound but thats it. When I put the headphones on, the sound is there. > I must also mention that I also installed Ubuntu Mate on it and faced the same problem. > I have run the Alsa diagnostic tool and the output is uploaded here: http://www.alsa-project.org/db/?f=1f6dd98e26f68e890258fa2fe5eb08ee343c5c13 >I will really appreciate any help. I am stuck bad here.. Any reason to use asus-mode1 ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem asus-mode1 is a broken pin fixup since node 0x21 is not any pin complex Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono + [ALC662_FIXUP_ASUS_MODE1] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x01a19c20 }, /* mic */ + { 0x19, 0x99a3092f }, /* int-mic */ + { 0x21, 0x0121401f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, Line out is muted when headphone jack is plugged control.30 { iface CARD name 'Front Headphone Jack' value true comment { access read type BOOLEAN count 1 } } Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0c

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Re: [alsa-devel] baytrail-pcm-audio: audio broken on Lenovo Thinkpad Tablet 10
by Sébastien Bourdeauducq 30 Mar '15

30 Mar '15

Hi, On 02/23/2015 08:58 PM, Jarkko Nikula wrote: >> Is there a way I can try sound/soc/codecs/rt5670.c instead? I edited it >> so that it attaches to my ACPI ID (and rt5640 doesn't), but that did not >> work. I noticed that the kernel has some Baytrail-specific modules for >> RT564x, and I guess those would need to be changed as well - but it's >> not straightforward. >> > Basically three changes are needed: > > 1. Hack add "10EC5640" to rt5670.c as you have done > 2. Add a new machine driver sound/soc/intel/sst-acpi.c/byt-rt5670 > I guess cht_bsw_rt5672.c can be used as a starting point for audio > routes and byt-rt5640.c for dai link definitions.mess > 3. Hack change sound/soc/intel/sst-acpi.c so that "10EC5640" loads > the new machine driver instead of byt-rt5640. Thanks. I did that and I get the following in the kernel log: [ 6.831422] byt-rt5670 byt-rt5670: ASoC: CPU DAI baytrail-pcm-audio not regis tered [ 6.831443] byt-rt5670 byt-rt5670: snd_soc_register_card failed -517 [ 6.831453] platform byt-rt5670: Driver byt-rt5670 requests probe deferral [ 6.884013] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 6.884023] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 Note: it seems to me that all Baytrail machines will have this "error: invalid DMA engine" message. AFAICT it is emitted from sst-firmware.c because sst->pdata->dma_engine is 0 and in sst-acpi.c the function sst_acpi_probe does not set it for Baytrail because resindex_dma_base is -1 in sst_acpi_baytrail_desc. [ 6.924094] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 6.924106] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 6.924109] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 6.931503] byt-rt5670 byt-rt5670: rt5670-aif1 <-> baytrail-pcm-audio mapping ok [ 6.931624] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Tx [ 6.931629] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Tx -> direct -> AIF1 Playback [ 6.931636] byt-rt5670 byt-rt5670: ASoC: no source widget found for codec_out0 [ 6.931640] byt-rt5670 byt-rt5670: ASoC: Failed to add route codec_out0 -> direct -> ssp2 Tx [ 6.931647] byt-rt5670 byt-rt5670: ASoC: no source widget found for codec_out1 [ 6.931650] byt-rt5670 byt-rt5670: ASoC: Failed to add route codec_out1 -> direct -> ssp2 Tx [ 6.931656] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Rx [ 6.931660] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Rx -> direct -> codec_in0 [ 6.931666] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Rx [ 6.931670] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Rx -> direct -> codec_in1 [ 6.931677] byt-rt5670 byt-rt5670: ASoC: no sink widget found for ssp2 Rx [ 6.931680] byt-rt5670 byt-rt5670: ASoC: Failed to add route AIF1 Capture -> direct -> ssp2 Rx My byt-rt5670 is there: https://gist.github.com/sbourdeauducq/8c5da8ac0da13e21ee31 I have to say that I am unfamiliar with Intel SST and I'm don't quite know what I am doing there. I wish sound devices were straightforward DMA-to-DAC devices instead of this complicated DSP architecture which AFAICT boils down to a glorified memcpy/resampler under Linux (this seems to be a pattern with this SoC though; I had a look at how to get the MIPI cameras to work on this machine and it's an even worse mess). > However, above changes may not make audio working since there has been > couple other new Baytrail based machines with correct codec but audio > seems to not move between the DSP and codec chip. Despite the error messages, the audio device is created and userspace applications can send data into it, but no sound is coming out. I also have to say that the settings in alsamixer are a terrible mess which seems not to originate from my hacks. > One assumption is if they use another SSP port than SSP2. Unfortunately > it's difficult to test since Linux DSP FW is hard-coded for SSP2. If only there were no DSP in the first place... Sébastien

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[alsa-devel] [PATCH 0/5 v2] ASoC: rsnd: DPCM base sampling rate convert support
by Kuninori Morimoto 30 Mar '15

30 Mar '15

Hi Mark These are v2 of remains of my previous patch set. Basically, these doesn't need soc-core's new feature, but it is assuming DPCM base sampling rate convert has both playback/capture in same time. 1 - 2: new Renesas sound card for DPCM 3 : bug fix patch. it can be applied in linus/master branch 4 - 5: new feature, but based on 3) patch Kuninori Morimoto (5): 1) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 2) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 3) ASoC: rsnd: call clk_prepare/unprepare() in probe/remove 4) ASoC: rsnd: remove useless debug message 5) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 - sound/soc/sh/rcar/core.c | 35 ++++++- sound/soc/sh/rcar/dma.c | 4 - sound/soc/sh/rcar/dvc.c | 31 +++--- sound/soc/sh/rcar/gen.c | 6 -- sound/soc/sh/rcar/rsnd.h | 19 +++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 93 +++++++++++------- sound/soc/sh/rcar/ssi.c | 36 ++++--- 12 files changed, 729 insertions(+), 86 deletions(-) _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] Atomic operations not multi-thread aware
by Joël Krähemann 30 Mar '15

30 Mar '15

Hi, please take a a look at gsequencer.org especially at: libags-audio * http://gsequencer.org/api/ags/ags-AgsPort.html gsequencer * http://gsequencer.org/api/ags/ags-AgsLineMember.html I can crash my application while modifying values of the LADSPA plugin assigned dial widget what uses atomic operations and a task engine ... Where can me read about state of Atomic Operations of linux or glibc? Me doing further investigation. bests Joël

2 2

[alsa-devel] [PATCH v2 0/2] ALSA: jack: Refactoring for jack kctls
by Jie Yang 30 Mar '15

30 Mar '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this implement, we are no longer need hda jack specific kctls, so here also remove jack kctls for hda. Any comments are welcome. Jie Yang (2): ALSA: jack: create jack kcontrols for every jack input device ALSA: hda - Remove jack kctls include/sound/jack.h | 8 ++++ sound/core/Kconfig | 3 -- sound/core/Makefile | 3 +- sound/core/jack.c | 99 +++++++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/Kconfig | 10 +---- sound/pci/hda/hda_jack.c | 45 ++-------------------- sound/pci/hda/hda_jack.h | 3 -- 7 files changed, 110 insertions(+), 61 deletions(-) -- 1.9.1

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Re: [alsa-devel] getting the sound to work on my asus m6s00v
by Raymond Yau 28 Mar '15

28 Mar '15

2015-3-28 下午5:52 於 "Christian Gleerup" <cromozon(a)swn.nu> 寫道： > > Trying again with my previous kernel since the new kernel seems broken regarding loading the sound module > > nothin plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = No > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > > Headphone plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = No > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff 0xffffffff = 0x80000000 + 0x7ffffff Try the modified version of hda_jack_sense_test.py which need hda_analyzer 's codec parser and alsa python https://bugzilla.kernel.org/show_bug.cgi?id=66621#c39 it need time for the impedance measurement to complete and become stable, You may need to increase time.sleep(0.02) if you still get 0xffffffff https://bugzilla.kernel.org/show_bug.cgi?id=66621#c41 # the hda_codec.py file comes from Jaroslav Kysela's hda_analyzer program. from hda_codec import * import time def parseoptions(): from optparse import OptionParser parser = OptionParser() parser.add_option("-c", "--card", dest="cardindex", default=0, metavar="CARD", help="card index (as can be seen in /proc/asound/cards)") parser.add_option("-i", "--codec", dest="codecindex", default=0, metavar="CODEC", help="codec index (as can be seen in /proc/asound/cardX/codecY)") (options, args) = parser.parse_args() return int(options.cardindex), int(options.codecindex) cardindex, codecindex = parseoptions() codec = HDACodec(cardindex, codecindex) codec.analyze() for nid in codec.nodes: node = codec.get_node(nid) if WIDGET_TYPE_IDS[node.wtype] == 'PIN': pd = " " impedance = " " if ((node.pincaps & 4) == 4): if (node.pincaps & 2): codec.rw(nid, VERBS['SET_PIN_SENSE'], 0) time.sleep(0.02) sense = codec.rw(nid, VERBS['GET_PIN_SENSE'], 0) if (node.pincaps & 1): impedance = "Imp=%d" % (sense & 0x7fffffff) if (sense & 0x80000000): pd = "PD=1" else: pd = "PD=0" print "Pin 0x%.2x [%s] %s %s at %s (%s) %s : %s %s" % (nid, node.jack_conn_name, node.jack_type_name, node.jack_location_name, node.jack_location2_name, node.jack_color_name, node.defcfg_misc , pd, impedance) > > Mic plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = Yes > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > > > > > >

1 0

[alsa-devel] getting the sound to work on my asus m6s00v
by Christian Gleerup 28 Mar '15

28 Mar '15

I had some problems getting the sound to work on my asus m6s00v (I think that is the model, it is a bit worn so I can't really read it) the symptoms was * on the integrated speakers, no sound at all * on the headphone speakers, sound for a very very short period that faded into nothing. with help from debianuser on freenode#alsa we managed to get it working. the option that i had to load the module with is the following: sudo modprobe snd-hda-intel model=3stack-digout the status now (perfect)... * sound on integrated speaker. * sound in headphones (disables integrated speaker output which is fine) * mic works, both for direct playback and recording * when recording it is the 'digital' channel that controls the volume of the recording level, all others Notice: I don't have OSS or pulseaudio installed. I am thinking it would be really nice if the driver automatically could detect that this option is required. this is the output from the alsa-info script http://www.alsa-project.org/db/?f=28b98a3bd1efc6a80dbd5c54b52c4472d8ea5496 let me know if I can do anything to help with this. Regards.

3 18

[alsa-devel] [PATCH RFT] ASoC: max98925: Fix bit-width 24 settings in max98925_dai_hw_params
by Axel Lin 27 Mar '15

27 Mar '15

Trivial typo fix. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Anish, This looks like a typo, can you confirm if this patch is correct. Thanks. sound/soc/codecs/max98925.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 0796dfa..9b5a17d 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -442,8 +442,8 @@ static int max98925_dai_hw_params(struct snd_pcm_substream *substream, case 24: regmap_update_bits(max98925->regmap, MAX98925_FORMAT, - M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); - max98925->ch_size = 32; + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_24); + max98925->ch_size = 24; break; case 32: regmap_update_bits(max98925->regmap, -- 1.9.1

2 1

Re: [alsa-devel] snd_usb_audio - siberia elite prism soundcard issues
by Raymond Yau 27 Mar '15

27 Mar '15

>> >> > >> >> > >> >> numid=11,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=74,step=0 : values=54,54,54,54,54,54,54,54 | dBminmax-min=-74.00dB,max=0.00dB >> >> >> >> You need workaround since pulseaudio don't like volume control with more than two channels >> >> >> >> https://bugs.freedesktop.org/show_bug.cgi?id=84983#c35 >> >> >> >> Can you post output of your usb audio device ? >> >> >> >> lsusb -vvvv >> > >> > >> > >> > Certainly! >> > >> > Here it is: >> > >> > Bus 001 Device 003: ID 1038:1225 Ideazon, Inc. >> > Device Descriptor: >> > bLength 18 >> > bDescriptorType 1 >> > bcdUSB 2.00 >> > bDeviceClass 0 (Defined at Interface level) >> > bDeviceSubClass 0 >> > bDeviceProtocol 0 >> > bMaxPacketSize0 64 >> > idVendor 0x1038 Ideazon, Inc. >> > idProduct 0x1225 >> > bcdDevice 0.03 >> > iManufacturer 1 SteelSeries >> > iProduct 2 SteelSeries Siberia Elite >> > iSerial 3 000000000000 >> > bNumConfigurations 1 >> >> > >> > AudioControl Interface Descriptor: >> > bLength 13 >> > bDescriptorType 36 >> > bDescriptorSubtype 4 (MIXER_UNIT) >> > Warning: Descriptor too short >> >> https://github.com/gregkh/usbutils/blob/master/lsusb.c >> >> Seem no warning for descriptor too short for uac2 and can dump those uac2 specific variables >> >> > bUnitID 10 >> > bNrInPins 2 >> > baSourceID( 0) 9 >> > baSourceID( 1) 1 >> > bNrChannels 8 >> > wChannelConfig 0x063f >> > Left Front (L) >> > Right Front (R) >> > Center Front (C) >> > Low Freqency Enhancement (LFE) >> > Left Surround (LS) >> > Right Surround (RS) >> > Side Left (SL) >> > Side Right (SR) >> > iChannelNames 0 >> > bmControls 0x00 >> > bmControls 0x00 >> > iMixer 33 (error) >> > > > > Maybe it's the language barrier, but I am not sure if you wanted me to do something. I, out of curiosity, tried to compile that lsusb but I'm missing headers such as config.h. > > Should you need any more feedback from me about this device, please do not hesitate asking through mailing list or directly, just consider I'm not dev! :) > Do you mean there is no sound using alsa hw device since pulseaudio does not support the multichannel speaker playback volume comtrol ? speaker-test -c8 -t wav -D hw:CARD=Elite speaker-test -c2 -t wav -D hw:CARD=Elite You need to know how to compile alsa driver for you running kernel if you want to fix the problem >> >> > > > amixer: Control hw:2 element read error: Invalid argument If alsamixer or amixer cannot set the speaker playback volume Try skip the problematic PCM Volume controls by adding if (state->chip->usb_id == USB_ID(0x1038,0x1225)) if(unitid == 10) ich_has_control = 0; If the driver really need this mixer unit for the playback path, you need to fix those controls bUnitID 10 bNrInPins 2 baSourceID( 0) 9 microphone baSourceID( 1) 1 usb streaming This look like the first four problematic controls related to microphone and the last control related to streaming numid=5,iface=MIXER,name='PCM Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=6,iface=MIXER,name='PCM Volume',index=1 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=7,iface=MIXER,name='PCM Volume',index=2 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=8,iface=MIXER,name='PCM Volume',index=3 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=9,iface=MIXER,name='PCM Volume',index=4 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument

2 1

[alsa-devel] [PATCH 1/4] ASoC: rt5645: Redefine format config for rt5650
by Bard Liao 27 Mar '15

27 Mar '15

rt5650 and rt5645 use different register bits for format configuration. This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, I am think if we should use different codec driver for rt5645 and rt5650. In fact, they are different Die. What do you think? --- --- sound/soc/codecs/rt5645.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c384a1..0133c8c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2049,7 +2049,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk; + unsigned int val_len = 0, val_clk, mask_clk, dl_sft; int pre_div, bclk_ms, frame_size; rt5645->lrck[dai->id] = params_rate(params); @@ -2063,6 +2063,16 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); return -EINVAL; } + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + dl_sft = 4; + break; + default: + dl_sft = 2; + break; + } + bclk_ms = frame_size > 32; rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); @@ -2075,13 +2085,13 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - val_len |= RT5645_I2S_DL_20; + val_len = 0x1; break; case 24: - val_len |= RT5645_I2S_DL_24; + val_len = 0x2; break; case 8: - val_len |= RT5645_I2S_DL_8; + val_len = 0x3; break; default: return -EINVAL; @@ -2093,7 +2103,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; case RT5645_AIF2: @@ -2101,7 +2111,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; default: @@ -2116,7 +2126,16 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0; + unsigned int reg_val = 0, pol_sft; + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + pol_sft = 8; + break; + default: + pol_sft = 7; + break; + } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2134,7 +2153,7 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_NF: - reg_val |= RT5645_I2S_BP_INV; + reg_val |= (1 << pol_sft); break; default: return -EINVAL; @@ -2158,12 +2177,12 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (dai->id) { case RT5645_AIF1: snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; default: -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/4] ASoC: rt5645: Redefine format config for rt5650
by Bard Liao 27 Mar '15

27 Mar '15

rt5650 and rt5645 use different register bits for format configuration. This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, I am think if we should use different codec driver for rt5645 and rt5650. In fact, they are different Die. What do you think? --- --- sound/soc/codecs/rt5645.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c384a1..0133c8c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2049,7 +2049,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk; + unsigned int val_len = 0, val_clk, mask_clk, dl_sft; int pre_div, bclk_ms, frame_size; rt5645->lrck[dai->id] = params_rate(params); @@ -2063,6 +2063,16 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); return -EINVAL; } + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + dl_sft = 4; + break; + default: + dl_sft = 2; + break; + } + bclk_ms = frame_size > 32; rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); @@ -2075,13 +2085,13 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - val_len |= RT5645_I2S_DL_20; + val_len = 0x1; break; case 24: - val_len |= RT5645_I2S_DL_24; + val_len = 0x2; break; case 8: - val_len |= RT5645_I2S_DL_8; + val_len = 0x3; break; default: return -EINVAL; @@ -2093,7 +2103,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; case RT5645_AIF2: @@ -2101,7 +2111,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; default: @@ -2116,7 +2126,16 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0; + unsigned int reg_val = 0, pol_sft; + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + pol_sft = 8; + break; + default: + pol_sft = 7; + break; + } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2134,7 +2153,7 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_NF: - reg_val |= RT5645_I2S_BP_INV; + reg_val |= (1 << pol_sft); break; default: return -EINVAL; @@ -2158,12 +2177,12 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (dai->id) { case RT5645_AIF1: snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; default: -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH v4 0/15] Those patches is used for dw_hdmi audio support
by Yakir Yang 27 Mar '15

27 Mar '15

We found Designware hdmi driver only support audio clock config, we can not play sound through it. To add Designware HDMI Audio support, we make those patch set: 1): fixed dw_hdmi irq bug, add irq control to suspend/resume interfaces. 2): add suspend/resume callback for dw_hdmi rockchip driver. 3): Warp irq control in functions. 4): Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() 5): Adjust n/cts config order. 6): Set ncts_atomic_write & cts_manual, according to dw_hdmi document. 7): Add audio support for more display resolutions(eg. 800x600). 8): Add audio support for No-CEA display resolutions. 9): Add Audio Sample Channel Status config interfaces to dw_hdmi driver. 10): add audio clock control interfaces to dw_hdmi driver. 11): creat "dw_hdmi-audio" platform device in dw_hdmi driver. 12): add codec driver for hdmi audio, callback dw_hdmi audio config functions. 13): add sound driver for hdmi audio, creat hdmi audio sound card. 14): add dt-bings file and add hdmi_audio node to corresponding dt file. Changes in v4: -Correct phy_type assignment bug - Combine CTS3 registers setting, reduce register operate times - Combine N3 registers setting - Add hdmi audio support when monitor support audio - Give HDMI_FC_AUD_SCHNL8 an readable value - Rename "hdmi_audio_*" to "dw_hdmi_audio_*" - Replace delaywork with irq thread, and add suspend/resume interfaces, Replace "dw-hdmi-audio" with consecutive strings. - Add ".pm = &snd_soc_pm_ops," Changes in v3: - Clear Hotplug interrupts before dw_hdmi_fb_register - Wrap irq control in fucntions - Setting the .pm member instead of suspend/resume - Add ID registers parse and record - Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() - Only adjust the n/cts setting order - Set ncts_atomic_write & cts_manual - Determine whether sample channel should set by desig_id. - Delete hdmi_audio_config interface and modify audio clock control functions - Remove audio_config & get_connect_status callback functions and add write/read/mod register callback functions - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Add irq control to suspend/resume interfaces - Add suspend/resume support for dw_hdmi_rockchip driver - add more n/cts combinations for more display resolutions - Enable audio support for No-CEA display mode - Add audio sample channel status setting - Add audio config interfaces to dw_hdmi driver - Update the audio control interfaces - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Daniel Kurtz (3): drm: bridge/dw_hdmi: adjust n/cts setting order drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual drm: bridge/dw_hdmi: add audio sample channel status setting Yakir Yang (12): drm: bridge/dw_hdmi: add irq control to suspend/resume drm: bridge/dw_hdmi: wrap irq control in fucntions drm: rockchip/dw_hdmi_rockchip: add resume/suspend support drm: bridge/dw_hdmi: add identification registers parse and record drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio support for No-CEA display resolutions drm: bridge/dw_hdmi: add enable/disable to dw_hdmi_audio callbacks drm: bridge/dw_hdmi: creat dw-hdmi-audio platform device ASoC: codec/dw-hdmi-audio: add codec driver for dw hdmi audio ASoC: rockchip/rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for Rockchip dw-hdmi-audio .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/dw_hdmi.c | 385 ++++++++++++++++++--- drivers/gpu/drm/bridge/dw_hdmi.h | 53 ++- drivers/gpu/drm/rockchip/dw_hdmi-rockchip.c | 16 + include/drm/bridge/dw_hdmi.h | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 379 ++++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 74 ++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 169 +++++++++ 12 files changed, 1076 insertions(+), 46 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

4 11

[alsa-devel] [PATCH] ASoC: Add support for NAU8824 codec to ASoC
by Wan Zongshun 27 Mar '15

27 Mar '15

Signed-off-by: Wan Zongshun <mcuos.com(a)gmail.com> Signed-off-by: Chih-Chiang Chang <ccchang12(a)nuvoton.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8824.c | 770 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8824.h | 379 ++++++++++++++++++++++ 4 files changed, 1156 insertions(+) create mode 100644 sound/soc/codecs/nau8824.c create mode 100644 sound/soc/codecs/nau8824.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c1..862b7bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8824 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -463,6 +464,10 @@ config SND_SOC_MAX98357A config SND_SOC_MAX9850 tristate +config SND_SOC_NAU8824 + tristate "Nuvoton NAU8824 CODEC" + depends on I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666..acb7bda 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-max98357a-objs := max98357a.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8824-objs := nau8824.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -250,6 +251,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c new file mode 100644 index 0000000..80c2283 --- /dev/null +++ b/sound/soc/codecs/nau8824.c @@ -0,0 +1,770 @@ +/* + * linux/sound/soc/codecs/nau8824.c + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/acpi.h> +#include <asm/div64.h> +#include <sound/jack.h> +#include <sound/core.h> +#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> +#include <linux/delay.h> + +#include "nau8824.h" + +static int nau8824_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +static int nau8824_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt); +static int nau8824_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); +static int nau8824_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); +static const DECLARE_TLV_DB_SCALE(out_spk_vol_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(out_hp_vol_tlv, -3000, 100, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -12800, 50, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -12800, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_left_vol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_right_vol_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new nau8824_snd_controls[] = { + /* SP Class-D mute control */ + SOC_DOUBLE("SP Playback Switch", NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_SFT, NAU8824_CLASS_D_SFT, 1, 0), + /* SP Class-D driver output stage gain control */ + SOC_SINGLE_TLV("SP Left Volume", NAU8824_CLASS_D_GAIN2, + NAU8824_CLASS_D_GAIN_L_SFT, + NAU8824_SPK_VOL_RSCL_RANGE, 0, out_spk_vol_tlv), + SOC_SINGLE_TLV("SP Right Volume", NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_GAIN_R_SFT, + NAU8824_SPK_VOL_RSCL_RANGE, 0, + out_spk_vol_tlv), + /* SP Class-D mute control */ + SOC_DOUBLE("HP Playback Switch", NAU8824_HP_MUTE, NAU8824_L_MUTE_SFT, + NAU8824_R_MUTE_SFT, 1, 1), + SOC_SINGLE_TLV("HP Left Volume", NAU8824_HP_VOL, NAU8824_L_VOL_SFT, + NAU8824_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), + SOC_SINGLE_TLV("HP Right Volume", NAU8824_HP_VOL, NAU8824_R_VOL_SFT, + NAU8824_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), + /* DMIC control */ + SOC_DOUBLE("DMIC L R Switch", NAU8824_ENA_CTRL, NAU8824_DMIC_CH0_SFT, + NAU8824_DMIC_CH1_SFT, 1, 0), + SOC_SINGLE("DMIC Enable", NAU8824_BIAS_ADJ, NAU8824_DMIC1_SFT, 1, 0), + SOC_SINGLE("VMID Switch", NAU8824_BIAS_ADJ, NAU8824_VMID_SFT, 1, 0), + SOC_SINGLE("BIAS Switch", NAU8824_BOOST, NAU8824_G_BIAS_SFT, 1, 0), + SOC_DOUBLE_R_TLV("MIC L R Gain", NAU8824_ADC_CH0_DGAIN_CTRL, + NAU8824_ADC_CH1_DGAIN_CTRL, 0, + NAU8824_ADC_VOL_RSCL_RANGE, 0, adc_vol_tlv), +}; + +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, NAU8824_CLK_DIVIDER, + NAU8824_CLK_DMIC_SRC_MASK, + NAU8824_CLK_DMIC_DIV_4); + break; + default: + break; + } + return 0; +} + +static const struct snd_kcontrol_new nau8824_rec_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_rec_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_spo_mix[] = { + SOC_DAPM_SINGLE("SP L Switch", SND_SOC_NOPM, + 0, 0, 0), + SOC_DAPM_SINGLE("SP R Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_hpo_mix[] = { + SOC_DAPM_SINGLE("HP L Switch", + NAU8824_HPO_MIXER, NAU8824_M_HPVOL_L_HM_SFT, + 1, 0), + SOC_DAPM_SINGLE("HP R Switch", + NAU8824_HPO_MIXER, NAU8824_M_HPVOL_R_HM_SFT, + 1, 0), +}; + +static const char * const nau8824_stereo_adc1_src[] = { "ADC", "DMIC" }; + +static const SOC_ENUM_SINGLE_DECL(nau8824_stereo_adc_l1_enum, NAU8824_ENA_CTRL, + NAU8824_DMIC_CH0_SFT, nau8824_stereo_adc1_src); + +static const SOC_ENUM_SINGLE_DECL(nau8824_stereo_adc_r1_enum, NAU8824_ENA_CTRL, + NAU8824_DMIC_CH1_SFT, nau8824_stereo_adc1_src); + +static const struct snd_kcontrol_new nau8824_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo ADC L1 source", + nau8824_stereo_adc_l1_enum); + +static const struct snd_kcontrol_new nau8824_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo ADC R1 source", + nau8824_stereo_adc_r1_enum); + +static int nau8824_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, NAU8824_ANALOG_CTRL2, + NAU8824_CLASS_D_CLAMP_MSK, + NAU8824_CLASS_D_CLAMP_BETTER); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_MASK, NAU8824_CLASS_D_DIS); + default: + break; + } + return 0; +} + +static int nau8824_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return 0; +} + +static int nau8824_dac1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return 0; +} + +static int micbias_power_on_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + snd_soc_update_bits(codec, NAU8824_BOOST, NAU8824_G_BIAS_MASK, + NAU8824_G_BIAS_EN); + else if (SND_SOC_DAPM_EVENT_OFF(event)) + snd_soc_update_bits(codec, NAU8824_BOOST, NAU8824_G_BIAS_MASK, + NAU8824_G_BIAS_DIS); + return 0; +} + +static int vmid_power_on_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_EN); + else if (SND_SOC_DAPM_EVENT_OFF(event)) + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_DIS); + return 0; +} + +static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_SUPPLY("micbias", SND_SOC_NOPM, 0, 0, + micbias_power_on_event, SND_SOC_DAPM_PRE_PMU + | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("vmid", SND_SOC_NOPM, 0, 0, vmid_power_on_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA("DMIC L1", NAU8824_ENA_CTRL, NAU8824_DMIC_CH0_SFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R1", NAU8824_ENA_CTRL, NAU8824_DMIC_CH1_SFT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, set_dmic_clk, + SND_SOC_DAPM_PRE_PMU), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", SND_SOC_NOPM, 0, 0, nau8824_rec_l_mix, + ARRAY_SIZE(nau8824_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", SND_SOC_NOPM, 0, 0, nau8824_rec_r_mix, + ARRAY_SIZE(nau8824_rec_r_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &nau8824_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &nau8824_sto_adc_r1_mux), + /* ADC IF1 */ + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC L1", NULL, NAU8824_DAC_CTRL, + NAU8824_DAC_L_SFT, 0, nau8824_dac1_event, + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC R1", NULL, NAU8824_DAC_CTRL, + NAU8824_DAC_R_SFT, 0, nau8824_dac1_event, + SND_SOC_DAPM_PRE_PMD), + + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPO MIX", SND_SOC_NOPM, 0, 0, nau8824_spo_mix, + ARRAY_SIZE(nau8824_spo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, nau8824_hpo_mix, + ARRAY_SIZE(nau8824_hpo_mix)), + + SND_SOC_DAPM_PGA_S("HP amp", 1, NAU8824_CLASS_G_CTRL, + NAU8824_CLASS_G_SHIFT, 0, nau8824_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 1, NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_SFT, 0, nau8824_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8824_dapm_routes[] = { + + {"DMIC L1", NULL, "DMIC1"}, + {"DMIC R1", NULL, "DMIC1"}, + + {"BST1", NULL, "IN1P"}, + {"BST1", NULL, "IN1N"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + {"IN2P", NULL, "micbias"}, + {"IN2P", NULL, "vmid"}, + + {"RECMIXL", "BST1 Switch", "BST1"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC R", NULL, "RECMIXR"}, + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC CLK"}, + + {"Stereo ADC L1 Mux", "ADC", "ADC L"}, + {"Stereo ADC L1 Mux", "DMIC", "DMIC L1"}, + {"Stereo ADC R1 Mux", "ADC", "ADC R"}, + {"Stereo ADC R1 Mux", "DMIC", "DMIC R1"}, + {"AIF1TX", NULL, "IF1 ADC"}, + + {"DAC L1", NULL, "AIF1RX"}, + {"DAC R1", NULL, "AIF1RX"}, + {"SPO MIX", "SP L Switch", "DAC L1"}, + {"SPO MIX", "SP R Switch", "DAC R1"}, + {"SPK amp", NULL, "SPO MIX"}, + {"SPK", NULL, "SPK amp"}, + + {"HPO MIX", "HP L Switch", "DAC L1"}, + {"HPO MIX", "HP R Switch", "DAC R1"}, + {"HP amp", NULL, "HPO MIX"}, + {"HPOL", NULL, "HP amp"}, + {"HPOR", NULL, "HP amp"}, +}; + +static int nau8824_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int val_len = 0; + + dev_dbg(codec->dev, "%s\n", __func__); + dev_dbg(codec->dev, "##- Data length: %d\n", params_format(params)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8824_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8824_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8824_I2S_DL_32; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_1, NAU8824_I2S_DL_MASK, + val_len); + return 0; +} + +static int nau8824_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) + snd_soc_update_bits(codec, NAU8824_DAC_MUTE_CTRL, + NAU8824_SOFT_MUTE_MASK, NAU8824_SOFT_MUTE_EN); + else + snd_soc_update_bits(codec, NAU8824_DAC_MUTE_CTRL, + NAU8824_SOFT_MUTE_MASK, NAU8824_SOFT_MUTE_DIS); + return 0; +} + +static int nau8824_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg_val = 0, reg_val_2 = 0; + + dev_dbg(codec->dev, "%s: Entered\n", __func__); + dev_dbg(codec->dev, "###nau8824_set_dai_fmt %x\n", fmt); + dev_dbg(codec->dev, "##+ nau8824_set_dai_fmt (%x)\n", fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg_val_2 |= NAU8824_I2S_MS_M; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + break; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= NAU8824_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg_val |= NAU8824_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= NAU8824_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + reg_val |= NAU8824_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= NAU8824_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= NAU8824_I2S_DF_PCM_B; + reg_val |= NAU8824_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_1, + NAU8824_I2S_DL_MASK | NAU8824_I2S_DF_MASK + | NAU8824_I2S_BP_MASK | NAU8824_I2S_PCMB_MASK, + reg_val); + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_2, NAU8824_I2S_MS_MASK, + reg_val_2); + + dev_dbg(codec->dev, "##-nau8824_set_dai_fmt Master\n"); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +static int nau8824_FLL_freerun_mode(struct snd_soc_codec *codec, int IsEnable) +{ + if (IsEnable == true) { + snd_soc_write(codec, 0x03, 0x0853); + snd_soc_write(codec, 0x09, 0xE000); + snd_soc_write(codec, 0x03, 0x8853); + } else { + snd_soc_write(codec, 0x03, 0x0853); + snd_soc_write(codec, 0x09, 0x6000); + snd_soc_write(codec, 0x03, 0x8853); + } + return 0; +} + +void set_sys_clk(struct snd_soc_codec *codec, int sys_clk) +{ + pr_debug("%s sys_clk=%x\n", __func__, sys_clk); + switch (sys_clk) { + case NAU8824_INTERNALCLOCK: + nau8824_FLL_freerun_mode(codec, true); + break; + + case NAU8824_MCLK: + default: + nau8824_FLL_freerun_mode(codec, false); + break; + } +} + +static int nau8824_set_dai_pll(struct snd_soc_dai *dai, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "In nau8824: dai_set_pll\n"); + return 0; +} + +static int nau8824_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + int divider = 1; + + if (clk_id == NAU8824_MCLK) { + set_sys_clk(codec, NAU8824_MCLK); + dev_dbg(codec->dev, "%s: input freq = %d divider = %d", + __func__, freq, divider); + + } else if (clk_id == NAU8824_INTERNALCLOCK) { + set_sys_clk(codec, NAU8824_INTERNALCLOCK); + } else { + dev_err(codec->dev, "Wrong clock src\n"); + return -EINVAL; + } + return 0; +} + +static int nau8824_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## nau8824_set_bias_level %d\n", level); + if (level == codec->dapm.bias_level) { + dev_dbg(codec->dev, "##set_bias_level: level returning...\r\n"); + return -EINVAL; + } + + switch (level) { + case SND_SOC_BIAS_ON: + /* All power is driven by DAPM system*/ + dev_dbg(codec->dev, "###nau8824_set_bias_level BIAS_ON\n"); + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_EN); + snd_soc_update_bits(codec, NAU8824_BOOST, + NAU8824_G_BIAS_MASK, NAU8824_G_BIAS_EN); + break; + + case SND_SOC_BIAS_PREPARE: + dev_dbg(codec->dev, "###nau8824_set_bias_level BIAS_PREPARE\n"); + break; + + case SND_SOC_BIAS_STANDBY: + dev_dbg(codec->dev, "###nau8824_set_bias_level STANDBY\n"); + break; + + case SND_SOC_BIAS_OFF: + dev_dbg(codec->dev, "###nau8824_set_bias_level OFF\n"); + set_sys_clk(codec, NAU8824_INTERNALCLOCK); + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, NAU8824_VMID_MASK, + NAU8824_VMID_DIS); + snd_soc_update_bits(codec, NAU8824_BOOST, + NAU8824_G_BIAS_MASK, NAU8824_G_BIAS_DIS); + break; + default: + break; + } + codec->dapm.bias_level = level; + dev_dbg(codec->dev, "## nau8824_set_bias_level %d\n", level); + + return 0; +} + +static int nau8824_suspend(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s: Entered\n", __func__); + nau8824_set_bias_level(codec, SND_SOC_BIAS_OFF); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +static int nau8824_resume(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s: Entered\n", __func__); + nau8824_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +struct nau8824_init_reg { + u8 reg; + u16 val; +}; + +static struct nau8824_init_reg init_list[] = { + {0x0000, 0x0001}, + {0x0000, 0x0000}, + {0x0066, 0x0060}, + {0x0076, 0x3000}, + {0x0004, 0x0002}, + {0x0005, 0xDD44}, + {0x0006, 0x0007}, + {0x0007, 0x0810}, + {0x0008, 0xC000}, + {0x0074, 0x1106}, + {0x0073, 0x330C}, + {0x0080, 0x0000}, + {0x006D, 0x0100}, + {0x006E, 0x0001}, + {0x006A, 0x0008}, + {0x006D, 0x0180}, + {0x006D, 0x1480}, + {0x006E, 0x0014}, + {0x0050, 0x2007}, + {0x0080, 0x3020}, + {0x006B, 0xC000}, + {0x007B, 0x1E1E}, + {0x006B, 0xC005}, + {0x0068, 0x8000}, + {0x0076, 0x300F}, + {0x007F, 0x000F}, + {0x0080, 0x0020}, + {0x006B, 0x0005}, + {0x007B, 0x0000}, + {0x0002, 0x0800}, + {0x0025, 0x1000}, + {0x0024, 0x0002}, + {0x0002, 0x0AA5}, + {0x0020, 0x0000}, + {0x0025, 0x1082}, + {0x0031, 0x0800}, + {0x0032, 0x0100}, + {0x0033, 0x0300}, + {0x0034, 0x0000}, + {0x001C, 0x000A}, + {0x001F, 0x2000}, + {0x0001, 0x0030}, + {0x0002, 0x3AA5}, + {0x0067, 0x4040}, + {0x007F, 0x300F}, + {0x0077, 0x0010}, + {0x0078, 0x3C00}, + {0x0076, 0x380F}, + {0x0076, 0x300F}, + {0x007A, 0x0202}, + {0x0067, 0x0000}, + {0x0072, 0x40A0}, + {0x0079, 0x0300}, + {0x0023, 0x1010}, + {0x0024, 0x0002}, + {0x0038, 0x1486}, + {0x003C, 0x1486}, + {0x0066, 0x0064}, + {0x0001, 0x40F0}, + {0x0001, 0x48F0}, + {0x0025, 0x3082}, + {0x0020, 0x0003}, + {0x0024, 0x8102}, + {0x002D, 0x0108}, + {0x002E, 0x0308}, + {0x002F, 0x0500}, + {0x0030, 0x0700}, + {0x001C, 0x000A}, + {0x001F, 0x2000}, + {0x0002, 0x3BA5}, + {0x0001, 0x48F3}, + {0x0002, 0xFBA5}, + {0x0014, 0x2210}, + {0x007c, 0x1e1e}, +}; + +#define NAU8824_INIT_REG_LEN ARRAY_SIZE(init_list) +static int nau8824_reg_init(struct snd_soc_codec *codec) +{ + int i; + + mdelay(1); + for (i = 0; i < NAU8824_INIT_REG_LEN; i++) { + if (init_list[i].reg == 0xFFFF) + mdelay(1); + else + snd_soc_write(codec, init_list[i].reg, + init_list[i].val); + } + return 0; +} + +static int nau8824_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct nau8824_priv *nau8824; + + nau8824 = snd_soc_codec_get_drvdata(codec); + codec->control_data = nau8824->regmap; + mutex_init(&nau8824->mutex); + nau8824_reg_init(codec); + + /* Dynamic Headset detection enabled */ + snd_soc_update_bits(codec, 0x01, 0x400, 0x400); + snd_soc_update_bits(codec, 0x02, 0x0008, 0x0008); + snd_soc_update_bits(codec, 0x0f, 0x0300, 0x0100); + snd_soc_write(codec, 0x09, 0xE000); + snd_soc_write(codec, NAU8824_IRQ_SETTING, 0x1006); + snd_soc_write(codec, 0x13, 0x1615); + snd_soc_write(codec, 0x15, 0x0414); + snd_soc_update_bits(codec, 0x16, 0xFF00, 0x5900); + snd_soc_update_bits(codec, 0x66, 0x0070, 0x0060); + + /* Program codec to use internal clock */ + nau8824_FLL_freerun_mode(codec, true); + return ret; +} + +static int nau8824_codec_remove(struct snd_soc_codec *codec) +{ + nau8824_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_nau8824 = { + .probe = nau8824_codec_probe, + .remove = nau8824_codec_remove, + .suspend = nau8824_suspend, + .resume = nau8824_resume, + .set_bias_level = nau8824_set_bias_level, + .controls = nau8824_snd_controls, + .num_controls = ARRAY_SIZE(nau8824_snd_controls), + .dapm_widgets = nau8824_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8824_dapm_widgets), + .dapm_routes = nau8824_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8824_dapm_routes), + .reg_word_size = sizeof(u16), + .reg_cache_default = init_list, + .set_sysclk = nau8824_set_sysclk, +}; + +static struct snd_soc_dai_ops nau8824_dai_ops = { + .hw_params = nau8824_hw_params, + .set_pll = nau8824_set_dai_pll, + .set_fmt = nau8824_set_dai_fmt, + .digital_mute = nau8824_dac_mute, +}; + +static struct snd_soc_dai_driver nau8824_dai_driver[] = { +{ + .name = "nau8824-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .ops = &nau8824_dai_ops, +} +}; + +static int nau8824_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct nau8824_priv *nau8824; + int ret; + + pr_debug("%s enter", __func__); + nau8824 = kzalloc(sizeof(*nau8824), GFP_KERNEL); + if (nau8824 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8824); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_nau8824, + nau8824_dai_driver, + ARRAY_SIZE(nau8824_dai_driver)); + if (ret < 0) + kfree(nau8824); + + return ret; +} + +static int nau8824_i2c_remove(struct i2c_client *i2c) +{ + struct nau8824_priv *nau8824 = dev_get_drvdata(&i2c->dev); + + snd_soc_unregister_codec(nau8824->dev); + + return 0; +} + +static const struct i2c_device_id nau8824_i2c_id[] = { + { "nau8824", 0}, + {"10508824:00", 0}, + {"10508824", 0}, + { } +}; + +MODULE_DEVICE_TABLE(i2c, nau8824_i2c_id); +static struct i2c_driver nau8824_i2c_driver = { + .driver = { + .name = "nau8824", + .owner = THIS_MODULE, + }, + .probe = nau8824_i2c_probe, + .remove = (nau8824_i2c_remove), + .id_table = nau8824_i2c_id, +}; +module_i2c_driver(nau8824_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8824 codec driver"); +MODULE_AUTHOR("Nuvoton"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h new file mode 100644 index 0000000..1745dac --- /dev/null +++ b/sound/soc/codecs/nau8824.h @@ -0,0 +1,379 @@ +/* + * linux/sound/soc/codecs/nau8824.h + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _NAU8824_H +#define _NAU8824_H + +#define NAU8824_RESET 0x00 +#define NAU8824_ENA_CTRL 0x01 +#define NAU8824_CLK_EN 0x02 +#define NAU8824_CLK_DIVIDER 0x03 +#define NAU8824_FL_1 0x04 +#define NAU8824_FL_2 0x05 +#define NAU8824_FL_3 0x06 +#define NAU8824_FL_4 0x07 +#define NAU8824_FL_5 0x08 +#define NAU8824_FL_6 0x09 +#define NAU8824_IRQ_STATUS 0x10 +#define NAU8824_IRQ_CLEAR 0x11 +#define NAU8824_IRQ_SETTING 0x12 +#define NAU8824_SAR_ADC 0x13 +#define NAU8824_VDET_COEFFICIENT 0x14 +#define NAU8824_I2S_PCM_CTRL_1 0x1C +#define NAU8824_I2S_PCM_CTRL_2 0x1D +#define NAU8824_ADC_CH0_DGAIN_CTRL 0x2D +#define NAU8824_ADC_CH1_DGAIN_CTRL 0x2E +#define NAU8824_DAC_MUTE_CTRL 0x31 +#define NAU8824_DAC0_DIG_VOL 0x32 +#define NAU8824_DAC1_DIG_VOL 0x33 +#define NAU8824_CLASS_G_CTRL 0x50 +#define NAU8824_SAR_ADC_OUTPUT 0x59 +#define NAU8824_BIAS_ADJ 0x66 +#define NAU8824_ANALOG_CTRL2 0x6A +#define NAU8824_HPO_MIXER 0x6B +#define NAU8824_CLASS_D_GAIN1 0x6D +#define NAU8824_CLASS_D_GAIN2 0x6E +#define NAU8824_DAC_CTRL 0x73 +#define NAU8824_MIC_BIAS 0x74 +#define NAU8824_HP_MUTE 0x75 +#define NAU8824_BOOST 0x76 +#define NAU8824_HP_VOL 0x7B + +/* reg. NAU8824_ENA_CTRL (0x01) */ +#define NAU8824_DMIC_EDGE_CH23_MASK (0x01 << 15) +#define NAU8824_DMIC_EDGE_CH23_SFT 15 +#define NAU8824_DMIC_EDGE_CH01_MASK (0x01 << 14) +#define NAU8824_DMIC_EDGE_CH01_SFT 14 +#define NAU8824_ADC_OP_EN_MASK (0x01 << 11) +#define NAU8824_ADC_OP_EN_SFT 11 +#define NAU8824_ADC_OP_EN (0x01 << 11) +#define NAU8824_ADC_OP_DIS (0x00 << 11) +#define NAU8824_INT_SLEEP_MASK (0x01 << 10) +#define NAU8824_INT_SLEEP_SFT 10 +#define NAU8824_INT_SLEEP_ON (0x01 << 10) +#define NAU8824_INT_SLEEP_OFF (0x00 << 10) +#define NAU8824_DMIC_CH3_MASK (0x01 << 9) +#define NAU8824_DMIC_CH3_SFT 9 +#define NAU8824_DMIC_CH3_ON (0x01 << 9) +#define NAU8824_DMIC_CH3_OFF (0x00 << 9) +#define NAU8824_DMIC_CH2_MASK (0x01 << 8) +#define NAU8824_DMIC_CH2_SFT 8 +#define NAU8824_DMIC_CH2_ON (0x01 << 8) +#define NAU8824_DMIC_CH2_OFF (0x00 << 8) +#define NAU8824_DMIC_CH1_MASK (0x01 << 7) +#define NAU8824_DMIC_CH1_SFT 7 +#define NAU8824_DMIC_CH1_ON (0x01 << 7) +#define NAU8824_DMIC_CH1_OFF (0x00 << 7) +#define NAU8824_DMIC_CH0_MASK (0x01 << 6) +#define NAU8824_DMIC_CH0_SFT 6 +#define NAU8824_DMIC_CH0_ON (0x01 << 6) +#define NAU8824_DMIC_CH0_OFF (0x00 << 6) +#define NAU8824_DAC_CH1_EN_MASK (0x01 << 5) +#define NAU8824_DAC_CH1_EN_SFT 5 +#define NAU8824_DAC_CH1_EN (0x01 << 5) +#define NAU8824_DAC_CH1_DIS (0x00 << 5) +#define NAU8824_DAC_CH0_EN_MASK (0x01 << 4) +#define NAU8824_DAC_CH0_EN_SFT 4 +#define NAU8824_DAC_CH0_EN (0x01 << 4) +#define NAU8824_DAC_CH0_DIS (0x00 << 4) +#define NAU8824_ADC_CH3_EN_MASK (0x01 << 3) +#define NAU8824_ADC_CH3_EN_SFT 3 +#define NAU8824_ADC_CH3_EN (0x01 << 3) +#define NAU8824_ADC_CH3_DIS (0x00 << 3) +#define NAU8824_ADC_CH2_EN_MASK (0x01 << 2) +#define NAU8824_ADC_CH2_EN_SFT 2 +#define NAU8824_ADC_CH2_EN (0x01 << 2) +#define NAU8824_ADC_CH2_DIS (0x00 << 2) +#define NAU8824_ADC_CH1_EN_MASK (0x01 << 1) +#define NAU8824_ADC_CH1_EN_SFT 1 +#define NAU8824_ADC_CH1_EN (0x01 << 1) +#define NAU8824_ADC_CH1_DIS (0x00 << 1) +#define NAU8824_ADC_CH0_EN_MASK (0x01 << 0) +#define NAU8824_ADC_CH0_EN_SFT 0 +#define NAU8824_ADC_CH0_EN (0x01 << 0) +#define NAU8824_ADC_CH0_DIS (0x00 << 0) + +/* reg. NAU8824_CLK_EN (0x02) */ +#define NAU8824_CLK_SAR_EN_MASK (0x1 << 3) +#define NAU8824_CLK_SAR_EN_SFT 3 +#define NAU8824_CLK_SAR_EN (0x1 << 3) +#define NAU8824_CLK_SAR_DIS (0x0 << 3) + +/* reg. NAU8824_CLK_DIVIDER (0x03) */ +#define NAU8824_SYSCLK_SRC_MASK (0x1 << 15) +#define NAU8824_SYSCLK_SRC_SFT 15 +#define NAU8824_SYSCLK_SRC_VCO (0x1 << 15) +#define NAU8824_SYSCLK_SRC_MCLK (0x0 << 15) +#define NAU8824_CLK_CODEC_SRC_MASK (0x1 << 13) +#define NAU8824_CLK_CODEC_SRC_SFT 13 +#define NAU8824_CODEC_SRC_SYSCLK (0x1 << 13) +#define NAU8824_CODEC_SRC_MCLK (0x0 << 13) +#define NAU8824_CLK_DMIC_SRC_MASK (0x7 << 10) +#define NAU8824_CLK_DMIC_SRC_SFT 10 +#define NAU8824_CLK_DMIC_DIV_4 (0x2 << 10) + +/* reg. NAU8824_IRQ_STATUS (0x10) */ +#define NAU8824_IRQ_KEY_RELEASE (0x1 << 5) +#define NAU8824_IRQ_LONGKEY (0x1 << 4) +#define NAU8824_IRQ_SHORTKEY (0x1 << 3) +#define NAU8824_IRQ_MIC_INSERTED (0x1 << 2) +#define NAU8824_IRQ_JACK_EJECTED (0x1 << 1) +#define NAU8824_IRQ_JACK_INSERTED (0x1 << 0) + +/* reg. NAU8824_IRQ_CLEAR (0x11) */ +#define NAU8824_IRQ_KEY_RELEASE_CLR (0x1 << 5) +#define NAU8824_IRQ_LONGKEY_CLR (0x1 << 4) +#define NAU8824_IRQ_SHORTKEY_CLR (0x1 << 3) +#define NAU8824_IRQ_MIC_INSERTED_CLR (0x1 << 2) +#define NAU8824_IRQ_JACK_EJECTED_CLR (0x1 << 1) +#define NAU8824_IRQ_JACK_INSERTED_CLR (0x1 << 0) + +/* reg NAU8824_SAR_ADC (0x13) */ +#define NAU8824_SAR_EN_MASK (0x1 << 12) +#define NAU8824_SAR_EN_SFT 12 +#define NAU8824_SAR_EN (0x1 << 12) +#define NAU8824_SAR_DIS (0x0 << 12) + +/* reg. NAU8824_I2S_PCM_CTRL_1 (0x1C) */ +#define NAU8824_I2S_BP_MASK (0x1 << 7) +#define NAU8824_I2S_BP_SFT 7 +#define NAU8824_I2S_BP_NOR (0x0 << 7) +#define NAU8824_I2S_BP_INV (0x1 << 7) +#define NAU8824_I2S_PCMB_MASK (0x1 << 6) +#define NAU8824_I2S_PCMB_SFT 6 +#define NAU8824_I2S_PCMB_EN (0x1 << 6) +#define NAU8824_I2S_PCMB_DIS (0x0 << 6) +#define NAU8824_I2S_DL_MASK (0x3 << 2) +#define NAU8824_I2S_DL_SFT 2 +#define NAU8824_I2S_DF_MASK (0x3) +#define NAU8824_I2S_DF_SFT 0 +#define NAU8824_I2S_DF_RIGHT (0x0) +#define NAU8824_I2S_DF_LEFT (0x1) +#define NAU8824_I2S_DF_I2S (0x2) +#define NAU8824_I2S_DF_PCM_A (0x3) +#define NAU8824_I2S_DF_PCM_B (0x3) + +/* reg. NAU8824_I2S_PCM_CTRL_2 (0x1D) */ +#define NAU8824_I2S_BCLKDIV_MASK 7 +#define NAU8824_I2S_BCLKDIV_SFT 0 + +#define NAU8824_I2S_MS_MASK (0x1 << 3) +#define NAU8824_I2S_MS_SFT 3 +#define NAU8824_I2S_MS_M (0x1 << 3) +#define NAU8824_I2S_MS_S (0x0 << 3) + +/* reg. NAU8824_DAC_MUTE_CTRL (0x31) */ +#define NAU8824_SOFT_MUTE_MASK (0x1 << 13) +#define NAU8824_SOFT_MUTE_SFT 13 +#define NAU8824_SOFT_MUTE_EN (0x1 << 13) +#define NAU8824_SOFT_MUTE_DIS (0x0 << 13) + +/* reg NAU8824_DAC0_DIG_VOL (0x32) */ +#define NAU8824_DAC0_GAIN_MASK (0x1FF << 0) + +/* reg NAU8824_DAC1_DIG_VOL (0x33) */ +#define NAU8824_DAC1_GAIN_MASK (0x1FF << 0) + +/* reg. NAU8824_CLASS_G_CTRL (0x50) */ +#define NAU8824_CLASS_G_CLK_SRC_MSK (0x3 << 14) +#define NAU8824_CLASS_G_CLK_DIS (0x3 << 14) +#define NAU8824_CLASS_G_CLK_SRC_MCLK (0x2 << 14) +#define NAU8824_CLASS_G_CLK_SRC_660K (0x1 << 14) +#define NAU8824_CLASS_G_CLK_SRC_2M (0x0 << 14) +#define NAU8824_CLASS_G_TIMER_MSK (0x3F << 8) +#define NAU8824_CLASS_G_TIMER_SFT 8 +#define NAU8824_CLASS_G_THRSLD_MSK (0x3 << 4) +#define NAU8824_CLASS_G_THRSLD_SFT 4 +#define NAU8824_CLASS_G_RIGHT_EN_MSK (0x1 << 2) +#define NAU8824_CLASS_G_RIGHT_EN (0x1 << 2) +#define NAU8824_CLASS_G_RIGHT_DIS (0x0 << 2) +#define NAU8824_CLASS_G_LEFT_EN_MSK (0x1 << 1) +#define NAU8824_CLASS_G_LEFT_EN (0x1 << 1) +#define NAU8824_CLASS_G_LEFT_DIS (0x0 << 1) +#define NAU8824_CLASS_G_EN_MSK (0x1 << 0) +#define NAU8824_CLASS_G_SHIFT 0 +#define NAU8824_CLASS_G_EN (0x1 << 0) +#define NAU8824_CLASS_G_DIS (0x0 << 0) + +/* reg NAU8824_SAR_ADC_OUTPUT (0x59) */ +#define NAU8824_ADC_OUTPUT_MASK (0xFF << 0) +#define NAU8824_ADC_OUTPUT_SFT 0 + +/* reg. NAU8824_BIAS_ADJ_2 (0x66) */ +#define NAU8824_VMID_MASK (0x1 << 6) +#define NAU8824_VMID_SFT 6 +#define NAU8824_VMID_EN (0x1 << 6) +#define NAU8824_VMID_DIS (0x0 << 6) +#define NAU8824_VMID_OPTION_MASK (0x1 << 4) +#define NAU8824_VMID_OPTION_SFT 4 +#define NAU8824_DMIC2_MASK (0x1 << 3) +#define NAU8824_DMIC2_SFT 3 +#define NAU8824_DMIC2_EN (0x1 << 3) +#define NAU8824_DMIC2_DIS (0x0 << 3) +#define NAU8824_DMIC1_MASK (0x1 << 2) +#define NAU8824_DMIC1_SFT 2 +#define NAU8824_DMIC1_EN (0x1 << 2) +#define NAU8824_DMIC1_DIS (0x0 << 2) + +/* reg. NAU8824_ANALOG_CTRL2 (0x6A) */ +#define NAU8824_CLASS_D_CLAMP_MSK (0x1 << 3) +#define NAU8824_CLASS_D_CLAMP_SFT 3 +#define NAU8824_CLASS_D_CLAMP_BETTER (0x1 << 3) +#define NAU8824_CLASS_D_CLAMP_DEFAULT (0x0 << 3) + +/* reg. NAU8824_HPO_MIXER (0x6B) */ +#define NAU8824_M_HPVOL_R_HM (0x1 << 2) +#define NAU8824_M_HPVOL_R_HM_SFT 2 +#define NAU8824_M_HPVOL_L_HM (0x1 << 0) +#define NAU8824_M_HPVOL_L_HM_SFT 0 + +/* reg. NAU8824_CLASS_D_GAIN1 (0x6D) */ +#define NAU8824_CLASS_D_GAIN_R_MSK (0x1F << 8) +#define NAU8824_CLASS_D_GAIN_R_SFT 8 +#define NAU8824_CLASS_D_MASK (0x1 << 7) +#define NAU8824_CLASS_D_SFT 7 +#define NAU8824_CLASS_D_EN_MSK (0x1 << 7) +#define NAU8824_CLASS_D_EN (0x1 << 7) +#define NAU8824_CLASS_D_DIS (0x0 << 7) + +/* reg. NAU8824_CLASS_D_GAIN2 (0x6E) */ +#define NAU8824_CLASS_D_GAIN_L_MSK (0x1F << 0) +#define NAU8824_CLASS_D_GAIN_L_SFT 0 + +/* reg. NAU8824_DAC_CTRL (0x73) */ +#define NAU8824_DAC_R_MASK (0x1 << 13) +#define NAU8824_DAC_R_SFT 13 +#define NAU8824_DAC_R_EN (0x1 << 13) +#define NAU8824_DAC_R_DIS (0x0 << 13) +#define NAU8824_DAC_L_MASK (0x1 << 12) +#define NAU8824_DAC_L_SFT 12 +#define NAU8824_DAC_L_EN (0x1 << 12) +#define NAU8824_DAC_L_DIS (0x0 << 12) + +/* reg. NAU8824_MIC_BIAS (0x74) */ +#define NAU8824_BIAS_POWER_MASK (0x1 << 8) +#define NAU8824_BIAS_POWER_SFT 8 +#define NAU8824_BIAS_POWER_ON (0x1 << 8) +#define NAU8824_BIAS_POWER_DOWN (0x0 << 8) +#define NAU8824_BIAS_LEVEL_MASK (0x7 << 0) +#define NAU8824_BIAS_LEVEL_SFT 0 +#define NAU8824_BIAS_LEVEL_VDDA (0x0 << 0) + +/* reg. NAU8824_HP_MUTE (0x75) */ +#define NAU8824_R_MUTE (0x1 << 14) +#define NAU8824_R_MUTE_SFT 14 +#define NAU8824_L_MUTE (0x1 << 6) +#define NAU8824_L_MUTE_SFT 6 + +/* reg. NAU8824_BOOST (0x76) */ +#define NAU8824_VMID_PRECHARGE_MASK (0x1 << 13) +#define NAU8824_VMID_PRECHARGE_SFT 13 +#define NAU8824_VMID_PRECHARGE_DIS (0x1 << 13) +#define NAU8824_VMID_PRECHARGE_EN (0x0 << 13) +#define NAU8824_G_BIAS_MASK (0x1 << 12) +#define NAU8824_G_BIAS_SFT 12 +#define NAU8824_G_BIAS_EN (0x1 << 12) +#define NAU8824_G_BIAS_DIS (0x0 << 12) + +/* reg. NAU8824_HP_VOL (0x7B) */ +#define NAU8824_R_VOL_MASK (0x1f << 8) +#define NAU8824_R_VOL_SFT 8 +#define NAU8824_SPK_L_VOL_MASK (0x1f) +#define NAU8824_SPK_R_VOL_MASK (0x1f << 8) +#define NAU8824_L_VOL_MASK (0x1f) +#define NAU8824_L_VOL_SFT 0 + +/* Volume Rescale */ +#define NAU8824_VOL_RSCL_MAX 0x1E +#define NAU8824_VOL_RSCL_RANGE 0x1E +#define NAU8824_SPK_VOL_RSCL_MAX 0x19 +#define NAU8824_SPK_VOL_RSCL_RANGE 0x19 +#define NAU8824_DAC_VOL_RSCL_RANGE 0x164 +#define NAU8824_ADC_VOL_RSCL_RANGE 0x164 + +/* Data Format */ +#define NAU8824_I2S_DL_32 (0x3 << 2) +#define NAU8824_I2S_DL_24 (0x2 << 2) +#define NAU8824_I2S_DL_20 (0x1 << 2) +#define NAU8824_I2S_DL_16 (0x0 << 2) + +#define NAU8824_FREQ_25000000 25000000 +#define NAU8824_PLL_CLKIN_MCLK 0 +#define NAU8824_BUTTONPRESS_MASK 0x20 +#define NAU8824_HSPLUG_MASK 0x10 + +enum { + NAU8824_INTERNALCLOCK = 0, + NAU8824_MCLK, +}; + +enum { + NAU8824_J_IN_EVENT, + NAU8824_J_OUT_EVENT, + NAU8824_BP_EVENT, + NAU8824_BR_EVENT, + NAU8824_UN_EVENT, +}; + +enum { + S_NAU8824_J_OUT, + S_NAU8824_J_IN_TEMP, + S_NAU8824_J_IN, + S_NAU8824_BUTTON_DECT, + S_NAU8824_NO_BUTTON_DECT, +}; + +#define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8824_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct nau8824_jack_data { + struct snd_soc_jack *jack; + int report; +}; + +struct nau8824_gpio_setup { + unsigned int reg; + u8 value; +}; + +struct nau8824_pdata { + unsigned int audio_mclk1; + unsigned int gpio_irq; + int naudint_irq; + int headset_detect; + int button_press_detect; +}; + +struct nau8824_priv { + struct nau8824_jack_data hs_jack; + struct workqueue_struct *workqueue; + struct delayed_work delayed_work; + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct mutex mutex; + struct nau8824_pdata pdata; + unsigned int irq; + bool jd_status; + bool bp_status; + int jack_type; +}; + +int nau8824_device_init(struct nau8824_priv *nau8824); +void nau8824_device_exit(struct nau8824_priv *nau8824); +void nau8824_enable_mic_bias(struct snd_soc_codec *codec, int enable); +int nau8824_query_jack_status(struct snd_soc_codec *codec); +int nau8824_query_btn_press(struct snd_soc_codec *codec); +void nau8824_btn_press_intr_enable(struct snd_soc_codec *codec, int enable); + +#endif /* _NAU8824_H */ -- 1.8.1.2

3 7

[alsa-devel] [PATCH] ASoC: atmel_pcm: use endian agnostic IO accessors
by Ben Dooks 26 Mar '15

26 Mar '15

The atmel-pcm.h is using __raw_readl and __raw_writel for IO accessors which do not work if the processor is in big endian mode (the peripheral will still expect data in little endian format). Note, I do not think this is being used in the AVR32 which would require a different fix (possible after atmel_io.h is merged). This has not been run-tested as I do not have a board with an audio code on. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> -- CC: Bo Shen <voice.shen(a)atmel.com> CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Mark Brown <broonie(a)kernel.org> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> CC: alsa-devel(a)alsa-project.org --- sound/soc/atmel/atmel-pcm.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 12ae814..8f5f551 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -80,8 +80,8 @@ struct atmel_pcm_dma_params { /* * SSC register access (since ssc_writel() / ssc_readl() require literal name) */ -#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) -#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) +#define ssc_readx(base, reg) (readl_relaxed((base) + (reg))) +#define ssc_writex(base, reg, value) writel_relaxed((value), (base) + (reg)) int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd); void atmel_pcm_free(struct snd_pcm *pcm); -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: davinci-evm: drop un-necessary remove function
by Jyri Sarha 26 Mar '15

26 Mar '15

From: Manish Badarkhe <manishvb(a)ti.com> As davinci card gets registered using 'devm_' api there is no need to unregister the card in 'remove' function. Hence drop the 'remove' function. Signed-off-by: Manish Badarkhe <manishvb(a)ti.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/davinci/davinci-evm.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b6bb594..8c2b9be 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -425,18 +425,8 @@ static int davinci_evm_probe(struct platform_device *pdev) return ret; } -static int davinci_evm_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static struct platform_driver davinci_evm_driver = { .probe = davinci_evm_probe, - .remove = davinci_evm_remove, .driver = { .name = "davinci_evm", .pm = &snd_soc_pm_ops, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: snd_soc_tas5086: Reinit register values on probe
by Pascal Huerst 26 Mar '15

26 Mar '15

If the machine driver has been un/reloaded, the register values of the codec driver have to be reinitialized in order to run properly. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/tas5086.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 249ef5c..cd19171 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -851,10 +851,16 @@ static int tas5086_probe(struct snd_soc_codec *codec) } tas5086_reset(priv); + regcache_mark_dirty(priv->regmap); + ret = tas5086_init(codec->dev, priv); if (ret < 0) goto exit_disable_regulators; + ret = regcache_sync(priv->regmap); + if (ret < 0) + goto exit_disable_regulators; + /* set master volume to 0 dB */ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); if (ret < 0) -- 1.9.3

2 4

[alsa-devel] [PATCH] ALSA: hda - Add one more node in the EAPD supporting candidate list
by Hui Wang 26 Mar '15

26 Mar '15

We have a HP machine which use the codec node 0x17 connecting the internal speaker, and from the node capability, we saw the EAPD, if we don't set the EAPD on for this node, the internal speaker can't output any sound. Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1436745 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- picked some content from the alsa-info.txt of this machine: Codec: Realtek ALC221 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0221 Subsystem Id: 0x103c8105 Revision Id: 0x100103 No Modem Function Group found ... ... ... Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00] Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0f sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8775389..75880fe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -396,7 +396,7 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) { /* We currently only handle front, HP */ static hda_nid_t pins[] = { - 0x0f, 0x10, 0x14, 0x15, 0 + 0x0f, 0x10, 0x14, 0x15, 0x17, 0 }; hda_nid_t *p; for (p = pins; *p; p++) -- 1.9.1

2 1

[alsa-devel] Add AZX_DCAPS_I915_POWEREWELL on SKL is OK?
by Yang, Libin 26 Mar '15

26 Mar '15

Hi Takashi, Is it OK to add AZX_DCAPS_I915_POWEREWELL for SKL in hda_intel.c? SKL HDA controller is not in powerwell domain. But its HDMI codec is in the domain. My understanding is if the flag is added here, it means the controller is in the domain. But I can't find a good place to add a flag to indicate the HDMI codec need the power domain. Regards, Libin

2 2

[alsa-devel] Fix a small typo in manpage cspctl.1 of alsa-tools/sb16_csp
by saulery＠legtux.org 26 Mar '15

26 Mar '15

Hello, I found an error in man pages cspctl.1. Could you please integrate this patch ? Regards,

2 2

[alsa-devel] [PATCH] ALSA: hda_intel: apply the Seperate stream_tag for Sunrise Point
by libin.yang＠intel.com 26 Mar '15

26 Mar '15

From: Libin Yang <libin.yang(a)intel.com> The total stream number of Sunrise Point's input and output stream exceeds 15, which will cause some streams do not work because of the overflow on SDxCTL.STRM field if using the legacy stream tag allocation method. This patch uses the new stream tag allocation method by add the flag AZX_DCAPS_SEPARATE_STREAM_TAG for Skylake platform. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index feebc1d..15a8299 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1963,7 +1963,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Sunrise Point */ { PCI_DEVICE(0x8086, 0xa170), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/5] ASoC: rsnd: DPCM base sampling rate convert support
by Kuninori Morimoto 26 Mar '15

26 Mar '15

Hi Mark These are remains of my previous patch set. Basically, these doesn't need soc-core's new feature, but it is assuming DPCM base sampling rate convert has both playback/capture in same time. 1 - 2: new Renesas sound card for DPCM 3 : bug fix patch. it can be applied in linus/master branch 4 - 5: new feature, but based on 3) patch Kuninori Morimoto (5): 1) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 2) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 3) ASoC: rsnd: call clk_prepare_enable/unprepare() in probe/remove 4) ASoC: rsnd: remove useless debug message 5) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 - sound/soc/sh/rcar/core.c | 25 ++++- sound/soc/sh/rcar/dma.c | 4 - sound/soc/sh/rcar/dvc.c | 28 +----- sound/soc/sh/rcar/gen.c | 4 - sound/soc/sh/rcar/rsnd.h | 12 ++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 84 +++++++++------- sound/soc/sh/rcar/ssi.c | 31 ++---- 12 files changed, 683 insertions(+), 96 deletions(-)

1 6

[alsa-devel] RawMIDI behaviour with MidiFace 4x4
by Kalvas, Taneli 25 Mar '15

25 Mar '15

Dear All, I am developing a patch librarian for a synthesizer. Because the communication happens with Midi sysex and timing is not relevant I chose RawMIDI interface and not the alsa sequencer. The problem I have encountered is that writing to the midi port with snd_rawmidi_write() seems to broken. The function accepts any size of data block and claims to have sent everything. In reality only a part of the message is sent. For a large block size I can see the activity light blinking for fraction of a second. This happens both in blocking and non-blocking mode. I would expect the snd_rawmidi_write() to accept only a part of the message. My hardware is Miditech Midiface 4x4, a USB midi interface: $ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xe0610000 irq 45 1 [MIDI4x4 ]: USB-Audio - MIDI4x4 MIDIPLUS MIDI4x4 at usb-0000:00:14.0-1, full speed usbid: 1acc:1a0b My alsa is quite recent (aplay shows version 1.0.27.2), I am running Ubuntu with 3.13.0-39-generic kernel. I have tested the system by connecting the output port to an input port on the hardware and having another program output the received bytes on stdout. My observations: - As long as the buffer size is 276 bytes or less it gets sent ok - Larger buffer sizes cause corruption and roughly 300 bytes of received. - Alsa reports midi ring buffer size of 4096. - snd_rawmidi_status_get_xruns() on both ends reports 0 errors. - If I write N=128 byte blocks and usleep for 352*N after each write, everything works ok. Is this due to hardware incompatibility with alsa driver or what is going on? Is there a way to connect two programs via a virtual RawMIDI port so that I could test without using the hardware? Any other diagnostics that I could do? My test programs: write.cpp: int main( void ) { int intstatus; snd_rawmidi_t *midiout; if( (intstatus = snd_rawmidi_open(NULL, &midiout, "hw:1,0,1", 0)) ) { std::cerr << "unable to open MIDI out port: " << snd_strerror(intstatus) << "\n"; exit( 1 ); } int N = 3000; // 276 bytes gets through ok. unsigned char msg[N]; for( int a = 0; a < N; a++ ) { msg[a] = a % 0x80; } msg[0] = 0xF0; msg[N-1] = 0xF7; ssize_t wrstatus; if( (wrstatus = snd_rawmidi_write(midiout, msg, N)) < 0 ) { std::cerr << "unable to write MIDI port: " << snd_strerror(wrstatus) << "\n"; return( -1 ); } std::cout << "write status = " << wrstatus << "\n"; if( (intstatus = snd_rawmidi_drain( midiout )) ) { std::cerr << "unable to drain: " << snd_strerror(intstatus) << "\n"; exit( 1 ); } // Sleep for the amount of time needed for transfer + 10 % extra + 100 ms usleep( 352*N + 100000 ); snd_rawmidi_status_t *stat; snd_rawmidi_status_malloc( &stat ); snd_rawmidi_status( midiout, stat ); int err = snd_rawmidi_status_get_xruns( stat ); std::cout << "xruns = " << err << "\n"; snd_rawmidi_close( midiout ); return( 0 ); } -------------------- read.cpp: bool quit = false; void signal_int( int ) { quit = true; } int main( void ) { int mode = SND_RAWMIDI_NONBLOCK; int intstat; std::string portname = "hw:1,0,0"; //std::string portname = "virtual"; snd_rawmidi_t* midiin; if( (intstat = snd_rawmidi_open(&midiin, NULL, portname.c_str(), mode)) < 0 ) { std::cerr << "unable to open MIDI port " << portname << ": " << snd_strerror(intstat) << "\n"; exit( 1 ); } int count = 0; unsigned char buf[1]; std::cout << std::dec << std::setfill(' ') << std::setw(6) << count << " "; signal( SIGINT, signal_int ); while( !quit ) { int read = snd_rawmidi_read( midiin, buf, 1 ); if( read == 0 || read == -EBUSY || read == -EAGAIN ) { usleep( 320 ); continue; } else if( read < 0 ) { std::cerr << "unable to read MIDI port" << portname << ": " << snd_strerror(read) << "\n"; exit( 1 ); } std::cout << std::hex << std::setfill('0') << "0x" << std::setw(2) << (int)buf[0] << " " << std::flush; count++; if( count % 8 == 0 ) { std::cout << "\n"; std::cout << std::dec << std::setfill(' ') << std::setw(6) << count << " " << std::flush; } } snd_rawmidi_status_t *stat; snd_rawmidi_status_malloc( &stat ); snd_rawmidi_status( midiin, stat ); int err = snd_rawmidi_status_get_xruns( stat ); std::cout << "\nxruns = " << err << "\n"; snd_rawmidi_close( midiin ); return( 0 ); } Thanks. Taneli -- Taneli Kalvas Ph.D., Researcher Department of Physics, room FL114 P.O. Box 35 (YFL) 40014 University of Jyväskylä, Finland Mobile: +358-44-314-1602 Fax: +358-14-617-411 Email: taneli.kalvas(a)jyu.fi

3 12

[alsa-devel] [PATCH resend] ASoC: fsl-ssi: Fix bitclock calculation for master mode
by Marc Kleine-Budde 25 Mar '15

25 Mar '15

From: Sascha Hauer <s.hauer(a)pengutronix.de> in fsl_ssi_set_bclk we are looking for the best divider by iterating over all possible dividers. The divider is in the range 1..256, so make this clear by using 'div' as name for the loop counter and iterate from 1 to 256 instead of 0 to 255. This makes obvious that tmprate is calculated wrong, it has to be freq * factor * div, not freq * factor * (i + 2). Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Marc Kleine-Budde <mkl(a)pengutronix.de> --- sound/soc/fsl/fsl_ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b9fabbf69db6..b8af08e49085 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -578,7 +578,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct regmap *regs = ssi_private->regs; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; - u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, div; unsigned long clkrate, baudrate, tmprate; u64 sub, savesub = 100000; unsigned int freq; @@ -602,8 +602,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; - for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + for (div = 1; div < 256; div++) { + tmprate = freq * factor * div; if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); @@ -618,7 +618,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, continue; clkrate /= factor; - afreq = clkrate / (i + 1); + afreq = clkrate / div; if (freq == afreq) sub = 0; @@ -636,7 +636,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, if (sub < savesub) { baudrate = tmprate; savesub = sub; - pm = i; + pm = div - 1; } /* We are lucky */ -- 2.1.4

2 2

[alsa-devel] [PATCH resend] ASoC: wm8741: Fix rates constraints values
by Sergej Sawazki 25 Mar '15

25 Mar '15

The WM8741 DAC supports the following typical audio sampling rates: 44.1kHz, 88.2kHz, 176.4kHz (eg: with a master clock of 22.5792MHz) 32kHz, 48kHz, 96kHz, 192kHz (eg: with a master clock of 24.576MHz) For the rates lists, we should use 82000 instead of 88235, 176400 instead of 1764000 and 192000 instead of 19200 (seems to be a typo). Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- sound/soc/codecs/wm8741.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 31bb480..9e71c76 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -123,7 +123,7 @@ static struct { }; static const unsigned int rates_11289[] = { - 44100, 88235, + 44100, 88200, }; static const struct snd_pcm_hw_constraint_list constraints_11289 = { @@ -150,7 +150,7 @@ static const struct snd_pcm_hw_constraint_list constraints_16384 = { }; static const unsigned int rates_16934[] = { - 44100, 88235, + 44100, 88200, }; static const struct snd_pcm_hw_constraint_list constraints_16934 = { @@ -168,7 +168,7 @@ static const struct snd_pcm_hw_constraint_list constraints_18432 = { }; static const unsigned int rates_22579[] = { - 44100, 88235, 1764000 + 44100, 88200, 176400 }; static const struct snd_pcm_hw_constraint_list constraints_22579 = { @@ -186,7 +186,7 @@ static const struct snd_pcm_hw_constraint_list constraints_24576 = { }; static const unsigned int rates_36864[] = { - 48000, 96000, 19200 + 48000, 96000, 192000 }; static const struct snd_pcm_hw_constraint_list constraints_36864 = { -- 1.9.1

3 2

[alsa-devel] [PATCH v3] ASoC: dapm: Fix build warning
by Charles Keepax 25 Mar '15

25 Mar '15

commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by switching to using devm_kasprintf. This also saves a couple of lines of code. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Use devm_kasprintf sound/soc/soc-dapm.c | 13 +++++-------- 1 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 049b300..4bd2a6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3346,7 +3346,6 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, { struct snd_soc_dapm_widget template; struct snd_soc_dapm_widget *w; - size_t len; char *link_name; int ret, count; unsigned long private_value; @@ -3366,28 +3365,26 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w_param_text) return -ENOMEM; - len = strlen(source->name) + strlen(sink->name) + 2; - link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); if (!link_name) { ret = -ENOMEM; goto outfree_w_param; } - snprintf(link_name, len, "%s-%s", source->name, sink->name); for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); - len = strlen("Anonymous Configuration ") + 3; w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); + devm_kasprintf(card->dev, GFP_KERNEL, + "Anonymous Configuration %d", + count); if (!w_param_text[count]) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, - "Anonymous Configuration %d", count); } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, -- 1.7.2.5

2 1

[alsa-devel] [ASoC] Free kcontrol memory while unregistering codec
by Vishal T 25 Mar '15

25 Mar '15

Hi, While going through the codec cleanup code (snd_soc_unregister_codec()), I could not find the code which frees memory allocated for snd_kcontrol (while adding the kcontrols to card). Am I missing anything obvious? Sorry if I have created noise. Thanks, V

1 0

[alsa-devel] [PATCH] ASoC: DAPM: Fix build warning
by Charles Keepax 25 Mar '15

25 Mar '15

commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by adding a non-const temporary variable to hold the value as it is created before assigning to the entry in w_param_text. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 049b300..7640370 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3376,18 +3376,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { + char *anon_name; + dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); len = strlen("Anonymous Configuration ") + 3; - w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!w_param_text[count]) { + anon_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!anon_name) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, + snprintf(anon_name, len, "Anonymous Configuration %d", count); + w_param_text[count] = anon_name; } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, -- 1.7.2.5

2 2

[alsa-devel] [RFC v2 0/3] ALSA: hda: add hda controller to hda core
by Vinod Koul 25 Mar '15

25 Mar '15

This series starts adding base hda controller code to hda core. There are few more changes like dsp loader etc planned to be added as well. This is based on Takashi's test/hda-core branch with patches posted last week, we have modified naming convention loosely based on takashi's work, Takshi feel free to recommend more changes if you think they are required Jeeja KP (2): ALSA: hda: move common hda controller register defines up ALSA: hda: add hda controller to hda Vinod Koul (1): ALSA: hda: add HDA_MAX_CODECS include/sound/hda_registers.h | 200 +++++++ include/sound/hda_verbs.h | 1 + include/sound/hdaudio.h | 215 +++++++ sound/hda/Makefile | 2 +- sound/hda/hdac_controller.c | 1284 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.c | 4 +- sound/pci/hda/hda_controller.h | 203 +------ 7 files changed, 1707 insertions(+), 202 deletions(-) create mode 100644 include/sound/hda_registers.h create mode 100644 sound/hda/hdac_controller.c -- 1.7.9.5

2 9

[alsa-devel] [PATCH v2] ASoC: dapm: Fix build warning
by Charles Keepax 25 Mar '15

25 Mar '15

commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by adding a non-const temporary variable to hold the value as it is created before assigning to the entry in w_param_text. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Just small fix to commit message sound/soc/soc-dapm.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 049b300..7640370 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3376,18 +3376,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { + char *anon_name; + dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); len = strlen("Anonymous Configuration ") + 3; - w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!w_param_text[count]) { + anon_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!anon_name) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, + snprintf(anon_name, len, "Anonymous Configuration %d", count); + w_param_text[count] = anon_name; } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, -- 1.7.2.5

1 0

[alsa-devel] Missing beep controls for Via HDA codec (snd_hda_codec_via)
by W. Trevor King 25 Mar '15

25 Mar '15

My VT1802 chip has a beep-generating node: $ cat /proc/asound/card1/codec#0 Codec: VIA VT1802 ... Vendor Id: 0x11068446 Subsystem Id: 0x15587410 Revision Id: 0x100000 ... Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1 Amp-Out vals: [0x0a] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 ... But I'm missing the: Control: name=... entries that I think I need to manage this widget from alsamixer. For example, sambrian posts codec output for a CX20561 chip which has [1]: Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] I can patch with something like this: diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 3de6d3d..85a0c08 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1520,8 +1520,10 @@ static int patch_vt2002P(struct hda_codec *codec) spec->gen.mixer_nid = 0x21; override_mic_boost(codec, 0x2b, 0, 3, 40); override_mic_boost(codec, 0x29, 0, 3, 40); - if (spec->codec_type == VT1802) + if (spec->codec_type == VT1802) { + spec->gen.beep_nid = 0x22; fix_vt1802_connections(codec); + } add_secret_dac_path(codec); snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); to get a dmesg entry like: input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card1/hdaudioC1D0/input15 but that doesn't add the beep controls. Looking at patch_analog.c, patch_conexant.c, and patch_realtek.c for inspiration, I expect we'll need an HDA_CODEC_MUTE_BEEP(_MONO) entry, and possibly an HDA_CODEC_VOLUME(_MONO) entry, but I'm not sure if these should be mono or not, or if they should be inputs (following patch_realtek.c) or outputs (following patch_analog.c and patch_conexant.c). I've filled out Via's datasheet-request form [2], but I don't know how likely that is to be accepted. Does anyone have a VT1802 datasheet that they're allowed to share (or some free time to fill in Via beep support on their own)? Or am I barking up the wrong tree? My only previous experience with this code-base was a beep-tweak for a previous box's Realtek chip [3], but I love my beeps and am happy to poke around with a datasheet until I get this working ;). Maybe folks with more experience can just guess the right code without needing to see specs for the VT1802? Cheers, Trevor [1]: http://forums.fedoraforum.org/archive/index.php/t-265822.html [2]: http://www.via.com.tw/en/support/contact_us/ [3]: http://thread.gmane.org/gmane.linux.alsa.devel/121300 -- This email may be signed or encrypted with GnuPG (http://www.gnupg.org). For more information, see http://en.wikipedia.org/wiki/Pretty_Good_Privacy

3 11

Re: [alsa-devel] ASoC: Max98090 Codec DAPM Widget setup in Open Embedded
by noman pouigt 25 Mar '15

25 Mar '15

On Mon, Mar 23, 2015 at 11:53 PM, noman pouigt <variksla(a)gmail.com> wrote: >> I would like to know which file I can use to connect/enable >> DPAM widgets permanently so a simple aplay/record can work? >> >> My environment is: >> - Yocto Linux Kernel 3.10 - (DTS based) >> - Max98090 Codec >> - imx6SL SoC >> >> I thought about doing this in alsa.conf or asound.conf but I couldn't >> find a single example with DAPM widgets . > > I am also stuck with the same problem. I am sure ALSA experts > can help here. > > What i want to know is how to set the dapm widgets in open > embedded? Is it alsa.conf or asound.conf ? > > I need to set something like this: > amixer cset name="LEFT MIXER DAC" 1 > > Using amixer i am able to playback but i don't know how can i > modify conf file to have this dapm widgets associated with > a device. > > Any example would suffice.

4 3

[alsa-devel] [PATCH v3 0/2] ALSA: jack: Refactoring for jack kctls
by Jie Yang 25 Mar '15

25 Mar '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this implement, we are no longer need hda jack specific kctls, so here also remove jack kctls for hda. Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. Jie Yang (2): ALSA: jack: create jack kcontrols for every jack input device ALSA: hda - Remove jack kctls include/sound/jack.h | 8 ++++ sound/core/Kconfig | 3 -- sound/core/Makefile | 3 +- sound/core/jack.c | 118 ++++++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/Kconfig | 10 +--- sound/pci/hda/hda_jack.c | 45 ++---------------- sound/pci/hda/hda_jack.h | 3 -- 7 files changed, 129 insertions(+), 61 deletions(-) -- 1.9.1

7 19

[alsa-devel] 4.0-rc3: sound broken in chromium but not in mpg123
by Pavel Machek 24 Mar '15

24 Mar '15

Hi! Intel desktop machine: 00:1b.0 Audio device: Intel Corporation NM10/ICH7 Family High Definition Audio Controller (rev 01) Sound worked ok in both in previous kernel versions. Any ideas? Thanks, Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

2 6

[alsa-devel] [PATCH] ASoC: soc-core: call soc_cleanup_card_debugfs() from snd_soc_unregister_card()
by Kuninori Morimoto 24 Mar '15

24 Mar '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Card debugfs is created in snd_soc_register_card(), but soc_cleanup_card_debugfs() is called from soc_cleanup_card_resources(). Cleanup function should be called from paired unregister function. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c26074..211783f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1751,8 +1751,6 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove and free each DAI */ soc_remove_dai_links(card); - soc_cleanup_card_debugfs(card); - /* remove the card */ if (card->remove) card->remove(card); @@ -2447,6 +2445,7 @@ int snd_soc_unregister_card(struct snd_soc_card *card) card->instantiated = false; snd_soc_dapm_shutdown(card); soc_cleanup_card_resources(card); + soc_cleanup_card_debugfs(card); dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); } -- 1.9.1

3 7

[alsa-devel] [PATCH v3] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 24 Mar '15

24 Mar '15

Set rule constraints to allow only combinations of sample-rate, sample-format, and channels counts that can be played/captured with reasonable sample-rate accuracy. The logic with tdm-slots and serializers (=i2s data wires) goes like this: The first wire will take all channels up to number of tdm-slots, before following wires (if any) are used. If the first wire is used fully, the remaining wires share the same clocks and the divider can be calculated for the first wire. Also, takes the number of tdm-slots into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Previous version of the patch: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-March/089319.html Since v2 I changed the SNDRV_PCM_HW_PARAM_SAMPLE_BITS rule in the to SNDRV_PCM_HW_PARAM_FORMAT rule. Somehow the constraints with a SNDRV_PCM_HW_PARAM_SAMPLE_BITS rule using snd_interval_list() converged to an empty set when ever I tried to play a format that could not pass the rule with any channel count or rate, resulting all too familiar assertion in alsa-lib. However, the direct tampering with SNDRV_PCM_HW_PARAM_FORMAT mask appears to work as it should. In addition to SAMPLE_BITS- to FORMAT-rule change I optimized the rate rule a bit so that it does not try the rates that are outside current range. According to my tests this does not affect the behavior in any other way but slightly speeds up the decision making. sound/soc/davinci/davinci-mcasp.c | 207 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 197 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d40b392..76156d18 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> #include <linux/platform_data/davinci_asp.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -65,6 +66,11 @@ struct davinci_mcasp_context { bool pm_state; }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; @@ -99,6 +105,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -868,6 +876,30 @@ 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 + (int)div64_long(1000000LL*rem, + (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) @@ -883,16 +915,20 @@ 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 channels = params_channels(params); + int rate = params_rate(params); + int sbits = params_width(params); + int ppm, div; + + if (channels > mcasp->tdm_slots) + channels = mcasp->tdm_slots; + + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + &ppm); + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -974,6 +1010,120 @@ 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_ruledata *rd = rule->private; + struct snd_interval *ri = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int sbits = params_width(params); + int channels = params_channels(params); + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int i, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + if (ri->min <= davinci_mcasp_dai_rates[i] && + ri->max >= davinci_mcasp_dai_rates[i]) { + uint bclk_freq = sbits*channels* + davinci_mcasp_dai_rates[i]; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + } + } + dev_dbg(rd->mcasp->dev, + "%d frequencies (%d-%d) for %d sbits and %d channels\n", + count, ri->min, ri->max, sbits, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask nfmt; + int rate = params_rate(params); + int channels = params_channels(params); + int i, count = 0; + + snd_mask_none(&nfmt); + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + if (snd_mask_test(fmt, i)) { + uint bclk_freq = snd_pcm_format_width(i)*channels*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + snd_mask_set(&nfmt, i); + count++; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible sample format for %d Hz and %d channels\n", + count, rate, channels); + + return snd_mask_refine(fmt, &nfmt); +} + +static int davinci_mcasp_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + int sbits = params_width(params); + int rate = params_rate(params); + int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? + rd->mcasp->tdm_slots : ci->max; + unsigned int list[ci->max - ci->min + 1]; + int c1, c, count = 0; + + for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { + uint bclk_freq = c1*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + /* If we can use all tdm_slots, we can put any + amount of channels to remaining wires as + long as they fit in. */ + if (c1 == rd->mcasp->tdm_slots) { + for (c = c1; c <= rd->serializers*c1 && + c <= ci->max; c++) + list[count++] = c; + } else { + list[count++] = c1; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible channel counts (%d-%d) for %d Hz and %d sbits\n", + count, ci->min, ci->max, rate, sbits); + + 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) { @@ -999,6 +1149,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1013,6 +1164,42 @@ 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) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + davinci_mcasp_hw_rule_format, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + davinci_mcasp_hw_rule_channels, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Fix duplicate const for DVC ramp rates
by Krzysztof Kozlowski 24 Mar '15

24 Mar '15

Replace duplicated const keyword for 'dvc_ramp_rate' with proper array of const pointers to const strings. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/soc/sh/rcar/dvc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index aeeef1352eee..6d85e36c6c2b 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -36,7 +36,7 @@ struct rsnd_dvc { ((pos) = (struct rsnd_dvc *)(priv)->dvc + i); \ i++) -static const char const *dvc_ramp_rate[] = { +static const char * const dvc_ramp_rate[] = { "128 dB/1 step", /* 00000 */ "64 dB/1 step", /* 00001 */ "32 dB/1 step", /* 00010 */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: ak4642: enable stereo line output power-save mode
by Kuninori Morimoto 24 Mar '15

24 Mar '15

From: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> ak4642 has power-save mode for stereo line to reduce pop noise. This patch enables it. Signed-off-by: Takeshi Kihara <takeshi.kihara.df(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 33 +++++++++++++++++++++++++++++++-- 1 file changed, 31 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7255f69..24ceb36 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -97,6 +97,9 @@ #define PMMP (1 << 2) /* MPWR pin Power Management */ #define MGAIN0 (1 << 0) /* MIC amp gain*/ +/* SG_SL2 */ +#define LOPS (1 << 6) /* Stero Line-out Power Save Mode */ + /* TIMER */ #define ZTM(param) ((param & 0x3) << 4) /* ALC Zero Crossing TimeOut */ #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) @@ -168,6 +171,29 @@ static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), }; +/* event handlers */ +static int ak4642_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_PRE_PMU: + /* Power save mode ON */ + snd_soc_update_bits(codec, SG_SL2, LOPS, LOPS); + break; + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + /* Power save mode OFF */ + mdelay(300); + snd_soc_update_bits(codec, SG_SL2, LOPS, 0); + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { /* Outputs */ @@ -182,9 +208,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, + SND_SOC_DAPM_MIXER_E("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], - ARRAY_SIZE(ak4642_lout_mixer_controls)), + ARRAY_SIZE(ak4642_lout_mixer_controls), + ak4642_lout_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), /* DAC */ SND_SOC_DAPM_DAC("DAC", NULL, PW_MGMT1, 2, 0), -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt286: check regmap_read result for ID check
by Bard Liao 24 Mar '15

24 Mar '15

It is worth to check the regmap_read result for ID check since it is the first regmap_read. And we can check if there is any i2c issue. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 098fc78..ccc0fa0 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1220,7 +1220,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret; + int i, ret, val; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1235,11 +1235,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { + ret = regmap_read(rt286->regmap, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret != 0) { + dev_err(&i2c->dev, "I2C error %d\n", ret); + return ret; + } + if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %x is not rt286\n", ret); + "Device with ID register %x is not rt286\n", val); return -ENODEV; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH RESEND] ARM: dts: Support audio on Exynos5422-odroidxu3 using simple-audio-card
by Inha Song 24 Mar '15

24 Mar '15

Add MAX98090 audio codec, I2S interface and the sound nodes to support audio on Exynos5422 SoC Based Odroid-XU3 board. Now we can support audio in Odroid-XU3 board using simple-audio-card DT binding. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- arch/arm/boot/dts/exynos5420.dtsi | 9 +++++ arch/arm/boot/dts/exynos5422-odroidxu3.dts | 57 ++++++++++++++++++++++++++++++ 2 files changed, 66 insertions(+) diff --git a/arch/arm/boot/dts/exynos5420.dtsi b/arch/arm/boot/dts/exynos5420.dtsi index 4eaeabe..e459c1d 100644 --- a/arch/arm/boot/dts/exynos5420.dtsi +++ b/arch/arm/boot/dts/exynos5420.dtsi @@ -415,6 +415,9 @@ <&clock_audss EXYNOS_I2S_BUS>, <&clock_audss EXYNOS_SCLK_I2S>; clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk0"; + #sound-dai-cells = <1>; samsung,idma-addr = <0x03000000>; pinctrl-names = "default"; pinctrl-0 = <&i2s0_bus>; @@ -429,6 +432,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S1>, <&clock CLK_SCLK_I2S1>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk1"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s1_bus>; status = "disabled"; @@ -442,6 +448,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S2>, <&clock CLK_SCLK_I2S2>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk2"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s2_bus>; status = "disabled"; diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3.dts b/arch/arm/boot/dts/exynos5422-odroidxu3.dts index edc25cf..9275ad6 100644 --- a/arch/arm/boot/dts/exynos5422-odroidxu3.dts +++ b/arch/arm/boot/dts/exynos5422-odroidxu3.dts @@ -11,6 +11,7 @@ */ /dts-v1/; +#include <dt-bindings/sound/samsung-i2s.h> #include "exynos5800.dtsi" / { @@ -285,6 +286,62 @@ rtc@101E0000 { status = "okay"; }; + + sound: sound { + compatible = "simple-audio-card"; + assigned-clocks = <&clock_audss EXYNOS_MOUT_AUDSS>, + <&clock_audss EXYNOS_MOUT_I2S>, + <&clock_audss EXYNOS_DOUT_AUD_BUS>; + assigned-clock-parents = <&clock CLK_FIN_PLL>, + <&clock_audss EXYNOS_MOUT_AUDSS>; + assigned-clock-rates = <0>, + <0>, + <19200000>; + + simple-audio-card,name = "Odroid-XU3"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack", + "Speakers", "Speakers"; + simple-audio-card,routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "Headphone Jack", "MICBIAS", + "IN1", "Headphone Jack", + "Speakers", "SPKL", + "Speakers", "SPKR"; + + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&link0_codec>; + simple-audio-card,frame-master = <&link0_codec>; + + simple-audio-card,cpu { + sound-dai = <&i2s0 0>; + system-clock-frequency = <19200000>; + }; + + link0_codec: simple-audio-card,codec { + sound-dai = <&max98090>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + }; + }; +}; + +&hsi2c_5 { + status = "okay"; + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx3>; + interrupts = <2 0>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; +}; + +&i2s0 { + status = "okay"; }; &hdmi { -- 2.0.0.390.gcb682f8

3 3

[alsa-devel] [PATCH] ALSA: asihpi: Fix duplicate const for clock sources
by Krzysztof Kozlowski 24 Mar '15

24 Mar '15

Replace duplicated const keyword for 'sampleclock_sources' with proper array of const pointers to const strings. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/pci/asihpi/asihpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e5cd7be85355..1039eccbb895 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2376,7 +2376,7 @@ static int snd_asihpi_cmode_add(struct snd_card_asihpi *asihpi, /*------------------------------------------------------------ Sampleclock source controls ------------------------------------------------------------*/ -static const char const *sampleclock_sources[] = { +static const char * const sampleclock_sources[] = { "N/A", "Local PLL", "Digital Sync", "Word External", "Word Header", "SMPTE", "Digital1", "Auto", "Network", "Invalid", "Prev Module", "BLU-Link", -- 1.9.1

2 1

[alsa-devel] [trivial PATCH] sound: Deparenthesize negative error returns
by Joe Perches 24 Mar '15

24 Mar '15

Make the returns a bit more kernel standard style. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/isa/wavefront/wavefront_fx.c | 6 +++--- sound/isa/wavefront/wavefront_synth.c | 26 +++++++++++++------------- sound/oss/dev_table.c | 6 +++--- sound/oss/v_midi.c | 4 ++-- 4 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c index b5a1970..0608a5a 100644 --- a/sound/isa/wavefront/wavefront_fx.c +++ b/sound/isa/wavefront/wavefront_fx.c @@ -79,13 +79,13 @@ wavefront_fx_memset (snd_wavefront_t *dev, if (page < 0 || page > 7) { snd_printk ("FX memset: " "page must be >= 0 and <= 7\n"); - return -(EINVAL); + return -EINVAL; } if (addr < 0 || addr > 0x7f) { snd_printk ("FX memset: " "addr must be >= 0 and <= 7f\n"); - return -(EINVAL); + return -EINVAL; } if (cnt == 1) { @@ -118,7 +118,7 @@ wavefront_fx_memset (snd_wavefront_t *dev, snd_printk ("FX memset " "(0x%x, 0x%x, 0x%lx, %d) incomplete\n", page, addr, (unsigned long) data, cnt); - return -(EIO); + return -EIO; } } diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 33f5ec1..69f76ff 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -793,7 +793,7 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PATCH, NULL, buf)) { snd_printk ("download patch failed\n"); - return -(EIO); + return -EIO; } return (0); @@ -831,7 +831,7 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PROGRAM, NULL, buf)) { snd_printk ("download patch failed\n"); - return -(EIO); + return -EIO; } return (0); @@ -952,7 +952,7 @@ wavefront_send_sample (snd_wavefront_t *dev, if (skip > 0 && header->hdr.s.SampleResolution != LINEAR_16BIT) { snd_printk ("channel selection only " "possible on 16-bit samples"); - return -(EINVAL); + return -EINVAL; } switch (skip) { @@ -1049,7 +1049,7 @@ wavefront_send_sample (snd_wavefront_t *dev, NULL, sample_hdr)) { snd_printk ("sample %sdownload refused.\n", header->size ? "" : "header "); - return -(EIO); + return -EIO; } if (header->size == 0) { @@ -1075,7 +1075,7 @@ wavefront_send_sample (snd_wavefront_t *dev, if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_BLOCK, NULL, NULL)) { snd_printk ("download block " "request refused.\n"); - return -(EIO); + return -EIO; } for (i = 0; i < blocksize; i++) { @@ -1135,12 +1135,12 @@ wavefront_send_sample (snd_wavefront_t *dev, if (dma_ack == -1) { snd_printk ("upload sample " "DMA ack timeout\n"); - return -(EIO); + return -EIO; } else { snd_printk ("upload sample " "DMA ack error 0x%x\n", dma_ack); - return -(EIO); + return -EIO; } } } @@ -1181,7 +1181,7 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_SAMPLE_ALIAS, NULL, alias_hdr)) { snd_printk ("download alias failed.\n"); - return -(EIO); + return -EIO; } dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_ALIAS); @@ -1232,7 +1232,7 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) msample_hdr)) { snd_printk ("download of multisample failed.\n"); kfree(msample_hdr); - return -(EIO); + return -EIO; } dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_MULTISAMPLE); @@ -1254,7 +1254,7 @@ wavefront_fetch_multisample (snd_wavefront_t *dev, if (snd_wavefront_cmd (dev, WFC_UPLOAD_MULTISAMPLE, log_ns, number)) { snd_printk ("upload multisample failed.\n"); - return -(EIO); + return -EIO; } DPRINT (WF_DEBUG_DATA, "msample %d has %d samples\n", @@ -1273,14 +1273,14 @@ wavefront_fetch_multisample (snd_wavefront_t *dev, if ((val = wavefront_read (dev)) == -1) { snd_printk ("upload multisample failed " "during sample loop.\n"); - return -(EIO); + return -EIO; } d[0] = val; if ((val = wavefront_read (dev)) == -1) { snd_printk ("upload multisample failed " "during sample loop.\n"); - return -(EIO); + return -EIO; } d[1] = val; @@ -1315,7 +1315,7 @@ wavefront_send_drum (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_EDRUM_PROGRAM, NULL, drumbuf)) { snd_printk ("download drum failed.\n"); - return -(EIO); + return -EIO; } return (0); diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c index d8cf3e5..6dad515 100644 --- a/sound/oss/dev_table.c +++ b/sound/oss/dev_table.c @@ -58,13 +58,13 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, if (vers != AUDIO_DRIVER_VERSION || driver_size > sizeof(struct audio_driver)) { printk(KERN_ERR "Sound: Incompatible audio driver for %s\n", name); - return -(EINVAL); + return -EINVAL; } num = sound_alloc_audiodev(); if (num == -1) { printk(KERN_ERR "sound: Too many audio drivers\n"); - return -(EBUSY); + return -EBUSY; } d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver))); sound_nblocks++; @@ -79,7 +79,7 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, if (d == NULL || op == NULL) { printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name); sound_unload_audiodev(num); - return -(ENOMEM); + return -ENOMEM; } init_waitqueue_head(&op->in_sleeper); init_waitqueue_head(&op->out_sleeper); diff --git a/sound/oss/v_midi.c b/sound/oss/v_midi.c index f0b4151..fc0ba27 100644 --- a/sound/oss/v_midi.c +++ b/sound/oss/v_midi.c @@ -49,13 +49,13 @@ static int v_midi_open (int dev, int mode, unsigned long flags; if (devc == NULL) - return -(ENXIO); + return -ENXIO; spin_lock_irqsave(&devc->lock,flags); if (devc->opened) { spin_unlock_irqrestore(&devc->lock,flags); - return -(EBUSY); + return -EBUSY; } devc->opened = 1; spin_unlock_irqrestore(&devc->lock,flags);

2 1

[alsa-devel] [PATCH] ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4
by Howard Mitchell 24 Mar '15

24 Mar '15

Currently GPIO4 is hardcoded to output the pll-lock signal. Unfortunately this is after the pll-out GPIO is configured which is selectable in the device tree. Therefore it is not possible to use GPIO4 for pll-out. Therefore this patch removes the configuration of GPIO4. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- sound/soc/codecs/pcm512x.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 8472099..5a30fdd 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1296,25 +1296,6 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, ret, pcm512x->pll_out); return ret; } - - gpio = PCM512x_G1OE << (4 - 1); - ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, - gpio, gpio); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable gpio %d: %d\n", - 4, ret); - return ret; - } - - gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1; - ret = regmap_update_bits(pcm512x->regmap, gpio, - PCM512x_GxSL, PCM512x_GxSL_PLLLK); - if (ret != 0) { - dev_err(codec->dev, - "Failed to output pll lock on %d: %d\n", - ret, 4); - return ret; - } } ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE, -- 1.7.9.5

3 3

[alsa-devel] [TINYCOMPRESS][PATCH 0/6] Convert build system to autotools
by Qais Yousef 24 Mar '15

24 Mar '15

This series converts the build system to use autotools and splits the files into lib and utils. Also patch 5 fixes error handling code. I can't test Android builds but both cross and native compiling worked for me. I left AUTHORS, ChangeLog and NEWS files empty. I used the license header to fill COPYING. Qais Yousef (6): Convert the build system to autotools add .gitignore file makefile.linux: delete as no longer necessary/used Android.mk: Update to use the new location of source files compress.c: fix check for errors from poll(), read() and write() src/lib/utils.c: remove this empty file .gitignore | 39 ++++ AUTHORS | 0 Android.mk | 6 +- COPYING | 51 ++++ ChangeLog | 0 INSTALL | 365 +++++++++++++++++++++++++++++ Makefile.am | 3 + NEWS | 0 autogen.sh | 6 + compress.c | 632 ------------------------------------------------- configure.ac | 31 +++ cplay.c | 330 -------------------------- crec.c | 489 -------------------------------------- m4/.place_holder | 0 makefile.linux | 41 ---- src/Makefile.am | 1 + src/lib/Makefile.am | 5 + src/lib/compress.c | 634 ++++++++++++++++++++++++++++++++++++++++++++++++++ src/utils/Makefile.am | 11 + src/utils/cplay.c | 330 ++++++++++++++++++++++++++ src/utils/crec.c | 489 ++++++++++++++++++++++++++++++++++++++ utils.c | 62 ----- 22 files changed, 1968 insertions(+), 1557 deletions(-) create mode 100644 .gitignore create mode 100644 AUTHORS create mode 100644 COPYING create mode 100644 ChangeLog create mode 100644 INSTALL create mode 100644 Makefile.am create mode 100644 NEWS create mode 100755 autogen.sh delete mode 100644 compress.c create mode 100644 configure.ac delete mode 100644 cplay.c delete mode 100644 crec.c create mode 100644 m4/.place_holder delete mode 100644 makefile.linux create mode 100644 src/Makefile.am create mode 100644 src/lib/Makefile.am create mode 100644 src/lib/compress.c create mode 100644 src/utils/Makefile.am create mode 100644 src/utils/cplay.c create mode 100644 src/utils/crec.c delete mode 100644 utils.c -- 2.1.0

2 8

[alsa-devel] [PATCH] MAINTAINERS: Add maintainer for maxim codecs
by Anish Kumar 24 Mar '15

24 Mar '15

This patch adds maintainer for maxim audio codecs. Signed-off-by: Anish Kumar <yesanishhere(a)gmail.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 8bdd7a7..2128586 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7387,6 +7387,13 @@ S: Supported F: sound/soc/ F: include/sound/soc* +MAXIM INTEGRATED SOC CODEC DRIVER +M: Anish Kumar <yesanishhere(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: sound/soc/codecs/max* +F: include/sound/max* + SPARC + UltraSPARC (sparc/sparc64) M: "David S. Miller" <davem(a)davemloft.net> L: sparclinux(a)vger.kernel.org -- 1.9.3

3 5

Re: [alsa-devel] Alsa-devel Digest, Vol 97, Issue 173
by noman pouigt 24 Mar '15

24 Mar '15

> I would like to know which file I can use to connect/enable > DPAM widgets permanently so a simple aplay/record can work? > > My environment is: > - Yocto Linux Kernel 3.10 - (DTS based) > - Max98090 Codec > - imx6SL SoC > > I thought about doing this in alsa.conf or asound.conf but I couldn't > find a single example with DAPM widgets . I am also stuck with the same problem. I am sure ALSA experts can help here. What i want to know is how to set the dapm widgets in open embedded? Is it alsa.conf or asound.conf ? I need to set something like this: amixer cset name="LEFT MIXER DAC" 1 Using amixer i am able to playback but i don't know how can i modify conf file to have this dapm widgets associated with a device. Any example would suffice.

1 0

[alsa-devel] [RFC][PATCH 00/11] digi00x: new driver for Digidesign 002/003 family
by Takashi Sakamoto 24 Mar '15

24 Mar '15

This patchset adds an experimental driver for Digidesign 002/003 family. If you have any models, please test with this driver and report the result. Currently, this driver correctly supports these major functionalities: (as long as I tested) * PCM sample capture * MIDI messages capture * MIDI messages playback And functionalities which need to be tested: * streaming with S/PDIF, ADAT and work clock source * MIDI device control And includes a bug which should be fixed: * PCM sample playback For a driver for these models, there was a preceding implementation by Robin Gareus and Damien Zammit. [alsa-devel] draft -- ALSA firewire + digi003 http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058177.ht… This patchset includes their implementation, called-as double-oh-three, to multiplex PCM samples into CIP packet. As long as I test, their implementation may includes any bugs and this driver still causes noisy sound in several cases, while I confirm to get the same data block pattern which Robin explains. http://gareus.org/wiki/digi003 If you interested in this bug and tries to fix it, my packet dump may be helpfull. Please see: http://subversion.ffado.org/wiki/digi003rack * 96khz-24bit-1ch.log (27.5 kB) * 44khz-24bit-1ch.log (49.8 kB) * 44khz-24bit-2ch.log (49.8 kB) And DKMS package is available on my github repository. https://github.com/takaswie/snd-firewire-improve Takashi Sakamoto (11): ALSA: digi00x: add skelton for Digi 002/003 device driver ALSA: digi00x: add streaming functionality ALSA: digi00x: add proc node for clock status ALSA: digi00x: add PCM functionality ALSA: digi00x: add MIDI functionality ALSA: digi00x: add hwdep interface ALSA: digi00x: support unknown asynchronous message ALSA: digi00x: support MIDI ports for device control ALSA: firewire-lib: allows to implement external MIDI callback function digi00x: improve MIDI capture/playback ALSA: digi00x: apply double-oh-three algorism to multiplex PCM samples include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 10 + sound/firewire/Makefile | 1 + sound/firewire/amdtp.c | 30 ++- sound/firewire/amdtp.h | 5 + sound/firewire/digi00x/Makefile | 3 + sound/firewire/digi00x/digi00x-hwdep.c | 192 ++++++++++++++ sound/firewire/digi00x/digi00x-midi.c | 199 +++++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 336 +++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 71 ++++++ sound/firewire/digi00x/digi00x-protocol.c | 361 +++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-stream.c | 398 ++++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x.c | 179 ++++++++++++++ sound/firewire/digi00x/digi00x.h | 126 ++++++++++ 15 files changed, 1907 insertions(+), 8 deletions(-) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-protocol.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h -- 2.1.0

7 39

[alsa-devel] [PATCH v2] ARM: dts: Support audio on Exynos5422-odroidxu3 using simple-audio-card
by Inha Song 24 Mar '15

24 Mar '15

Add MAX98090 audio codec, I2S interface and the sound nodes to support audio on Exynos5422 SoC Based Odroid-XU3 board. Now we can support audio in Odroid-XU3 board using simple-audio-card DT binding. Signed-off-by: Inha Song <ideal.song(a)samsung.com> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- arch/arm/boot/dts/exynos5420.dtsi | 9 +++++ arch/arm/boot/dts/exynos5422-odroidxu3.dts | 59 ++++++++++++++++++++++++++++++ 2 files changed, 68 insertions(+) diff --git a/arch/arm/boot/dts/exynos5420.dtsi b/arch/arm/boot/dts/exynos5420.dtsi index 4eaeabe..e459c1d 100644 --- a/arch/arm/boot/dts/exynos5420.dtsi +++ b/arch/arm/boot/dts/exynos5420.dtsi @@ -415,6 +415,9 @@ <&clock_audss EXYNOS_I2S_BUS>, <&clock_audss EXYNOS_SCLK_I2S>; clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk0"; + #sound-dai-cells = <1>; samsung,idma-addr = <0x03000000>; pinctrl-names = "default"; pinctrl-0 = <&i2s0_bus>; @@ -429,6 +432,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S1>, <&clock CLK_SCLK_I2S1>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk1"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s1_bus>; status = "disabled"; @@ -442,6 +448,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S2>, <&clock CLK_SCLK_I2S2>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk2"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s2_bus>; status = "disabled"; diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3.dts b/arch/arm/boot/dts/exynos5422-odroidxu3.dts index edc25cf..c038f0d 100644 --- a/arch/arm/boot/dts/exynos5422-odroidxu3.dts +++ b/arch/arm/boot/dts/exynos5422-odroidxu3.dts @@ -11,6 +11,7 @@ */ /dts-v1/; +#include <dt-bindings/sound/samsung-i2s.h> #include "exynos5800.dtsi" / { @@ -285,6 +286,64 @@ rtc@101E0000 { status = "okay"; }; + + sound: sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "Odroid-XU3"; + simple-audio-card,widgets = + "Headphone", "Headphone Jack", + "Speakers", "Speakers"; + simple-audio-card,routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "Headphone Jack", "MICBIAS", + "IN1", "Headphone Jack", + "Speakers", "SPKL", + "Speakers", "SPKR"; + + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&link0_codec>; + simple-audio-card,frame-master = <&link0_codec>; + + simple-audio-card,cpu { + sound-dai = <&i2s0 0>; + system-clock-frequency = <19200000>; + }; + + link0_codec: simple-audio-card,codec { + sound-dai = <&max98090>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + }; + }; +}; + +&clock_audss { + assigned-clocks = <&clock_audss EXYNOS_MOUT_AUDSS>, + <&clock_audss EXYNOS_MOUT_I2S>, + <&clock_audss EXYNOS_DOUT_AUD_BUS>; + assigned-clock-parents = <&clock CLK_FIN_PLL>, + <&clock_audss EXYNOS_MOUT_AUDSS>; + assigned-clock-rates = <0>, + <0>, + <19200000>; +}; + +&hsi2c_5 { + status = "okay"; + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx3>; + interrupts = <2 0>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; +}; + +&i2s0 { + status = "okay"; }; &hdmi { -- 2.0.0.390.gcb682f8

1 0

[alsa-devel] [PATCH 00/10 v3] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 24 Mar '15

24 Mar '15

Hi Mark These are v3 of DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, basically it was copied from simple-card. v2 patch-set had ak4642 patch, but, it is already applied. v3 doesn't include it. 1 : prepare patch for Renesas new sound card driver 2 - 3 : new Renesas sound card driver 4 - 8 : fixup/tidyup patches for rsnd 9 - 10 : DPCM support patch for rsnd Kuninori Morimoto (10): 1) ASoC: core: add snd_soc_get_dai_drv() to get DAI driver 2) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 3) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 4) ASoC: rsnd: no more SSI restart when unusual situation 5) ASoC: rsnd: no more SRC restart when unusual situation 6) ASoC: rsnd: tidyup error message format 7) ASoC: rsnd: show debug info for sampling rate convert 8) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 9) ASoC: rsnd: add dai_link stream name 10) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ include/sound/soc.h | 1 + sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 +++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd.h | 16 +++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 69 ++++++++++---- 12 files changed, 777 insertions(+), 54 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 22

[alsa-devel] [PATCH] ASoC:pcm512x: Make PLL lock output selectable via device tree.
by Howard Mitchell 23 Mar '15

23 Mar '15

Currently the PLL Lock output signal is hardcoded to GPIO4. This makes it seletable in the same way as pll-in and pll-out. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- .../devicetree/bindings/sound/pcm512x.txt | 3 ++ sound/soc/codecs/pcm512x.c | 47 +++++++++++++------- 2 files changed, 33 insertions(+), 17 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt index 3aae3b4..432f186 100644 --- a/Documentation/devicetree/bindings/sound/pcm512x.txt +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -26,6 +26,8 @@ Optional properties: given pll-in pin and PLL output on the given pll-out pin. An external connection from the pll-out pin to the SCLK pin is assumed. + - pll-lock : gpio pin used to output the PLL lock flag. + Examples: pcm5122: pcm5122@4c { @@ -49,4 +51,5 @@ Examples: clocks = <&sck>; pll-in = <3>; pll-out = <6>; + pll-lock = <4>; }; diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 8472099..a4217d7 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -49,6 +49,7 @@ struct pcm512x_priv { int fmt; int pll_in; int pll_out; + int pll_lock; int pll_r; int pll_j; int pll_d; @@ -1296,24 +1297,26 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, ret, pcm512x->pll_out); return ret; } + } - gpio = PCM512x_G1OE << (4 - 1); - ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, - gpio, gpio); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable gpio %d: %d\n", - 4, ret); - return ret; - } - - gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1; - ret = regmap_update_bits(pcm512x->regmap, gpio, - PCM512x_GxSL, PCM512x_GxSL_PLLLK); - if (ret != 0) { - dev_err(codec->dev, - "Failed to output pll lock on %d: %d\n", - ret, 4); - return ret; + if (pcm512x->pll_lock) { + gpio = PCM512x_G1OE << (pcm512x->pll_lock - 1); + ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, + gpio, gpio); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable gpio %d: %d\n", + pcm512x->pll_lock, ret); + return ret; + } + + gpio = PCM512x_GPIO_OUTPUT_1 + pcm512x->pll_lock - 1; + ret = regmap_update_bits(pcm512x->regmap, gpio, + PCM512x_GxSL, PCM512x_GxSL_PLLLK); + if (ret != 0) { + dev_err(codec->dev, + "Failed to output pll lock on %d: %d\n", + ret, pcm512x->pll_lock); + return ret; } } @@ -1518,6 +1521,16 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) ret = -EINVAL; goto err_clk; } + + if (of_property_read_u32(np, "pll-lock", &val) >= 0) { + if (val > 6) { + dev_err(dev, "Invalid pll-lock\n"); + ret = -EINVAL; + goto err_clk; + } + pcm512x->pll_lock = val; + } + } #endif -- 1.7.9.5

3 7

[alsa-devel] [PATCH v2] ALSA: hda - Add dock support for Thinkpad T450s (17aa:5036)
by Sebastian Wicki 23 Mar '15

23 Mar '15

This model uses the same dock port as the previous generation. Signed-off-by: Sebastian Wicki <gandro(a)gmx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eee4532..8775389 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4995,6 +4995,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 2.3.3

2 1

[alsa-devel] [PATCH] ALSA: hda - Add dock support for Thinkpad T450s (17aa:5036)
by Sebastian Wicki 23 Mar '15

23 Mar '15

This model uses the same dock port as the previous generation. Signed-off-by: Sebastian Wicki <gandro(a)gmx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eee4532..a9e2e6e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4988,6 +4988,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.3.3

2 2

[alsa-devel] [PATCH 0/7] Yet more HD-audio restrucruing
by Takashi Iwai 23 Mar '15

23 Mar '15

Hi, this is the patchset I mentioned in the previous post -- moving some HD-audio core stuff into sound/hda/ directory. As of now, only the codec drivers are split to sound/hda/. The legacy bus (controller) driver still remains almost intact, but the split of the core controller driver is also planned, and will be finished hopefully before 4.1 merge window. The original patches are found in topic/hda-core branch of sound git tree. The patches posted here are the ones rebased on top of the previous widget PM patches, and found in test/hda-core branch. Once when the widget PM patches get merged, I'll adjust this patchset on top of it. Takashi === Takashi Iwai (7): ALSA: hda - Make snd_hda_bus_type public ALSA: hda - Move some codes up to hdac_bus struct ALSA: hda - Move a part of hda_codec stuff into hdac_device ALSA: hda - Add widget sysfs tree ALSA: hda - Support indirect execution of verbs ALSA: hda - Fix possible runtime PM refcount unbalance ALSA: hda - Re-add tracepoints to HD-audio core driver include/sound/hdaudio.h | 181 ++++++++++++ sound/Kconfig | 2 + sound/Makefile | 2 +- sound/hda/Kconfig | 2 + sound/hda/Makefile | 6 + sound/hda/hda_bus_type.c | 42 +++ sound/hda/hdac_bus.c | 186 +++++++++++++ sound/hda/hdac_device.c | 527 +++++++++++++++++++++++++++++++++++ sound/hda/hdac_sysfs.c | 404 +++++++++++++++++++++++++++ sound/hda/local.h | 23 ++ sound/hda/trace.c | 6 + sound/hda/trace.h | 62 +++++ sound/pci/hda/Kconfig | 1 + sound/pci/hda/Makefile | 1 - sound/pci/hda/hda_auto_parser.c | 33 ++- sound/pci/hda/hda_beep.c | 4 +- sound/pci/hda/hda_bind.c | 157 +++-------- sound/pci/hda/hda_codec.c | 600 +++++++++------------------------------- sound/pci/hda/hda_codec.h | 87 ++---- sound/pci/hda/hda_controller.c | 8 +- sound/pci/hda/hda_generic.c | 25 +- sound/pci/hda/hda_intel.c | 4 +- sound/pci/hda/hda_local.h | 16 +- sound/pci/hda/hda_proc.c | 42 +-- sound/pci/hda/hda_sysfs.c | 60 ++-- sound/pci/hda/hda_trace.h | 119 -------- sound/pci/hda/local.h | 39 +++ sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0132.c | 6 +- sound/pci/hda/patch_conexant.c | 20 +- sound/pci/hda/patch_hdmi.c | 20 +- sound/pci/hda/patch_realtek.c | 78 +++--- sound/pci/hda/patch_si3054.c | 6 +- sound/pci/hda/patch_sigmatel.c | 84 +++--- sound/pci/hda/patch_via.c | 23 +- sound/pci/hda/thinkpad_helper.c | 2 +- 36 files changed, 1895 insertions(+), 985 deletions(-) create mode 100644 include/sound/hdaudio.h create mode 100644 sound/hda/Kconfig create mode 100644 sound/hda/Makefile create mode 100644 sound/hda/hda_bus_type.c create mode 100644 sound/hda/hdac_bus.c create mode 100644 sound/hda/hdac_device.c create mode 100644 sound/hda/hdac_sysfs.c create mode 100644 sound/hda/local.h create mode 100644 sound/hda/trace.c create mode 100644 sound/hda/trace.h delete mode 100644 sound/pci/hda/hda_trace.h create mode 100644 sound/pci/hda/local.h -- 2.3.3

1 8

[alsa-devel] [patch] ALSA: echoaudio: read past end of array
by Dan Carpenter 23 Mar '15

23 Mar '15

We need to cap "ucontrol->id.index / num_busses_in(chip)" so the we don't read beyond the end of the array. I also adding a check on "in" and changing the type in snd_echo_mixer_put() from short to unsigned int. Those changes are done for symmetry and are cosmetic. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index a962de0..862db9a 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1283,12 +1283,14 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, static int snd_echo_mixer_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct echoaudio *chip; + struct echoaudio *chip = snd_kcontrol_chip(kcontrol); + unsigned int out = ucontrol->id.index / num_busses_in(chip); + unsigned int in = ucontrol->id.index % num_busses_in(chip); - chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = - chip->monitor_gain[ucontrol->id.index / num_busses_in(chip)] - [ucontrol->id.index % num_busses_in(chip)]; + if (out >= ECHO_MAXAUDIOOUTPUTS || in >= ECHO_MAXAUDIOINPUTS) + return -EINVAL; + + ucontrol->value.integer.value[0] = chip->monitor_gain[out][in]; return 0; } @@ -1297,12 +1299,14 @@ static int snd_echo_mixer_put(struct snd_kcontrol *kcontrol, { struct echoaudio *chip; int changed, gain; - short out, in; + unsigned int out, in; changed = 0; chip = snd_kcontrol_chip(kcontrol); out = ucontrol->id.index / num_busses_in(chip); in = ucontrol->id.index % num_busses_in(chip); + if (out >= ECHO_MAXAUDIOOUTPUTS || in >= ECHO_MAXAUDIOINPUTS) + return -EINVAL; gain = ucontrol->value.integer.value[0]; if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT) return -EINVAL;

2 1

[alsa-devel] [patch] ALSA: ak411x: simplify snd_ak4113_create() a bit
by Dan Carpenter 23 Mar '15

23 Mar '15

"err" is always a negative error code here, so there is no point in checking. Removing the check silences a static checker warning and makes the code a bit more clear. Also we don't need to initialize "err". Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index 8884488..2183e9e 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -73,7 +73,7 @@ int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, void *private_data, struct ak4113 **r_ak4113) { struct ak4113 *chip; - int err = 0; + int err; unsigned char reg; static struct snd_device_ops ops = { .dev_free = snd_ak4113_dev_free, @@ -109,7 +109,7 @@ int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, __fail: snd_ak4113_free(chip); - return err < 0 ? err : -EIO; + return err; } EXPORT_SYMBOL_GPL(snd_ak4113_create);

2 1

[alsa-devel] [PATCH 0/5 v4] ASoC: rsnd: tidyup .remove method
by Kuninori Morimoto 23 Mar '15

23 Mar '15

Hi Mark These are v4 of unbind/bind issue fixup patches. It adds new snd_soc_remove_card() and temporally remove card from system. and rebind it again if next component was binded. Then, it indicates warning Kuninori Morimoto (5): ASoC: soc-core: add snd_soc_remove_card() ASoC: soc-core: deactivate pins in snd_soc_instantiate_card() ASoC: soc-core: call soc_cleanup_card_debugfs() from snd_soc_unregister_card() ASoC: soc-core: call snd_soc_remove_card() when component del ASoC: soc-core: indicate warning for unbind/bind issue include/sound/soc.h | 1 + sound/soc/soc-core.c | 81 +++++++++++++++++++++++++++++++++++++------------- 2 files changed, 61 insertions(+), 21 deletions(-) _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 13

[alsa-devel] simplify simple-card
by Kuninori Morimoto 23 Mar '15

23 Mar '15

Hi Mark, all I want to create new sound card driver which supports DAPM. But I can't use simple-card driver for this purpose because DT binding issue. I guess, this new sound card can use/share simple-card functions. OTOH, simple-card is not "simple" today :( Basically, main purpose of simple-card / new sound card is binding CPU/Codec settings to ALSA SoC framework. But we have many pattern for it. From DT bindings point of view, we can/should use same DT bindings for same settings. Now, simple-card is supporting - single CPU/Codec - multi CPU/Codec - more feature... My opinion is that complex code has un-understandable bug. So, how about simplify simple-card, what do you think about this idea ? We separate current simple-card to common parts, and, simple-card specific parts. basically, common parts controls DT bindings and ALSA SoC settings. sound card specific parts cares each own feature. I don't know about naming, but these audio card can use same bindings via common functions. - common : basic common functions - simple-audio-card : existing driver, keep as-is for compatible reason - single-audio-card : single CPU/Codec sound card. simplified from simple-audio-card - multi-audio-card : multi CPU/Codec sound card. simplified from simple-audio-card - fixrate-audio-card : new sound card same as single-audio-card, but it needs sampling-rate-convert via DAPM, - chain-audio-card : DAPM card (?) - .... My idea is taht we can replace current "simple-audio-card" to "single-audio-card", or, "simple-audio-card" to "multi-audio-card" on DT. Best regards --- Kuninori Morimoto

2 6

[alsa-devel] [PATCH 1/2] [resend] snd-waveartist: Introduce Rockwell WaveArtist RWA010 driver
by Ondrej Zary 22 Mar '15

22 Mar '15

This is a new ALSA driver driver for Rockwell WaveArtist RWA010 chips found on some (rare) ISA sound cards, such as DCS Multimedia S717. I wasn't able to get the old OSS WaveArtist driver to work with my card but it was a great source of information about the chip (as the full datasheet is not available - only a brief one and a design guide). However, the OSS driver only supports few of the mixer registers so I had to install the card in Windows and dump mixer registers while changing the mixer settings. Then tried what the rest of the registers do and the result is a fully working mixer which even supports more controls than the Windows driver. Someone with a NetWinder can add support for it to this driver and then remove the old OSS one. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- include/sound/mpu401.h | 1 + sound/isa/Kconfig | 11 + sound/isa/Makefile | 2 + sound/isa/waveartist.c | 1399 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/waveartist.h | 74 +++ 5 files changed, 1487 insertions(+) create mode 100644 sound/isa/waveartist.c create mode 100644 sound/isa/waveartist.h diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h index e942096..8fceeba 100644 --- a/include/sound/mpu401.h +++ b/include/sound/mpu401.h @@ -44,6 +44,7 @@ #define MPU401_HW_INTEL8X0 17 /* Intel8x0 driver */ #define MPU401_HW_PC98II 18 /* Roland PC98II */ #define MPU401_HW_AUREAL 19 /* Aureal Vortex */ +#define MPU401_HW_WAVEARTIST 20 /* Rockwell WaveArtist */ #define MPU401_INFO_INPUT (1 << 0) /* input stream */ #define MPU401_INFO_OUTPUT (1 << 1) /* output stream */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 0216475..7a3b4a2 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -454,5 +454,16 @@ config SND_MSND_CLASSIC To compile this driver as a module, choose M here: the module will be called snd-msnd-classic. +config SND_WAVEARTIST + tristate "Rockwell WaveArtist RWA010" + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_PCM + help + Say Y here to include support for Rockwell WaveArtist RWA010 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-waveartist. + endif # SND_ISA diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 9a15f14..23e11e2 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -12,6 +12,7 @@ snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o snd-sscape-objs := sscape.o +snd-waveartist-objs := waveartist.o # Toplevel Module Dependency obj-$(CONFIG_SND_ADLIB) += snd-adlib.o @@ -23,6 +24,7 @@ obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o +obj-$(CONFIG_SND_WAVEARTIST) += snd-waveartist.o obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ galaxy/ gus/ msnd/ opti9xx/ \ sb/ wavefront/ wss/ diff --git a/sound/isa/waveartist.c b/sound/isa/waveartist.c new file mode 100644 index 0000000..e5bda5c --- /dev/null +++ b/sound/isa/waveartist.c @@ -0,0 +1,1399 @@ +/* + * Driver for Rockwell WaveArtist RWA010 soundcards + * + * Copyright (c) 2015 Ondrej Zary + * + * HW-related parts based on OSS WaveArtist driver by Hannu Savolainen + * + * ALSA code based on ES18xx driver by Christian Fischbach & Abramo Bagnara + * and also by OPL3-SA2 driver by Jaroslav Kysela + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/isa.h> +#include <linux/pnp.h> +#include <linux/isapnp.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> +#include "waveartist.h" + +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("Driver for Rockwell WaveArtist RWA010 sound cards"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; +#ifdef CONFIG_PNP +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +#endif +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x250-0x3f0 */ +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x388-0x3f0 */ +static long midi_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;/* 0x300-0x3f0 */ +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,10,11 */ +static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 5,6,7 */ +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for WaveArtist soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for WaveArtist soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable WaveArtist soundcard."); +#ifdef CONFIG_PNP +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); +#endif +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for WaveArtist driver."); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for WaveArtist driver."); +module_param_array(midi_port, long, NULL, 0444); +MODULE_PARM_DESC(midi_port, "MIDI port # for WaveArtist driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for WaveArtist driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "DMA1 # for WaveArtist driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "DMA2 # for WaveArtist driver."); + +#ifdef CONFIG_PNP +static int isa_registered; +static int pnp_registered; +#endif + +#define PFX "waveartist: " + +#ifdef CONFIG_PNP +#define WA_DEVICE(pnpid) { \ + .id = pnpid, \ + .devs = { {"RSS5000"}, {"RSS5001"}, {"RSS5002"} } \ +} +/* + * RSS5000 = WaveArtist, RSS5001 = SB, RSS5002 = MPU-401, RSS5003 = IDE, + * RSS5004 = gameport, RSS5005 = modem, RSS5006 = 3D + */ + +static struct pnp_card_device_id snd_waveartist_pnpids[] = { + WA_DEVICE("RSS5000"), /* 16-bit decode */ + WA_DEVICE("RSS5100"), /* 16-bit decode + modem */ + WA_DEVICE("RSS5200"), /* 10-bit decode + modem */ + WA_DEVICE("RSS5300"), /* 10-bit decode */ + WA_DEVICE("RSS5400"), /* 16-bit decode + IDE */ + WA_DEVICE("RSS5500"), /* 16-bit decode + modem + IDE */ + WA_DEVICE("RSS5600"), /* 10-bit decode + IDE */ + WA_DEVICE("RSS5700"), /* 10-bit decode + modem + IDE */ + WA_DEVICE("RSS5800"), /* 10-bit decode + modem + 3D */ + WA_DEVICE("RSS5900"), /* 10-bit decode + 3D */ + WA_DEVICE("RSS5A00"), /* 16-bit decode + modem + 3D */ + WA_DEVICE("RSS5B00"), /* 16-bit decode + 3D */ + { .id = "" } /* end */ +}; +MODULE_DEVICE_TABLE(pnp_card, snd_waveartist_pnpids); +#endif /* CONFIG_PNP */ + +struct snd_waveartist { +#ifdef CONFIG_PNP + struct pnp_dev *wa; /* WaveArtist device */ + struct pnp_dev *sb; /* SB emulation device */ + struct pnp_dev *mpu; /* MPU-401 device */ +#endif + unsigned long port; /* base port */ + struct resource *res_port; /* base port resource */ + int irq; + int dma_playback; + int dma_capture; + + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + + spinlock_t reg_lock; + struct snd_hwdep *synth; + struct snd_rawmidi *rmidi; + + u16 image[20]; /* mixer registers image */ +}; + +static inline void wa_outb(struct snd_waveartist *chip, u8 reg, u8 val) +{ + outb(val, chip->port + reg); +} + +static inline u8 wa_inb(struct snd_waveartist *chip, u8 reg) +{ + return inb(chip->port + reg); +} + +static inline void wa_outw(struct snd_waveartist *chip, u8 reg, u16 val) +{ + outw(val, chip->port + reg); +} + +static inline u16 wa_inw(struct snd_waveartist *chip, u8 reg) +{ + return inw(chip->port + reg); +} + +static inline void waveartist_set_ctlr(struct snd_waveartist *chip, u8 clear, + u8 set) +{ + clear = ~clear & wa_inb(chip, CTLR); + wa_outb(chip, CTLR, clear | set); +} + +/* acknowledge IRQ */ +static inline void waveartist_iack(struct snd_waveartist *chip) +{ + u8 old_ctlr = wa_inb(chip, CTLR) & ~IRQ_ACK; + + wa_outb(chip, CTLR, old_ctlr | IRQ_ACK); + wa_outb(chip, CTLR, old_ctlr); +} + +static int waveartist_reset(struct snd_waveartist *chip) +{ + unsigned int timeout, res = -1; + + waveartist_set_ctlr(chip, -1, RESET); + msleep(200); + waveartist_set_ctlr(chip, RESET, 0); + + timeout = 500; + do { + mdelay(2); + + if (wa_inb(chip, STATR) & CMD_RF) { + res = wa_inw(chip, CMDR); + if (res == 0x55aa) + break; + } + } while (--timeout); + + if (timeout == 0) { + dev_warn(chip->card->dev, "WaveArtist: reset timeout (res=0x%x)\n", + res); + return 1; + } + + return 0; +} + +/* Helper function to send and receive words + * from WaveArtist. It handles all the handshaking + * and can send or receive multiple words. + */ +static int waveartist_cmd(struct snd_waveartist *chip, + int nr_cmd, u16 *cmd, + int nr_resp, u16 *resp) +{ + unsigned long flags; + unsigned int timed_out = 0, i; + + spin_lock_irqsave(&chip->reg_lock, flags); + /* + * The chip can hang if we access the STATR register too quickly + * after a write. Do a dummy read to slow down. + */ + wa_inb(chip, CTLR); + + if (wa_inb(chip, STATR) & CMD_RF) { + /* flush the port */ + wa_inw(chip, CMDR); + udelay(10); + } + + for (i = 0; !timed_out && i < nr_cmd; i++) { + int count; + + for (count = 5000; count; count--) + if (wa_inb(chip, STATR) & CMD_WE) + break; + + if (!count) + timed_out = 1; + else + wa_outw(chip, CMDR, cmd[i]); + /* Another dummy read */ + wa_inb(chip, CTLR); + } + + for (i = 0; !timed_out && i < nr_resp; i++) { + int count; + + for (count = 5000; count; count--) + if (wa_inb(chip, STATR) & CMD_RF) + break; + + if (!count) + timed_out = 1; + else + resp[i] = wa_inw(chip, CMDR); + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + + return timed_out ? 1 : 0; +} + +/* Send one command word */ +static inline int waveartist_cmd1(struct snd_waveartist *chip, u16 cmd) +{ + return waveartist_cmd(chip, 1, &cmd, 0, NULL); +} + +/* Send one command, receive one word */ +static inline u16 waveartist_cmd1_r(struct snd_waveartist *chip, u16 cmd) +{ + u16 ret; + + waveartist_cmd(chip, 1, &cmd, 1, &ret); + + return ret; +} + +/* Send a double command, receive one word (and throw it away) */ +static inline int waveartist_cmd2(struct snd_waveartist *chip, u16 cmd, u16 arg) +{ + u16 vals[2] = { cmd, arg }; + + return waveartist_cmd(chip, 2, vals, 1, vals); +} + +/* Send a triple command */ +static inline int waveartist_cmd3(struct snd_waveartist *chip, u16 cmd, + u16 arg1, u16 arg2) +{ + u16 vals[3] = { cmd, arg1, arg2 }; + + return waveartist_cmd(chip, 3, vals, 0, NULL); +} + +static u16 waveartist_getrev(struct snd_waveartist *chip) +{ + u16 temp[2]; + u16 cmd = WACMD_GETREV; + + waveartist_cmd(chip, 1, &cmd, 2, temp); + + return temp[0]; +} + +static irqreturn_t snd_waveartist_interrupt(int irq, void *dev_id) +{ + u8 status, irqstatus; + struct snd_card *card = dev_id; + struct snd_waveartist *chip; + + if (card == NULL) + return IRQ_NONE; + + chip = card->private_data; + + irqstatus = wa_inb(chip, IRQSTAT); + status = wa_inb(chip, STATR); + + if (status & IRQ_REQ) /* clear interrupt */ + waveartist_iack(chip); + + if (irqstatus & IRQ_PCM) { /* PCM buffer done */ + if ((status & DMA1) && chip->playback_substream) + snd_pcm_period_elapsed(chip->playback_substream); + if ((status & DMA0) && chip->capture_substream) + snd_pcm_period_elapsed(chip->capture_substream); + if (!(status & (DMA0 | DMA1))) + dev_warn(chip->card->dev, "Unknown PCM interrupt\n"); + } + + if (irqstatus & IRQ_SB) /* we do not use SB mode */ + dev_warn(chip->card->dev, "Unexpected SB interrupt\n"); + + if ((irqstatus & IRQ_MPU) && chip->rmidi) + snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); + + return IRQ_HANDLED; +} + +#ifdef CONFIG_PM +static int snd_waveartist_suspend(struct snd_card *card, pm_message_t state) +{ + struct snd_waveartist *chip = card->private_data; + int i; + + if (!card) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + + /* save mixer registers */ + for (i = 0; i < ARRAY_SIZE(chip->image); i++) + chip->image[i] = waveartist_cmd1_r(chip, + WACMD_GET_LEVEL | i << 8); + + return 0; +} + +static int snd_waveartist_resume(struct snd_card *card) +{ + struct snd_waveartist *chip; + int i; + + if (!card) + return 0; + + chip = card->private_data; + + /* restore mixer registers */ + for (i = 0; i < 10; i += 2) + waveartist_cmd3(chip, WACMD_SET_MIXER, + chip->image[i], chip->image[i + 1]); + for (i = 10; i < ARRAY_SIZE(chip->image); i += 2) + waveartist_cmd3(chip, WACMD_SET_LEVEL | + ((i - 10) / 2) << 8, + chip->image[i], chip->image[i + 1]); + + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif /* CONFIG_PM */ + +#ifdef CONFIG_PNP +static void snd_waveartist_set_irq(struct pnp_dev *pdev, int irq) +{ + if (!pdev->active) + return; + isapnp_cfg_begin(isapnp_card_number(pdev), isapnp_csn_number(pdev)); + isapnp_write_byte(0x70, irq); /* ISAPNP_CFG_IRQ */ + isapnp_cfg_end(); +} + +static int snd_waveartist_pnp(int dev, struct snd_waveartist *chip, + struct pnp_card_link *card, + const struct pnp_card_device_id *id) +{ + chip->wa = pnp_request_card_device(card, id->devs[0].id, NULL); + if (!chip->wa) + return -EBUSY; + + chip->sb = pnp_request_card_device(card, id->devs[1].id, NULL); + if (!chip->sb) + return -EBUSY; + + chip->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); + if (!chip->mpu) + return -EBUSY; + + if (pnp_activate_dev(chip->wa) < 0) { + dev_err(chip->card->dev, "WA PnP configure failure\n"); + return -EBUSY; + } + if (pnp_activate_dev(chip->sb) < 0) { + dev_err(chip->card->dev, "SB PnP configure failure\n"); + return -EBUSY; + } + port[dev] = pnp_port_start(chip->wa, 0); + dma2[dev] = pnp_dma(chip->wa, 0); + fm_port[dev] = pnp_port_start(chip->sb, 1); + dma1[dev] = pnp_dma(chip->sb, 0); + irq[dev] = pnp_irq(chip->sb, 0); + + /* + * The card uses only one IRQ (listed in the resources of SB device) + * which needs to be shared by WaveArtist and MPU-401 devices. They + * don't have an IRQ resource so it must be forced. + */ + snd_waveartist_set_irq(chip->wa, irq[dev]); + + /* allocate MPU-401 resources */ + if (pnp_activate_dev(chip->mpu) < 0) + dev_err(chip->card->dev, "MPU-401 PnP configure failure: will be disabled\n"); + else { + midi_port[dev] = pnp_port_start(chip->mpu, 0); + snd_waveartist_set_irq(chip->mpu, irq[dev]); + } + + dev_dbg(chip->card->dev, "PnP WaveArtist: port=0x%lx, fm port=0x%lx, midi port=0x%lx\n", + port[dev], fm_port[dev], midi_port[dev]); + dev_dbg(chip->card->dev, "PnP WaveArtist: dma1=%i, dma2=%i, irq=%i\n", + dma1[dev], dma2[dev], irq[dev]); + + return 0; +} +#endif /* CONFIG_PNP */ + +static int snd_waveartist_playback_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + if (err < 0) + return err; + + return 0; +} + +static int snd_waveartist_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static enum wa_format waveartist_format(snd_pcm_format_t format) +{ + if (snd_pcm_format_width(format) == 16) + return WA_FMT_S16; + if (snd_pcm_format_unsigned(format)) + return WA_FMT_U8; + else + return WA_FMT_S8; +} + +static u16 waveartist_rate(struct snd_pcm_runtime *runtime) +{ + return (runtime->rate << 16) / 44100; +} + +static int snd_waveartist_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + /* Set rate */ + if (waveartist_cmd2(chip, WACMD_OUTPUTSPEED, waveartist_rate(runtime))) + dev_warn(chip->card->dev, "error setting playback rate %dHz\n", + runtime->rate); + /* Set channel count */ + if (waveartist_cmd2(chip, WACMD_OUTPUTCHANNELS, runtime->channels)) + dev_warn(chip->card->dev, "error setting playback %d channels\n", + runtime->channels); + /* Set DMA channel */ + if (waveartist_cmd2(chip, WACMD_OUTPUTDMA, + chip->dma_playback > 3 ? WA_DMA_16BIT : WA_DMA_8BIT)) + dev_warn(chip->card->dev, "error setting playback data path\n"); + /* Set format */ + if (waveartist_cmd2(chip, WACMD_OUTPUTFORMAT, + waveartist_format(runtime->format))) + dev_warn(chip->card->dev, "error setting playback format %d\n", + runtime->format); + /* Set sample count */ + if (waveartist_cmd2(chip, WACMD_OUTPUTSIZE, count - 1)) + dev_warn(chip->card->dev, "error setting playback count %d\n", + count); + /* Configure DMA controller */ + snd_dma_program(chip->dma_playback, runtime->dma_addr, size, + DMA_MODE_WRITE | DMA_AUTOINIT); + + return 0; +} + +static int snd_waveartist_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + waveartist_cmd1(chip, WACMD_OUTPUTSTART); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + waveartist_cmd1(chip, WACMD_OUTPUTSTOP); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + waveartist_cmd1(chip, WACMD_OUTPUTPAUSE); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + waveartist_cmd1(chip, WACMD_OUTPUTRESUME); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int snd_waveartist_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + if (err < 0) + return err; + + return 0; +} + +static int snd_waveartist_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + /* Set rate */ + if (waveartist_cmd2(chip, WACMD_INPUTSPEED, waveartist_rate(runtime))) + dev_warn(chip->card->dev, "error setting capture rate %dHz\n", + runtime->rate); + /* Set channel count */ + if (waveartist_cmd2(chip, WACMD_INPUTCHANNELS, runtime->channels)) + dev_warn(chip->card->dev, "error setting capture %d channels\n", + runtime->channels); + /* Set DMA channel */ + if (waveartist_cmd2(chip, WACMD_INPUTDMA, + chip->dma_capture > 3 ? WA_DMA_16BIT : WA_DMA_8BIT)) + dev_warn(chip->card->dev, "error setting capture data path\n"); + /* Set format */ + if (waveartist_cmd2(chip, WACMD_INPUTFORMAT, + waveartist_format(runtime->format))) + dev_warn(chip->card->dev, "error setting capture format %d\n", + runtime->format); + /* Set sample count */ + if (waveartist_cmd2(chip, WACMD_INPUTSIZE, count - 1)) + dev_warn(chip->card->dev, "error setting capture count %d\n", + count); + /* Configure DMA controller */ + snd_dma_program(chip->dma_capture, runtime->dma_addr, size, + DMA_MODE_READ | DMA_AUTOINIT); + + return 0; +} + +static int snd_waveartist_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + waveartist_cmd1(chip, WACMD_INPUTSTART); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + waveartist_cmd1(chip, WACMD_INPUTSTOP); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + waveartist_cmd1(chip, WACMD_INPUTPAUSE); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + waveartist_cmd1(chip, WACMD_INPUTRESUME); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_waveartist_playback_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + size_t size = snd_pcm_lib_buffer_bytes(substream); + size_t ptr = snd_dma_pointer(chip->dma_playback, size); + + return bytes_to_frames(substream->runtime, ptr); +} + +static snd_pcm_uframes_t snd_waveartist_capture_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + size_t size = snd_pcm_lib_buffer_bytes(substream); + size_t ptr = snd_dma_pointer(chip->dma_capture, size); + + return bytes_to_frames(substream->runtime, ptr); +} + +static struct snd_pcm_hardware snd_waveartist_playback = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_44100, + .rate_min = 4000, + .rate_max = 44100, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static struct snd_pcm_hardware snd_waveartist_capture = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_44100, + .rate_min = 4000, + .rate_max = 44100, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static int snd_waveartist_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = substream; + substream->runtime->hw = snd_waveartist_playback; + + snd_pcm_limit_isa_dma_size(chip->dma_playback, + &substream->runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma_playback, + &substream->runtime->hw.period_bytes_max); + + return 0; +} + +static int snd_waveartist_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = substream; + substream->runtime->hw = snd_waveartist_capture; + + snd_pcm_limit_isa_dma_size(chip->dma_capture, + &substream->runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma_capture, + &substream->runtime->hw.period_bytes_max); + + return 0; +} + +static int snd_waveartist_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = NULL; + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int snd_waveartist_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_waveartist *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = NULL; + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static struct snd_pcm_ops snd_waveartist_playback_ops = { + .open = snd_waveartist_playback_open, + .close = snd_waveartist_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_waveartist_playback_hw_params, + .hw_free = snd_waveartist_pcm_hw_free, + .prepare = snd_waveartist_playback_prepare, + .trigger = snd_waveartist_playback_trigger, + .pointer = snd_waveartist_playback_pointer, +}; + +static struct snd_pcm_ops snd_waveartist_capture_ops = { + .open = snd_waveartist_capture_open, + .close = snd_waveartist_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_waveartist_capture_hw_params, + .hw_free = snd_waveartist_pcm_hw_free, + .prepare = snd_waveartist_capture_prepare, + .trigger = snd_waveartist_capture_trigger, + .pointer = snd_waveartist_capture_pointer, +}; + +static int snd_waveartist_pcm(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(card, "WaveArtist", 0, 1, 1, &pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_waveartist_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_waveartist_capture_ops); + + /* global setup */ + pcm->private_data = chip; + pcm->info_flags = 0; + if (chip->dma_playback == chip->dma_capture) + pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX; + strcpy(pcm->name, card->shortname); + chip->pcm = pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), 64 * 1024, + chip->dma_playback > 3 || chip->dma_capture > 3 ? + 128 * 1024 : 64 * 1024); + + return 0; +} + +static void snd_waveartist_free(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + + release_and_free_resource(chip->res_port); + if (chip->irq >= 0) + free_irq(chip->irq, card); + if (chip->dma_playback >= 0) { + disable_dma(chip->dma_playback); + free_dma(chip->dma_playback); + } + if (chip->dma_capture >= 0 && chip->dma_capture != chip->dma_playback) { + disable_dma(chip->dma_capture); + free_dma(chip->dma_capture); + } +} + +static int snd_waveartist_dev_free(struct snd_device *device) +{ + snd_waveartist_free(device->card); + + return 0; +} + +static int snd_waveartist_init(struct snd_waveartist *chip) +{ + if (waveartist_reset(chip)) + return -ENODEV; + dev_dbg(chip->card->dev, "chip rev. 0x%x\n", waveartist_getrev(chip)); + + waveartist_cmd1(chip, WACMD_RST_MIXER); /* reset mixer */ + waveartist_iack(chip); /* clear any pending interrupt */ + waveartist_set_ctlr(chip, 0, DMA1_IE | DMA0_IE); /* enable DMA IRQs */ + waveartist_iack(chip); /* clear any pending interrupt */ + + return 0; +} + +static int snd_waveartist_new_device(struct snd_card *card, + unsigned long port, + unsigned long mpu_port, + unsigned long fm_port, + int irq, int dma1, int dma2) +{ + struct snd_waveartist *chip = card->private_data; + static struct snd_device_ops ops = { + .dev_free = snd_waveartist_dev_free, + }; + int err; + + spin_lock_init(&chip->reg_lock); + chip->card = card; + chip->port = port; + chip->irq = -1; + chip->dma_playback = -1; + chip->dma_capture = -1; + + chip->res_port = request_region(port, 16, "WaveArtist"); + if (!chip->res_port) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab ports 0x%lx-0x%lx\n", + port, port + 16 - 1); + return -EBUSY; + } + + if (snd_waveartist_init(chip) < 0) { + snd_waveartist_free(card); + return -ENODEV; + } + + if (request_irq(irq, snd_waveartist_interrupt, 0, "WaveArtist", card)) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab IRQ %d\n", irq); + return -EBUSY; + } + chip->irq = irq; + + if (dma1 == dma2 || dma1 == SNDRV_AUTO_DMA || dma2 == SNDRV_AUTO_DMA) { + /* we have only one DMA channel */ + if (dma1 == SNDRV_AUTO_DMA) + dma1 = dma2; + if (request_dma(dma1, "WaveArtist")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab DMA %d\n", + dma1); + return -EBUSY; + } + chip->dma_playback = chip->dma_capture = dma1; + } else { + /* + * 2 channels: use 16-bit for playback and 8-bit for capture as + * full-duplex works better this way. However, the chip seems to + * have some band-width limit so full-duplex at 44kHz/16-bit/2ch + * is not possible - some frames are dropped during capture, + * resulting in too-fast recording. If capture is done using + * lower rate or 8-bit or mono, everything is fine. + */ + if (dma2 > 3) + swap(dma1, dma2); + + if (request_dma(dma1, "WaveArtist playback")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab playback DMA %d\n", + dma1); + return -EBUSY; + } + chip->dma_playback = dma1; + + if (dma2 != dma1 && request_dma(dma2, "WaveArtist capture")) { + snd_waveartist_free(card); + dev_err(chip->card->dev, "unable to grab capture DMA %d\n", + dma2); + return -EBUSY; + } + chip->dma_capture = dma2; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_waveartist_free(card); + return err; + } + + return 0; +} + +static int snd_waveartist_card_new(struct device *pdev, int dev, + struct snd_card **cardp) +{ + return snd_card_new(pdev, index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_waveartist), cardp); +} + +static int snd_waveartist_info_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + + return 0; +} + +static int snd_waveartist_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u16 val; + + val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | reg << 8); + ucontrol->value.integer.value[0] = (val >> shift) & mask; + + return 0; +} + +static int snd_waveartist_put_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u16 val, old_val, new_val; + + val = (ucontrol->value.integer.value[0] & mask); + mask <<= shift; + val <<= shift; + + old_val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | reg << 8); + new_val = (old_val & ~mask) | (val & mask); + + if (new_val == old_val) + return 0; + + /* new_val already contains register number (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_val, new_val); + + return 1; +} + +static int snd_waveartist_info_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask ? mask : 0x7fff; + + return 0; +} + +static int snd_waveartist_get_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x0f; + int shift_right = (kcontrol->private_value >> 20) & 0x0f; + int mask = (kcontrol->private_value >> 24) & 0xff; + u16 left, right; + + if (mask == 0) + mask = 0x7fff; + + left = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | left_reg << 8); + if (left_reg != right_reg) + right = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | + right_reg << 8); + else + right = left; + + ucontrol->value.integer.value[0] = (left >> shift_left) & mask; + ucontrol->value.integer.value[1] = (right >> shift_right) & mask; + + return 0; +} + +static int snd_waveartist_put_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x0f; + int shift_right = (kcontrol->private_value >> 20) & 0x0f; + int mask = (kcontrol->private_value >> 24) & 0xff; + u16 val1, val2, mask1, mask2; + u16 old_left, old_right, new_left, new_right; + + if (mask == 0) + mask = 0x7fff; + + val1 = ucontrol->value.integer.value[0] & mask; + val2 = ucontrol->value.integer.value[1] & mask; + val1 <<= shift_left; + val2 <<= shift_right; + mask1 = mask << shift_left; + mask2 = mask << shift_right; + + old_left = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | left_reg << 8); + if (left_reg != right_reg) + old_right = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | + right_reg << 8); + else + old_right = old_left; + + new_left = (old_left & ~mask1) | (val1 & mask1); + new_right = (old_right & ~mask2) | (val2 & mask2); + + if (new_left == old_left && new_right == old_right) + return 0; + + if (left_reg < 10) + /* new_* already contain reg. num. (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_left, new_right); + else + waveartist_cmd3(chip, WACMD_SET_LEVEL | + ((left_reg - 10) / 2) << 8, + new_left, new_right); + + return 1; +} + +static int snd_waveartist_info_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const mux_texts[] = { + "None", "Mix", "Line", "Phone", "CD", "Mic" + }; + + return snd_ctl_enum_info(uinfo, 2, ARRAY_SIZE(mux_texts), mux_texts); +} + +static int snd_waveartist_get_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + int mux = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | 0x08 << 8); + + ucontrol->value.enumerated.item[0] = mux & 0x07; + ucontrol->value.enumerated.item[1] = (mux >> 3) & 0x07; + + return 0; +} + +static int snd_waveartist_put_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_waveartist *chip = snd_kcontrol_chip(kcontrol); + u16 old_val, new_val; + u16 val1 = ucontrol->value.enumerated.item[0]; + u16 val2 = ucontrol->value.enumerated.item[1]; + + old_val = waveartist_cmd1_r(chip, WACMD_GET_LEVEL | 0x08 << 8); + new_val = (old_val & ~0x3f) | (val1 & 0x07) | ((val2 & 0x07) << 3); + if (new_val == old_val) + return 0; + + /* new_val already contains register number (bits 12..15), bit 11 set */ + waveartist_cmd3(chip, WACMD_SET_MIXER, new_val, new_val); + + return 1; +} + +#define WAVEARTIST_SINGLE(xname, reg, shift, mask) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_waveartist_info_single, \ + .get = snd_waveartist_get_single, .put = snd_waveartist_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) \ +} + +#define WAVEARTIST_DOUBLE(xname, left_reg, right_reg, shift_left, shift_right, \ + mask) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_waveartist_info_double, \ + .get = snd_waveartist_get_double, .put = snd_waveartist_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 20) | (mask << 24) \ +} + +static struct snd_kcontrol_new snd_waveartist_controls[] = { +WAVEARTIST_DOUBLE("Master Playback Volume", 2, 6, 1, 1, 0x07), +WAVEARTIST_DOUBLE("Master Playback Switch", 0, 4, 0, 0, 1), +WAVEARTIST_DOUBLE("PCM Playback Volume", 10, 11, 0, 0, 0x00),/* 0x00 = 0x7fff */ +WAVEARTIST_DOUBLE("FM Playback Volume", 12, 13, 0, 0, 0x00),/* 0x00 = 0x7fff */ +/* WAVEARTIST_DOUBLE("Wavetable? Playback Volume", 14, 15, 0, 0, 0x00), */ +/* WAVEARTIST_DOUBLE("? Playback Volume", 16, 17, 0, 0, 0x00), */ +/* WAVEARTIST_DOUBLE("? Playback Volume", 18, 19, 0, 0, 0x00), */ +WAVEARTIST_DOUBLE("Digital Playback Switch", 3, 7, 10, 10, 1), /* PCM + FM */ +WAVEARTIST_DOUBLE("CD Playback Volume", 0, 4, 1, 1, 0x1f), +WAVEARTIST_DOUBLE("CD Playback Switch", 3, 7, 6, 6, 1), +WAVEARTIST_DOUBLE("Line Playback Volume", 0, 4, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Line Playback Switch", 3, 7, 4, 4, 1), +WAVEARTIST_DOUBLE("Phone Playback Volume", 1, 5, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Phone Playback Switch", 3, 7, 5, 5, 1), +WAVEARTIST_SINGLE("Mono Playback Volume", 8, 5, 0x1f), +WAVEARTIST_DOUBLE("Mono Playback Switch", 3, 7, 9, 9, 1), +WAVEARTIST_DOUBLE("Mic Playback Volume", 2, 6, 6, 6, 0x1f), +WAVEARTIST_DOUBLE("Mic 2 Playback Volume", 1, 5, 1, 1, 0x1f), +WAVEARTIST_DOUBLE("Mic Playback Switch", 3, 7, 7, 7, 1), +WAVEARTIST_DOUBLE("Mic 2 Playback Switch", 3, 7, 8, 8, 1), +WAVEARTIST_DOUBLE("Mic Gain", 2, 6, 4, 4, 0x03), +WAVEARTIST_DOUBLE("Capture Volume", 3, 7, 0, 0, 0x0f), +WAVEARTIST_SINGLE("Mono Output Playback Switch", 1, 0, 1), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_waveartist_info_mux, + .get = snd_waveartist_get_mux, + .put = snd_waveartist_put_mux, +} +}; + +static int snd_waveartist_mixer(struct snd_card *card) +{ + struct snd_waveartist *chip = card->private_data; + int err; + unsigned int idx; + + strcpy(card->mixername, chip->pcm->name); + + for (idx = 0; idx < ARRAY_SIZE(snd_waveartist_controls); idx++) { + struct snd_kcontrol *kctl; + + kctl = snd_ctl_new1(&snd_waveartist_controls[idx], chip); + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + } + + return 0; +} + +static int snd_waveartist_probe(struct snd_card *card, int dev) +{ + struct snd_waveartist *chip = card->private_data; + struct snd_opl3 *opl3; + int err; + + err = snd_waveartist_new_device(card, + port[dev], midi_port[dev], fm_port[dev], + irq[dev], dma1[dev], dma2[dev]); + if (err < 0) + return err; + + strcpy(card->driver, "WaveArtist"); + strcpy(card->shortname, "Rockwell WaveArtist RWA010"); + sprintf(card->longname, "%s at 0x%lx, irq %d, dma1 %d, dma2 %d", + card->shortname, chip->port, irq[dev], dma1[dev], dma2[dev]); + + err = snd_waveartist_pcm(card); + if (err < 0) + return err; + err = snd_waveartist_mixer(card); + if (err < 0) + return err; + + if (fm_port[dev] >= 0x388 && fm_port[dev] < 0x3f0) { + err = snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, + OPL3_HW_OPL3, 0, &opl3); + if (err < 0) + return err; + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + return err; + err = snd_opl3_hwdep_new(opl3, 0, 1, &chip->synth); + if (err < 0) + return err; + } + if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x3f0) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_WAVEARTIST, + midi_port[dev], MPU401_INFO_IRQ_HOOK, + -1, &chip->rmidi); + if (err < 0) + return err; + } + + return snd_card_register(card); +} + +static int snd_waveartist_isa_match(struct device *pdev, + unsigned int dev) +{ + if (!enable[dev]) + return 0; +#ifdef CONFIG_PNP + if (isapnp[dev]) + return 0; +#endif + if (port[dev] == SNDRV_AUTO_PORT) { + dev_err(pdev, "specify port\n"); + return 0; + } + if (irq[dev] == SNDRV_AUTO_IRQ) { + dev_err(pdev, "specify irq\n"); + return 0; + } + if (dma1[dev] == SNDRV_AUTO_DMA) { + dev_err(pdev, "specify dma1\n"); + return 0; + } + + return 1; +} + +static int snd_waveartist_isa_probe(struct device *pdev, + unsigned int dev) +{ + struct snd_card *card; + int err; + + err = snd_waveartist_card_new(pdev, dev, &card); + if (err < 0) + return err; + err = snd_waveartist_probe(card, dev); + if (err < 0) { + snd_card_free(card); + return err; + } + dev_set_drvdata(pdev, card); + + return 0; +} + +static int snd_waveartist_isa_remove(struct device *devptr, + unsigned int dev) +{ + snd_card_free(dev_get_drvdata(devptr)); + + return 0; +} + +#ifdef CONFIG_PM +static int snd_waveartist_isa_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + return snd_waveartist_suspend(dev_get_drvdata(dev), state); +} + +static int snd_waveartist_isa_resume(struct device *dev, unsigned int n) +{ + return snd_waveartist_resume(dev_get_drvdata(dev)); +} +#endif + +static struct isa_driver waveartist_isa_driver = { + .match = snd_waveartist_isa_match, + .probe = snd_waveartist_isa_probe, + .remove = snd_waveartist_isa_remove, +#ifdef CONFIG_PM + .suspend = snd_waveartist_isa_suspend, + .resume = snd_waveartist_isa_resume, +#endif + .driver = { .name = "waveartist" }, +}; + +#ifdef CONFIG_PNP +static int snd_waveartist_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + static int dev; + int err; + struct snd_card *card; + + for (; dev < SNDRV_CARDS; dev++) { + if (enable[dev] && isapnp[dev]) + break; + } + if (dev >= SNDRV_CARDS) + return -ENODEV; + + err = snd_waveartist_card_new(&pcard->card->dev, dev, &card); + if (err < 0) + return err; + err = snd_waveartist_pnp(dev, card->private_data, pcard, pid); + if (err < 0) { + dev_err(&pcard->card->dev, "PnP detection failed\n"); + snd_card_free(card); + return err; + } + err = snd_waveartist_probe(card, dev); + if (err < 0) { + snd_card_free(card); + return err; + } + pnp_set_card_drvdata(pcard, card); + dev++; + + return 0; +} + +static void snd_waveartist_pnp_remove(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + /* disable forced IRQs */ + snd_waveartist_set_irq(chip->wa, 0); + snd_waveartist_set_irq(chip->mpu, 0); + + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); +} + +#ifdef CONFIG_PM +static int snd_waveartist_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + snd_waveartist_suspend(card, state); + + /* disable forced IRQs to prevent opps */ + snd_waveartist_set_irq(chip->wa, 0); + snd_waveartist_set_irq(chip->mpu, 0); + + return 0; +} +static int snd_waveartist_pnp_resume(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_waveartist *chip = card->private_data; + + /* re-enable forced IRQs */ + snd_waveartist_set_irq(chip->wa, chip->irq); + snd_waveartist_set_irq(chip->mpu, chip->irq); + + return snd_waveartist_resume(pnp_get_card_drvdata(pcard)); +} +#endif + +static struct pnp_card_driver waveartist_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = "waveartist", + .id_table = snd_waveartist_pnpids, + .probe = snd_waveartist_pnp_detect, + .remove = snd_waveartist_pnp_remove, +#ifdef CONFIG_PM + .suspend = snd_waveartist_pnp_suspend, + .resume = snd_waveartist_pnp_resume, +#endif +}; +#endif /* CONFIG_PNP */ + +static int __init alsa_card_waveartist_init(void) +{ + int err; + + err = isa_register_driver(&waveartist_isa_driver, SNDRV_CARDS); +#ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&waveartist_pnpc_driver); + if (!err) + pnp_registered = 1; + + if (isa_registered) + err = 0; +#endif + return err; +} + +static void __exit alsa_card_waveartist_exit(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&waveartist_pnpc_driver); + + if (isa_registered) +#endif + isa_unregister_driver(&waveartist_isa_driver); +} + +module_init(alsa_card_waveartist_init) +module_exit(alsa_card_waveartist_exit) diff --git a/sound/isa/waveartist.h b/sound/isa/waveartist.h new file mode 100644 index 0000000..72254cf --- /dev/null +++ b/sound/isa/waveartist.h @@ -0,0 +1,74 @@ +/* + * Rockwell WaveArtist RWA010 chip register definitions + * + * Based on OSS WaveArtist driver by Hannu Savolainen + */ + +/* registers */ +#define CMDR 0 /* command (16-bit) */ +#define DATR 2 /* data (for PIO?) (16-bit) */ +#define CTLR 4 /* control */ +#define STATR 5 /* status */ +#define EXPCR1 6 /* expansion control 1 */ +#define EXPCR2 7 /* expansion control 2 */ +#define EXPDAT1 8 /* expansion data 1 (16-bit) */ +#define EXPDAT2 10 /* expansion data 2 (16-bit) */ +#define IRQSTAT 12 /* IRQ status */ + +/* STATR register bit definitions */ +#define CMD_WE BIT(7) /* CMDR write empty (ready for write) */ +#define CMD_RF BIT(6) /* CMDR read full (ready for read) */ +#define DAT_WE BIT(5) /* DATR write empty (ready for write) */ +#define DAT_RF BIT(4) /* DATR read full (ready for read) */ +#define IRQ_REQ BIT(3) /* IRQ was requested */ +#define DMA1 BIT(2) /* DMA1 IRQ was requested */ +#define DMA0 BIT(1) /* DMA0 IRQ was requested */ + +/* CTLR register bit definitions */ +#define CMD_WEIE BIT(7) /* CMDR write empty interrupt enable */ +#define CMD_RFIE BIT(6) /* CMDR read full interrupt enable */ +#define DAT_WEIE BIT(5) /* DATR write empty interrupt enable */ +#define DAT_RFIE BIT(4) /* DATR read full interrupt enable */ +#define RESET BIT(3) /* chip reset */ +#define DMA1_IE BIT(2) /* DMA1 interrupt enable */ +#define DMA0_IE BIT(1) /* DMA0 interrupt enable */ +#define IRQ_ACK BIT(0) /* IRQ acknowlege */ + +/* IRQSTAT register bit definitions */ +#define IRQ_MPU BIT(2) /* MPU-401 */ +#define IRQ_SB BIT(1) /* SB emulation */ +#define IRQ_PCM BIT(0) /* native PCM */ + +/* commands */ +#define WACMD_GETREV 0x00 + +#define WACMD_INPUTFORMAT 0x10 /* 0=S8, 1=S16, 2=U8 */ +#define WACMD_INPUTCHANNELS 0x11 /* 1=mono, 2=Stereo */ +#define WACMD_INPUTSPEED 0x12 /* sampling rate */ +#define WACMD_INPUTDMA 0x13 /* 0=8bit, 1=16bit, 2=PIO */ +#define WACMD_INPUTSIZE 0x14 /* samples to interrupt */ +#define WACMD_INPUTSTART 0x15 /* start ADC */ +#define WACMD_INPUTPAUSE 0x16 /* pause ADC */ +#define WACMD_INPUTSTOP 0x17 /* stop ADC */ +#define WACMD_INPUTRESUME 0x18 /* resume ADC */ +#define WACMD_INPUTPIO 0x19 /* PIO ADC */ + +#define WACMD_OUTPUTFORMAT 0x20 /* 0=S8, 1=S16, 2=U8 */ +#define WACMD_OUTPUTCHANNELS 0x21 /* 1=mono, 2=stereo */ +#define WACMD_OUTPUTSPEED 0x22 /* sampling rate */ +#define WACMD_OUTPUTDMA 0x23 /* 0=8bit, 1=16bit, 2=PIO */ +#define WACMD_OUTPUTSIZE 0x24 /* samples to interrupt */ +#define WACMD_OUTPUTSTART 0x25 /* start DAC */ +#define WACMD_OUTPUTPAUSE 0x26 /* pause DAC */ +#define WACMD_OUTPUTSTOP 0x27 /* stop DAC */ +#define WACMD_OUTPUTRESUME 0x28 /* resume DAC */ +#define WACMD_OUTPUTPIO 0x29 /* PIO DAC */ + +#define WACMD_GET_LEVEL 0x30 /* read mixer reg */ +#define WACMD_SET_LEVEL 0x31 /* set mixer regs (10..19) */ +#define WACMD_SET_MIXER 0x32 /* set mixer regs (0..9) */ +#define WACMD_RST_MIXER 0x33 /* mixer reset */ +#define WACMD_SET_MONO 0x34 /* set mono mode (|0x000=L, |0x100=R) */ + +enum wa_format { WA_FMT_S8 = 0, WA_FMT_S16, WA_FMT_U8 }; +enum wa_dma { WA_DMA_8BIT = 0, WA_DMA_16BIT, WA_DMA_PIO }; -- Ondrej Zary

1 1

[alsa-devel] [PATCH 0/7 linux-next] sound: constify of_device_id array
by Fabian Frederick 22 Mar '15

22 Mar '15

This small patchset adds const to of_device_id arrays in sound branch. Fabian Frederick (7): ALSA: aoa: constify of_device_id array ASoC: fsl: constify of_device_id array ASoC: kirkwood: constify of_device_id array ASoC: rt5631: constify of_device_id array ASoC: ak4554: constify of_device_id array ASoC: fsi: constify of_device_id array ASoC: rsnd: constify of_device_id array sound/aoa/soundbus/i2sbus/core.c | 2 +- sound/soc/codecs/ak4554.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/sh/fsi.c | 4 ++-- sound/soc/sh/rcar/core.c | 2 +- 9 files changed, 10 insertions(+), 10 deletions(-) -- 1.9.1

3 14

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Peter Ujfalusi 22 Mar '15

22 Mar '15

On 03/19/2015 04:34 PM, Greg Knight wrote: >> SO is it so that the codec's MCLK is coming from McASP AHCLKX (or R) and this >> clock need to be present all the time? W/o the MCLK the registers are not >> accessible? > > Correct, the SGTL5000's I2C registers are not accessible without MCLK. Yeah, according to the datasheet, it needs MCLK in order to communicate with it. > From my testing, the clock need only be present when the audio chip is > in use. When not in use, the chip appears perfectly happy to not be > clocked. If you change audio controls while you don't have audio activity, you will still need to have the MCLK running. > Is there a better place to pull a ~12MHz clock signal for use as a > master clock from an AM335X? For a codec such as the SGTL5000 I would connect a clock, which is running all the time. If you take a look at the driver, it enables the clock at probe and leaves it running as long as the driver is loaded. But AM335x has CLKOUT1,2 outputs as well. They are not high quality clocks, but can be enabled/disabled. >> This will not work in mainline kernel. The McASP is turned off when it is not >> in use (no audio activity) so the AHCLKX/R is not going to be there on the >> ball output. > > The clock-to-talk hack in the patch enables the AHCLKX line during > codec initialization. From my testing, we don't seem to spend much > time communicating with the device when we're not using it. Disabling > the clock when it's not used should not be a problem. As I said, this is not going to work in upstream kernel. The McASP is not even powered up during probe time anymore. It is only powered up when it is needed, this means that you will not have clocks for the codec coming from McASP pin. The driver is not designed to support such a use case, when you use one of it's clock as a master clock for external chip. In other words the McASP driver is not a clock driver. It might be possible to create a clock driver on top of the McASP driver as child, but it is really something which should not be supported. > >> How do you configure the frequency of the AHCLKX/R, this hack only enables it, >> so you are going to have _some_ clock going out. > > The frequency is the 12 MHz (I think) I get out from AHCLKX without > any further configuration; I have not investigated changing that > frequency as it suited my needs. But I can look into doing > set_clkdiv/set_sysclk during the clock-to-talk enable if that's more > appropriate? > >> You should be using set_clkdiv and set_sysclk calls from the machine driver to >> configure the AHCLKX (we do not have AHCLKR supported in the driver ATM). > > So, ehm, the SGTL5000 component logic (is this the machine driver?) > should adjust the McASP's clocks? Am I misunderstanding? Would this > not impact binding the SGTL5000 to devices that are *not* McASPs/break > abstraction? > > TBH I'm not clear on how the SGTL5000 bindings to other hardware work. It works in other HW because the designers connected the SGTL5000's MCLK to a clock source which can be controlled or it is running all the time. In your case the designers chosen a clock line, which is not working as 'normal' clock. > Is it safe to assume set_sysclk is always going to adjust the I2S MCLK > speed? Please pardon my crude questions, I'm not deeply familiar with > the structure of ALSA. With the setclkdiv/sysclk you can configure the AHCLKX direction and configure the divider to be used to generate the clock which goes out, or to divide down the clock which is coming in. Another issue this hack has is: what happens if for some reason the SGTL5000 driver loads first? McASP will not enable the AHCLKX so the codec driver will fail reading the CHIP_ID and fails to load. You need to provide a dummy clock node to the codec, so that will not fail, but you will not have the MCLK. > I'll reformat the patches and try from a mainline kernel when I get > some spare time. I'm not sure if this can be accepted, sorry. It is really a hack and there are too many corner cases where the system can fail. -- Péter

3 4

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Mark Brown 22 Mar '15

22 Mar '15

On Thu, Mar 19, 2015 at 01:48:08PM -0400, Greg Knight wrote: > Is there a straightforward way we could enable the clock on-demand - > when we're communicating with the codec/mixer or actually using the > audio subsystem - while keeping it disabled when not in use? We don't > expect to be using the audio subsystem frequently. This is what the clock API is there for - the driver can enable the clock whenver it needs it and disable it when it doesn't.

1 0

[alsa-devel] [PATCH] ASoC: Intel: remove misleading DMA error messages on Baytrail platforms
by Pierre-Louis Bossart 22 Mar '15

22 Mar '15

During probe, the Baytrail audio driver reports errors such as: [44.172040] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [44.172137] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed Those error messages are misleading, there is no error since the DMA is explicitly not configured for Baytrail. Add a test to remove DMA error checks when DMA is not configured and return silently. Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- With Mark in cc: this time, sorry for the noise. sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-dsp.h | 1 + sound/soc/intel/sst-firmware.c | 4 ++++ 3 files changed, 6 insertions(+) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index b3d8456..42f293f 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -142,6 +142,7 @@ static int sst_acpi_probe(struct platform_device *pdev) sst_acpi->desc = desc; sst_acpi->mach = mach; + sst_pdata->resindex_dma_base = desc->resindex_dma_base; if (desc->resindex_dma_base >= 0) { sst_pdata->dma_engine = desc->dma_engine; sst_pdata->dma_base = desc->resindex_dma_base; diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32..148d8c5 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -206,6 +206,7 @@ struct sst_pdata { const struct firmware *fw; /* DMA */ + int resindex_dma_base; /* other fields invalid if equals to -1 */ u32 dma_base; u32 dma_size; int dma_engine; diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5f71ef6..4a993d1 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -271,6 +271,10 @@ int sst_dma_new(struct sst_dsp *sst) const char *dma_dev_name; int ret = 0; + if (sst->pdata->resindex_dma_base == -1) + /* DMA is not used, return and squelsh error messages */ + return 0; + /* configure the correct platform data for whatever DMA engine * is attached to the ADSP IP. */ switch (sst->pdata->dma_engine) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC:pcm512x: Fix divide by zero issue.
by Howard Mitchell 22 Mar '15

22 Mar '15

If den=1 and pllin_rate>20MHz then den and num are adjusted to 0 causing a divide by zero error a few lines further on. Therefore this patch correctly scales num and den such that pllin_rate/den < 20MHz as required in the device data sheet. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- sound/soc/codecs/pcm512x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5301c4a..8472099 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -713,8 +713,8 @@ static int pcm512x_find_pll_coeff(struct snd_soc_dai *dai, /* pllin_rate / P (or here, den) cannot be greater than 20 MHz */ if (pllin_rate / den > 20000000 && num < 8) { - num *= 20000000 / (pllin_rate / den); - den *= 20000000 / (pllin_rate / den); + num *= DIV_ROUND_UP(pllin_rate / den, 20000000); + den *= DIV_ROUND_UP(pllin_rate / den, 20000000); } dev_dbg(dev, "num / den = %lu / %lu\n", num, den); -- 1.7.9.5

3 2

Re: [alsa-devel] dummy codec and the dummy platform use the same name: snd-soc-dummy
by Mark Brown 22 Mar '15

22 Mar '15

On Fri, Mar 20, 2015 at 11:27:35PM -0700, Zhao Ye wrote: > Then in > static void soc_init_codec_debugfs(struct snd_soc_codec *codec) > and, > static void soc_init_platform_debugfs(struct snd_soc_platform *platform) > this two place will make two debugfs_create_dir() with the " snd-soc-dummy" name twice. > So I think we should change one of the names. Well, upstream what I'd expect to be happening here is that we merge everything into a single dummy component which avoids the problem another way.

1 0

Re: [alsa-devel] [PATCH] ASoC: wm8741: Fix rates constraints values
by Charles Keepax 22 Mar '15

22 Mar '15

On Fri, Mar 13, 2015 at 10:20:14PM +0100, Sergej Sawazki wrote: > According to table 11 in the datasheet (Rev 4.3): > Example: With a master clock (MCLK) of 11.2896MHz, the following sampling rates (LRCLK) are supported: > 44.1kHz (MCLK/256), 88.2kHz (MCLK/128) > > For the rates lists, we should use 82000 instead of 88235 and 176400 instead of 1764000 (seems to be a typo). > > Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> > --- Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Thanks, Charles

2 1

[alsa-devel] [PATCH 1/2] ALSA:ControlName.txt: Add 'Analogue' as a prefix for volume controls
by Howard Mitchell 21 Mar '15

21 Mar '15

Currently 'Playback Volume' is the correct way to express an analogue volume control. However, this control name has initialisation defaults applied when using 'alsactl restore' and in some cases this is not appropriate. An example would be a control that has a selection of 0db and -6dB of gain that is intended to set the fullscale ouput voltage of a DAC. The TI pcm512x family of DAcs have such a control. In this case the device/driver reset defaults are preferred. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- Documentation/sound/alsa/ControlNames.txt | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index 79a6127..3fc1cf50 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -71,11 +71,11 @@ SOURCE: HDMI/DP (either HDMI or DisplayPort) Exceptions (deprecated): - [Digital] Capture Source - [Digital] Capture Switch (aka input gain switch) - [Digital] Capture Volume (aka input gain volume) - [Digital] Playback Switch (aka output gain switch) - [Digital] Playback Volume (aka output gain volume) + [Analogue|Digital] Capture Source + [Analogue|Digital] Capture Switch (aka input gain switch) + [Analogue|Digital] Capture Volume (aka input gain volume) + [Analogue|Digital] Playback Switch (aka output gain switch) + [Analogue|Digital] Playback Volume (aka output gain volume) Tone Control - Switch Tone Control - Bass Tone Control - Treble -- 1.7.9.5

3 3

[alsa-devel] [PATCH 0/4] More aggressive PM for HD-audio
by Takashi Iwai 21 Mar '15

21 Mar '15

Hi, here is a patchset for supporting more aggressive PM for HD-audio. This allows to change the power state of each widget more dynamically with jack and stream states. It's activated only when the codec driver (or via sysfs or f/w patch) sets codec->power_mgmt flag. In theory, this should work for the recent Realtek codecs, but currently I have no machine for test. David, could you or your team check whether this works for ALC282 or such? Just add like: --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5415,6 +5415,7 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; + codec->power_mgmt = 1; snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); The patchset is for for-next branch of sound git tree, but they might be applicable to 4.0-rc (or even older), too. The current patches are found in topic/hda-power branch. thanks, Takashi === Takashi Iwai (4): ALSA: hda - Simplify PCM setup overrides ALSA: hda - Support advanced power state controls ALSA: hda - Use the new power control for VIA codecs ALSA: hda - Adjust power of beep widget and outputs sound/pci/hda/hda_beep.c | 29 +- sound/pci/hda/hda_beep.h | 1 + sound/pci/hda/hda_codec.c | 4 + sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/hda_generic.c | 480 ++++++++++++++++++++++++------ sound/pci/hda/hda_generic.h | 5 +- sound/pci/hda/patch_realtek.c | 41 --- sound/pci/hda/patch_sigmatel.c | 5 + sound/pci/hda/patch_via.c | 662 +---------------------------------------- 9 files changed, 427 insertions(+), 802 deletions(-) -- 2.3.3

3 11

[alsa-devel] ASoC: Max98090 Codec DAPM Widget setup in Open Embedded
by C. Omer Rafique 20 Mar '15

20 Mar '15

I would like to know which file I can use to connect/enable DPAM widgets permanently so a simple aplay/record can work? My environment is: - Yocto Linux Kernel 3.10 - (DTS based) - Max98090 Codec - imx6SL SoC I thought about doing this in alsa.conf or asound.conf but I couldn't find a single example with DAPM widgets . Thank you Choudhary

1 0

[alsa-devel] [PATCH] ALSA: hda - Rename power_mgmt flag with power_save_node
by Takashi Iwai 20 Mar '15

20 Mar '15

David suggested that the name "power_mgmt" is too ambiguous. Rename the flag with a bit clearer one "power_save_node". Also, add the corresponding description to HD-Audio.txt, too. Reported-by: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio.txt | 6 +++++- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_generic.c | 30 +++++++++++++++--------------- sound/pci/hda/patch_sigmatel.c | 8 ++++---- sound/pci/hda/patch_via.c | 12 ++++++------ 5 files changed, 31 insertions(+), 27 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39b77e6..e7193aac669c 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -466,7 +466,11 @@ The generic parser supports the following hints: - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities -- power_down_unused (bool): power down the unused widgets +- power_save_node (bool): advanced power management for each widget, + controlling the power sate (D0/D3) of each widget node depending on + the actual pin and stream states +- power_down_unused (bool): power down the unused widgets, a subset of + power_save_node, and will be dropped in future - add_hp_mic (bool): add the headphone to capture source if possible - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 148e84ce61cf..ccf355d4a8fa 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -371,7 +371,7 @@ struct hda_codec { unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ - unsigned int power_mgmt:1; /* advanced PM for each widget */ + unsigned int power_save_node:1; /* advanced PM for each widget */ #ifdef CONFIG_PM unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ atomic_t in_pm; /* suspend/resume being performed */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 94c7f43ad182..de4024db27b4 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -140,9 +140,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "single_adc_amp"); if (val >= 0) codec->single_adc_amp = !!val; - val = snd_hda_get_bool_hint(codec, "power_mgmt"); + val = snd_hda_get_bool_hint(codec, "power_save_node"); if (val >= 0) - codec->power_mgmt = !!val; + codec->power_save_node = !!val; val = snd_hda_get_bool_hint(codec, "auto_mute"); if (val >= 0) @@ -661,7 +661,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) continue; - if (codec->power_mgmt) { + if (codec->power_save_node) { if (!path->stream_enabled && !path->pin_fixed) continue; /* ignore unplugged paths except for DAC/ADC */ @@ -879,8 +879,8 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, path->active = false; /* make sure the widget is powered up */ - if (enable && (spec->power_down_unused || codec->power_mgmt)) - path_power_update(codec, path, codec->power_mgmt); + if (enable && (spec->power_down_unused || codec->power_save_node)) + path_power_update(codec, path, codec->power_save_node); for (i = path->depth - 1; i >= 0; i--) { hda_nid_t nid = path->path[i]; @@ -905,7 +905,7 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - if (!(spec->power_down_unused || codec->power_mgmt) || path->active) + if (!(spec->power_down_unused || codec->power_save_node) || path->active) return; sync_power_state_change(codec, path_power_update(codec, path, true)); } @@ -3981,7 +3981,7 @@ static hda_nid_t set_pin_power_jack(struct hda_codec *codec, hda_nid_t pin, { bool on; - if (!codec->power_mgmt) + if (!codec->power_save_node) return 0; on = snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; @@ -4038,7 +4038,7 @@ static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4067,7 +4067,7 @@ static void sync_all_pin_power_ctls(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; sync_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4111,7 +4111,7 @@ static int add_fake_beep_paths(struct hda_codec *codec) hda_nid_t nid = spec->beep_nid; int err; - if (!codec->power_mgmt || !nid) + if (!codec->power_save_node || !nid) return 0; err = add_fake_paths(codec, nid, cfg->line_outs, cfg->line_out_pins); if (err < 0) @@ -4232,7 +4232,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } set_pin_eapd(codec, nid, !mute); - if (codec->power_mgmt) { + if (codec->power_save_node) { bool on = !mute; if (on) on = snd_hda_jack_detect_state(codec, nid) @@ -4740,11 +4740,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) * @nid: audio widget * @on: power on/off flag * - * Set this in patch_ops.stream_pm. Only valid with power_mgmt flag. + * Set this in patch_ops.stream_pm. Only valid with power_save_node flag. */ void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on) { - if (codec->power_mgmt) + if (codec->power_save_node) set_path_power(codec, nid, -1, on); } EXPORT_SYMBOL_GPL(snd_hda_gen_stream_pm); @@ -4915,14 +4915,14 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused || codec->power_mgmt) + if (spec->power_down_unused || codec->power_save_node) codec->power_filter = snd_hda_gen_path_power_filter; if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); if (err < 0) return err; - if (codec->beep && codec->power_mgmt) { + if (codec->beep && codec->power_save_node) { err = add_fake_beep_paths(codec); if (err < 0) return err; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7e531d5cde51..5b7c173adcb8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4494,7 +4494,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; @@ -4600,7 +4600,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4650,7 +4650,7 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4692,7 +4692,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d5d1dca4f11b..485663bb9101 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -133,7 +133,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->gen.power_down_unused = 1; return spec; } @@ -236,7 +236,7 @@ static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = codec->power_mgmt; + ucontrol->value.enumerated.item[0] = codec->power_save_node; return 0; } @@ -247,9 +247,9 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; bool val = !!ucontrol->value.enumerated.item[0]; - if (val == codec->power_mgmt) + if (val == codec->power_save_node) return 0; - codec->power_mgmt = val; + codec->power_save_node = val; spec->gen.power_down_unused = val; analog_low_current_mode(codec); return 1; @@ -295,7 +295,7 @@ static void __analog_low_current_mode(struct hda_codec *codec, bool force) bool enable; unsigned int verb, parm; - if (!codec->power_mgmt) + if (!codec->power_save_node) enable = false; else enable = is_aa_path_mute(codec) && !spec->gen.active_streams; @@ -517,7 +517,7 @@ static int via_parse_auto_config(struct hda_codec *codec) return err; /* disable widget PM at start for compatibility */ - codec->power_mgmt = 0; + codec->power_save_node = 0; spec->gen.power_down_unused = 0; return 0; } -- 2.3.3

1 0

[alsa-devel] [PATCH] ASoC: max98925: Constify regmap config and other codec data
by Krzysztof Kozlowski 20 Mar '15

20 Mar '15

Constify local structures (snd_soc_dai_ops, snd_soc_codec_driver, regmap_config) and array (reg_defaults) which are not modified by the driver and passed to core as pointer to const. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/soc/codecs/max98925.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 1f8e80315749..0796dfaeb139 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -34,7 +34,7 @@ static const char *const hpf_text[] = { "Disable", "DC Block", "100Hz", "200Hz", "400Hz", "800Hz", }; -static struct reg_default max98925_reg[] = { +static const struct reg_default max98925_reg[] = { { 0x0B, 0x00 }, /* IRQ Enable0 */ { 0x0C, 0x00 }, /* IRQ Enable1 */ { 0x0D, 0x00 }, /* IRQ Enable2 */ @@ -491,7 +491,7 @@ static int max98925_dai_set_sysclk(struct snd_soc_dai *dai, #define MAX98925_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops max98925_dai_ops = { +static const struct snd_soc_dai_ops max98925_dai_ops = { .set_sysclk = max98925_dai_set_sysclk, .set_fmt = max98925_dai_set_fmt, .hw_params = max98925_dai_hw_params, @@ -541,7 +541,7 @@ static int max98925_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_max98925 = { +static const struct snd_soc_codec_driver soc_codec_dev_max98925 = { .probe = max98925_probe, .controls = max98925_snd_controls, .num_controls = ARRAY_SIZE(max98925_snd_controls), @@ -551,7 +551,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98925 = { .num_dapm_widgets = ARRAY_SIZE(max98925_dapm_widgets), }; -static struct regmap_config max98925_regmap = { +static const struct regmap_config max98925_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = MAX98925_REV_VERSION, -- 1.9.1

2 1

[alsa-devel] [RFC][PATCH 00/13] alsa-tools: libhinawa for control applications of FireWire devices
by Takashi Sakamoto 20 Mar '15

20 Mar '15

This is RFC for a new library into alsa-tools. ALSA in Linux 3.16 or later extends a support for FireWire sound devices. Currently ALSA drivers supports streaming functionality only, while most of these devices require software implementation to control its internal DSP. The way to achieve this is to transfer byte sequence to the unit and wait byte sequence which the unit transfers if required. This library, libhinawa, just support these operations, nothing others. The supported types of transactions are: - IEEE 1394 read transaction - IEEE 1394 write transaction - IEEE 1394 lock transaction - IEC 61883-1 FCP transaction - Echo Fireworks transaction (with a help of snd-fireworks kernel driver) - Dice notification (with a help of snd-dice kernel driver) To help new developers, this library supports GObject Introspection for language bindings. The main logic of applications can be written with preferred languages such as Python, Ruby, Perl and so on. In the end of this patchset, some Python scripts with Gtk+, Qt4 and Qt5 are committed as samples. This is my first development with GNU Autotools, GLib/GObject and GObject Introspection. Furthermore, I'm a beginner of FireWire subsystem programming. I'm happy to receive your comments, especially: - the way to write Linux version dependency in configure.ac - the way of libtool versioning - the value of poll timeout in fw_unit/snd_unit - the value of thread condition timeout in fw_fcp/snd_dice/snd_efw - improvements of 'unit_query' object - any programming mistakes (threading and so on...) Regards Takashi Sakamoto (13): libhinawa: add build definitions libhinawa: add hinawa context libhinawa: support GTK-Doc to generate references libhinawa: add 'fw_unit' object as a listener for FireWire unit libhinawa: support GObject Introspection for language bindings libhinawa: add 'fw_resp' object as a responder for FireWire transaction libhinawa: add 'fw_req' object as requester for FireWire transaction libhinawa: add 'fw_fcp' object as a helper of FCP transaction libhinawa: add 'snd_unit' object as a listener for ALSA FireWire devices libhinawa: add 'snd_dice' object as a helper for Dice notification libhinawa: add 'snd_efw' object as a helper for EFW transaction libhinawa: add 'unit_query' as a query for ALSA FireWire devices libhinawa: add sample scripts Makefile | 2 +- libhinawa/AUTHORS | 1 + libhinawa/COPYING | 504 ++++++++++++++++++++++++++++ libhinawa/ChangeLog | 5 + libhinawa/Makefile.am | 6 + libhinawa/NEWS | 0 libhinawa/README | 30 ++ libhinawa/autogen.sh | 11 + libhinawa/configure.ac | 68 ++++ libhinawa/doc/Makefile.am | 2 + libhinawa/doc/reference/Makefile.am | 46 +++ libhinawa/doc/reference/hinawa-docs.sgml | 47 +++ libhinawa/doc/reference/version.xml.in | 1 + libhinawa/samples/gtk3.py | 190 +++++++++++ libhinawa/samples/qt4.py | 206 ++++++++++++ libhinawa/samples/qt5.py | 192 +++++++++++ libhinawa/samples/run.sh | 15 + libhinawa/src/Makefile.am | 97 ++++++ libhinawa/src/backport.h | 41 +++ libhinawa/src/fw_fcp.c | 282 ++++++++++++++++ libhinawa/src/fw_fcp.h | 58 ++++ libhinawa/src/fw_req.c | 265 +++++++++++++++ libhinawa/src/fw_req.h | 56 ++++ libhinawa/src/fw_resp.c | 232 +++++++++++++ libhinawa/src/fw_resp.h | 51 +++ libhinawa/src/fw_unit.c | 343 +++++++++++++++++++ libhinawa/src/fw_unit.h | 52 +++ libhinawa/src/hinawa_context.c | 60 ++++ libhinawa/src/hinawa_context.h | 10 + libhinawa/src/internal.h | 31 ++ libhinawa/src/snd_dice.c | 195 +++++++++++ libhinawa/src/snd_dice.h | 53 +++ libhinawa/src/snd_efw.c | 315 ++++++++++++++++++ libhinawa/src/snd_efw.h | 54 +++ libhinawa/src/snd_unit.c | 548 +++++++++++++++++++++++++++++++ libhinawa/src/snd_unit.h | 69 ++++ libhinawa/src/unit_query.c | 116 +++++++ libhinawa/src/unit_query.h | 48 +++ 38 files changed, 4301 insertions(+), 1 deletion(-) create mode 100644 libhinawa/AUTHORS create mode 100644 libhinawa/COPYING create mode 100644 libhinawa/ChangeLog create mode 100644 libhinawa/Makefile.am create mode 100644 libhinawa/NEWS create mode 100644 libhinawa/README create mode 100755 libhinawa/autogen.sh create mode 100644 libhinawa/configure.ac create mode 100644 libhinawa/doc/Makefile.am create mode 100644 libhinawa/doc/reference/Makefile.am create mode 100644 libhinawa/doc/reference/hinawa-docs.sgml create mode 100644 libhinawa/doc/reference/version.xml.in create mode 100755 libhinawa/samples/gtk3.py create mode 100755 libhinawa/samples/qt4.py create mode 100755 libhinawa/samples/qt5.py create mode 100755 libhinawa/samples/run.sh create mode 100644 libhinawa/src/Makefile.am create mode 100644 libhinawa/src/backport.h create mode 100644 libhinawa/src/fw_fcp.c create mode 100644 libhinawa/src/fw_fcp.h create mode 100644 libhinawa/src/fw_req.c create mode 100644 libhinawa/src/fw_req.h create mode 100644 libhinawa/src/fw_resp.c create mode 100644 libhinawa/src/fw_resp.h create mode 100644 libhinawa/src/fw_unit.c create mode 100644 libhinawa/src/fw_unit.h create mode 100644 libhinawa/src/hinawa_context.c create mode 100644 libhinawa/src/hinawa_context.h create mode 100644 libhinawa/src/internal.h create mode 100644 libhinawa/src/snd_dice.c create mode 100644 libhinawa/src/snd_dice.h create mode 100644 libhinawa/src/snd_efw.c create mode 100644 libhinawa/src/snd_efw.h create mode 100644 libhinawa/src/snd_unit.c create mode 100644 libhinawa/src/snd_unit.h create mode 100644 libhinawa/src/unit_query.c create mode 100644 libhinawa/src/unit_query.h -- 2.1.0

5 20

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Peter Ujfalusi 20 Mar '15

20 Mar '15

Greg, On 03/19/2015 07:48 PM, Greg Knight wrote: >> For a codec such as the SGTL5000 I would connect a clock, which is > running all >> the time. If you take a look at the driver, it enables the clock at > probe and >> leaves it running as long as the driver is loaded. > > Speaking briefly to my "electrical" colleague, he's a bit concerned > about the extra power load that a crystal oscillator might incur, while > I'm concerned about audio quality from a source like CLKOUT2. > > Is there a straightforward way we could enable the clock on-demand - > when we're communicating with the codec/mixer or actually using the > audio subsystem - while keeping it disabled when not in use? We don't > expect to be using the audio subsystem frequently. > > Worst-case we can set up a GPIO and enable/disable it as needed in > user-space - acceptable, if annoying, for our application. See my reply to Nikolay. -- Péter

1 0

[alsa-devel] Can iMX6SL supports different sampling rate for different SSIs?
by Xuebing Wang 20 Mar '15

20 Mar '15

Nicolin, Thank you very much for your always help. I am using 3 SSIs (as I2S master mode) on iMX6SL for 3 different codec. 1) If I read iMX6SL reference manual correctly, I am deriving SSI sys_clock from pll4. -- pll4 = Fref * (DIV_SELECT + NUM/DENOM) ** I am using a value of 786.432000 MHz -- pll4_main_clk = pll4 / (PLL_AUDIOn: POST_DIV_SELECT) / CCM_ANALOG_MISC2n: MSB:LSB ** I am using a value of 196.608 MHz -- SSI1_CLK_ROOT = pll4_main_clk / (CS1CDR: ssi1_clk_pred) / CS1CDR: ssi1_clk_podf ** I am using a value of 3.072 MHz (for 48k sampling rate) 2) As pll4_main_clk is global for all 3 SSIs, I think that we can *not* use 8k for ssi1 and 44.1k for ssi2, right? Because 8k and 44.1k requires different pll4 clock, right? -- However, as 48k, 32k, 16k, 8k can use same pll4, thus different SSI can use different sampling rate in this subset (48k, 32k, 16k, 8k). -- Same principle applies for 44.1k, 22.050, 11.025 subset of sampling rates. Is my understanding correct? -- Xuebing

2 3

[alsa-devel] [PATCH v2.1] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 19 Mar '15

19 Mar '15

Set rule constraints to allow only combinations of sample-rate, sample-bits, and channels that can be played/captured with reasonable sample-rate accuracy. The logic with tdm-slots and serializers (=i2s data wires) goes like this: The first wire will take all channels up to number of tdm-slots, before following wires (if any) are used. If the first wire is used fully, the remaining wires share the same clocks and the divider can be calculated for the first wire. Also, takes the number of tdm-slots into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- This is othetwise the same patch as v2, but I added Peter's ack and rebased on top of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/davinci sound/soc/davinci/davinci-mcasp.c | 195 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 185 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d40b392..625cca1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> #include <linux/platform_data/davinci_asp.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -65,6 +66,11 @@ struct davinci_mcasp_context { bool pm_state; }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; @@ -99,6 +105,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -868,6 +876,30 @@ 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 + (int)div64_long(1000000LL*rem, + (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) @@ -883,16 +915,20 @@ 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 channels = params_channels(params); + int rate = params_rate(params); + int sbits = params_width(params); + int ppm, div; + + if (channels > mcasp->tdm_slots) + channels = mcasp->tdm_slots; + + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + &ppm); + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -974,6 +1010,108 @@ 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_ruledata *rd = rule->private; + int sbits = params_width(params); + int channels = params_channels(params); + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int i, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + uint bclk_freq = sbits*channels*davinci_mcasp_dai_rates[i]; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + + } + dev_dbg(rd->mcasp->dev, "%d frequencies for %d sbits and %d channels\n", + count, sbits, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_samplebits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + int channels = params_channels(params); + int rate = params_rate(params); + unsigned int list[4]; + int sbits, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (sbits = 8; sbits <= 32; sbits += 8) { + uint bclk_freq = channels*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = sbits; + } + dev_dbg(rd->mcasp->dev, + "%d possible sample size for %d Hz and %d channels\n", + count, rate, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + int sbits = params_width(params); + int rate = params_rate(params); + int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? + rd->mcasp->tdm_slots : ci->max; + unsigned int list[ci->max - ci->min + 1]; + int c1, c, count = 0; + + for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { + uint bclk_freq = c1*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + /* If we can use all tdm_slots, we can put any + amount of channels to remaining wires as + long as they fit in. */ + if (c1 == rd->mcasp->tdm_slots) { + for (c = c1; c <= rd->serializers*c1 && + c <= ci->max; c++) + list[count++] = c; + } else { + list[count++] = c1; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible channel counts for %d Hz and %d sbits\n", + count, rate, sbits); + + 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) { @@ -999,6 +1137,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1013,6 +1152,42 @@ 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) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + davinci_mcasp_hw_rule_samplebits, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + davinci_mcasp_hw_rule_channels, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

1 1

[alsa-devel] [PATCH] ASoC: Intel: remove misleading DMA error messages on Baytrail platforms
by Pierre-Louis Bossart 19 Mar '15

19 Mar '15

During probe, the Baytrail audio driver reports errors such as: [44.172040] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [44.172137] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed Those error messages are misleading, there is no error since the DMA is explicitly not configured for Baytrail. Add a test to remove DMA error checks when DMA is not configured and return silently. Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-dsp.h | 1 + sound/soc/intel/sst-firmware.c | 4 ++++ 3 files changed, 6 insertions(+) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index b3d8456..42f293f 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -142,6 +142,7 @@ static int sst_acpi_probe(struct platform_device *pdev) sst_acpi->desc = desc; sst_acpi->mach = mach; + sst_pdata->resindex_dma_base = desc->resindex_dma_base; if (desc->resindex_dma_base >= 0) { sst_pdata->dma_engine = desc->dma_engine; sst_pdata->dma_base = desc->resindex_dma_base; diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32..148d8c5 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -206,6 +206,7 @@ struct sst_pdata { const struct firmware *fw; /* DMA */ + int resindex_dma_base; /* other fields invalid if equals to -1 */ u32 dma_base; u32 dma_size; int dma_engine; diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5f71ef6..4a993d1 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -271,6 +271,10 @@ int sst_dma_new(struct sst_dsp *sst) const char *dma_dev_name; int ret = 0; + if (sst->pdata->resindex_dma_base == -1) + /* DMA is not used, return and squelsh error messages */ + return 0; + /* configure the correct platform data for whatever DMA engine * is attached to the ADSP IP. */ switch (sst->pdata->dma_engine) { -- 1.9.1

1 0

[alsa-devel] [PATCH] ALSA: jack: create jack kcontrols for every jack input device
by Jie Yang 19 Mar '15

19 Mar '15

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Currently the ALSA jack core registers only input devices for each jack registered. These jack input devices are not readable by userspace devices that run as non root. This patch adds support for additionally registering jack kcontrol devices for every input jack registered. This allows non root userspace to read jack status. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Modified-by: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reveiwed-by: Mark Brown <broonie(a)kernel.org> --- include/sound/control.h | 2 ++ include/sound/jack.h | 2 ++ sound/core/jack.c | 89 +++++++++++++++++++++++++++++++++++++++++++++++-- 3 files changed, 91 insertions(+), 2 deletions(-) mode change 100644 => 100755 sound/core/jack.c diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054..023b70a 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -66,6 +66,7 @@ struct snd_kcontrol_volatile { struct snd_kcontrol { struct list_head list; /* list of controls */ + struct list_head jack_list; /* list of controls belong to the same jack*/ struct snd_ctl_elem_id id; unsigned int count; /* count of same elements */ snd_kcontrol_info_t *info; @@ -75,6 +76,7 @@ struct snd_kcontrol { snd_kcontrol_tlv_rw_t *c; const unsigned int *p; } tlv; + unsigned int jack_bit_idx; /*the corresponding jack type bit index */ unsigned long private_value; void *private_data; void (*private_free)(struct snd_kcontrol *kcontrol); diff --git a/include/sound/jack.h b/include/sound/jack.h index 2182350..477dcb4 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -73,6 +73,8 @@ enum snd_jack_types { struct snd_jack { struct input_dev *input_dev; + struct list_head kctl_list; + struct snd_card *card; int registered; int type; const char *id; diff --git a/sound/core/jack.c b/sound/core/jack.c old mode 100644 new mode 100755 index 8658578..c067b6a --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -24,6 +24,7 @@ #include <linux/module.h> #include <sound/jack.h> #include <sound/core.h> +#include <sound/control.h> static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { SW_HEADPHONE_INSERT, @@ -54,7 +55,13 @@ static int snd_jack_dev_disconnect(struct snd_device *device) static int snd_jack_dev_free(struct snd_device *device) { struct snd_jack *jack = device->device_data; + struct snd_card *card = device->card; + struct snd_kcontrol *kctl; + list_for_each_entry(kctl, &jack->kctl_list, jack_list) { + list_del(&kctl->jack_list); + snd_ctl_remove(card, kctl); + } if (jack->private_free) jack->private_free(jack); @@ -100,6 +107,63 @@ static int snd_jack_dev_register(struct snd_device *device) return err; } + +/* get the first unused/available index number for the given kctl name */ +static int get_available_index(struct snd_card *card, const char *name) +{ + struct snd_kcontrol *kctl; + int idx = 0; + int len = strlen(name); + + down_write(&card->controls_rwsem); + next: + list_for_each_entry(kctl, &card->controls, list) { + if (!strncmp(name, kctl->id.name, len) && + !strcmp(" Jack", kctl->id.name + len) && + kctl->id.index == idx) { + idx++; + goto next; + } + } + up_write(&card->controls_rwsem); + return idx; +} + +static void kctl_private_free(struct snd_kcontrol *kctl) +{ + list_del(&kctl->jack_list); +} + +static int snd_jack_new_kctl(struct snd_card *card, struct snd_jack *jack, int type) +{ + struct snd_kcontrol *kctl; + int i, err, index, state = 0 /* use 0 for default state ?? */; + + INIT_LIST_HEAD(&jack->kctl_list); + for (i = 0; i < fls(SND_JACK_BTN_0); i++) { + int testbit = 1 << i; + if (type & testbit) { + index = get_available_index(card,jack->id); + kctl = snd_kctl_jack_new(jack->id, index, card); + if (!kctl) + return -ENOMEM; + kctl->private_free = kctl_private_free; + /* use jack_bit_idx for the kctl type bit */ + kctl->jack_bit_idx = i; + + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + list_add_tail(&kctl->jack_list, &jack->kctl_list); + + /* set initial jack state */ + snd_kctl_jack_report(card, kctl, state & testbit); + } + } + + return 0; +} + /** * snd_jack_new - Create a new jack * @card: the card instance @@ -138,7 +202,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, } jack->input_dev->phys = "ALSA"; - + jack->card = card; jack->type = type; for (i = 0; i < SND_JACK_SWITCH_TYPES; i++) @@ -150,10 +214,17 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (err < 0) goto fail_input; + /* register jack kcontrols */ + err = snd_jack_new_kctl(card, jack, type); + if (err < 0) + goto fail_kctl; + *jjack = jack; return 0; +fail_kctl: + snd_device_free(card, jack); fail_input: input_free_device(jack->input_dev); kfree(jack->id); @@ -230,6 +301,7 @@ EXPORT_SYMBOL(snd_jack_set_key); */ void snd_jack_report(struct snd_jack *jack, int status) { + struct snd_kcontrol *kctl; int i; if (!jack) @@ -245,13 +317,26 @@ void snd_jack_report(struct snd_jack *jack, int status) for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) { int testbit = 1 << i; - if (jack->type & testbit) + if (jack->type & testbit) { input_report_switch(jack->input_dev, jack_switch_types[i], status & testbit); + } } input_sync(jack->input_dev); + + for (i = 0; i < fls(SND_JACK_BTN_0); i++) { + int testbit = 1 << i; + if (jack->type & testbit) { + list_for_each_entry(kctl, &jack->kctl_list, jack_list) { + if (kctl->jack_bit_idx == i) { + snd_kctl_jack_report(jack->card, kctl, + status & testbit); + } + } + } + } } EXPORT_SYMBOL(snd_jack_report); -- 1.9.1

5 8

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Peter Ujfalusi 19 Mar '15

19 Mar '15

On 03/19/2015 07:07 AM, Greg Knight wrote: > Hi, Mark, Jaroslav, > > I've attached a couple patches to provide support for the Freescale > SGTL5000 chip when using the McASP on the AM3359. It is more correct to say that this is needed in a custom hardware which uses McASP and SGTL5000 codec. > This adds an optional hack, "clock-to-talk" to the McASP subsystem which > can start AHCLKX/AHCLKR prior to initializing the codec. This is needed > on the SGTL5000, which depends on AHCLKX for its master clock. SO is it so that the codec's MCLK is coming from McASP AHCLKX (or R) and this clock need to be present all the time? W/o the MCLK the registers are not accessible? > These patches are based off of 3.14.31-ti-r49. What is the process for > getting these merged? First off, please send the patches with git send-email so people can comment on them. This will not work in mainline kernel. The McASP is turned off when it is not in use (no audio activity) so the AHCLKX/R is not going to be there on the ball output. How do you configure the frequency of the AHCLKX/R, this hack only enables it, so you are going to have _some_ clock going out. You should be using set_clkdiv and set_sysclk calls from the machine driver to configure the AHCLKX (we do not have AHCLKR supported in the driver ATM). -- Péter

1 0

Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Mark Brown 19 Mar '15

19 Mar '15

On Thu, Mar 19, 2015 at 01:07:39AM -0400, Greg Knight wrote: > These patches are based off of 3.14.31-ti-r49. What is the process for > getting these merged? Please submit patches using the process in SubmittingPatches, you need to work against current development versions of the kernel rather than an old vendor kernel.

1 0

[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 19 Mar '15

19 Mar '15

Set rule constraints to allow only combinations of sample-rate, sample-bits, and channels that can be played/captured with reasonable sample-rate accuracy. The logic with tdm-slots and serializers (=i2s data wires) goes like this: The first wire will take all channels up to number of tdm-slots, before following wires (if any) are used. If the first wire is used fully, the remaining wires share the same clocks and the divider can be calculated for the first wire. Also, takes the number of tdm-slots into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- I hope the logic with tdm-slots and serializers is now correct. I do not have HW to test multiple serializers case, so please review davinci_mcasp_hw_rule_channels() extra carefully. Best regards, Jyri sound/soc/davinci/davinci-mcasp.c | 195 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 185 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155..3b2fb56 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/of_device.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -64,6 +65,11 @@ struct davinci_mcasp_context { u32 *xrsr_regs; /* for serializer configuration */ }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; @@ -98,6 +104,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -855,6 +863,30 @@ 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 + (int)div64_long(1000000LL*rem, + (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 +904,20 @@ 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 channels = params_channels(params); + int rate = params_rate(params); + int sbits = params_width(params); + int ppm, div; + + if (channels > mcasp->tdm_slots) + channels = mcasp->tdm_slots; + + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + &ppm); + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -973,6 +1009,108 @@ 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_ruledata *rd = rule->private; + int sbits = params_width(params); + int channels = params_channels(params); + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int i, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + uint bclk_freq = sbits*channels*davinci_mcasp_dai_rates[i]; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + + } + dev_dbg(rd->mcasp->dev, "%d frequencies for %d sbits and %d channels\n", + count, sbits, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_samplebits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + int channels = params_channels(params); + int rate = params_rate(params); + unsigned int list[4]; + int sbits, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (sbits = 8; sbits <= 32; sbits += 8) { + uint bclk_freq = channels*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = sbits; + } + dev_dbg(rd->mcasp->dev, + "%d possible sample size for %d Hz and %d channels\n", + count, rate, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + int sbits = params_width(params); + int rate = params_rate(params); + int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? + rd->mcasp->tdm_slots : ci->max; + unsigned int list[ci->max - ci->min + 1]; + int c1, c, count = 0; + + for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { + uint bclk_freq = c1*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + /* If we can use all tdm_slots, we can put any + amount of channels to remaining wires as + long as they fit in. */ + if (c1 == rd->mcasp->tdm_slots) { + for (c = c1; c <= rd->serializers*c1 && + c <= ci->max; c++) + list[count++] = c; + } else { + list[count++] = c1; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible channel counts for %d Hz and %d sbits\n", + count, rate, sbits); + + 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) { @@ -998,6 +1136,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1012,6 +1151,42 @@ 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) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + davinci_mcasp_hw_rule_samplebits, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + davinci_mcasp_hw_rule_channels, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

3 2

[alsa-devel] snd_usb_audio - siberia elite prism soundcard issues
by Daniel 19 Mar '15

19 Mar '15

Hello, First of all, since this is my first message I hope am not mistaken reporting this two issues about this piece of hardware (I was pointed here by alsa irc support channel) and bug tracker url does not seem to work, so I decided to post it here. Problem: Errors with mixer controls, breaking alsamixer, and not working 7.1 support. Scenario: Siberia Elite Prism soundcard supports 7.1 channels but it seems it can only play FRONT LEFT and FRONT RIGHT channels, no matter the setup. Here is the alsa-info script output: http://www.alsa-project.org/db/?f=301e6045129665d086d2571652c62648de0eac99 Additionally: aplay -L output: front:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Front speakers surround40:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Direct sample mixing device dsnoop:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Direct sample snooping device hw:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Direct hardware device without any conversions plughw:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Hardware device with all software conversions and amixer output for this device: # for i in {1..32}; do amixer -cElite cget numid=$i; done numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | | chmap-fixed=FL,FR numid=2,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | | chmap-fixed=FL,FR numid=3,iface=MIXER,name='Sidetone Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=4,iface=MIXER,name='Sidetone Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=56,step=0 : values=0 | dBminmax-min=-74.00dB,max=-18.00dB numid=5,iface=MIXER,name='PCM Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=6,iface=MIXER,name='PCM Volume',index=1 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=7,iface=MIXER,name='PCM Volume',index=2 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=8,iface=MIXER,name='PCM Volume',index=3 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=9,iface=MIXER,name='PCM Volume',index=4 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=10,iface=MIXER,name='Speaker Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=11,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=74,step=0 : values=54,54,54,54,54,54,54,54 | dBminmax-min=-74.00dB,max=0.00dB numid=12,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=13,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=74,step=0 : values=73,73 | dBminmax-min=-74.00dB,max=0.00dB amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 And hw-params from aplay: aplay --dump-hw-params -d1 -Dhw:Elite /dev/zero Playing raw data '/dev/zero' : Unsigned 8 bit, Rate 8000 Hz, Mono HW Params of device "hw:Elite": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE S24_3LE SUBFORMAT: STD SAMPLE_BITS: [16 24] FRAME_BITS: [32 192] CHANNELS: [2 8] RATE: [8000 96000] PERIOD_TIME: [1000 16384000] PERIOD_SIZE: [8 131072] PERIOD_BYTES: [64 524288] PERIODS: [2 1024] BUFFER_TIME: (166 32768000] BUFFER_SIZE: [16 262144] BUFFER_BYTES: [64 1048576] TICK_TIME: ALL -------------------- aplay: set_params:1233: Sample format non available Available formats: - S16_LE - S24_3LE Also in syslog I often can see messages related to this device: usb 1-1.3: 10:0: cannot get min/max values for control 3 (id 10) usb 1-1.3: 2:1: cannot get freq at ep 0x1 Loading alsamixer and trying to select the soundcard exits the program with this message: cannot load mixer controls: Invalid argument If anyone wants to check this issue, should you need any type of info regarding this, please do not hesitate asking, I will post anything you require. PS: I don't think it is really necessary to say either but I don't expect anyone to force-commit to this either, I mainly want to know if there is a solution to this or if reporting this issue I can help to get it fixed. Thanks for your time. -- *Daniel Ferradal*

3 5

[alsa-devel] [PATCH] ASoC: Intel: acpi_probe: fix error return path
by Vinod Koul 19 Mar '15

19 Mar '15

Fix the sst_acpi_probe memory allocation error path by setting right error code and initiating the cleanup insteadof just returning Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index b782dfdcdbba..b536ddd2517d 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -309,7 +309,7 @@ static int sst_acpi_probe(struct platform_device *pdev) ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), GFP_KERNEL); if (!ctx->shim_regs64) { - return -ENOMEM; + ret = -ENOMEM; goto do_sst_cleanup; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] Intel: ASoC: Add condition check before set param to waves
by han.lu＠intel.com 19 Mar '15

19 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Check waves state before set parameter through ipc to prevent unexpected operation. Also remove redundant check. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 43fb5f3..20b629a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -2055,6 +2055,11 @@ int sst_hsw_launch_param_buf(struct sst_hsw *hsw) { int ret, idx; + if (!sst_hsw_is_module_active(hsw, SST_HSW_MODULE_WAVES)) { + dev_dbg(hsw->dev, "module waves is not active\n"); + return 0; + } + /* put all param lines to DSP through ipc */ for (idx = 0; idx < hsw->param_idx_w; idx++) { ret = sst_hsw_module_set_param(hsw, diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index b40ec74..6c6229a 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -399,13 +399,9 @@ static int hsw_waves_param_put(struct snd_kcontrol *kcontrol, if (ret < 0) return ret; - if (sst_hsw_is_module_loaded(hsw, id)) { - if (!sst_hsw_is_module_active(hsw, id)) - return 0; - + if (sst_hsw_is_module_active(hsw, id)) ret = sst_hsw_module_set_param(hsw, id, 0, param_id, param_size, ucontrol->value.bytes.data); - } return ret; } -- 1.9.1

2 1

[alsa-devel] [PATCH 0/12 v2] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 19 Mar '15

19 Mar '15

Hi Mark These are v2 of DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, but it is copied from simple-card. 1 - 2 : prepare patch for Renesas new sound card driver 3 - 4 : new Renesas sound card driver 5 : tidyup for ak4642 6 - 10 : fixup/tidyup patches for rsnd 11 - 12 : DPCM support patch for rsnd Kuninori Morimoto (12): 1) ASoC: core: remove of_xlate_dai_name from snd_soc_component_driver 2) ASoC: core: add snd_soc_get_dai_info() to get driver infomation 3) ASoC: rsnd-dpcm-card: add Renesas sound card support for DPCM 4) ASoC: rsnd-dpcm-card: add .be_hw_params_fixup support for convert rate 5) ASoC: ak4642: tidyup DAPM route for playback 6) ASoC: rsnd: no more SSI restart when unusual situation 7) ASoC: rsnd: no more SRC restart when unusual situation 8) ASoC: rsnd: tidyup error message format 9) ASoC: rsnd: show debug info for sampling rate convert 10) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 11) ASoC: rsnd: add dai_link stream name 12) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsnd-dpcm-card.txt | 72 ++++++++++++++ include/sound/soc.h | 6 +- sound/soc/codecs/ak4642.c | 4 +- sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 ++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd-dpcm-card.c | 496 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 16 ++- sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 71 ++++++++------ 13 files changed, 765 insertions(+), 66 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

3 17

[alsa-devel] bug in alsa-plugins: #if SND_PCM_EXTPLUG_VERSION wrong
by Ethan Grammatikidis 18 Mar '15

18 Mar '15

in alsa-plugins 1.0.28 and 1.0.29, mix/*.c contain the following line: #if SND_PCM_EXTPLUG_VERSION >= 0x102 that 0x102 needs to be 0x10002. i have alsa-lib 1.0.26 installed. its headers define SND_PCM_EXTPLUG_VERSION to 0x10001, corresponding to version number 1.0.1. that #if effectively tests for >= 0.1.2, so attempting to build alsa-plugins 1.0.28 or 1.0.29 results in a long list of errors. SND_CHMAP_MONO, SND_CHMAP_FL, etc are undeclared and snd_pcm_chmap_query_t is an unknown type. in alsa-plugins 1.0.26 SND_PCM_EXTPLUG_VERSION is not tested. i didn't look at 1.0.27 reporting here because the bug tracker is down. -- Wir mussen wissen, wir werden wissen.

2 1

[alsa-devel] [PATCH 1/6] ESI W192M : Correct copy/paste from prodigy driver
by klem.dev＠gmail.com 18 Mar '15

18 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

3 2

[alsa-devel] [PATCH 1/2] ASoC: Intel: Remove support for Intel MID DMA from firmware loader
by Jarkko Nikula 18 Mar '15

18 Mar '15

Intel MID DMA driver is going to be removed by the coming commit 36111da7838e ("dmaengine: intel-mid-dma: remove the driver") in spi.git tree. Since there are no users for SST_DMA_TYPE_MID type the support for it can be removed from here in advance. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-dsp.h | 1 - sound/soc/intel/sst-firmware.c | 3 --- 2 files changed, 4 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32f0077..3412474083ff 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -28,7 +28,6 @@ /* Supported SST DMA Devices */ #define SST_DMA_TYPE_DW 1 -#define SST_DMA_TYPE_MID 2 /* autosuspend delay 5s*/ #define SST_RUNTIME_SUSPEND_DELAY (5 * 1000) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5e5800897da2..38881f1fb990 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -277,9 +277,6 @@ int sst_dma_new(struct sst_dsp *sst) case SST_DMA_TYPE_DW: dma_dev_name = "dw_dmac"; break; - case SST_DMA_TYPE_MID: - dma_dev_name = "Intel MID DMA"; - break; default: dev_err(sst->dev, "error: invalid DMA engine %d\n", sst->pdata->dma_engine); -- 2.1.4

3 6

[alsa-devel] [PATCH] ASoC: samsung: Enable SND_SIMPLE_CARD for Odroid X2/U3
by Sylwester Nawrocki 18 Mar '15

18 Mar '15

Odroid X2/U3 sound support can now be specified in device tree using the simple card binding, make SND_SOC_ODROIDX2 select SND_SIMPLE_CARD to ensure there are always required drivers in place. Reported-by: Tobias Jakobi <liquid.acid(a)gmx.net> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 0632a36..25caadf 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -226,6 +226,7 @@ config SND_SOC_ODROIDX2 depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S + select SND_SIMPLE_CARD help Say Y here to enable audio support for the Odroid-X2/U3. -- 1.7.9.5

2 1

Re: [alsa-devel] ASoC: Intel: mrfld- add ACPI module
by Dan Carpenter 18 Mar '15

18 Mar '15

Hello Vinod Koul, The patch 336cfbb05edf: "ASoC: Intel: mrfld- add ACPI module" from Nov 11, 2014, leads to the following static checker warning: sound/soc/intel/sst/sst_acpi.c:313 sst_acpi_probe() info: ignoring unreachable code. sound/soc/intel/sst/sst_acpi.c 304 ret = sst_context_init(ctx); 305 if (ret < 0) 306 return ret; 307 308 /* need to save shim registers in BYT */ 309 ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), 310 GFP_KERNEL); 311 if (!ctx->shim_regs64) { 312 return -ENOMEM; ^^^^^^^^^^^^^^^ 313 goto do_sst_cleanup; ^^^^^^^^^^^^^^^^^^^^ 314 } 315 316 sst_configure_runtime_pm(ctx); 317 platform_set_drvdata(pdev, ctx); 318 return ret; 319 320 do_sst_cleanup: 321 sst_context_cleanup(ctx); 322 platform_set_drvdata(pdev, NULL); 323 dev_err(ctx->dev, "failed with %d\n", ret); 324 return ret; 325 } regards, dan carpenter

2 2

[alsa-devel] [GIT PULL] ASoC fixes for v4.0
by Mark Brown 18 Mar '15

18 Mar '15

The following changes since commit 06e5801b8cb3fc057d88cb4dc03c0b64b2744cda: Linux 4.0-rc4 (2015-03-15 17:38:20 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.0-rc4 for you to fetch changes up to 33484c67398295108ab31e17b50b5a6fba1f5eef: Merge remote-tracking branches 'asoc/fix/sgtl5000' and 'asoc/fix/sn95031' into asoc-linus (2015-03-16 12:03:17 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.0 As well as the usual collection of driver specific fixes there's a few more generic things: - Lots of fixes from Takashi for drivers using the wrong field in the control union to communicate with userspace, leading to potential errors on 64 bit systems. - A fix from Lars for locking of the lists of devices we maintain, mostly only likely to trigger during device probe and removal. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt286: Change the DMI mapping for Dino Eric Nelson (1): ASoC: sgtl5000: remove useless register write clearing CHRGPUMP_POWERUP Lars-Peter Clausen (4): ASoC: ak4671: Fix control-less DAPM routes ASoC: da732x: Fix control-less DAPM routes ASoC: sn95031: Fix control-less DAPM routes ASoC: Fix component lists locking Lu, Han (1): ASoC: Intel: remove conflicts when load/unload multiple firmware images Maciej S. Szmigiero (1): fsl_ssi: fix of_property_read_u32_array return value check Mark Brown (3): Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/ak4671', 'asoc/fix/control', 'asoc/fix/da732x', 'asoc/fix/fsl-ssi', 'asoc/fix/lock' and 'asoc/fix/rt286' into asoc-linus Merge remote-tracking branches 'asoc/fix/sgtl5000' and 'asoc/fix/sn95031' into asoc-linus Nicolin Chen (1): ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation Takashi Iwai (14): ASoC: adav80x: Fix wrong value references for boolean kctl ASoC: ak4641: Fix wrong value references for boolean kctl ASoC: cs4271: Fix wrong value references for boolean kctl ASoC: es8238: Fix wrong value references for boolean kctl ASoC: pcm1681: Fix wrong value references for boolean kctl ASoC: tas5086: Fix wrong value references for boolean kctl ASoC: wm2000: Fix wrong value references for boolean kctl ASoC: wm8731: Fix wrong value references for boolean kctl ASoC: wm8903: Fix wrong value references for boolean kctl ASoC: wm8904: Fix wrong value references for boolean kctl ASoC: wm8955: Fix wrong value references for boolean kctl ASoC: wm8960: Fix wrong value references for boolean kctl ASoC: wm9712: Fix wrong value references for boolean kctl ASoC: wm9713: Fix wrong value references for boolean kctl sound/soc/codecs/adav80x.c | 4 ++-- sound/soc/codecs/ak4641.c | 4 ++-- sound/soc/codecs/ak4671.c | 44 +++++++++++++++++++-------------------- sound/soc/codecs/cs4271.c | 4 ++-- sound/soc/codecs/da732x.c | 8 +++---- sound/soc/codecs/es8328.c | 4 ++-- sound/soc/codecs/pcm1681.c | 4 ++-- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/sgtl5000.c | 8 +------ sound/soc/codecs/sn95031.c | 4 ++-- sound/soc/codecs/tas5086.c | 4 ++-- sound/soc/codecs/wm2000.c | 8 +++---- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8955.c | 4 ++-- sound/soc/codecs/wm8960.c | 4 ++-- sound/soc/codecs/wm9712.c | 6 +++--- sound/soc/codecs/wm9713.c | 6 +++--- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/intel/sst-haswell-dsp.c | 3 --- sound/soc/intel/sst-haswell-ipc.c | 32 +++++++++++++++++++++------- sound/soc/soc-core.c | 41 ++++++++++++++++++++++++++---------- 23 files changed, 118 insertions(+), 92 deletions(-)

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 18 Mar '15

18 Mar '15

Set a rule constraints to allow only combinations of sample-rate and number of frame-bits that can be played/captured with a reasonable sample-rate accuracy. Also, takes the number of serializers into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Now this works as it should. Thanks, Jyri sound/soc/davinci/davinci-mcasp.c | 150 +++++++++++++++++++++++++++++++++++--- 1 file changed, 140 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155..3ecee3b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/of_device.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -64,6 +65,11 @@ struct davinci_mcasp_context { u32 *xrsr_regs; /* for serializer configuration */ }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; @@ -98,6 +104,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -855,6 +863,30 @@ 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 + (int)div64_long(1000000LL*rem, + (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 +904,19 @@ 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); - } + uint bclk_freq = snd_soc_params_to_bclk(params); + int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + TX_MODE : RX_MODE; + int ppm, div; + + bclk_freq /= mcasp->ruledata[dir].serializers; + + div = davinci_mcasp_calc_clk_div(mcasp, bclk_freq, &ppm); + + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -973,6 +1008,74 @@ 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_ruledata *rd = 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]/ + rd->serializers; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + + } + dev_dbg(rd->mcasp->dev, "%d frequencies for %u fsize (%d)\n", + count, frame_size, rd->serializers); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_framebits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + unsigned int max_chan_per_wire = ci->max / rd->serializers; + unsigned int list[max_chan_per_wire*4]; + int rate = params_rate(params); + int chans, fbits, count = 0; + + for (chans = ci->min; chans <= max_chan_per_wire; chans++) { + for (fbits = 8; fbits <= 32; fbits += 8) { + uint bclk_freq = chans*fbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + list[count++] = chans*fbits*rd->serializers; + BUG_ON(count > ARRAY_SIZE(list)); + } + } + } + + dev_dbg(rd->mcasp->dev, + "%d possible frame size for %d Hz (%d)\n", + count, rate, rd->serializers); + + 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) { @@ -998,6 +1101,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1012,6 +1116,32 @@ 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) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + davinci_mcasp_hw_rule_framebits, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

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[alsa-devel] Support two wm8962 codec in i.mx6 board.
by Shawn Embedded 18 Mar '15

18 Mar '15

Hello everyone, I'm new to ALSA. I have a question here. Now there is a wm8962 on the i.mx6 board, running well, and we want to add an extra wm8962 to the board, connected to i.mx6 on another SSI port through a different i.mx6's AUDMUX port. The two wm8962 will work independently completely, at the same time possiblly. Before sending this mail to the list, I've already googled some materials about my requirements. I've found there is a structure named "struct snd_soc_aux_dev". For my applicantion, if I want to support another wm8962, could I can use this structure for my intention please? But in this link http://www.spinics.net/lists/alsa-devel/msg40127.html , it's said that This makes possible to register auxiliary dailess codecs in a machine driver. Term dailess is used here for amplifiers and codecs without DAI or DAI being unused. then, does it mean that I couldn't use this structure for my application? Or, I should declare another structure "struct snd_soc_card" for the second wm8962, just as the now-working wm8962? or other ways? In fact, I'm a little confused if I could declare two structure snd_soc_card at the same? If it's OK, what should I do next to meet my requirement? Could anyone give me any guidance on this? Any links, and materials will also be much appreciated. Thank you in advance. ------------------- Best Regards, Shawn

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[alsa-devel] [PATCH 6/6] ESI W192M : Add sampling rate control of the ADC/DAC
by klem.dev＠gmail.com 18 Mar '15

18 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 6d3412f..9906119 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; } @@ -388,6 +446,44 @@ 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); @@ -487,21 +583,32 @@ 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 5/6] ESI W192M : Add text Line in/Mic for selecting input gain state
by klem.dev＠gmail.com 18 Mar '15

18 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add text Line in/Mic for selecting input gain state in mixer for 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 c7ffafa..6d3412f 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -339,8 +339,14 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Line In", "Mic" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -354,7 +360,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -370,7 +376,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) @@ -411,7 +417,7 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "MIC/Line switch", + .name = "MIC/Line Input Enum", .count = 2, .info = stac9460_mic_sw_info, .get = stac9460_mic_sw_get, -- 2.1.4

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[alsa-devel] [PATCH 4/6] ESI W192M : Add TLV support for control value in dB scale
by klem.dev＠gmail.com 18 Mar '15

18 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 3/6] ESI W192M : Enable midi i/o of port envy24 chip as available
by klem.dev＠gmail.com 18 Mar '15

18 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 18 Mar '15

18 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

1 0

[alsa-devel] [PATCH v9] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 17 Mar '15

17 Mar '15

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 164 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 166 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5c0a798..a2098e4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade1..4636a05 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -941,6 +941,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ededb97..893a6e5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1230,7 +1230,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1242,7 +1243,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bc2d7a0..b0c5951 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -853,6 +853,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* create control for links with > 1 config */ + if (w->num_params <= 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. @@ -2724,6 +2754,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3198,7 +3231,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, { struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; - const struct snd_soc_pcm_stream *config = w->params; + const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; @@ -3290,8 +3323,39 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3299,14 +3363,61 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) { + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + len = strlen("Anonymous Configuration ") + 3; + w_param_text[count] = + devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + snprintf(w_param_text[count], len, + "Anonymous Configuration %d", count); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, + GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3314,6 +3425,30 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3321,15 +3456,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + devm_kfree(card->dev, w); +outfree_kcontrol_news: + devm_kfree(card->dev, (void *)template.kcontrol_news); +outfree_private_value: + devm_kfree(card->dev, (void *)private_value); +outfree_link_name: + devm_kfree(card->dev, link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1

[alsa-devel] pcm512x driver: Mixer control naming issue
by Howard Mitchell 17 Mar '15

17 Mar '15

Hi, As you may be aware the Raspbian distribution for the RaspberryPi has recently been upgraded to kernel version 3.18. It was previously at version 3.12. This has had the unfortunate side effect that users of the PCM5122 DAC have found that the output volume has been significantly reduced. The issue is related to this commit <https://git.kernel.org/cgit/linux/kernel/git/stable/linux-stable.git/commit…> which adds support for a couple of analogue gain controls. One control (Playback Volume) sets a gain of either 0db or -6dB and is intended to set the full scale output at 2Vrms or 1Vrms respectively. The other (Playback Boost Volume) provides a boost of +0.8dB. The hardware reset value of both of these gain controls is 0dB, however, in the Raspbian distribution 'Playback Volume' is being defaulted to -6dB. After some investigation I figured out that this was caused by the alsa restore mechanism. If '/var/lib/alsa/asound.state' doesn't exist or there is no saved state for a particular control then the defaults in '/usr/share/alsa/init/default' are used. In this file the default for a control named 'Playback Volume' is -20dB which is forcing the control to it's minimum value of -6dB. This seems like incorrect behaviour for something that is essentially an implementation detail, i.e. a choice that should be made by the board designer not the end user. Therefore I'd like to see what you think of the following suggestions (which I'd be more than happy to implement and submit patches for): * Either change the names of these controls to something that's not affected by the alsa restore mechanism, * or remove them as mixer controls, but make them accessible some other way - e.g. device tree. Howard Mitchell

3 9

[alsa-devel] [PATCH 0/2] ASoC: Intel: fixup patches for Braswell machine driver
by Jin Yao 17 Mar '15

17 Mar '15

Hi Mark, These are Braswell machine driver fixup patches. Jin Yao (2): 1) ASoC: Intel: Add suspend_pre and resume_post for Braswell snd_soc_card 2) ASoC: Intel: move the jack creation to Braswell machine driver For 1), it requires following dependency patches to build: ASoC: rt5670: Add IRQ function ASoC: rt5670: export jack suspend/resume APIs For 2), it requires following dependency patch to build: ASoC: Allow to register jacks at the card level These dependency patches have been integrated in for-next branch but not in intel branch. Thanks Jin Yao sound/soc/intel/cht_bsw_rt5672.c | 55 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 55 insertions(+) -- 1.9.1

2 3

[alsa-devel] [PATCH 1/2] ASoC: rt286: Drop unnecessary dapm bias_level initialization
by Lars-Peter Clausen 17 Mar '15

17 Mar '15

The default value for the bias_level is SND_SOC_BIAS_OFF when probe is being called, there is no need to initialize it explicitly. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/rt286.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index fb3f5d3..7bbe714 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1048,7 +1048,6 @@ static int rt286_probe(struct snd_soc_codec *codec) struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); rt286->codec = codec; - codec->dapm.bias_level = SND_SOC_BIAS_OFF; if (rt286->i2c->irq) { regmap_update_bits(rt286->regmap, -- 1.8.0

3 4

[alsa-devel] [PATCH 1/1] Intel: ASoC: remove duplicated code on merge
by han.lu＠intel.com 17 Mar '15

17 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index d4865a6..43fb5f3 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -2363,11 +2363,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* init param buffer */ sst_hsw_reset_param_buf(hsw); - /* allocate scratch mem regions */ - ret = sst_block_alloc_scratch(hsw->dsp); - if (ret < 0) - goto boot_err; - /* wait for DSP boot completion */ sst_dsp_boot(hsw->dsp); ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, -- 1.9.1

3 2

[alsa-devel] HD-audio restructuring (and future changes)
by Takashi Iwai 16 Mar '15

16 Mar '15

Hi, I now merged the branches topic/hda-bus and topic/hda-unbind into for-next branch that is targeted for 4.1 kernel. They are the changes of HD-audio I posted in the last month, moving the HD-audio binding to standard bus and its relevant fixes. At this moment, there should be no big visible (nor audible) changes. It's merely an internal structure change. Now you see more sysfs entries and controls, but the basic functionality via API remains identical. Also, hda-emu was updated together with this merge. Let me know if you hit any issue with these updates. About the future plan: there are already more pending changes for HD-audio and I'd like to merge them for 4.1, too. Namely, many basic HD-audio codes are split to a smaller core library in sound/hda, so that we can build the upcoming ASoC HD-audio driver with a smaller subset of HD-audio codes. You can find the current code in topic/hda-core branch. In this branch, the bus and HD-audio codec core are moved into sound/hda. You may grasp an image from the main header file include/sound/hdaudio.h. The split and cleanup of controller driver code is in development, and some pieces are found in sound-unstable git tree as of now. The patches will be submitted once when all settled down. Meanwhile, there is another patchset I've set previously for the regmap support. This is found in topic/hda-regmap branch, which is built on top of topic/hda-core branch. Unless any serious problem is found, I'm going to merge this as well. Note that this needs some fixes in regmap side, so this branch might be rebased in future to merge the regmap fixes into it. One undecided thing is whether we keep codes in sound/pci/hda directory, or move everything into sound/hda, e.g. sound/hda/legacy/pci/hda_intel.c, sound/hda/legacy/codecs/*.c. The obvious drawback of moving files is that it makes backport difficult, so it's bad for stable kernels. As of now, I'm inclined to keep them as is now, but I'm open about this. Let me know if you have any suggestions. thanks, Takashi

1 0

[alsa-devel] [PATCH v2] ASoC: wm8994: Manage wm8994's MCLK in codec driver
by Inha Song 16 Mar '15

16 Mar '15

Previously, When we use wm8994 codec, we should control its MCLK in machine driver. But, It should be managed by wm8994 codec driver, not machine driver. This patch add MCLK clock DT parsing logic and control own MCLK. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- sound/soc/codecs/wm8994.c | 22 +++++++++++++++++++++- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4fbc768..acb6b44 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/gcd.h> #include <linux/i2c.h> #include <linux/platform_device.h> @@ -2471,6 +2472,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + int i; wm_hubs_set_bias_level(codec, level); @@ -2492,8 +2494,15 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; } - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) + clk_prepare_enable(wm8994->clk_mclk[i]); + active_reference(codec); + } else if (codec->dapm.bias_level == SND_SOC_BIAS_ON) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) + clk_disable_unprepare(wm8994->clk_mclk[i]); + } break; case SND_SOC_BIAS_STANDBY: @@ -4449,6 +4458,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { static int wm8994_probe(struct platform_device *pdev) { struct wm8994_priv *wm8994; + int i; wm8994 = devm_kzalloc(&pdev->dev, sizeof(struct wm8994_priv), GFP_KERNEL); @@ -4460,6 +4470,16 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) { + char mclk_name[] = "MCLKx"; + + snprintf(mclk_name, sizeof(mclk_name), "MCLK%d", i + 1); + + wm8994->clk_mclk[i] = devm_clk_get(pdev->dev.parent, mclk_name); + if (IS_ERR(wm8994->clk_mclk[i])) + wm8994->clk_mclk[i] = NULL; + } + pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index dd73387..d426c44 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -79,6 +79,7 @@ struct wm8994_priv { int sysclk[2]; int sysclk_rate[2]; int mclk[2]; + struct clk *clk_mclk[2]; int aifclk[2]; int aifdiv[2]; int channels[2]; -- 2.0.0.390.gcb682f8

2 1

[alsa-devel] [RFC PATCH] ASoC: wm8731: let codec to manage clock by itself
by Songjun Wu 16 Mar '15

16 Mar '15

Enable WM8731 to support common clock framework. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- sound/soc/codecs/wm8731.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 098c143..8df1550 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -25,6 +25,7 @@ #include <linux/spi/spi.h> #include <linux/of_device.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -45,6 +46,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; @@ -390,6 +392,8 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8731_SYSCLK_XTAL: case WM8731_SYSCLK_MCLK: + if (wm8731->mclk && clk_set_rate(wm8731->mclk, freq)) + return -EINVAL; wm8731->sysclk_type = clk_id; break; default: @@ -491,6 +495,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (wm8731->mclk) + clk_prepare_enable(wm8731->mclk); break; case SND_SOC_BIAS_PREPARE: break; @@ -509,6 +515,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: + if (wm8731->mclk) + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -667,6 +675,19 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&spi->dev, "Assuming static MCLK\n"); + } else { + dev_err(&spi->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); @@ -718,6 +739,19 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&i2c->dev, "Assuming static MCLK\n"); + } else { + dev_err(&i2c->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); -- 1.7.9.5

3 2

[alsa-devel] [PATCH 5/6] ESI W192M : Add text Line in/Mic for selecting input gain state
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add text Line in/Mic for selecting input gain state in mixer for 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 c7ffafa..a6e7afa 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -339,8 +339,14 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Line In", "Mic" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -354,7 +360,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -370,7 +376,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) -- 2.1.4

3 3

[alsa-devel] [PATCH] ASoC: rt5677: add API to select ASRC clock source
by Oder Chiou 16 Mar '15

16 Mar '15

This patch defines an API to select the clock source for specified filters. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 163 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 79 ++++++++++++++++++++++ 2 files changed, 242 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c2a6e40..af18258 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1034,6 +1034,169 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source, return 0; } +/** + * rt5677_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5677 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc3_mask = 0, asrc3_value = 0; + unsigned int asrc4_mask = 0, asrc4_value = 0; + unsigned int asrc5_mask = 0, asrc5_value = 0; + unsigned int asrc6_mask = 0, asrc6_value = 0; + unsigned int asrc7_mask = 0, asrc7_value = 0; + + switch (clk_src) { + case RT5677_CLK_SEL_SYS: + case RT5677_CLK_SEL_I2S1_ASRC: + case RT5677_CLK_SEL_I2S2_ASRC: + case RT5677_CLK_SEL_I2S3_ASRC: + case RT5677_CLK_SEL_I2S4_ASRC: + case RT5677_CLK_SEL_I2S5_ASRC: + case RT5677_CLK_SEL_I2S6_ASRC: + case RT5677_CLK_SEL_SYS2: + case RT5677_CLK_SEL_SYS3: + case RT5677_CLK_SEL_SYS4: + case RT5677_CLK_SEL_SYS5: + case RT5677_CLK_SEL_SYS6: + case RT5677_CLK_SEL_SYS7: + break; + + default: + return -EINVAL; + } + + /* ASRC 3 */ + if (filter_mask & RT5677_DA_STEREO_FILTER) { + asrc3_mask |= RT5677_DA_STO_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5677_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO2_L_FILTER) { + asrc3_mask |= RT5677_DA_MONO2L_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_MONO2L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO2L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO2_R_FILTER) { + asrc3_mask |= RT5677_DA_MONO2R_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_MONO2R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO2R_CLK_SEL_SFT); + } + + if (asrc3_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_3, asrc3_mask, + asrc3_value); + + /* ASRC 4 */ + if (filter_mask & RT5677_DA_MONO3_L_FILTER) { + asrc4_mask |= RT5677_DA_MONO3L_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO3L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO3L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO3_R_FILTER) { + asrc4_mask |= RT5677_DA_MONO3R_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO3R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO3R_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO4_L_FILTER) { + asrc4_mask |= RT5677_DA_MONO4L_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO4L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO4L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO4_R_FILTER) { + asrc4_mask |= RT5677_DA_MONO4R_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO4R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO4R_CLK_SEL_SFT); + } + + if (asrc4_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_4, asrc4_mask, + asrc4_value); + + /* ASRC 5 */ + if (filter_mask & RT5677_AD_STEREO1_FILTER) { + asrc5_mask |= RT5677_AD_STO1_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO2_FILTER) { + asrc5_mask |= RT5677_AD_STO2_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO2_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO2_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO3_FILTER) { + asrc5_mask |= RT5677_AD_STO3_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO3_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO3_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO4_FILTER) { + asrc5_mask |= RT5677_AD_STO4_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO4_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO4_CLK_SEL_SFT); + } + + if (asrc5_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_5, asrc5_mask, + asrc5_value); + + /* ASRC 6 */ + if (filter_mask & RT5677_AD_MONO_L_FILTER) { + asrc6_mask |= RT5677_AD_MONOL_CLK_SEL_MASK; + asrc6_value = (asrc6_value & ~RT5677_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5677_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_MONO_R_FILTER) { + asrc6_mask |= RT5677_AD_MONOR_CLK_SEL_MASK; + asrc6_value = (asrc6_value & ~RT5677_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5677_AD_MONOR_CLK_SEL_SFT); + } + + if (asrc6_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_6, asrc6_mask, + asrc6_value); + + /* ASRC 7 */ + if (filter_mask & RT5677_DSP_OB_0_3_FILTER) { + asrc7_mask |= RT5677_DSP_OB_0_3_CLK_SEL_MASK; + asrc7_value = (asrc7_value & ~RT5677_DSP_OB_0_3_CLK_SEL_MASK) + | (clk_src << RT5677_DSP_OB_0_3_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DSP_OB_4_7_FILTER) { + asrc7_mask |= RT5677_DSP_OB_4_7_CLK_SEL_MASK; + asrc7_value = (asrc7_value & ~RT5677_DSP_OB_4_7_CLK_SEL_MASK) + | (clk_src << RT5677_DSP_OB_4_7_CLK_SEL_SFT); + } + + if (asrc7_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_7, asrc7_mask, + asrc7_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5677_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5677_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 07df96b..9dceb41 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1406,6 +1406,46 @@ #define RT5677_DSP_CLK_SRC_PLL2 (0x0 << 7) #define RT5677_DSP_CLK_SRC_BYPASS (0x1 << 7) +/* ASRC Control 3 (0x85) */ +#define RT5677_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5677_DA_STO_CLK_SEL_SFT 12 +#define RT5677_DA_MONO2L_CLK_SEL_MASK (0xf << 4) +#define RT5677_DA_MONO2L_CLK_SEL_SFT 4 +#define RT5677_DA_MONO2R_CLK_SEL_MASK (0xf << 0) +#define RT5677_DA_MONO2R_CLK_SEL_SFT 0 + +/* ASRC Control 4 (0x86) */ +#define RT5677_DA_MONO3L_CLK_SEL_MASK (0xf << 12) +#define RT5677_DA_MONO3L_CLK_SEL_SFT 12 +#define RT5677_DA_MONO3R_CLK_SEL_MASK (0xf << 8) +#define RT5677_DA_MONO3R_CLK_SEL_SFT 8 +#define RT5677_DA_MONO4L_CLK_SEL_MASK (0xf << 4) +#define RT5677_DA_MONO4L_CLK_SEL_SFT 4 +#define RT5677_DA_MONO4R_CLK_SEL_MASK (0xf << 0) +#define RT5677_DA_MONO4R_CLK_SEL_SFT 0 + +/* ASRC Control 5 (0x87) */ +#define RT5677_AD_STO1_CLK_SEL_MASK (0xf << 12) +#define RT5677_AD_STO1_CLK_SEL_SFT 12 +#define RT5677_AD_STO2_CLK_SEL_MASK (0xf << 8) +#define RT5677_AD_STO2_CLK_SEL_SFT 8 +#define RT5677_AD_STO3_CLK_SEL_MASK (0xf << 4) +#define RT5677_AD_STO3_CLK_SEL_SFT 4 +#define RT5677_AD_STO4_CLK_SEL_MASK (0xf << 0) +#define RT5677_AD_STO4_CLK_SEL_SFT 0 + +/* ASRC Control 6 (0x88) */ +#define RT5677_AD_MONOL_CLK_SEL_MASK (0xf << 12) +#define RT5677_AD_MONOL_CLK_SEL_SFT 12 +#define RT5677_AD_MONOR_CLK_SEL_MASK (0xf << 8) +#define RT5677_AD_MONOR_CLK_SEL_SFT 8 + +/* ASRC Control 7 (0x89) */ +#define RT5677_DSP_OB_0_3_CLK_SEL_MASK (0xf << 12) +#define RT5677_DSP_OB_0_3_CLK_SEL_SFT 12 +#define RT5677_DSP_OB_4_7_CLK_SEL_MASK (0xf << 8) +#define RT5677_DSP_OB_4_7_CLK_SEL_SFT 8 + /* VAD Function Control 4 (0x9f) */ #define RT5677_VAD_SRC_MASK (0x7 << 8) #define RT5677_VAD_SRC_SFT 8 @@ -1670,6 +1710,42 @@ enum rt5677_type { RT5676, }; +/* ASRC clock source selection */ +enum { + RT5677_CLK_SEL_SYS, + RT5677_CLK_SEL_I2S1_ASRC, + RT5677_CLK_SEL_I2S2_ASRC, + RT5677_CLK_SEL_I2S3_ASRC, + RT5677_CLK_SEL_I2S4_ASRC, + RT5677_CLK_SEL_I2S5_ASRC, + RT5677_CLK_SEL_I2S6_ASRC, + RT5677_CLK_SEL_SYS2, + RT5677_CLK_SEL_SYS3, + RT5677_CLK_SEL_SYS4, + RT5677_CLK_SEL_SYS5, + RT5677_CLK_SEL_SYS6, + RT5677_CLK_SEL_SYS7, +}; + +/* filter mask */ +enum { + RT5677_DA_STEREO_FILTER = 0x1, + RT5677_DA_MONO2_L_FILTER = (0x1 << 1), + RT5677_DA_MONO2_R_FILTER = (0x1 << 2), + RT5677_DA_MONO3_L_FILTER = (0x1 << 3), + RT5677_DA_MONO3_R_FILTER = (0x1 << 4), + RT5677_DA_MONO4_L_FILTER = (0x1 << 5), + RT5677_DA_MONO4_R_FILTER = (0x1 << 6), + RT5677_AD_STEREO1_FILTER = (0x1 << 7), + RT5677_AD_STEREO2_FILTER = (0x1 << 8), + RT5677_AD_STEREO3_FILTER = (0x1 << 9), + RT5677_AD_STEREO4_FILTER = (0x1 << 10), + RT5677_AD_MONO_L_FILTER = (0x1 << 11), + RT5677_AD_MONO_R_FILTER = (0x1 << 12), + RT5677_DSP_OB_0_3_FILTER = (0x1 << 13), + RT5677_DSP_OB_4_7_FILTER = (0x1 << 14), +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1696,4 +1772,7 @@ struct rt5677_priv { bool is_vref_slow; }; +int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + #endif /* __RT5677_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: nuc900: No need to track the dma buffer in the driver state struct
by Lars-Peter Clausen 16 Mar '15

16 Mar '15

The DMA buffer and address can be accessed through the snd_pcm_runtime. There is no need to manually track them in the driver's state struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/nuc900/nuc900-audio.h | 3 --- sound/soc/nuc900/nuc900-pcm.c | 31 +++++-------------------------- 2 files changed, 5 insertions(+), 29 deletions(-) diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h index 59f7e8e..d0b7257 100644 --- a/sound/soc/nuc900/nuc900-audio.h +++ b/sound/soc/nuc900/nuc900-audio.h @@ -100,10 +100,7 @@ struct nuc900_audio { void __iomem *mmio; spinlock_t lock; - dma_addr_t dma_addr[2]; - unsigned long buffersize[2]; unsigned long irq_num; - struct snd_pcm_substream *substream; struct resource *res; struct clk *clk; struct device *dev; diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index b809fa9..5ae5ca1 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -42,29 +42,10 @@ static const struct snd_pcm_hardware nuc900_pcm_hardware = { static int nuc900_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct nuc900_audio *nuc900_audio = runtime->private_data; - unsigned long flags; - int ret = 0; - - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) - return ret; - - spin_lock_irqsave(&nuc900_audio->lock, flags); - - nuc900_audio->substream = substream; - nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr; - nuc900_audio->buffersize[substream->stream] = - params_buffer_bytes(params); - - spin_unlock_irqrestore(&nuc900_audio->lock, flags); - - return ret; + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } -static void nuc900_update_dma_register(struct snd_pcm_substream *substream, - dma_addr_t dma_addr, size_t count) +static void nuc900_update_dma_register(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct nuc900_audio *nuc900_audio = runtime->private_data; @@ -78,8 +59,8 @@ static void nuc900_update_dma_register(struct snd_pcm_substream *substream, mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH; } - AUDIO_WRITE(mmio_addr, dma_addr); - AUDIO_WRITE(mmio_len, count); + AUDIO_WRITE(mmio_addr, runtime->dma_addr); + AUDIO_WRITE(mmio_len, runtime->dma_bytes); } static void nuc900_dma_start(struct snd_pcm_substream *substream) @@ -170,9 +151,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream) spin_lock_irqsave(&nuc900_audio->lock, flags); - nuc900_update_dma_register(substream, - nuc900_audio->dma_addr[substream->stream], - nuc900_audio->buffersize[substream->stream]); + nuc900_update_dma_register(substream); val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: Make snd_soc_dapm_kcontrol_codec() inline
by Lars-Peter Clausen 16 Mar '15

16 Mar '15

snd_soc_dapm_kcontrol_codec() is a extremely simple function and inlining it typically results in less code than necessary for calling the non-inlined version of the function. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 1 - include/sound/soc.h | 13 +++++++++++++ sound/soc/soc-dapm.c | 10 ---------- 3 files changed, 13 insertions(+), 11 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8d7416e..78633ef 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -440,7 +440,6 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list); -struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/include/sound/soc.h b/include/sound/soc.h index bc6cfab..92b92ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1268,6 +1268,19 @@ static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( return component->dapm_ptr; } +/** + * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol + * @kcontrol: The kcontrol + * + * This function must only be used on DAPM contexts that are known to be part of + * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined. + */ +static inline struct snd_soc_codec *snd_soc_dapm_kcontrol_codec( + struct snd_kcontrol *kcontrol) +{ + return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); +} + /* codec IO */ unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b6f8820..95337c8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -473,16 +473,6 @@ struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( } EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_dapm); -/** - * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol - * @kcontrol: The kcontrol - */ -struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) -{ - return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec); - static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; -- 1.8.0

2 1

[alsa-devel] [PATCH V8 0/5] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 16 Mar '15

16 Mar '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This patch series adds support for I2S audio playback on the Storm board, which contains a Qualcomm Technologies ipq806x SOC and a Maxim max98357a DAC/amp. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This contains an MI2S port, which is what these drivers are configured to use. The LPAIF also contains a DMA engine that is dedicated to moving audio samples into the transmit FIFO of the MI2S port. One bus from the MI2S port of the SOC is connected to the DAC/amp for stereo playback. This bus is configured so that the SOC is bus master and consists of DATA, LRCLK, and BCLK. The DAC/amp does not need MCLK to operate. In addition, a single GPIO pin from the SOC is connected to the same DAC/amp, which gives enable/disable control over the DAC/amp. The specific drivers added are: * a codec DAI driver for controlling the DAC/amp (applied from V4) * a CPU DAI driver for controlling the MI2S port (applied from V7) * a platform driver for controlling the LPAIF DMA engine (applied from V7) * a machine driver that instantiates a dai-link for playback The LPASS also contains clocks that need to be controlled. Those drivers have been submitted as several separate patch series: * [PATCH v3 0/8] qcom audio clock control drivers http://lkml.org/lkml/2015/1/19/656 * [PATCH] arm: dts: qcom: Add LCC nodes https://lkml.org/lkml/2015/1/28/557 * [PATCH 1/2] arm: qcom_defconfig: Enable lpass clock driver https://lkml.org/lkml/2015/1/28/582 * [PATCH] clk: qcom: Fix ipq806x LCC frequency tables https://lkml.org/lkml/2015/2/26/774 * [PATCH v2] clk: qcom: Properly change rates for ahbix clock https://lkml.org/lkml/2015/3/6/878 Even though the ipq806x LPASS does not contain an audio DSP, other SOCs do have one. For those SOCs, the audio DSP typically controls the hardware blocks in the LPASS. Hence, different CPU DAI driver(s) would need to be used in order to facilitate audio with the DSP. As such, the LPASS DT binding description alludes to an optional adsp phandle, which is not present for this SOC. For other SOCs that do contain an audio DSP, the phandle should exist and refer to a an adsp node. Not using the audio DSP on an SOC that does have one would require different CPU DAI driver(s), in addition to possible bootloader and/or firmware changes. Note that the use of a phandle was a modification that has been submitted as part of a separate patch series. Corresponding additions to the device tree for the ipq806x SOC and its documentation have also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well (applied from V7). = Changes since V7 [Patch V7 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-March/088546.html * Changed the compatible property of the machine driver to make it specific to audio * Removed the lcc node definition from the DT, as it was submitted separately = Changes since V6 [Patch V6 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/088218.h… * Added REGMAP_MMIO selection for the CPU and platform drivers * Modified the AHBIX frequency to match the corresponding LCC fix * Tweaked the logging in the CPU driver probe. = Changes since V5 [Patch V5 00/12] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087832.h… * Correctly use Storm as the build target label and DT binding label. * Added audio DSP sub-node to the LPASS device tree, disabled for this SOC. * Added logic to CPU DAI driver to fail the probe() if a DSP is present. * Use the standard naming convention for the DAI link properties. * General code cleanup. = Changes since V4 [Patch V4 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087499.h… * Replaced simple-card with a machine driver to resolve the system clock configuration, rather than having the CPU DAI driver do it. * Added header files to avoid indirect header dependencies and implicit forward declarations. * Tweaked the ISR to match the conventions of the surrounding code. * Removed the usage of the low-power memory as it is not needed. * Removed the use of the DRV_NAME constant. * Added explicit dependency on gpiolib for the codec driver. * Moved the MODULE_DEVICE_TABLE macro inside the CONFIG_OF conditional. * Modified the documentation to reflect the changes. * General code cleanup. = Changes since V3 [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085694.h… * Placed the content of the inline functions into the callbacks. * Replaced use of readl/writel register access functions with regmap access functions. Notable exception is the ISR, which uses ioread32/iowrite32. * Rearranged the sequencing of the hardware block enables to fit within the ASoC framework callbacks, while remaining functional. REQ 1: The hardware requires the enable sequence to be: LPAIF-DMA[enable],then LPAIF-MI2S[enable], then DAC-GPIO[enable] REQ 2: The hardware requires the disable sequence to be: DAC-GPIO[disable], then LPAIF-MI2S[disable] * Corrected the implementation of the pointer callback. * Utilize the LPM to buffer audio samples, rather than memory external to LPASS. * Corrected the interrupt clearing in the ISR. * Implemented a default system clock (defined by the simple-card DT node), and optional LPASS DT node modifiers that can alter the system clock in order to expand the range of available bit clock frequencies. * Addressed all of the remaining issues raised by Mark Brown. * General code cleanup. = Changes since V2 [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085186.h… * Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. * Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. * Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. * Removed machine driver and tied DAI drivers to simple-audio-card. * Packaged the build files and Maxim codec files together in one change. * Removed QCOM as vendor from Maxim code and documentation. * Separated the SOC and board definitions into the correct DTS files. * Update device tree documentation to reflect changes. * General code cleanup. = Changes since V1 [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084322.h… * Remove the native LPAIF driver, and move its functionality to the CPU DAI driver. * Add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver). * Use devm_* and dev_* where possible. * ISR only handles relevant DMA channel now. * Update device tree documentation to reflect changes. * General code cleanup. Kenneth Westfield (5): ASoC: qcom: Document Storm bindings ASoC: qcom: Add Storm machine driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware Documentation/devicetree/bindings/sound/storm.txt | 23 +++ arch/arm/boot/dts/qcom-ipq8064.dtsi | 16 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/qcom/Kconfig | 25 ++++ sound/soc/qcom/Makefile | 11 ++ sound/soc/qcom/storm.c | 162 ++++++++++++++++++++++ 7 files changed, 239 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/storm.txt create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/storm.c -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 6

[alsa-devel] [PATCH 1/2] ASoC: qcom: Change qcom, adsp in LPASS CPU bindings
by Kenneth Westfield 16 Mar '15

16 Mar '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Change the representation of the audio DSP, in the LPASS CPU bindings description, from a required subnode to an optional phandle. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt index e7c6e9321863f022ebf0d51b75d7bb83c10b9062..e00732dac9393e95f0530505c9f38127f58980a3 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -22,13 +22,9 @@ Required properties: - reg-names : A list which must include the following entries: * "lpass-lpaif" -Required subnodes: +Optional properties: -- qcom,adsp : Audio DSP sub-node - -Optional Audio DSP subnode properties: - -- status : "disabled" indicates the adsp is not available. +- qcom,adsp : Phandle for the audio DSP node Example: @@ -43,7 +39,5 @@ lpass@28100000 { pinctrl-1 = <&mi2s_idle>; reg = <0x28100000 0x10000>; reg-names = "lpass-lpaif"; - qcom,adsp { - status = "disabled"; - }; + qcom,adsp = <&adsp>; }; -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 2

[alsa-devel] [PATCH] ALSA: hda - Don't access stereo amps for mono channel widgets
by Takashi Iwai 16 Mar '15

16 Mar '15

The current HDA generic parser initializes / modifies the amp values always in stereo, but this seems causing the problem on ALC3229 codec that has a few mono channel widgets: namely, these mono widgets react to actions for both channels equally. In the driver code, we do care the mono channel and create a control only for the left channel (as defined in HD-audio spec) for such a node. When the control is updated, only the left channel value is changed. However, in the resume, the right channel value is also restored from the initial value we took as stereo, and this overwrites the left channel value. This ends up being the silent output as the right channel has been never touched and remains muted. This patch covers the places where unconditional stereo amp accesses are done and converts to the conditional accesses. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94581 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 30 ++++++++++++++++++++++-------- 1 file changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4ec6331..fe18071bf93a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) { unsigned int caps = query_amp_caps(codec, nid, dir); int val = get_amp_val_to_activate(codec, nid, dir, caps, false); - snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + else + snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val); +} + +/* update the amp, doing in stereo or mono depending on NID */ +static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, + unsigned int mask, unsigned int val) +{ + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + return snd_hda_codec_amp_stereo(codec, nid, dir, idx, + mask, val); + else + return snd_hda_codec_amp_update(codec, nid, 0, dir, idx, + mask, val); } /* calculate amp value mask we can modify; @@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, return; val &= mask; - snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val); + update_amp(codec, nid, dir, idx, mask, val); } static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, @@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) has_amp = nid_has_mute(codec, mix, HDA_INPUT); for (i = 0; i < nums; i++) { if (has_amp) - snd_hda_codec_amp_stereo(codec, mix, - HDA_INPUT, i, - 0xff, HDA_AMP_MUTE); + update_amp(codec, mix, HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) - snd_hda_codec_amp_stereo(codec, conn[i], - HDA_OUTPUT, 0, - 0xff, HDA_AMP_MUTE); + update_amp(codec, conn[i], HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); } } -- 2.3.2

2 4

[alsa-devel] [PATCH 0/13] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 16 Mar '15

16 Mar '15

Hi Mark These are DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, but it is copied from simple-card. Current ASoC DPCM needs route settings for FE/BE, but sometimes it will have trouble, because many drivers are using same name for this routing. This patch set solved this issue too. 1 - 2 : prepare patch for Renesas new sound card driver 3 - 4 : DAPM routing issue solution patches 5 - 6 : new Renesas sound card driver 7 - 11 : fixup/tidyup patches for rsnd 12 - 13 : DPCM support patch for rsnd Kuninori Morimoto (13): 1) ASoC: core: remove of_xlate_dai_name from snd_soc_component_driver 2) ASoC: core: add snd_soc_get_dai_info() to get driver infomation 3) ASoC: dapm: add snd_soc_dapm_route_scan macro for DAPM route 4) ASoC: dapm: add driver prefix matching for snd_soc_dapm_add_route() 5) ASoC: rsnd-dpcm-card: add Renesas sound card support for DPCM 6) ASoC: rsnd-dpcm-card: add .be_hw_params_fixup support for convert rate 7) ASoC: rsnd: no more SSI restart when unusual situation 8) ASoC: rsnd: no more SRC restart when unusual situation 9) ASoC: rsnd: tidyup error message format 10) ASoC: rsnd: show debug info for sampling rate convert 11) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 12) ASoC: rsnd: add dai_link stream name 13) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsnd-dpcm-card.txt | 72 ++++++++++++++ include/sound/soc.h | 6 +- sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 ++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd-dpcm-card.c | 491 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 16 ++- sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 71 ++++++++------ sound/soc/soc-dapm.c | 33 +++++-- 13 files changed, 780 insertions(+), 75 deletions(-) Best regards --- Kuninori Morimoto

2 16

[alsa-devel] [PATCH 6/6] ESI W192M : Add sampling rate control of the ADC/DAC
by klem.dev＠gmail.com 16 Mar '15

16 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 a6e7afa..f3e3879 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; } @@ -388,6 +446,44 @@ 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); @@ -487,21 +583,32 @@ 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

1 0

[alsa-devel] [PATCH 4/6] ESI W192M : Add TLV support for control value in dB scale
by klem.dev＠gmail.com 16 Mar '15

16 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

1 0

[alsa-devel] [PATCH 3/6] ESI W192M : Enable midi i/o of port envy24 chip as available
by klem.dev＠gmail.com 16 Mar '15

16 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

1 0

[alsa-devel] [PATCH 2/6] ESI W192M : Update eeprom structure to C99 standard
by klem.dev＠gmail.com 16 Mar '15

16 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

1 0

[alsa-devel] [PATCH 1/6] ESI W192M : Correct copy/paste from prodigy driver
by klem.dev＠gmail.com 16 Mar '15

16 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

1 0

[alsa-devel] Issues and/or possible bugs in alsa
by Yomi Ogunwumi 15 Mar '15

15 Mar '15

Hello. I seem to have run into a few issues with alsa. First, when I plug my headphones into my laptop, sound continues to play out of the speakers and doesn't switch over to (or unmute) the headphones. Auto-Mute Mode is enabled in alsamixer. The same thing happens when I unplug the headphones, it doesn't switch over to the speakers. I have to do the same thing I described above, (no unmuting, I just have to turn up the volume for the speakers in alsamixer.) Second. When I plug in these headphones, I can't use the internal microphone on this laptop, it mutes. (AFAICT) These headphones do not have a microphone built in, they're just an ordinary pair of headphones. However, I was given a workaround in #alsa on freenode. In order to use the laptop's internal microphone with my headphones plugged in, I have to do : sudo hda-verb /dev/snd/hwC1D0 0x22 SET_CONNECT_SEL 6 — it works, but it isn't really ideal. alsa-info is here : https://gist.github.com/Yomi0/9bfc3e72588cf8a35811 The first file in that gist is alsa-info with headphones and the second is without headphones. Third. This is just a question. Is this : https://bugs.freedesktop.org/show_bug.cgi?id=86676 actually a pulseaudio bug, or is it an issue with alsa? I'm only asking since Raymond linked something that seemed to belong to the alsa project. Thank you. -- *Yomi*

5 12

[alsa-devel] [PATCH] ASoC: fsl-ssi: Fix bitclock calculation for master mode
by Marc Kleine-Budde 15 Mar '15

15 Mar '15

From: Sascha Hauer <s.hauer(a)pengutronix.de> in fsl_ssi_set_bclk we are looking for the best divider by iterating over all possible dividers. The divider is in the range 1..256, so make this clear by using 'div' as name for the loop counter and iterate from 1 to 256 instead of 0 to 255. This makes obvious that tmprate is calculated wrong, it has to be freq * factor * div, not freq * factor * (i + 2). Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Marc Kleine-Budde <mkl(a)pengutronix.de> --- sound/soc/fsl/fsl_ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b9fabbf69db6..b8af08e49085 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -578,7 +578,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct regmap *regs = ssi_private->regs; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; - u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, div; unsigned long clkrate, baudrate, tmprate; u64 sub, savesub = 100000; unsigned int freq; @@ -602,8 +602,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; - for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + for (div = 1; div < 256; div++) { + tmprate = freq * factor * div; if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); @@ -618,7 +618,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, continue; clkrate /= factor; - afreq = clkrate / (i + 1); + afreq = clkrate / div; if (freq == afreq) sub = 0; @@ -636,7 +636,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, if (sub < savesub) { baudrate = tmprate; savesub = sub; - pm = i; + pm = div - 1; } /* We are lucky */ -- 2.1.4

1 0

Re: [alsa-devel] snd_usb_audio - siberia elite prism soundcard issues
by Raymond Yau 15 Mar '15

15 Mar '15

>> >> >> >> >> > > >> >> >> > > > amixer: Control hw:2 element read error: Invalid argument >> >> >> > > > usb 1-1.3: 10:0: cannot get min/max values for control 3 (id 10) >> >> >> > > > usb 1-1.3: 2:1: cannot get freq at ep 0x1 >> >> >> >> numid=5,iface=MIXER,name='PCM Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> numid=6,iface=MIXER,name='PCM Volume',index=1 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> numid=7,iface=MIXER,name='PCM Volume',index=2 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> numid=8,iface=MIXER,name='PCM Volume',index=3 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> numid=9,iface=MIXER,name='PCM Volume',index=4 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> > AudioControl Interface Descriptor: >> >> > bLength 13 >> >> > bDescriptorType 36 >> >> > bDescriptorSubtype 4 (MIXER_UNIT) >> >> > Warning: Descriptor too short >> >> > bUnitID 10 >> >> > bNrInPins 2 >> >> > baSourceID( 0) 9 >> >> > baSourceID( 1) 1 >> >> > bNrChannels 8 >> >> > wChannelConfig 0x063f >> >> > Left Front (L) >> >> > Right Front (R) >> >> > Center Front (C) >> >> > Low Freqency Enhancement (LFE) >> >> > Left Surround (LS) >> >> > Right Surround (RS) >> >> > Side Left (SL) >> >> > Side Right (SR) >> >> > iChannelNames 0 >> >> > bmControls 0x00 >> >> > bmControls 0x00 >> >> > iMixer 33 (error) Refer to usb audio specification 1.0 Table 4-5: Mixer Unit Descriptor bLength = 10 + p + N = 14 Number of Input Pins of this Unit: p > >> > bNrInPins 2 The ID of the Unit or Terminal to which the first and second Input Pins of this Mixer Unit is connected. > >> > baSourceID( 0) 9 >> >> > baSourceID( 1) 1 Number of logical output channels in the Mixer’s output audio channel cluster. >> >> > bNrChannels 8 Describes the spatial location of the logical channels. >> >> > wChannelConfig 0x063f >> >> > Left Front (L) >> >> > Right Front (R) >> >> > Center Front (C) >> >> > Low Freqency Enhancement (LFE) >> >> > Left Surround (LS) >> >> > Right Surround (RS) >> >> > Side Left (SL) >> >> > Side Right (SR) Bit map indicating which mixing Controls are programmable. >> >> > bmControls 0x00 >> >> > bmControls 0x00 The Mixer Unit Descriptor reports which Controls are programmable in the bmControls bitmap field. This bitmap must be interpreted as a two-dimensional bit array that has a row for each logical input channel and a column for each logical output channel. If a bit at position [u, v] is set, this means that the Mixer Unit contains a programmable mixing Control that connects input channel u to output channel v. If bit [u, v] is clear, this indicates that the connection between input channel u and output channel v is non-programmable. Its fixed value can be retrieved through the appropriate request. The valid range for u is from one to n. The valid range for v is from one to m. The bmControls field stores the bit array row after row where the MSb of the first byte corresponds to the connection between input channel 1 and output channel 1. If (n x m) is not an integer multiple of 8, the bit array is padded with zeros until an integer number of bytes is occupied. The number of bytes used to store the bit array, N, can be calculated as follows: IF ((n x m) MOD 8) <> 0 THEN N = ((n x m) DIV 8) + 1 ELSE N = ((n x m) DIV 8) Index of a string descriptor, describing the name of the first logical channel. >> > iChannelNames 0 Index of a string descriptor, describing the Mixer Unit. >> >> > iMixer 33 >> >> >> if (check_matrix_bitmap(c, ich, och, num_outs)) { >> ich_has_controls = 1; >> break; >> } >> } This mean value returned by check_matrix_bitmap() is incorrect as >> >> > bmControls 0x00 >> >> > bmControls 0x00 >> if (ich_has_controls) >> build_mixer_unit_ctl(state, desc, pin, ich, >> unitid, &iterm); >> } >> } >> return 0; >> } > >

2 5

[alsa-devel] [PATCH 5/6] ESI W192M : Add text Line in/Mic for selecting input gain state
by klem.dev＠gmail.com 14 Mar '15

14 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add text Line in/Mic for selecting input gain state in mixer for 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 3af9daf..497ff1a 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -339,8 +339,23 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Line In", "Mic" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items + - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -354,7 +369,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -370,7 +385,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) -- 2.1.4

3 2

[alsa-devel] [PATCH 4/6] ESI W192M : Add TLV support to control volume using dB scale
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..3af9daf 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 + .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

3 2

[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

3 3

[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

1 0

[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

2 1

[alsa-devel] [RESEND RFC PATCH] ASoC: wm8731: let codec to manage clock by itself
by Songjun Wu 13 Mar '15

13 Mar '15

Enable WM8731 to support common clock framework. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- sound/soc/codecs/wm8731.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 098c143..8df1550 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -25,6 +25,7 @@ #include <linux/spi/spi.h> #include <linux/of_device.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -45,6 +46,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; @@ -390,6 +392,8 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8731_SYSCLK_XTAL: case WM8731_SYSCLK_MCLK: + if (wm8731->mclk && clk_set_rate(wm8731->mclk, freq)) + return -EINVAL; wm8731->sysclk_type = clk_id; break; default: @@ -491,6 +495,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (wm8731->mclk) + clk_prepare_enable(wm8731->mclk); break; case SND_SOC_BIAS_PREPARE: break; @@ -509,6 +515,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: + if (wm8731->mclk) + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -667,6 +675,19 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&spi->dev, "Assuming static MCLK\n"); + } else { + dev_err(&spi->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); @@ -718,6 +739,19 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&i2c->dev, "Assuming static MCLK\n"); + } else { + dev_err(&i2c->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ASoC: Intel: Add suspend_pre and resume_post for Braswell snd_soc_card
by Jin Yao 13 Mar '15

13 Mar '15

On Braswell, we need to add some machine specific setting before suspend and after resume. For example, disable/enable jack detection in codec so use snd_soc_card suspend_pre and resume_post ops for this purpose. Signed-off-by: Jin Yao <yao.jin(a)linux.intel.com> --- sound/soc/intel/cht_bsw_rt5672.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index 279df4c..c41fae3 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -267,6 +267,35 @@ static struct snd_soc_dai_link cht_dailink[] = { }, }; +static int cht_suspend_pre(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); + rt5670_jack_suspend(codec); + break; + } + } + return 0; +} + +static int cht_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + dev_dbg(codec->dev, "enabling jack detect for resume.\n"); + rt5670_jack_resume(codec); + break; + } + } + + return 0; +} + /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "cherrytrailcraudio", @@ -278,6 +307,8 @@ static struct snd_soc_card snd_soc_card_cht = { .num_dapm_routes = ARRAY_SIZE(cht_audio_map), .controls = cht_mc_controls, .num_controls = ARRAY_SIZE(cht_mc_controls), + .suspend_pre = cht_suspend_pre, + .resume_post = cht_resume_post, }; static int snd_cht_mc_probe(struct platform_device *pdev) -- 1.9.1

4 7

[alsa-devel] Issues w/ Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG
by Alnie 13 Mar '15

13 Mar '15

At Wed, 25 Feb 2015 00:21:15 -0800, Alnie wrote: > > > > On 02/25/2015 12:06 AM, Takashi Iwai wrote: > > At Tue, 24 Feb 2015 23:55:25 -0800, > > Alnie wrote: > >> > >> [ 2619.199658] pciehp 0000:00:1c.3:pcie04: Card not present on Slot(3) > >> [ 2619.199666] pciehp 0000:00:1c.3:pcie04: slot(3): Link Down event > >> [ 2619.199674] pciehp 0000:00:1c.3:pcie04: Link Down event ignored on > >> slot(3): already powering off > >> [ 2619.199911] pci_bus 0000:06: busn_res: [bus 06-0c] is released > >> [ 2620.757928] pciehp 0000:00:1c.3:pcie04: Card present on Slot(3) > >> [ 2620.767950] pciehp 0000:00:1c.3:pcie04: slot(3): Link Up event > >> [ 2620.767985] pciehp 0000:00:1c.3:pcie04: Link Up event ignored on > >> slot(3): already powering on > > > > pciehp reacted power off and on here, and... > > > >> [ 2620.876160] pci 0000:05:00.0: [1102:7006] type 01 class 0x060400 > >> [ 2620.876426] pci 0000:05:00.0: supports D1 D2 > >> [ 2620.876719] pci 0000:05:00.0: bridge configuration invalid ([bus > >> 00-00]), reconfiguring > >> [ 2620.876928] pci_bus 0000:06: busn_res: can not insert [bus 06-ff] > >> under [bus 05-0c] (conflicts with (null) [bus 05-0c]) > > > > .... and something failed here for the PCI bridge device. > > > >> [ 2620.876967] pci 0000:06:00.0: [1102:0009] type 00 class 0x040300 > >> [ 2620.877013] pci 0000:06:00.0: reg 0x10: [mem 0x00000000-0x00003fff] > >> [ 2620.877203] pci 0000:06:00.0: supports D1 D2 > >> [ 2620.877443] pci 0000:05:00.0: PCI bridge to [bus 06-ff] > >> [ 2620.877462] pci 0000:05:00.0: bridge window [io 0x0000-0x0fff] > >> [ 2620.877474] pci 0000:05:00.0: bridge window [mem 0x00000000-0x000fffff] > >> [ 2620.877492] pci 0000:05:00.0: bridge window [mem > >> 0x00000000-0x000fffff 64bit pref] > >> [ 2620.877500] pci_bus 0000:06: busn_res: [bus 06-ff] end is updated to 06 > >> [ 2620.877559] pci 0000:05:00.0: BAR 14: assigned [mem > >> 0xf4300000-0xf43fffff] > >> [ 2620.877568] pci 0000:05:00.0: BAR 15: assigned [mem > >> 0xf4000000-0xf40fffff 64bit pref] > >> [ 2620.877574] pci 0000:05:00.0: BAR 13: assigned [io 0x2000-0x2fff] > >> [ 2620.877582] pci 0000:06:00.0: BAR 0: assigned [mem 0xf4300000-0xf4303fff] > >> [ 2620.877595] pci 0000:05:00.0: PCI bridge to [bus 06] > >> [ 2620.877603] pci 0000:05:00.0: bridge window [io 0x2000-0x2fff] > >> [ 2620.877618] pci 0000:05:00.0: bridge window [mem 0xf4300000-0xf43fffff] > >> [ 2620.877630] pci 0000:05:00.0: bridge window [mem > >> 0xf4000000-0xf40fffff 64bit pref] > >> [ 2620.877647] pcieport 0000:00:1c.3: PCI bridge to [bus 05-0c] > >> [ 2620.877655] pcieport 0000:00:1c.3: bridge window [io 0x2000-0x2fff] > >> [ 2620.877667] pcieport 0000:00:1c.3: bridge window [mem > >> 0xf4300000-0xf7efffff] > >> [ 2620.877677] pcieport 0000:00:1c.3: bridge window [mem > >> 0xf4000000-0xf40fffff 64bit pref] > >> [ 2620.877990] pci 0000:05:00.0: enabling device (0000 -> 0003) > >> [ 2620.878013] snd_hda_intel 0000:06:00.0: enabling device (0000 -> 0002) > >> [ 2620.882747] azx_single_wait_for_response: 108 callbacks suppressed > >> [ 2620.882753] snd_hda_intel 0000:06:00.0: Codec #1 probe error; > >> disabling it... > >> [ 2620.887052] snd_hda_intel 0000:06:00.0: no AFG or MFG node found > >> [ 2620.887058] snd_hda_intel 0000:06:00.0: no codecs initialized > > > > What if you once unload snd-hda-intel module and reload again while > > keeping the card plugged? I guess it still doesn't work, but just to > > be sure... > > > > Overall, this look more like a problem of PCI core stuff. Better to > > toss the issue to PCI people. > > > > > > Takashi > > > > I did alsa force-reload and the same error comes up. Do you know of a > good starting point for presenting PCI issues? > >Ask on linux-pci at vger.kernel.org ? > > >Takashi Bjorn has suggested it is not a PCI issue... http://article.gmane.org/gmane.linux.kernel.pci/39492 Any other ideas in regards to this? -Alnie

2 23

[alsa-devel] [PATCH v2] ALSA: control: Add sanity checks for user ctl id name string
by Takashi Iwai 13 Mar '15

13 Mar '15

There was no check about the id string of user control elements, so we accepted even a control element with an empty string, which is obviously bogus. This patch adds more sanity checks of id strings. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: fix typos, thinkos sound/core/control.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index 35324a8e83c8..eeb691d1911f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1) return -EINVAL; + if (!*info->id.name) + return -EINVAL; + if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name)) + return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| -- 2.3.2

2 3

[alsa-devel] [PATCH] ALSA: control: Fix breakage of user ctl element addition
by Takashi Iwai 13 Mar '15

13 Mar '15

In the commit [2225e79b9b03: 'ALSA: core: reduce stack usage related to snd_ctl_new()'], the id field of the newly added kctl is untouched, thus all attribute like name string remain empty. The fix is just to add the forgotten memcpy of the id field. Fixes: 2225e79b9b03 ('ALSA: core: reduce stack usage related to snd_ctl_new()') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 54a412af3224..d677c27746e9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1267,6 +1267,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, err = snd_ctl_new(&kctl, count, access, file); if (err < 0) return err; + memcpy(&kctl->id, &info->id, sizeof(kctl->id)); kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL); if (kctl->private_data == NULL) { -- 2.3.2

2 1

[alsa-devel] Microphone Not Working after Suspend on Apple MacBookAir5, 2 Cirrus Logic CS4206
by Vasil Zlatanov 12 Mar '15

12 Mar '15

Issue as seen on reddit thread: https://www.reddit.com/r/linux/comments/1kccbk/alsa_bug_driving_me_crazy_pl… A common problem for Apple Macbook Air 5.2 users where microphone does not work after system resume. As users have noted microphone starts working ones again when the volume is changed in alsa-mixer. alsa-info dumps are attached showing before and after being adjusted (where before causes microphone not to work while in after it does) IRC user 'debianuser' kindly helped out and found that using the two following commands fix the issue: amixer -cPCH set "Capture" 0% nocap amixer -cPCH set "Capture" 100% cap In my experience both are required and they seem to have the same effect as manually changing the volume level in alsamixer, and as such resolve the issue (until another suspend when they have to be run again). As of now I'm resolving this issue by creating a service that is to run after resume from susspend with the given amixer commands to renable the microphone. The second issue, which is also in the reddit thread, is that the microphine is mono and when recording only right channel audio is seen. For me this broke programs such as mumble and skype. A alsa-info output with coefficients is attached. 'debianuser' managed to find a solution to this issue well, resolving it by running the following two commands: hda-verb /dev/snd/hwC0D0 0x11 SET_COEF_INDEX 2 hda-verb /dev/snd/hwC0D0 0x11 SET_PROC_COEF 0x102a After one runs them a single time after boot, the channels collectly show and programs issues are resolved. 'debianuser' mentions that the sources for the commands are available in the kernel and applied to the macbook 8.1 model, but not to mine.

2 2

Re: [alsa-devel] Microphone Not Working after Suspend on Apple MacBookAir5, 2 Cirrus Logic CS4206
by Takashi Iwai 12 Mar '15

12 Mar '15

At Thu, 12 Mar 2015 20:13:05 +0100, Vasil Zlatanov wrote: > > Applied both patches against 3.18.6-1 and correctly resolved all > issues. Good to hear, I queued both patches now. They'll be backported to stable kernels once when merged to upstream. thanks, Takashi > > On Thu, Mar 12, 2015 at 11:38:39AM +0100, Takashi Iwai wrote: > > At Wed, 11 Mar 2015 23:26:00 +0100, > > Vasil Zlatanov wrote: > > > > > > Issue as seen on reddit thread: > > > https://www.reddit.com/r/linux/comments/1kccbk/alsa_bug_driving_me_crazy_pl… > > > > > > A common problem for Apple Macbook Air 5.2 users where microphone does > > > not work after system resume. As users have noted microphone starts > > > working ones again when the volume is changed in alsa-mixer. > > > > > > alsa-info dumps are attached showing before and after being adjusted > > > (where before causes microphone not to work while in after it does) > > > > > > IRC user 'debianuser' kindly helped out and found that using the two > > > following commands fix the issue: > > > > > > amixer -cPCH set "Capture" 0% nocap > > > amixer -cPCH set "Capture" 100% cap > > > > > > In my experience both are required and they seem to have the same effect > > > as manually changing the volume level in alsamixer, and as such resolve > > > the issue (until another suspend when they have to be run again). > > > > This might be due to the single ADC amp. The ADC widgets have > > multiple inputs and usually they have individual amp volumes. But > > some codecs give only a single volume for all inputs. > > > > There is a flag in the driver to change the control for that. The > > patch below enables it. > > > > diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c > > index 1589c9bcce3e..dd2b3d92071f 100644 > > --- a/sound/pci/hda/patch_cirrus.c > > +++ b/sound/pci/hda/patch_cirrus.c > > @@ -584,6 +584,7 @@ static int patch_cs420x(struct hda_codec *codec) > > return -ENOMEM; > > > > spec->gen.automute_hook = cs_automute; > > + codec->single_adc_amp = 1; > > > > snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl, > > cs420x_fixups); > > > > Alternatively, you can pass "single_adc_smp = true" hint in the > > firmware patch passed via patch option. See > > Documentation/sound/alsa/HD-Audio.txt for details. > > > > > > > As of now I'm resolving this issue by creating a service that is to run > > > after resume from susspend with the given amixer commands to renable the > > > microphone. > > > > > > The second issue, which is also in the reddit thread, is that the > > > microphine is mono and when recording only right channel audio is seen. > > > For me this broke programs such as mumble and skype. > > > > > > A alsa-info output with coefficients is attached. > > > > > > 'debianuser' managed to find a solution to this issue well, resolving it > > > by running the following two commands: > > > > > > hda-verb /dev/snd/hwC0D0 0x11 SET_COEF_INDEX 2 > > > hda-verb /dev/snd/hwC0D0 0x11 SET_PROC_COEF 0x102a > > > > > > After one runs them a single time after boot, the channels collectly > > > show and programs issues are resolved. 'debianuser' mentions that the > > > sources for the commands are available in the kernel and applied to the > > > macbook 8.1 model, but not to mine. > > > > Yes, it's the very same workaround for MBP8,1. You can pass > > model=mbp81 option. If it works, you can just add a sline to > > cs420x_fixup_tbl[] something like > > > > @@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { > > SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), > > SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), > > SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), > > + SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81), > > SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42), > > SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), > > {} /* terminator */ > > > > > > > > Takashi > [2 <application/pgp-signature (7bit)>] >

1 0

[alsa-devel] [PATCH] ASoC: rt5645: Add TDM support for rt5650
by Bard Liao 12 Mar '15

12 Mar '15

rt5650 and rt5645 use different register bits for TDM configuration. This patch modifies rt5645_set_tdm_slot to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 41 ++++++++++++++++++++++++++++++----------- 1 file changed, 30 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index debf16c..af468ae 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2283,23 +2283,42 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; - unsigned int val = 0; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int i_slot_sft, o_slot_sft, i_width_sht, o_width_sht, en_sft; + unsigned int mask, val = 0; + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + en_sft = 15; + i_slot_sft = 10; + o_slot_sft = 8; + i_width_sht = 6; + o_width_sht = 4; + mask = 0x8ff0; + break; + default: + en_sft = 14; + i_slot_sft = o_slot_sft = 12; + i_width_sht = o_width_sht = 10; + mask = 0x7c00; + break; + } if (rx_mask || tx_mask) { - val |= (1 << 14); - snd_soc_update_bits(codec, RT5645_BASS_BACK, - RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); + val |= (1 << en_sft); + if (rt5645->codec_type == CODEC_TYPE_RT5645) + snd_soc_update_bits(codec, RT5645_BASS_BACK, + RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); } switch (slots) { case 4: - val |= (1 << 12); + val |= (1 << i_slot_sft) | (1 << o_slot_sft); break; case 6: - val |= (2 << 12); + val |= (2 << i_slot_sft) | (2 << o_slot_sft); break; case 8: - val |= (3 << 12); + val |= (3 << i_slot_sft) | (3 << o_slot_sft); break; case 2: default: @@ -2308,20 +2327,20 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (slot_width) { case 20: - val |= (1 << 10); + val |= (1 << i_width_sht) | (1 << o_width_sht); break; case 24: - val |= (2 << 10); + val |= (2 << i_width_sht) | (2 << o_width_sht); break; case 32: - val |= (3 << 10); + val |= (3 << i_width_sht) | (3 << o_width_sht); break; case 16: default: break; } - snd_soc_update_bits(codec, RT5645_TDM_CTRL_1, 0x7c00, val); + snd_soc_update_bits(codec, RT5645_TDM_CTRL_1, mask, val); return 0; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3 0/3] add sound effect module waves
by han.lu＠intel.com 12 Mar '15

12 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Merged 5 patches(0004 to 0008) of last revision into 3 new patches Lu, Han (3): ASoC: Intel: add kcontrol to enable/disable sound effect module waves ASoC: Intel: add function to set parameter to sound effect module waves ASoC: Intel: add kcontrol to set parameter to sound effect module waves sound/soc/intel/sst-haswell-ipc.c | 142 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 35 ++++++++++ sound/soc/intel/sst-haswell-pcm.c | 118 +++++++++++++++++++++++++++++++ 3 files changed, 295 insertions(+) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: wm8994: Manage wm8994's MCLK in codec driver
by Inha Song 12 Mar '15

12 Mar '15

Previously, When we use wm8994 codec, we should control its MCLK in machine driver. But, It should be managed by wm8994 codec driver, not machine driver. This patch add MCLK clock DT parsing logic and control own MCLK. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- sound/soc/codecs/wm8994.c | 27 ++++++++++++++++++++++++++- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b390..d80dcc6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/gcd.h> #include <linux/i2c.h> #include <linux/platform_device.h> @@ -2470,6 +2471,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + int i; wm_hubs_set_bias_level(codec, level); @@ -2491,8 +2493,19 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; } - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) { + if (!IS_ERR(wm8994->clk_mclk[i])) + clk_prepare_enable(wm8994->clk_mclk[i]); + } active_reference(codec); + } else if (codec->dapm.bias_level == SND_SOC_BIAS_ON) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) { + if (!IS_ERR(wm8994->clk_mclk[i])) + clk_disable_unprepare( + wm8994->clk_mclk[i]); + } + } break; case SND_SOC_BIAS_STANDBY: @@ -4451,6 +4464,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { static int wm8994_probe(struct platform_device *pdev) { struct wm8994_priv *wm8994; + int i; wm8994 = devm_kzalloc(&pdev->dev, sizeof(struct wm8994_priv), GFP_KERNEL); @@ -4460,6 +4474,17 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) { + char mclk_name[] = "MCLKx"; + + snprintf(mclk_name, sizeof(mclk_name), "MCLK%d", i + 1); + + wm8994->clk_mclk[i] = devm_clk_get(pdev->dev.parent, mclk_name); + if (IS_ERR(wm8994->clk_mclk[i])) + if (PTR_ERR(wm8994->clk_mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 6536f8d..8cf06a4 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -78,6 +78,7 @@ struct wm8994_priv { int sysclk[2]; int sysclk_rate[2]; int mclk[2]; + struct clk *clk_mclk[2]; int aifclk[2]; int aifdiv[2]; int channels[2]; -- 2.0.0.390.gcb682f8

3 4

[alsa-devel] ti, codec property in sound/soc/omap/omap-twl4030.c not used
by Belisko Marek 12 Mar '15

12 Mar '15

Hi, I'm looking at sources for omap-twl4030.c driver which is used by many boards. In DT bindings documentation is property ti,codec described as mandatory but it's never parsed in driver code. I did small experiment in gta04 board and remove ti,codec from devicetree and sound was working. I think it can be removed from documentation and also from various device trees. Is my assumption correct? Thanks. BR, marek -- as simple and primitive as possible ------------------------------------------------- Marek Belisko - OPEN-NANDRA Freelance Developer Ruska Nova Ves 219 | Presov, 08005 Slovak Republic Tel: +421 915 052 184 skype: marekwhite twitter: #opennandra web: http://open-nandra.com

2 2

[alsa-devel] [PATCH] ALSA: snd-usb: add quirks for Roland UA-22
by Daniel Mack 12 Mar '15

12 Mar '15

The device complies to the UAC1 standard but hides that fact with proprietary descriptors. The autodetect quirk for Roland devices catches the audio interface but misses the MIDI part, so a specific quirk is needed. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Reported-by: Rafa Lafuente <rafalafuente(a)gmail.com> Tested-by: Raphaël Doursenaud <raphael(a)doursenaud.fr> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 67d4765..07f984d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0159), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UA-22", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* this catches most recent vendor-specific Roland devices */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | -- 2.3.1

2 1

[alsa-devel] [PATCH 41/45] include/uapi/sound/emu10k1.h: hide gpr_valid, tram_valid and code_valid in userspace
by Mikko Rapeli 12 Mar '15

12 Mar '15

The DECLARE_BITMAP macro is not available in userspace headers. Fixes userspace compile error: error: expected specifier-qualifier-list before ‘DECLARE_BITMAP’ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/emu10k1.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index ec1535b..f2fd870 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -300,7 +300,9 @@ struct snd_emu10k1_fx8010_control_old_gpr { struct snd_emu10k1_fx8010_code { char name[128]; +#ifdef __KERNEL__ DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ +#endif __u32 __user *gpr_map; /* initializers */ unsigned int gpr_add_control_count; /* count of GPR controls to add/replace */ @@ -313,11 +315,15 @@ struct snd_emu10k1_fx8010_code { unsigned int gpr_list_control_total; /* total count of GPR controls */ struct snd_emu10k1_fx8010_control_gpr __user *gpr_list_controls; /* listed GPR controls */ +#ifdef __KERNEL__ DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ +#endif __u32 __user *tram_data_map; /* data initializers */ __u32 __user *tram_addr_map; /* map initializers */ +#ifdef __KERNEL__ DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ +#endif __u32 __user *code; /* one instruction - 64 bits */ }; -- 2.1.4

3 7

[alsa-devel] [PATCH] rate: Allow LGPL for libsamplerate commercial licensee
by Takashi Iwai 12 Mar '15

12 Mar '15

alsa-plugins-rate is tagged with GPL just for following the license of libsamplerate. However, libsamplerate provides a commercial license to allow it used with less restrictions. In that case, we also don't have to stick with GPL. Instead, the code follows LGPL like other plugin codes in this package. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- If anyone sees a legal problem with this, please let me know. I find it OK, but IANAL. rate/rate_samplerate.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/rate/rate_samplerate.c b/rate/rate_samplerate.c index 53a0627b21dc..0b14a59000bf 100644 --- a/rate/rate_samplerate.c +++ b/rate/rate_samplerate.c @@ -3,6 +3,14 @@ * * Copyright (c) 2006 by Takashi Iwai <tiwai(a)suse.de> * + * This plugin code is supposed to be used and distributed primarily + * under GPL v2 or later, in order to follow the license of libsamplerate. + * However, if you already own a commercial license to use libsamplerate + * for dynamic linking, this plugin code can be used and distributed also + * under LGPL v2.1 or later. + * + * For GPL-2.0+: + * * 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 @@ -13,9 +21,17 @@ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + * For LGPL-2.1+: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. */ #include <stdio.h> -- 2.3.2

1 0

[alsa-devel] [PATCH] sound:pci:Change return type of function, shift_bits in xonar_dg.c to avoid overflow
by Nicholas Krause 12 Mar '15

12 Mar '15

This changes the return type of the function,shift_bits to unsigned long in order to avoid overflowing issues with this function when called as found when running sparse on the kernel source tree. Signed-off-by: Nicholas Krause <xerofoify(a)gmail.com> --- sound/pci/oxygen/xonar_dg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200..2337a76 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -238,7 +238,7 @@ void set_cs4245_adc_params(struct oxygen *chip, cs4245_write_spi(chip, CS4245_MCLK_FREQ); } -static inline unsigned int shift_bits(unsigned int value, +static inline unsigned long shift_bits(unsigned int value, unsigned int shift_from, unsigned int shift_to, unsigned int mask) -- 2.1.0

3 2

[alsa-devel] [PATCH] ALSA: control: Add sanity checks for user ctl id name string
by Takashi Iwai 12 Mar '15

12 Mar '15

There was no check about the id string of user control elements, so we accepted even a control element with an empty string, which is obviously bogus. This patch adds more sanity checks of id strings. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index 35324a8e83c8..7ed2b214b16d 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1) return -EINVAL; + if (*info->id.name) + return -EINVAL; + if (strnlen(info->id.name, sizeof(info->id.name)) > sizeof(info->id.name)) + return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| -- 2.3.2

2 2

[alsa-devel] [PATCH 03/22] sound: %pF is only for function pointers
by Scott Wood 12 Mar '15

12 Mar '15

Use %pS for actual addresses, otherwise you'll get bad output on arches like ppc64 where %pF expects a function descriptor. Signed-off-by: Scott Wood <scottwood(a)freescale.com> Cc: alsa-devel(a)alsa-project.org --- sound/core/device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/device.c b/sound/core/device.c index 41bec30..4879711 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -127,7 +127,7 @@ void snd_device_free(struct snd_card *card, void *device_data) if (dev) __snd_device_free(dev); else - dev_dbg(card->dev, "device free %p (from %pF), not found\n", + dev_dbg(card->dev, "device free %p (from %pS), not found\n", device_data, __builtin_return_address(0)); } EXPORT_SYMBOL(snd_device_free); -- 2.1.0

1 0

[alsa-devel] [PATCH 0/8] Fix struct clk pointer comparing
by Shawn Guo 12 Mar '15

12 Mar '15

On the first day back from Chinese new year holiday, I got a regression report from rmk, saying Ethernet stops working on HimmingBoard with v4.0-rc1. I read through the thread [1] and found a couple of i.MX audio drivers are also affected per Stephen's Coccinelle report. That's why I came up with this series based on Quentin's structclk.cocci, Stephen's result and Mike's input (thanks all). The first two patches are run-time tested and all others are compile-tested only. [1] http://thread.gmane.org/gmane.linux.kernel/1872604/focus=1880862 Shawn Guo (8): clk: add helper function clk_is_match() ARM: imx: fix struct clk pointer comparing drm: armada: fix struct clk pointer comparing pwm: atmel-hlcdc: fix struct clk pointer comparing serial: samsung: fix struct clk pointer comparing ASoC: fsl_esai: fix struct clk pointer comparing ASoC: fsl_spdif: fix struct clk pointer comparing ASoC: kirkwood: fix struct clk pointer comparing arch/arm/mach-imx/mach-imx6q.c | 5 +++-- drivers/clk/clk.c | 6 ++++++ drivers/gpu/drm/armada/armada_510.c | 2 +- drivers/pwm/pwm-atmel-hlcdc.c | 2 +- drivers/tty/serial/samsung.c | 2 +- include/linux/clk.h | 16 ++++++++++++++++ sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 4 ++-- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 9 files changed, 32 insertions(+), 9 deletions(-) -- 1.9.1

15 40

[alsa-devel] [asoc:topic/max98925 1/1] sound/soc/codecs/max98925.c:189:1: sparse: symbol 'max98925_spk_tlv' was not declared. Should it be static?
by kbuild test robot 11 Mar '15

11 Mar '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/max98925 head: 31bdf73603f61a22c1643d1d887cf36eaa3a6834 commit: 31bdf73603f61a22c1643d1d887cf36eaa3a6834 [1/1] ASoC: Add max98925 codec driver reproduce: # apt-get install sparse git checkout 31bdf73603f61a22c1643d1d887cf36eaa3a6834 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/max98925.c:189:1: sparse: symbol 'max98925_spk_tlv' was not declared. Should it be static? Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

2 2

[alsa-devel] [PATCH 0/15] don't export static symbol
by Julia Lawall 11 Mar '15

11 Mar '15

These patches remove EXPORT_SYMBOL or EXPORT_SYMBOL_GPL declarations on static functions. This was done using the following semantic patch: (http://coccinelle.lip6.fr/) // <smpl> @r@ type T; identifier f; @@ static T f (...) { ... } @@ identifier r.f; declarer name EXPORT_SYMBOL; @@ -EXPORT_SYMBOL(f); @@ identifier r.f; declarer name EXPORT_SYMBOL_GPL; @@ -EXPORT_SYMBOL_GPL(f); // </smpl>

2 2

[alsa-devel] [PATCH] ASoC: Add max98925 codec driver
by Anish Kumar 11 Mar '15

11 Mar '15

Signed-off-by: Anish Kumar <yesanishhere(a)gmail.com> --- changes in this version: - addressed mark brown comments regarding probe function. - changed the mixer namings - added device tree documentation changes since v1: - addressed mark brown comments regarding open coding functions. - changed the mixer namings - added some dapm widgets .../devicetree/bindings/sound/max98925.txt | 22 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98925.c | 655 ++++++++++++++++ sound/soc/codecs/max98925.h | 832 +++++++++++++++++++++ 5 files changed, 1515 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98925.txt create mode 100644 sound/soc/codecs/max98925.c create mode 100644 sound/soc/codecs/max98925.h diff --git a/Documentation/devicetree/bindings/sound/max98925.txt b/Documentation/devicetree/bindings/sound/max98925.txt new file mode 100644 index 0000000..27be63e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98925.txt @@ -0,0 +1,22 @@ +max98925 audio CODEC + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98925" + + - vmon-slot-no : slot number used to send voltage information + + - imon-slot-no : slot number used to send current information + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98925@1a { + compatible = "maxim,max98925"; + vmon-slot-no = <0>; + imon-slot-no = <2>; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0bddd92..a7f44e3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,6 +68,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C + select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C @@ -455,6 +456,9 @@ config SND_SOC_MAX98088 config SND_SOC_MAX98090 tristate +config SND_SOC_MAX98925 + tristate + config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7acb6c1..1658d19 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -63,6 +63,7 @@ snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o +snd-soc-max98925-objs := max98925.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o snd-soc-max9850-objs := max9850.o @@ -247,6 +248,7 @@ obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o +obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c new file mode 100644 index 0000000..74f4f0b --- /dev/null +++ b/sound/soc/codecs/max98925.c @@ -0,0 +1,655 @@ +/* + * max98925.c -- ALSA SoC Stereo MAX98925 driver + * Copyright 2013-15 Maxim Integrated Products + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98925.h" + +static const char *const dai_text[] = { + "Left", "Right", "LeftRight", "LeftRightDiv2", +}; + +static const char * const max98925_boost_voltage_text[] = { + "8.5V", "8.25V", "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V", + "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V" +}; + +static SOC_ENUM_SINGLE_DECL(max98925_boost_voltage, + MAX98925_CONFIGURATION, M98925_BST_VOUT_SHIFT, + max98925_boost_voltage_text); + +static const char *const hpf_text[] = { + "Disable", "DC Block", "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static struct reg_default max98925_reg[] = { + { 0x0B, 0x00 }, /* IRQ Enable0 */ + { 0x0C, 0x00 }, /* IRQ Enable1 */ + { 0x0D, 0x00 }, /* IRQ Enable2 */ + { 0x0E, 0x00 }, /* IRQ Clear0 */ + { 0x0F, 0x00 }, /* IRQ Clear1 */ + { 0x10, 0x00 }, /* IRQ Clear2 */ + { 0x11, 0xC0 }, /* Map0 */ + { 0x12, 0x00 }, /* Map1 */ + { 0x13, 0x00 }, /* Map2 */ + { 0x14, 0xF0 }, /* Map3 */ + { 0x15, 0x00 }, /* Map4 */ + { 0x16, 0xAB }, /* Map5 */ + { 0x17, 0x89 }, /* Map6 */ + { 0x18, 0x00 }, /* Map7 */ + { 0x19, 0x00 }, /* Map8 */ + { 0x1A, 0x06 }, /* DAI Clock Mode 1 */ + { 0x1B, 0xC0 }, /* DAI Clock Mode 2 */ + { 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */ + { 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */ + { 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */ + { 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */ + { 0x20, 0x50 }, /* Format */ + { 0x21, 0x00 }, /* TDM Slot Select */ + { 0x22, 0x00 }, /* DOUT Configuration VMON */ + { 0x23, 0x00 }, /* DOUT Configuration IMON */ + { 0x24, 0x00 }, /* DOUT Configuration VBAT */ + { 0x25, 0x00 }, /* DOUT Configuration VBST */ + { 0x26, 0x00 }, /* DOUT Configuration FLAG */ + { 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */ + { 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */ + { 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */ + { 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */ + { 0x2B, 0x02 }, /* DOUT Drive Strength */ + { 0x2C, 0x90 }, /* Filters */ + { 0x2D, 0x00 }, /* Gain */ + { 0x2E, 0x02 }, /* Gain Ramping */ + { 0x2F, 0x00 }, /* Speaker Amplifier */ + { 0x30, 0x0A }, /* Threshold */ + { 0x31, 0x00 }, /* ALC Attack */ + { 0x32, 0x80 }, /* ALC Atten and Release */ + { 0x33, 0x00 }, /* ALC Infinite Hold Release */ + { 0x34, 0x92 }, /* ALC Configuration */ + { 0x35, 0x01 }, /* Boost Converter */ + { 0x36, 0x00 }, /* Block Enable */ + { 0x37, 0x00 }, /* Configuration */ + { 0x38, 0x00 }, /* Global Enable */ + { 0x3A, 0x00 }, /* Boost Limiter */ +}; + +static const struct soc_enum max98925_dai_enum = + SOC_ENUM_SINGLE(MAX98925_GAIN, 5, ARRAY_SIZE(dai_text), dai_text); + +static const struct soc_enum max98925_hpf_enum = + SOC_ENUM_SINGLE(MAX98925_FILTERS, 0, ARRAY_SIZE(hpf_text), hpf_text); + +static const struct snd_kcontrol_new max98925_hpf_sel_mux = + SOC_DAPM_ENUM("Rc Filter MUX Mux", max98925_hpf_enum); + +static const struct snd_kcontrol_new max98925_dai_sel_mux = + SOC_DAPM_ENUM("DAI IN MUX Mux", max98925_dai_enum); + +static int max98925_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(max98925->regmap, + MAX98925_BLOCK_ENABLE, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(max98925->regmap, + MAX98925_BLOCK_ENABLE, M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, 0); + break; + default: + return 0; + } + return 0; +} + +static const struct snd_soc_dapm_widget max98925_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("DAI IN MUX", SND_SOC_NOPM, 0, 0, + &max98925_dai_sel_mux), + SND_SOC_DAPM_MUX("Rc Filter MUX", SND_SOC_NOPM, 0, 0, + &max98925_hpf_sel_mux), + SND_SOC_DAPM_DAC_E("Amp Enable", NULL, MAX98925_BLOCK_ENABLE, + M98925_SPK_EN_SHIFT, 0, max98925_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98925_GLOBAL_ENABLE, + M98925_EN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static const struct snd_soc_dapm_route max98925_audio_map[] = { + {"DAI IN MUX", "Left", "DAI_OUT"}, + {"DAI IN MUX", "Right", "DAI_OUT"}, + {"DAI IN MUX", "LeftRight", "DAI_OUT"}, + {"DAI IN MUX", "LeftRightDiv2", "DAI_OUT"}, + {"Rc Filter MUX", "Disable", "DAI IN MUX"}, + {"Rc Filter MUX", "DC Block", "DAI IN MUX"}, + {"Rc Filter MUX", "100Hz", "DAI IN MUX"}, + {"Rc Filter MUX", "200Hz", "DAI IN MUX"}, + {"Rc Filter MUX", "400Hz", "DAI IN MUX"}, + {"Rc Filter MUX", "800Hz", "DAI IN MUX"}, + {"Amp Enable", NULL, "Rc Filter MUX"}, + {"BE_OUT", NULL, "Amp Enable"}, + {"BE_OUT", NULL, "Global Enable"}, +}; + +static bool max98925_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_VBAT_DATA: + case MAX98925_VBST_DATA: + case MAX98925_LIVE_STATUS0: + case MAX98925_LIVE_STATUS1: + case MAX98925_LIVE_STATUS2: + case MAX98925_STATE0: + case MAX98925_STATE1: + case MAX98925_STATE2: + case MAX98925_FLAG0: + case MAX98925_FLAG1: + case MAX98925_FLAG2: + case MAX98925_REV_VERSION: + return true; + default: + return false; + } +} + +static bool max98925_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_IRQ_CLEAR0: + case MAX98925_IRQ_CLEAR1: + case MAX98925_IRQ_CLEAR2: + case MAX98925_ALC_HOLD_RLS: + return false; + default: + return true; + } +} + +DECLARE_TLV_DB_SCALE(max98925_spk_tlv, -600, 100, 0); + +static const struct snd_kcontrol_new max98925_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98925_GAIN, + M98925_SPK_GAIN_SHIFT, (1<<M98925_SPK_GAIN_WIDTH)-1, 0, + max98925_spk_tlv), + SOC_SINGLE("Ramp Switch", MAX98925_GAIN_RAMPING, + M98925_SPK_RMP_EN_SHIFT, 1, 0), + SOC_SINGLE("ZCD Switch", MAX98925_GAIN_RAMPING, + M98925_SPK_ZCD_EN_SHIFT, 1, 0), + SOC_SINGLE("ALC Switch", MAX98925_THRESHOLD, + M98925_ALC_EN_SHIFT, 1, 0), + SOC_SINGLE("ALC Threshold", MAX98925_THRESHOLD, M98925_ALC_TH_SHIFT, + (1<<M98925_ALC_TH_WIDTH)-1, 0), + SOC_ENUM("Boost Output Voltage", max98925_boost_voltage), +}; + +/* codec sample rate and n/m dividers parameter table */ +static const struct { + int rate; + int sr; + int divisors[3][2]; +} rate_table[] = { + { + .rate = 8000, + .sr = 0, + .divisors = { {1, 375}, {5, 1764}, {1, 384} } + }, + { + .rate = 11025, + .sr = 1, + .divisors = { {147, 40000}, {1, 256}, {147, 40960} } + }, + { + .rate = 12000, + .sr = 2, + .divisors = { {1, 250}, {5, 1176}, {1, 256} } + }, + { + .rate = 16000, + .sr = 3, + .divisors = { {2, 375}, {5, 882}, {1, 192} } + }, + { + .rate = 22050, + .sr = 4, + .divisors = { {147, 20000}, {1, 128}, {147, 20480} } + }, + { + .rate = 24000, + .sr = 5, + .divisors = { {1, 125}, {5, 588}, {1, 128} } + }, + { + .rate = 32000, + .sr = 6, + .divisors = { {4, 375}, {5, 441}, {1, 96} } + }, + { + .rate = 44100, + .sr = 7, + .divisors = { {147, 10000}, {1, 64}, {147, 10240} } + }, + { + .rate = 48000, + .sr = 8, + .divisors = { {2, 125}, {5, 294}, {1, 64} } + }, +}; + +static inline int max98925_rate_value(struct snd_soc_codec *codec, + int rate, int clock, int *value, int *n, int *m) +{ + int ret = -EINVAL; + int i; + + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i].rate >= rate) { + *value = rate_table[i].sr; + *n = rate_table[i].divisors[clock][0]; + *m = rate_table[i].divisors[clock][1]; + ret = 0; + break; + } + } + dev_dbg(codec->dev, "%s: sample rate is %d, returning %d\n", + __func__, rate_table[i].rate, *value); + return ret; +} + +static void max98925_set_sense_data(struct max98925_priv *max98925) +{ + /* set VMON slots */ + regmap_update_bits(max98925->regmap, + MAX98925_DOUT_CFG_VMON, + M98925_DAI_VMON_EN_MASK, M98925_DAI_VMON_EN_MASK); + regmap_update_bits(max98925->regmap, + MAX98925_DOUT_CFG_VMON, + M98925_DAI_VMON_SLOT_MASK, + max98925->v_slot << M98925_DAI_VMON_SLOT_SHIFT); + /* set IMON slots */ + regmap_update_bits(max98925->regmap, + MAX98925_DOUT_CFG_IMON, + M98925_DAI_IMON_EN_MASK, M98925_DAI_IMON_EN_MASK); + regmap_update_bits(max98925->regmap, + MAX98925_DOUT_CFG_IMON, + M98925_DAI_IMON_SLOT_MASK, + max98925->i_slot << M98925_DAI_IMON_SLOT_SHIFT); +} + +static int max98925_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* set DAI to slave mode */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, 0); + max98925_set_sense_data(max98925); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* + * set left channel DAI to master mode, + * right channel always slave + */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, M98925_DAI_MAS_MASK); + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + invert = M98925_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + invert = M98925_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert = M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98925->regmap, MAX98925_FORMAT, + M98925_DAI_BCI_MASK | M98925_DAI_BCI_MASK, invert); + return 0; +} + +static int max98925_set_clock(struct max98925_priv *max98925, + struct snd_pcm_hw_params *params) +{ + unsigned int dai_sr = 0, clock, mdll, n, m; + struct snd_soc_codec *codec = max98925->codec; + int rate = params_rate(params); + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98925->ch_size; + + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (max98925->sysclk) { + case 6000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx16; + break; + case 11289600: + clock = 1; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12288000: + clock = 2; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + default: + dev_info(max98925->codec->dev, "unsupported sysclk %d\n", + max98925->sysclk); + return -EINVAL; + } + + if (max98925_rate_value(codec, rate, clock, &dai_sr, &n, &m)) + return -EINVAL; + + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_SR_MASK, dai_sr << M98925_DAI_SR_SHIFT); + /* set DAI m divider */ + regmap_write(max98925->regmap, + MAX98925_DAI_CLK_DIV_M_MSBS, m >> 8); + regmap_write(max98925->regmap, + MAX98925_DAI_CLK_DIV_M_LSBS, m & 0xFF); + /* set DAI n divider */ + regmap_write(max98925->regmap, + MAX98925_DAI_CLK_DIV_N_MSBS, n >> 8); + regmap_write(max98925->regmap, + MAX98925_DAI_CLK_DIV_N_LSBS, n & 0xFF); + /* set MDLL */ + regmap_update_bits(max98925->regmap, MAX98925_DAI_CLK_MODE1, + M98925_MDLL_MULT_MASK, mdll << M98925_MDLL_MULT_SHIFT); + return 0; +} + +static int max98925_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + switch (snd_pcm_format_width(params_format(params))) { + case 16: + regmap_update_bits(max98925->regmap, + MAX98925_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_16); + max98925->ch_size = 16; + break; + case 24: + regmap_update_bits(max98925->regmap, + MAX98925_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + max98925->ch_size = 32; + break; + case 32: + regmap_update_bits(max98925->regmap, + MAX98925_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + max98925->ch_size = 32; + break; + default: + pr_err("%s: format unsupported %d", + __func__, params_format(params)); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: format supported %d", + __func__, params_format(params)); + return max98925_set_clock(max98925, params); +} + +static int max98925_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case 0: + /* use MCLK for Left channel, right channel always BCLK */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, 0); + break; + case 1: + /* configure dai clock source to BCLK instead of MCLK */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, + M98925_DAI_CLK_SOURCE_MASK); + break; + default: + return -EINVAL; + } + max98925->sysclk = freq; + return 0; +} + +#define MAX98925_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops max98925_dai_ops = { + .set_sysclk = max98925_dai_set_sysclk, + .set_fmt = max98925_dai_set_fmt, + .hw_params = max98925_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98925_dai[] = { + { + .name = "max98925-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .ops = &max98925_dai_ops, + } +}; + +static int max98925_probe(struct snd_soc_codec *codec) +{ + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + max98925->codec = codec; + codec->control_data = max98925->regmap; + regmap_write(max98925->regmap, MAX98925_GLOBAL_ENABLE, 0x00); + /* It's not the default but we need to set DAI_DLY */ + regmap_write(max98925->regmap, + MAX98925_FORMAT, M98925_DAI_DLY_MASK); + regmap_write(max98925->regmap, MAX98925_TDM_SLOT_SELECT, 0xC8); + regmap_write(max98925->regmap, MAX98925_DOUT_HIZ_CFG1, 0xFF); + regmap_write(max98925->regmap, MAX98925_DOUT_HIZ_CFG2, 0xFF); + regmap_write(max98925->regmap, MAX98925_DOUT_HIZ_CFG3, 0xFF); + regmap_write(max98925->regmap, MAX98925_DOUT_HIZ_CFG4, 0xF0); + regmap_write(max98925->regmap, MAX98925_FILTERS, 0xD8); + regmap_write(max98925->regmap, MAX98925_ALC_CONFIGURATION, 0xF8); + regmap_write(max98925->regmap, MAX98925_CONFIGURATION, 0xF0); + /* Disable ALC muting */ + regmap_write(max98925->regmap, MAX98925_BOOST_LIMITER, 0xF8); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98925 = { + .probe = max98925_probe, + .controls = max98925_snd_controls, + .num_controls = ARRAY_SIZE(max98925_snd_controls), + .dapm_routes = max98925_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98925_audio_map), + .dapm_widgets = max98925_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98925_dapm_widgets), +}; + +static struct regmap_config max98925_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98925_REV_VERSION, + .reg_defaults = max98925_reg, + .num_reg_defaults = ARRAY_SIZE(max98925_reg), + .volatile_reg = max98925_volatile_register, + .readable_reg = max98925_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98925_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + int ret, reg; + u32 value; + struct max98925_priv *max98925; + + max98925 = devm_kzalloc(&i2c->dev, + sizeof(*max98925), GFP_KERNEL); + if (!max98925) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98925); + max98925->regmap = devm_regmap_init_i2c(i2c, &max98925_regmap); + if (IS_ERR(max98925->regmap)) { + ret = PTR_ERR(max98925->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + goto err_out; + } + + if (!of_property_read_u32(i2c->dev.of_node, "vmon-slot-no", &value)) { + if (value > M98925_DAI_VMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "vmon slot number is wrong:\n"); + return -EINVAL; + } + max98925->v_slot = value; + } + if (!of_property_read_u32(i2c->dev.of_node, "imon-slot-no", &value)) { + if (value > M98925_DAI_IMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "imon slot number is wrong:\n"); + return -EINVAL; + } + max98925->i_slot = value; + } + ret = regmap_read(max98925->regmap, + MAX98925_REV_VERSION, &reg); + if ((ret < 0) || + ((reg != MAX98925_VERSION) && + (reg != MAX98925_VERSION1))) { + dev_err(&i2c->dev, + "device initialization error (%d 0x%02X)\n", + ret, reg); + goto err_out; + } + dev_info(&i2c->dev, "device version 0x%02X\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98925, + max98925_dai, ARRAY_SIZE(max98925_dai)); + if (ret < 0) + dev_err(&i2c->dev, + "Failed to register codec: %d\n", ret); +err_out: + return ret; +} + +static int max98925_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98925_i2c_id[] = { + { "max98925", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98925_i2c_id); + +static const struct of_device_id max98925_of_match[] = { + { .compatible = "maxim,max98925", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98925_of_match); + +static struct i2c_driver max98925_i2c_driver = { + .driver = { + .name = "max98925", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98925_of_match), + .pm = NULL, + }, + .probe = max98925_i2c_probe, + .remove = max98925_i2c_remove, + .id_table = max98925_i2c_id, +}; + +module_i2c_driver(max98925_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98925 driver"); +MODULE_AUTHOR("Ralph Birt <rdbirt(a)gmail.com>, Anish kumar <anish.kumar(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98925.h b/sound/soc/codecs/max98925.h new file mode 100644 index 0000000..3783248 --- /dev/null +++ b/sound/soc/codecs/max98925.h @@ -0,0 +1,832 @@ +/* + * max98925.h -- MAX98925 ALSA SoC Audio driver + * + * Copyright 2013-2015 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98925_H +#define _MAX98925_H + +#define MAX98925_VERSION 0x51 +#define MAX98925_VERSION1 0x80 +#define MAX98925_VBAT_DATA 0x00 +#define MAX98925_VBST_DATA 0x01 +#define MAX98925_LIVE_STATUS0 0x02 +#define MAX98925_LIVE_STATUS1 0x03 +#define MAX98925_LIVE_STATUS2 0x04 +#define MAX98925_STATE0 0x05 +#define MAX98925_STATE1 0x06 +#define MAX98925_STATE2 0x07 +#define MAX98925_FLAG0 0x08 +#define MAX98925_FLAG1 0x09 +#define MAX98925_FLAG2 0x0A +#define MAX98925_IRQ_ENABLE0 0x0B +#define MAX98925_IRQ_ENABLE1 0x0C +#define MAX98925_IRQ_ENABLE2 0x0D +#define MAX98925_IRQ_CLEAR0 0x0E +#define MAX98925_IRQ_CLEAR1 0x0F +#define MAX98925_IRQ_CLEAR2 0x10 +#define MAX98925_MAP0 0x11 +#define MAX98925_MAP1 0x12 +#define MAX98925_MAP2 0x13 +#define MAX98925_MAP3 0x14 +#define MAX98925_MAP4 0x15 +#define MAX98925_MAP5 0x16 +#define MAX98925_MAP6 0x17 +#define MAX98925_MAP7 0x18 +#define MAX98925_MAP8 0x19 +#define MAX98925_DAI_CLK_MODE1 0x1A +#define MAX98925_DAI_CLK_MODE2 0x1B +#define MAX98925_DAI_CLK_DIV_M_MSBS 0x1C +#define MAX98925_DAI_CLK_DIV_M_LSBS 0x1D +#define MAX98925_DAI_CLK_DIV_N_MSBS 0x1E +#define MAX98925_DAI_CLK_DIV_N_LSBS 0x1F +#define MAX98925_FORMAT 0x20 +#define MAX98925_TDM_SLOT_SELECT 0x21 +#define MAX98925_DOUT_CFG_VMON 0x22 +#define MAX98925_DOUT_CFG_IMON 0x23 +#define MAX98925_DOUT_CFG_VBAT 0x24 +#define MAX98925_DOUT_CFG_VBST 0x25 +#define MAX98925_DOUT_CFG_FLAG 0x26 +#define MAX98925_DOUT_HIZ_CFG1 0x27 +#define MAX98925_DOUT_HIZ_CFG2 0x28 +#define MAX98925_DOUT_HIZ_CFG3 0x29 +#define MAX98925_DOUT_HIZ_CFG4 0x2A +#define MAX98925_DOUT_DRV_STRENGTH 0x2B +#define MAX98925_FILTERS 0x2C +#define MAX98925_GAIN 0x2D +#define MAX98925_GAIN_RAMPING 0x2E +#define MAX98925_SPK_AMP 0x2F +#define MAX98925_THRESHOLD 0x30 +#define MAX98925_ALC_ATTACK 0x31 +#define MAX98925_ALC_ATTEN_RLS 0x32 +#define MAX98925_ALC_HOLD_RLS 0x33 +#define MAX98925_ALC_CONFIGURATION 0x34 +#define MAX98925_BOOST_CONVERTER 0x35 +#define MAX98925_BLOCK_ENABLE 0x36 +#define MAX98925_CONFIGURATION 0x37 +#define MAX98925_GLOBAL_ENABLE 0x38 +#define MAX98925_BOOST_LIMITER 0x3A +#define MAX98925_REV_VERSION 0xFF + +#define MAX98925_REG_CNT (MAX98925_R03A_BOOST_LIMITER+1) + +/* MAX98925 Register Bit Fields */ + +/* MAX98925_R002_LIVE_STATUS0 */ +#define M98925_THERMWARN_STATUS_MASK (1<<3) +#define M98925_THERMWARN_STATUS_SHIFT 3 +#define M98925_THERMWARN_STATUS_WIDTH 1 +#define M98925_THERMSHDN_STATUS_MASK (1<<1) +#define M98925_THERMSHDN_STATUS_SHIFT 1 +#define M98925_THERMSHDN_STATUS_WIDTH 1 + +/* MAX98925_R003_LIVE_STATUS1 */ +#define M98925_SPKCURNT_STATUS_MASK (1<<5) +#define M98925_SPKCURNT_STATUS_SHIFT 5 +#define M98925_SPKCURNT_STATUS_WIDTH 1 +#define M98925_WATCHFAIL_STATUS_MASK (1<<4) +#define M98925_WATCHFAIL_STATUS_SHIFT 4 +#define M98925_WATCHFAIL_STATUS_WIDTH 1 +#define M98925_ALCINFH_STATUS_MASK (1<<3) +#define M98925_ALCINFH_STATUS_SHIFT 3 +#define M98925_ALCINFH_STATUS_WIDTH 1 +#define M98925_ALCACT_STATUS_MASK (1<<2) +#define M98925_ALCACT_STATUS_SHIFT 2 +#define M98925_ALCACT_STATUS_WIDTH 1 +#define M98925_ALCMUT_STATUS_MASK (1<<1) +#define M98925_ALCMUT_STATUS_SHIFT 1 +#define M98925_ALCMUT_STATUS_WIDTH 1 +#define M98925_ACLP_STATUS_MASK (1<<0) +#define M98925_ACLP_STATUS_SHIFT 0 +#define M98925_ACLP_STATUS_WIDTH 1 + +/* MAX98925_R004_LIVE_STATUS2 */ +#define M98925_SLOTOVRN_STATUS_MASK (1<<6) +#define M98925_SLOTOVRN_STATUS_SHIFT 6 +#define M98925_SLOTOVRN_STATUS_WIDTH 1 +#define M98925_INVALSLOT_STATUS_MASK (1<<5) +#define M98925_INVALSLOT_STATUS_SHIFT 5 +#define M98925_INVALSLOT_STATUS_WIDTH 1 +#define M98925_SLOTCNFLT_STATUS_MASK (1<<4) +#define M98925_SLOTCNFLT_STATUS_SHIFT 4 +#define M98925_SLOTCNFLT_STATUS_WIDTH 1 +#define M98925_VBSTOVFL_STATUS_MASK (1<<3) +#define M98925_VBSTOVFL_STATUS_SHIFT 3 +#define M98925_VBSTOVFL_STATUS_WIDTH 1 +#define M98925_VBATOVFL_STATUS_MASK (1<<2) +#define M98925_VBATOVFL_STATUS_SHIFT 2 +#define M98925_VBATOVFL_STATUS_WIDTH 1 +#define M98925_IMONOVFL_STATUS_MASK (1<<1) +#define M98925_IMONOVFL_STATUS_SHIFT 1 +#define M98925_IMONOVFL_STATUS_WIDTH 1 +#define M98925_VMONOVFL_STATUS_MASK (1<<0) +#define M98925_VMONOVFL_STATUS_SHIFT 0 +#define M98925_VMONOVFL_STATUS_WIDTH 1 + +/* MAX98925_R005_STATE0 */ +#define M98925_THERMWARN_END_STATE_MASK (1<<3) +#define M98925_THERMWARN_END_STATE_SHIFT 3 +#define M98925_THERMWARN_END_STATE_WIDTH 1 +#define M98925_THERMWARN_BGN_STATE_MASK (1<<2) +#define M98925_THERMWARN_BGN_STATE_SHIFT 1 +#define M98925_THERMWARN_BGN_STATE_WIDTH 1 +#define M98925_THERMSHDN_END_STATE_MASK (1<<1) +#define M98925_THERMSHDN_END_STATE_SHIFT 1 +#define M98925_THERMSHDN_END_STATE_WIDTH 1 +#define M98925_THERMSHDN_BGN_STATE_MASK (1<<0) +#define M98925_THERMSHDN_BGN_STATE_SHIFT 0 +#define M98925_THERMSHDN_BGN_STATE_WIDTH 1 + +/* MAX98925_R006_STATE1 */ +#define M98925_SPRCURNT_STATE_MASK (1<<5) +#define M98925_SPRCURNT_STATE_SHIFT 5 +#define M98925_SPRCURNT_STATE_WIDTH 1 +#define M98925_WATCHFAIL_STATE_MASK (1<<4) +#define M98925_WATCHFAIL_STATE_SHIFT 4 +#define M98925_WATCHFAIL_STATE_WIDTH 1 +#define M98925_ALCINFH_STATE_MASK (1<<3) +#define M98925_ALCINFH_STATE_SHIFT 3 +#define M98925_ALCINFH_STATE_WIDTH 1 +#define M98925_ALCACT_STATE_MASK (1<<2) +#define M98925_ALCACT_STATE_SHIFT 2 +#define M98925_ALCACT_STATE_WIDTH 1 +#define M98925_ALCMUT_STATE_MASK (1<<1) +#define M98925_ALCMUT_STATE_SHIFT 1 +#define M98925_ALCMUT_STATE_WIDTH 1 +#define M98925_ALCP_STATE_MASK (1<<0) +#define M98925_ALCP_STATE_SHIFT 0 +#define M98925_ALCP_STATE_WIDTH 1 + +/* MAX98925_R007_STATE2 */ +#define M98925_SLOTOVRN_STATE_MASK (1<<6) +#define M98925_SLOTOVRN_STATE_SHIFT 6 +#define M98925_SLOTOVRN_STATE_WIDTH 1 +#define M98925_INVALSLOT_STATE_MASK (1<<5) +#define M98925_INVALSLOT_STATE_SHIFT 5 +#define M98925_INVALSLOT_STATE_WIDTH 1 +#define M98925_SLOTCNFLT_STATE_MASK (1<<4) +#define M98925_SLOTCNFLT_STATE_SHIFT 4 +#define M98925_SLOTCNFLT_STATE_WIDTH 1 +#define M98925_VBSTOVFL_STATE_MASK (1<<3) +#define M98925_VBSTOVFL_STATE_SHIFT 3 +#define M98925_VBSTOVFL_STATE_WIDTH 1 +#define M98925_VBATOVFL_STATE_MASK (1<<2) +#define M98925_VBATOVFL_STATE_SHIFT 2 +#define M98925_VBATOVFL_STATE_WIDTH 1 +#define M98925_IMONOVFL_STATE_MASK (1<<1) +#define M98925_IMONOVFL_STATE_SHIFT 1 +#define M98925_IMONOVFL_STATE_WIDTH 1 +#define M98925_VMONOVFL_STATE_MASK (1<<0) +#define M98925_VMONOVFL_STATE_SHIFT 0 +#define M98925_VMONOVFL_STATE_WIDTH 1 + +/* MAX98925_R008_FLAG0 */ +#define M98925_THERMWARN_END_FLAG_MASK (1<<3) +#define M98925_THERMWARN_END_FLAG_SHIFT 3 +#define M98925_THERMWARN_END_FLAG_WIDTH 1 +#define M98925_THERMWARN_BGN_FLAG_MASK (1<<2) +#define M98925_THERMWARN_BGN_FLAG_SHIFT 2 +#define M98925_THERMWARN_BGN_FLAG_WIDTH 1 +#define M98925_THERMSHDN_END_FLAG_MASK (1<<1) +#define M98925_THERMSHDN_END_FLAG_SHIFT 1 +#define M98925_THERMSHDN_END_FLAG_WIDTH 1 +#define M98925_THERMSHDN_BGN_FLAG_MASK (1<<0) +#define M98925_THERMSHDN_BGN_FLAG_SHIFT 0 +#define M98925_THERMSHDN_BGN_FLAG_WIDTH 1 + +/* MAX98925_R009_FLAG1 */ +#define M98925_SPKCURNT_FLAG_MASK (1<<5) +#define M98925_SPKCURNT_FLAG_SHIFT 5 +#define M98925_SPKCURNT_FLAG_WIDTH 1 +#define M98925_WATCHFAIL_FLAG_MASK (1<<4) +#define M98925_WATCHFAIL_FLAG_SHIFT 4 +#define M98925_WATCHFAIL_FLAG_WIDTH 1 +#define M98925_ALCINFH_FLAG_MASK (1<<3) +#define M98925_ALCINFH_FLAG_SHIFT 3 +#define M98925_ALCINFH_FLAG_WIDTH 1 +#define M98925_ALCACT_FLAG_MASK (1<<2) +#define M98925_ALCACT_FLAG_SHIFT 2 +#define M98925_ALCACT_FLAG_WIDTH 1 +#define M98925_ALCMUT_FLAG_MASK (1<<1) +#define M98925_ALCMUT_FLAG_SHIFT 1 +#define M98925_ALCMUT_FLAG_WIDTH 1 +#define M98925_ALCP_FLAG_MASK (1<<0) +#define M98925_ALCP_FLAG_SHIFT 0 +#define M98925_ALCP_FLAG_WIDTH 1 + +/* MAX98925_R00A_FLAG2 */ +#define M98925_SLOTOVRN_FLAG_MASK (1<<6) +#define M98925_SLOTOVRN_FLAG_SHIFT 6 +#define M98925_SLOTOVRN_FLAG_WIDTH 1 +#define M98925_INVALSLOT_FLAG_MASK (1<<5) +#define M98925_INVALSLOT_FLAG_SHIFT 5 +#define M98925_INVALSLOT_FLAG_WIDTH 1 +#define M98925_SLOTCNFLT_FLAG_MASK (1<<4) +#define M98925_SLOTCNFLT_FLAG_SHIFT 4 +#define M98925_SLOTCNFLT_FLAG_WIDTH 1 +#define M98925_VBSTOVFL_FLAG_MASK (1<<3) +#define M98925_VBSTOVFL_FLAG_SHIFT 3 +#define M98925_VBSTOVFL_FLAG_WIDTH 1 +#define M98925_VBATOVFL_FLAG_MASK (1<<2) +#define M98925_VBATOVFL_FLAG_SHIFT 2 +#define M98925_VBATOVFL_FLAG_WIDTH 1 +#define M98925_IMONOVFL_FLAG_MASK (1<<1) +#define M98925_IMONOVFL_FLAG_SHIFT 1 +#define M98925_IMONOVFL_FLAG_WIDTH 1 +#define M98925_VMONOVFL_FLAG_MASK (1<<0) +#define M98925_VMONOVFL_FLAG_SHIFT 0 +#define M98925_VMONOVFL_FLAG_WIDTH 1 + +/* MAX98925_R00B_IRQ_ENABLE0 */ +#define M98925_THERMWARN_END_EN_MASK (1<<3) +#define M98925_THERMWARN_END_EN_SHIFT 3 +#define M98925_THERMWARN_END_EN_WIDTH 1 +#define M98925_THERMWARN_BGN_EN_MASK (1<<2) +#define M98925_THERMWARN_BGN_EN_SHIFT 2 +#define M98925_THERMWARN_BGN_EN_WIDTH 1 +#define M98925_THERMSHDN_END_EN_MASK (1<<1) +#define M98925_THERMSHDN_END_EN_SHIFT 1 +#define M98925_THERMSHDN_END_EN_WIDTH 1 +#define M98925_THERMSHDN_BGN_EN_MASK (1<<0) +#define M98925_THERMSHDN_BGN_EN_SHIFT 0 +#define M98925_THERMSHDN_BGN_EN_WIDTH 1 + +/* MAX98925_R00C_IRQ_ENABLE1 */ +#define M98925_SPKCURNT_EN_MASK (1<<5) +#define M98925_SPKCURNT_EN_SHIFT 5 +#define M98925_SPKCURNT_EN_WIDTH 1 +#define M98925_WATCHFAIL_EN_MASK (1<<4) +#define M98925_WATCHFAIL_EN_SHIFT 4 +#define M98925_WATCHFAIL_EN_WIDTH 1 +#define M98925_ALCINFH_EN_MASK (1<<3) +#define M98925_ALCINFH_EN_SHIFT 3 +#define M98925_ALCINFH_EN_WIDTH 1 +#define M98925_ALCACT_EN_MASK (1<<2) +#define M98925_ALCACT_EN_SHIFT 2 +#define M98925_ALCACT_EN_WIDTH 1 +#define M98925_ALCMUT_EN_MASK (1<<1) +#define M98925_ALCMUT_EN_SHIFT 1 +#define M98925_ALCMUT_EN_WIDTH 1 +#define M98925_ALCP_EN_MASK (1<<0) +#define M98925_ALCP_EN_SHIFT 0 +#define M98925_ALCP_EN_WIDTH 1 + +/* MAX98925_R00D_IRQ_ENABLE2 */ +#define M98925_SLOTOVRN_EN_MASK (1<<6) +#define M98925_SLOTOVRN_EN_SHIFT 6 +#define M98925_SLOTOVRN_EN_WIDTH 1 +#define M98925_INVALSLOT_EN_MASK (1<<5) +#define M98925_INVALSLOT_EN_SHIFT 5 +#define M98925_INVALSLOT_EN_WIDTH 1 +#define M98925_SLOTCNFLT_EN_MASK (1<<4) +#define M98925_SLOTCNFLT_EN_SHIFT 4 +#define M98925_SLOTCNFLT_EN_WIDTH 1 +#define M98925_VBSTOVFL_EN_MASK (1<<3) +#define M98925_VBSTOVFL_EN_SHIFT 3 +#define M98925_VBSTOVFL_EN_WIDTH 1 +#define M98925_VBATOVFL_EN_MASK (1<<2) +#define M98925_VBATOVFL_EN_SHIFT 2 +#define M98925_VBATOVFL_EN_WIDTH 1 +#define M98925_IMONOVFL_EN_MASK (1<<1) +#define M98925_IMONOVFL_EN_SHIFT 1 +#define M98925_IMONOVFL_EN_WIDTH 1 +#define M98925_VMONOVFL_EN_MASK (1<<0) +#define M98925_VMONOVFL_EN_SHIFT 0 +#define M98925_VMONOVFL_EN_WIDTH 1 + +/* MAX98925_R00E_IRQ_CLEAR0 */ +#define M98925_THERMWARN_END_CLR_MASK (1<<3) +#define M98925_THERMWARN_END_CLR_SHIFT 3 +#define M98925_THERMWARN_END_CLR_WIDTH 1 +#define M98925_THERMWARN_BGN_CLR_MASK (1<<2) +#define M98925_THERMWARN_BGN_CLR_SHIFT 2 +#define M98925_THERMWARN_BGN_CLR_WIDTH 1 +#define M98925_THERMSHDN_END_CLR_MASK (1<<1) +#define M98925_THERMSHDN_END_CLR_SHIFT 1 +#define M98925_THERMSHDN_END_CLR_WIDTH 1 +#define M98925_THERMSHDN_BGN_CLR_MASK (1<<0) +#define M98925_THERMSHDN_BGN_CLR_SHIFT 0 +#define M98925_THERMSHDN_BGN_CLR_WIDTH 1 + +/* MAX98925_R00F_IRQ_CLEAR1 */ +#define M98925_SPKCURNT_CLR_MASK (1<<5) +#define M98925_SPKCURNT_CLR_SHIFT 5 +#define M98925_SPKCURNT_CLR_WIDTH 1 +#define M98925_WATCHFAIL_CLR_MASK (1<<4) +#define M98925_WATCHFAIL_CLR_SHIFT 4 +#define M98925_WATCHFAIL_CLR_WIDTH 1 +#define M98925_ALCINFH_CLR_MASK (1<<3) +#define M98925_ALCINFH_CLR_SHIFT 3 +#define M98925_ALCINFH_CLR_WIDTH 1 +#define M98925_ALCACT_CLR_MASK (1<<2) +#define M98925_ALCACT_CLR_SHIFT 2 +#define M98925_ALCACT_CLR_WIDTH 1 +#define M98925_ALCMUT_CLR_MASK (1<<1) +#define M98925_ALCMUT_CLR_SHIFT 1 +#define M98925_ALCMUT_CLR_WIDTH 1 +#define M98925_ALCP_CLR_MASK (1<<0) +#define M98925_ALCP_CLR_SHIFT 0 +#define M98925_ALCP_CLR_WIDTH 1 + +/* MAX98925_R010_IRQ_CLEAR2 */ +#define M98925_SLOTOVRN_CLR_MASK (1<<6) +#define M98925_SLOTOVRN_CLR_SHIFT 6 +#define M98925_SLOTOVRN_CLR_WIDTH 1 +#define M98925_INVALSLOT_CLR_MASK (1<<5) +#define M98925_INVALSLOT_CLR_SHIFT 5 +#define M98925_INVALSLOT_CLR_WIDTH 1 +#define M98925_SLOTCNFLT_CLR_MASK (1<<4) +#define M98925_SLOTCNFLT_CLR_SHIFT 4 +#define M98925_SLOTCNFLT_CLR_WIDTH 1 +#define M98925_VBSTOVFL_CLR_MASK (1<<3) +#define M98925_VBSTOVFL_CLR_SHIFT 3 +#define M98925_VBSTOVFL_CLR_WIDTH 1 +#define M98925_VBATOVFL_CLR_MASK (1<<2) +#define M98925_VBATOVFL_CLR_SHIFT 2 +#define M98925_VBATOVFL_CLR_WIDTH 1 +#define M98925_IMONOVFL_CLR_MASK (1<<1) +#define M98925_IMONOVFL_CLR_SHIFT 1 +#define M98925_IMONOVFL_CLR_WIDTH 1 +#define M98925_VMONOVFL_CLR_MASK (1<<0) +#define M98925_VMONOVFL_CLR_SHIFT 0 +#define M98925_VMONOVFL_CLR_WIDTH 1 + +/* MAX98925_R011_MAP0 */ +#define M98925_ER_THERMWARN_EN_MASK (1<<7) +#define M98925_ER_THERMWARN_EN_SHIFT 7 +#define M98925_ER_THERMWARN_EN_WIDTH 1 +#define M98925_ER_THERMWARN_MAP_MASK (0x07<<4) +#define M98925_ER_THERMWARN_MAP_SHIFT 4 +#define M98925_ER_THERMWARN_MAP_WIDTH 3 + +/* MAX98925_R012_MAP1 */ +#define M98925_ER_ALCMUT_EN_MASK (1<<7) +#define M98925_ER_ALCMUT_EN_SHIFT 7 +#define M98925_ER_ALCMUT_EN_WIDTH 1 +#define M98925_ER_ALCMUT_MAP_MASK (0x07<<4) +#define M98925_ER_ALCMUT_MAP_SHIFT 4 +#define M98925_ER_ALCMUT_MAP_WIDTH 3 +#define M98925_ER_ALCP_EN_MASK (1<<3) +#define M98925_ER_ALCP_EN_SHIFT 3 +#define M98925_ER_ALCP_EN_WIDTH 1 +#define M98925_ER_ALCP_MAP_MASK (0x07<<0) +#define M98925_ER_ALCP_MAP_SHIFT 0 +#define M98925_ER_ALCP_MAP_WIDTH 3 + +/* MAX98925_R013_MAP2 */ +#define M98925_ER_ALCINFH_EN_MASK (1<<7) +#define M98925_ER_ALCINFH_EN_SHIFT 7 +#define M98925_ER_ALCINFH_EN_WIDTH 1 +#define M98925_ER_ALCINFH_MAP_MASK (0x07<<4) +#define M98925_ER_ALCINFH_MAP_SHIFT 4 +#define M98925_ER_ALCINFH_MAP_WIDTH 3 +#define M98925_ER_ALCACT_EN_MASK (1<<3) +#define M98925_ER_ALCACT_EN_SHIFT 3 +#define M98925_ER_ALCACT_EN_WIDTH 1 +#define M98925_ER_ALCACT_MAP_MASK (0x07<<0) +#define M98925_ER_ALCACT_MAP_SHIFT 0 +#define M98925_ER_ALCACT_MAP_WIDTH 3 + +/* MAX98925_R014_MAP3 */ +#define M98925_ER_SPKCURNT_EN_MASK (1<<7) +#define M98925_ER_SPKCURNT_EN_SHIFT 7 +#define M98925_ER_SPKCURNT_EN_WIDTH 1 +#define M98925_ER_SPKCURNT_MAP_MASK (0x07<<4) +#define M98925_ER_SPKCURNT_MAP_SHIFT 4 +#define M98925_ER_SPKCURNT_MAP_WIDTH 3 + +/* MAX98925_R015_MAP4 */ +/* RESERVED */ + +/* MAX98925_R016_MAP5 */ +#define M98925_ER_IMONOVFL_EN_MASK (1<<7) +#define M98925_ER_IMONOVFL_EN_SHIFT 7 +#define M98925_ER_IMONOVFL_EN_WIDTH 1 +#define M98925_ER_IMONOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_IMONOVFL_MAP_SHIFT 4 +#define M98925_ER_IMONOVFL_MAP_WIDTH 3 +#define M98925_ER_VMONOVFL_EN_MASK (1<<3) +#define M98925_ER_VMONOVFL_EN_SHIFT 3 +#define M98925_ER_VMONOVFL_EN_WIDTH 1 +#define M98925_ER_VMONOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VMONOVFL_MAP_SHIFT 0 +#define M98925_ER_VMONOVFL_MAP_WIDTH 3 + +/* MAX98925_R017_MAP6 */ +#define M98925_ER_VBSTOVFL_EN_MASK (1<<7) +#define M98925_ER_VBSTOVFL_EN_SHIFT 7 +#define M98925_ER_VBSTOVFL_EN_WIDTH 1 +#define M98925_ER_VBSTOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_VBSTOVFL_MAP_SHIFT 4 +#define M98925_ER_VBSTOVFL_MAP_WIDTH 3 +#define M98925_ER_VBATOVFL_EN_MASK (1<<3) +#define M98925_ER_VBATOVFL_EN_SHIFT 3 +#define M98925_ER_VBATOVFL_EN_WIDTH 1 +#define M98925_ER_VBATOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VBATOVFL_MAP_SHIFT 0 +#define M98925_ER_VBATOVFL_MAP_WIDTH 3 + +/* MAX98925_R018_MAP7 */ +#define M98925_ER_INVALSLOT_EN_MASK (1<<7) +#define M98925_ER_INVALSLOT_EN_SHIFT 7 +#define M98925_ER_INVALSLOT_EN_WIDTH 1 +#define M98925_ER_INVALSLOT_MAP_MASK (0x07<<4) +#define M98925_ER_INVALSLOT_MAP_SHIFT 4 +#define M98925_ER_INVALSLOT_MAP_WIDTH 3 +#define M98925_ER_SLOTCNFLT_EN_MASK (1<<3) +#define M98925_ER_SLOTCNFLT_EN_SHIFT 3 +#define M98925_ER_SLOTCNFLT_EN_WIDTH 1 +#define M98925_ER_SLOTCNFLT_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTCNFLT_MAP_SHIFT 0 +#define M98925_ER_SLOTCNFLT_MAP_WIDTH 3 + +/* MAX98925_R019_MAP8 */ +#define M98925_ER_SLOTOVRN_EN_MASK (1<<3) +#define M98925_ER_SLOTOVRN_EN_SHIFT 3 +#define M98925_ER_SLOTOVRN_EN_WIDTH 1 +#define M98925_ER_SLOTOVRN_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTOVRN_MAP_SHIFT 0 +#define M98925_ER_SLOTOVRN_MAP_WIDTH 3 + +/* MAX98925_R01A_DAI_CLK_MODE1 */ +#define M98925_DAI_CLK_SOURCE_MASK (1<<6) +#define M98925_DAI_CLK_SOURCE_SHIFT 6 +#define M98925_DAI_CLK_SOURCE_WIDTH 1 +#define M98925_MDLL_MULT_MASK (0x0F<<0) +#define M98925_MDLL_MULT_SHIFT 0 +#define M98925_MDLL_MULT_WIDTH 4 + +#define M98925_MDLL_MULT_MCLKx8 6 +#define M98925_MDLL_MULT_MCLKx16 8 + +/* MAX98925_R01B_DAI_CLK_MODE2 */ +#define M98925_DAI_SR_MASK (0x0F<<4) +#define M98925_DAI_SR_SHIFT 4 +#define M98925_DAI_SR_WIDTH 4 +#define M98925_DAI_MAS_MASK (1<<3) +#define M98925_DAI_MAS_SHIFT 3 +#define M98925_DAI_MAS_WIDTH 1 +#define M98925_DAI_BSEL_MASK (0x07<<0) +#define M98925_DAI_BSEL_SHIFT 0 +#define M98925_DAI_BSEL_WIDTH 3 + +#define M98925_DAI_BSEL_32 (0 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_48 (1 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_64 (2 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_256 (6 << M98925_DAI_BSEL_SHIFT) + +/* MAX98925_R01C_DAI_CLK_DIV_M_MSBS */ +#define M98925_DAI_M_MSBS_MASK (0xFF<<0) +#define M98925_DAI_M_MSBS_SHIFT 0 +#define M98925_DAI_M_MSBS_WIDTH 8 + +/* MAX98925_R01D_DAI_CLK_DIV_M_LSBS */ +#define M98925_DAI_M_LSBS_MASK (0xFF<<0) +#define M98925_DAI_M_LSBS_SHIFT 0 +#define M98925_DAI_M_LSBS_WIDTH 8 + +/* MAX98925_R01E_DAI_CLK_DIV_N_MSBS */ +#define M98925_DAI_N_MSBS_MASK (0x7F<<0) +#define M98925_DAI_N_MSBS_SHIFT 0 +#define M98925_DAI_N_MSBS_WIDTH 7 + +/* MAX98925_R01F_DAI_CLK_DIV_N_LSBS */ +#define M98925_DAI_N_LSBS_MASK (0xFF<<0) +#define M98925_DAI_N_LSBS_SHIFT 0 +#define M98925_DAI_N_LSBS_WIDTH 8 + +/* MAX98925_R020_FORMAT */ +#define M98925_DAI_CHANSZ_MASK (0x03<<6) +#define M98925_DAI_CHANSZ_SHIFT 6 +#define M98925_DAI_CHANSZ_WIDTH 2 +#define M98925_DAI_EXTBCLK_HIZ_MASK (1<<4) +#define M98925_DAI_EXTBCLK_HIZ_SHIFT 4 +#define M98925_DAI_EXTBCLK_HIZ_WIDTH 1 +#define M98925_DAI_WCI_MASK (1<<3) +#define M98925_DAI_WCI_SHIFT 3 +#define M98925_DAI_WCI_WIDTH 1 +#define M98925_DAI_BCI_MASK (1<<2) +#define M98925_DAI_BCI_SHIFT 2 +#define M98925_DAI_BCI_WIDTH 1 +#define M98925_DAI_DLY_MASK (1<<1) +#define M98925_DAI_DLY_SHIFT 1 +#define M98925_DAI_DLY_WIDTH 1 +#define M98925_DAI_TDM_MASK (1<<0) +#define M98925_DAI_TDM_SHIFT 0 +#define M98925_DAI_TDM_WIDTH 1 + +#define M98925_DAI_CHANSZ_16 (1 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_24 (2 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_32 (3 << M98925_DAI_CHANSZ_SHIFT) + +/* MAX98925_R021_TDM_SLOT_SELECT */ +#define M98925_DAI_DO_EN_MASK (1<<7) +#define M98925_DAI_DO_EN_SHIFT 7 +#define M98925_DAI_DO_EN_WIDTH 1 +#define M98925_DAI_DIN_EN_MASK (1<<6) +#define M98925_DAI_DIN_EN_SHIFT 6 +#define M98925_DAI_DIN_EN_WIDTH 1 +#define M98925_DAI_INR_SOURCE_MASK (0x07<<3) +#define M98925_DAI_INR_SOURCE_SHIFT 3 +#define M98925_DAI_INR_SOURCE_WIDTH 3 +#define M98925_DAI_INL_SOURCE_MASK (0x07<<0) +#define M98925_DAI_INL_SOURCE_SHIFT 0 +#define M98925_DAI_INL_SOURCE_WIDTH 3 + +/* MAX98925_R022_DOUT_CFG_VMON */ +#define M98925_DAI_VMON_EN_MASK (1<<5) +#define M98925_DAI_VMON_EN_SHIFT 5 +#define M98925_DAI_VMON_EN_WIDTH 1 +#define M98925_DAI_VMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VMON_SLOT_SHIFT 0 +#define M98925_DAI_VMON_SLOT_WIDTH 5 + +#define M98925_DAI_VMON_SLOT_00_01 (0 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_01_02 (1 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_02_03 (2 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_03_04 (3 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_04_05 (4 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_05_06 (5 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_06_07 (6 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_07_08 (7 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_08_09 (8 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_09_0A (9 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0A_0B (10 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0B_0C (11 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0C_0D (12 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0D_0E (13 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0E_0F (14 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0F_10 (15 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_10_11 (16 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_11_12 (17 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_12_13 (18 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_13_14 (19 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_14_15 (20 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_15_16 (21 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_16_17 (22 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_17_18 (23 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_18_19 (24 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_19_1A (25 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1A_1B (26 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1B_1C (27 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1C_1D (28 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1D_1E (29 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1E_1F (30 << M98925_DAI_VMON_SLOT_SHIFT) + +/* MAX98925_R023_DOUT_CFG_IMON */ +#define M98925_DAI_IMON_EN_MASK (1<<5) +#define M98925_DAI_IMON_EN_SHIFT 5 +#define M98925_DAI_IMON_EN_WIDTH 1 +#define M98925_DAI_IMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_IMON_SLOT_SHIFT 0 +#define M98925_DAI_IMON_SLOT_WIDTH 5 + +#define M98925_DAI_IMON_SLOT_00_01 (0 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_01_02 (1 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_02_03 (2 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_03_04 (3 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_04_05 (4 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_05_06 (5 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_06_07 (6 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_07_08 (7 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_08_09 (8 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_09_0A (9 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0A_0B (10 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0B_0C (11 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0C_0D (12 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0D_0E (13 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0E_0F (14 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0F_10 (15 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_10_11 (16 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_11_12 (17 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_12_13 (18 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_13_14 (19 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_14_15 (20 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_15_16 (21 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_16_17 (22 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_17_18 (23 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_18_19 (24 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_19_1A (25 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1A_1B (26 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1B_1C (27 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1C_1D (28 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1D_1E (29 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1E_1F (30 << M98925_DAI_IMON_SLOT_SHIFT) + +/* MAX98925_R024_DOUT_CFG_VBAT */ +#define M98925_DAI_VBAT_EN_MASK (1<<5) +#define M98925_DAI_VBAT_EN_SHIFT 5 +#define M98925_DAI_VBAT_EN_WIDTH 1 +#define M98925_DAI_VBAT_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBAT_SLOT_SHIFT 0 +#define M98925_DAI_VBAT_SLOT_WIDTH 5 + +/* MAX98925_R025_DOUT_CFG_VBST */ +#define M98925_DAI_VBST_EN_MASK (1<<5) +#define M98925_DAI_VBST_EN_SHIFT 5 +#define M98925_DAI_VBST_EN_WIDTH 1 +#define M98925_DAI_VBST_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBST_SLOT_SHIFT 0 +#define M98925_DAI_VBST_SLOT_WIDTH 5 + +/* MAX98925_R026_DOUT_CFG_FLAG */ +#define M98925_DAI_FLAG_EN_MASK (1<<5) +#define M98925_DAI_FLAG_EN_SHIFT 5 +#define M98925_DAI_FLAG_EN_WIDTH 1 +#define M98925_DAI_FLAG_SLOT_MASK (0x1F<<0) +#define M98925_DAI_FLAG_SLOT_SHIFT 0 +#define M98925_DAI_FLAG_SLOT_WIDTH 5 + +/* MAX98925_R027_DOUT_HIZ_CFG1 */ +#define M98925_DAI_SLOT_HIZ_CFG1_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG1_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG1_WIDTH 8 + +/* MAX98925_R028_DOUT_HIZ_CFG2 */ +#define M98925_DAI_SLOT_HIZ_CFG2_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG2_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG2_WIDTH 8 + +/* MAX98925_R029_DOUT_HIZ_CFG3 */ +#define M98925_DAI_SLOT_HIZ_CFG3_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG3_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG3_WIDTH 8 + +/* MAX98925_R02A_DOUT_HIZ_CFG4 */ +#define M98925_DAI_SLOT_HIZ_CFG4_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG4_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG4_WIDTH 8 + +/* MAX98925_R02B_DOUT_DRV_STRENGTH */ +#define M98925_DAI_OUT_DRIVE_MASK (0x03<<0) +#define M98925_DAI_OUT_DRIVE_SHIFT 0 +#define M98925_DAI_OUT_DRIVE_WIDTH 2 + +/* MAX98925_R02C_FILTERS */ +#define M98925_ADC_DITHER_EN_MASK (1<<7) +#define M98925_ADC_DITHER_EN_SHIFT 7 +#define M98925_ADC_DITHER_EN_WIDTH 1 +#define M98925_IV_DCB_EN_MASK (1<<6) +#define M98925_IV_DCB_EN_SHIFT 6 +#define M98925_IV_DCB_EN_WIDTH 1 +#define M98925_DAC_DITHER_EN_MASK (1<<4) +#define M98925_DAC_DITHER_EN_SHIFT 4 +#define M98925_DAC_DITHER_EN_WIDTH 1 +#define M98925_DAC_FILTER_MODE_MASK (1<<3) +#define M98925_DAC_FILTER_MODE_SHIFT 3 +#define M98925_DAC_FILTER_MODE_WIDTH 1 +#define M98925_DAC_HPF_MASK (0x07<<0) +#define M98925_DAC_HPF_SHIFT 0 +#define M98925_DAC_HPF_WIDTH 3 +#define M98925_DAC_HPF_DISABLE (0 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_DC_BLOCK (1 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_100 (2 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_200 (3 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_400 (4 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_800 (5 << M98925_DAC_HPF_SHIFT) + +/* MAX98925_R02D_GAIN */ +#define M98925_DAC_IN_SEL_MASK (0x03<<5) +#define M98925_DAC_IN_SEL_SHIFT 5 +#define M98925_DAC_IN_SEL_WIDTH 2 +#define M98925_SPK_GAIN_MASK (0x1F<<0) +#define M98925_SPK_GAIN_SHIFT 0 +#define M98925_SPK_GAIN_WIDTH 5 + +#define M98925_DAC_IN_SEL_LEFT_DAI (0 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_RIGHT_DAI (1 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_SUMMED_DAI (2 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_DIV2_SUMMED_DAI (3 << M98925_DAC_IN_SEL_SHIFT) + +/* MAX98925_R02E_GAIN_RAMPING */ +#define M98925_SPK_RMP_EN_MASK (1<<1) +#define M98925_SPK_RMP_EN_SHIFT 1 +#define M98925_SPK_RMP_EN_WIDTH 1 +#define M98925_SPK_ZCD_EN_MASK (1<<0) +#define M98925_SPK_ZCD_EN_SHIFT 0 +#define M98925_SPK_ZCD_EN_WIDTH 1 + +/* MAX98925_R02F_SPK_AMP */ +#define M98925_SPK_MODE_MASK (1<<0) +#define M98925_SPK_MODE_SHIFT 0 +#define M98925_SPK_MODE_WIDTH 1 + +/* MAX98925_R030_THRESHOLD */ +#define M98925_ALC_EN_MASK (1<<5) +#define M98925_ALC_EN_SHIFT 5 +#define M98925_ALC_EN_WIDTH 1 +#define M98925_ALC_TH_MASK (0x1F<<0) +#define M98925_ALC_TH_SHIFT 0 +#define M98925_ALC_TH_WIDTH 5 + +/* MAX98925_R031_ALC_ATTACK */ +#define M98925_ALC_ATK_STEP_MASK (0x0F<<4) +#define M98925_ALC_ATK_STEP_SHIFT 4 +#define M98925_ALC_ATK_STEP_WIDTH 4 +#define M98925_ALC_ATK_RATE_MASK (0x7<<0) +#define M98925_ALC_ATK_RATE_SHIFT 0 +#define M98925_ALC_ATK_RATE_WIDTH 3 + +/* MAX98925_R032_ALC_ATTEN_RLS */ +#define M98925_ALC_MAX_ATTEN_MASK (0x0F<<4) +#define M98925_ALC_MAX_ATTEN_SHIFT 4 +#define M98925_ALC_MAX_ATTEN_WIDTH 4 +#define M98925_ALC_RLS_RATE_MASK (0x7<<0) +#define M98925_ALC_RLS_RATE_SHIFT 0 +#define M98925_ALC_RLS_RATE_WIDTH 3 + +/* MAX98925_R033_ALC_HOLD_RLS */ +#define M98925_ALC_RLS_TGR_MASK (1<<0) +#define M98925_ALC_RLS_TGR_SHIFT 0 +#define M98925_ALC_RLS_TGR_WIDTH 1 + +/* MAX98925_R034_ALC_CONFIGURATION */ +#define M98925_ALC_MUTE_EN_MASK (1<<7) +#define M98925_ALC_MUTE_EN_SHIFT 7 +#define M98925_ALC_MUTE_EN_WIDTH 1 +#define M98925_ALC_MUTE_DLY_MASK (0x07<<4) +#define M98925_ALC_MUTE_DLY_SHIFT 4 +#define M98925_ALC_MUTE_DLY_WIDTH 3 +#define M98925_ALC_RLS_DBT_MASK (0x07<<0) +#define M98925_ALC_RLS_DBT_SHIFT 0 +#define M98925_ALC_RLS_DBT_WIDTH 3 + +/* MAX98925_R035_BOOST_CONVERTER */ +#define M98925_BST_SYNC_MASK (1<<7) +#define M98925_BST_SYNC_SHIFT 7 +#define M98925_BST_SYNC_WIDTH 1 +#define M98925_BST_PHASE_MASK (0x03<<4) +#define M98925_BST_PHASE_SHIFT 4 +#define M98925_BST_PHASE_WIDTH 2 +#define M98925_BST_SKIP_MODE_MASK (0x03<<0) +#define M98925_BST_SKIP_MODE_SHIFT 0 +#define M98925_BST_SKIP_MODE_WIDTH 2 + +/* MAX98925_R036_BLOCK_ENABLE */ +#define M98925_BST_EN_MASK (1<<7) +#define M98925_BST_EN_SHIFT 7 +#define M98925_BST_EN_WIDTH 1 +#define M98925_WATCH_EN_MASK (1<<6) +#define M98925_WATCH_EN_SHIFT 6 +#define M98925_WATCH_EN_WIDTH 1 +#define M98925_CLKMON_EN_MASK (1<<5) +#define M98925_CLKMON_EN_SHIFT 5 +#define M98925_CLKMON_EN_WIDTH 1 +#define M98925_SPK_EN_MASK (1<<4) +#define M98925_SPK_EN_SHIFT 4 +#define M98925_SPK_EN_WIDTH 1 +#define M98925_ADC_VBST_EN_MASK (1<<3) +#define M98925_ADC_VBST_EN_SHIFT 3 +#define M98925_ADC_VBST_EN_WIDTH 1 +#define M98925_ADC_VBAT_EN_MASK (1<<2) +#define M98925_ADC_VBAT_EN_SHIFT 2 +#define M98925_ADC_VBAT_EN_WIDTH 1 +#define M98925_ADC_IMON_EN_MASK (1<<1) +#define M98925_ADC_IMON_EN_SHIFT 1 +#define M98925_ADC_IMON_EN_WIDTH 1 +#define M98925_ADC_VMON_EN_MASK (1<<0) +#define M98925_ADC_VMON_EN_SHIFT 0 +#define M98925_ADC_VMON_EN_WIDTH 1 + +/* MAX98925_R037_CONFIGURATION */ +#define M98925_BST_VOUT_MASK (0x0F<<4) +#define M98925_BST_VOUT_SHIFT 4 +#define M98925_BST_VOUT_WIDTH 4 +#define M98925_THERMWARN_LEVEL_MASK (0x03<<2) +#define M98925_THERMWARN_LEVEL_SHIFT 2 +#define M98925_THERMWARN_LEVEL_WIDTH 2 +#define M98925_WATCH_TIME_MASK (0x03<<0) +#define M98925_WATCH_TIME_SHIFT 0 +#define M98925_WATCH_TIME_WIDTH 2 + +/* MAX98925_R038_GLOBAL_ENABLE */ +#define M98925_EN_MASK (1<<7) +#define M98925_EN_SHIFT 7 +#define M98925_EN_WIDTH 1 + +/* MAX98925_R03A_BOOST_LIMITER */ +#define M98925_BST_ILIM_MASK (0x1F<<3) +#define M98925_BST_ILIM_SHIFT 3 +#define M98925_BST_ILIM_WIDTH 5 + +/* MAX98925_R0FF_VERSION */ +#define M98925_REV_ID_MASK (0xFF<<0) +#define M98925_REV_ID_SHIFT 0 +#define M98925_REV_ID_WIDTH 8 + +struct max98925_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct max98925_pdata *pdata; + unsigned int sysclk; + unsigned int v_slot; + unsigned int i_slot; + unsigned int spk_gain; + unsigned int ch_size; +}; +#endif -- 1.9.3

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix built-in mic on Compaq Presario CQ60
by Takashi Iwai 11 Mar '15

11 Mar '15

Compaq Presario CQ60 laptop with CX20561 gives a wrong pin for the built-in mic NID 0x17 instead of NID 0x1d, and it results in the non-working mic. This patch just remaps the pin correctly via fixup. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=920604 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fd3ed18670e9..da67ea8645a6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -223,6 +223,7 @@ enum { CXT_PINCFG_LENOVO_TP410, CXT_PINCFG_LEMOTE_A1004, CXT_PINCFG_LEMOTE_A1205, + CXT_PINCFG_COMPAQ_CQ60, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, @@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lemote, }, + [CXT_PINCFG_COMPAQ_CQ60] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* 0x17 was falsely set up as a mic, it should 0x1d */ + { 0x17, 0x400001f0 }, + { 0x1d, 0x97a70120 }, + { } + } + }, [CXT_FIXUP_STEREO_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, @@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = { }; static const struct snd_pci_quirk cxt5051_fixups[] = { + SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), {} }; -- 2.3.2

1 0

[alsa-devel] [PATCH v2 1/8] ASoC: Intel: add function to load firmware image
by han.lu＠intel.com 11 Mar '15

11 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Add a general method to load firmware image, and apply to base firmware image loading. With the method, the driver will support loading multiple different modules in order to support different features. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index b9da030..396d545 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -173,6 +173,16 @@ struct sst_module_runtime_context { }; /* + * Audio DSP Module State + */ +enum sst_module_state { + SST_MODULE_STATE_UNLOADED = 0, /* default state */ + SST_MODULE_STATE_LOADED, + SST_MODULE_STATE_INITIALIZED, /* and inactive */ + SST_MODULE_STATE_ACTIVE, +}; + +/* * Audio DSP Generic Module. * * Each Firmware file can consist of 1..N modules. A module can span multiple @@ -203,6 +213,9 @@ struct sst_module { struct list_head list; /* DSP list of modules */ struct list_head list_fw; /* FW list of modules */ struct list_head runtime_list; /* list of runtime module objects*/ + + /* state */ + enum sst_module_state state; }; /* diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5f71ef6..5e58008 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -498,6 +498,7 @@ struct sst_module *sst_module_new(struct sst_fw *sst_fw, sst_module->scratch_size = template->scratch_size; sst_module->persistent_size = template->persistent_size; sst_module->entry = template->entry; + sst_module->state = SST_MODULE_STATE_UNLOADED; INIT_LIST_HEAD(&sst_module->block_list); INIT_LIST_HEAD(&sst_module->runtime_list); diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 402b728..8ad733b 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -169,6 +169,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, block = (void *)block + sizeof(*block) + block->size; } + mod->state = SST_MODULE_STATE_LOADED; return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 863a9ca..ec688f5 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1844,6 +1844,8 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) if (ret < 0) dev_err(dev, "error: audio DSP boot failure\n"); + sst_hsw_init_module_state(hsw); + ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, msecs_to_jiffies(IPC_BOOT_MSECS)); if (ret == 0) { @@ -1886,6 +1888,74 @@ struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw) return hsw->dsp; } +void sst_hsw_init_module_state(struct sst_hsw *hsw) +{ + struct sst_module *module; + enum sst_hsw_module_id id; + + /* the base fw contains several modules */ + for (id = SST_HSW_MODULE_BASE_FW; id < SST_HSW_MAX_MODULE_ID; id++) { + module = sst_module_get_from_id(hsw->dsp, id); + if (module) + module->state = SST_MODULE_STATE_ACTIVE; + } +} + +int sst_hsw_module_load(struct sst_hsw *hsw, + u32 module_id, u32 instance_id, char *name) +{ + int ret = 0; + const struct firmware *fw = NULL; + struct sst_fw *hsw_sst_fw; + struct sst_module *module; + struct device *dev = hsw->dev; + struct sst_dsp *dsp = hsw->dsp; + + dev_dbg(dev, "sst_hsw_module_load id=%d, name='%s'", module_id, name); + + module = sst_module_get_from_id(dsp, module_id); + if (module == NULL) { + /* loading for the first time */ + if (module_id == SST_HSW_MODULE_BASE_FW) { + /* for base module: use fw requested in acpi probe */ + fw = dsp->pdata->fw; + if (!fw) { + dev_err(dev, "request Base fw failed\n"); + return -ENODEV; + } + } else { + /* try and load any other optional modules if they are + * available. Use dev_info instead of dev_err in case + * request firmware failed */ + ret = request_firmware(&fw, name, dev); + if (ret) { + dev_info(dev, "fw image %s not available(%d)\n", + name, ret); + return ret; + } + } + hsw_sst_fw = sst_fw_new(dsp, fw, hsw); + if (hsw_sst_fw == NULL) { + dev_err(dev, "error: failed to load firmware\n"); + ret = -ENOMEM; + goto out; + } + module = sst_module_get_from_id(dsp, module_id); + if (module == NULL) { + dev_err(dev, "error: no module %d in firmware %s\n", + module_id, name); + } + } else + dev_info(dev, "module %d (%s) already loaded\n", + module_id, name); +out: + /* release fw, but base fw should be released by acpi driver */ + if (fw && module_id != SST_HSW_MODULE_BASE_FW) + release_firmware(fw); + + return ret; +} + static struct sst_dsp_device hsw_dev = { .thread = hsw_irq_thread, .ops = &haswell_ops, @@ -1947,12 +2017,10 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* keep the DSP in reset state for base FW loading */ sst_dsp_reset(hsw->dsp); - hsw->sst_fw = sst_fw_new(hsw->dsp, pdata->fw, hsw); - if (hsw->sst_fw == NULL) { - ret = -ENODEV; - dev_err(dev, "error: failed to load firmware\n"); + /* load base module and other modules in base firmware image */ + ret = sst_hsw_module_load(hsw, SST_HSW_MODULE_BASE_FW, 0, "Base"); + if (ret < 0) goto fw_err; - } /* allocate scratch mem regions */ ret = sst_block_alloc_scratch(hsw->dsp); @@ -1971,6 +2039,9 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) goto boot_err; } + /* init module state after boot */ + sst_hsw_init_module_state(hsw); + /* get the FW version */ sst_hsw_fw_get_version(hsw, &version); @@ -1986,7 +2057,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) boot_err: sst_dsp_reset(hsw->dsp); - sst_fw_free(hsw->sst_fw); + sst_fw_free_all(hsw->dsp); fw_err: dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, hsw->dx_context, hsw->dx_context_paddr); diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 8580960..e071b3a 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -467,6 +467,12 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata); struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw); +/* fw module function */ +void sst_hsw_init_module_state(struct sst_hsw *hsw); + +int sst_hsw_module_load(struct sst_hsw *hsw, + u32 module_id, u32 instance_id, char *name); + /* runtime module management */ struct sst_module_runtime *sst_hsw_runtime_module_create(struct sst_hsw *hsw, int mod_id, int offset); -- 1.9.1

3 13

[alsa-devel] [PATCH v2 1/2] ASoC: rt5670: Add IRQ function
by Bard Liao 11 Mar '15

11 Mar '15

This patch adds the IRQ function support of rt5670. We use a flag named dev_gpio in platform data to inform codec driver if the IRQ function is used or not. Also, we export rt5670_set_jack_detect for machine driver to pass the jack point. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * change struct snd_soc_jack *jack = &rt5670->jack; to struct snd_soc_jack *jack = rt5670->jack; --- include/sound/rt5670.h | 1 + sound/soc/codecs/rt5670.c | 176 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5670.h | 4 ++ 3 files changed, 179 insertions(+), 2 deletions(-) diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h index bd31119..b7d6051 100644 --- a/include/sound/rt5670.h +++ b/include/sound/rt5670.h @@ -14,6 +14,7 @@ struct rt5670_platform_data { int jd_mode; bool in2_diff; + bool dev_gpio; bool dmic_en; unsigned int dmic1_data_pin; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 4b57c85..23ca049 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -403,6 +403,171 @@ static bool rt5670_readable_register(struct device *dev, unsigned int reg) } } +/** + * rt5670_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert) +{ + int val; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_DET_MODE | RT5670_CBJ_MN_JD, + RT5670_CBJ_MN_JD); + snd_soc_write(codec, RT5670_GPIO_CTRL2, 0x0004); + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + snd_soc_update_bits(codec, RT5670_CJ_CTRL1, + RT5670_CBJ_BST1_EN, RT5670_CBJ_BST1_EN); + snd_soc_write(codec, RT5670_JD_CTRL3, 0x00f0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, RT5670_CBJ_MN_JD); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, 0); + msleep(300); + val = snd_soc_read(codec, RT5670_CJ_CTRL3) & 0x7; + if (val == 0x1 || val == 0x2) { + rt5670->jack_type = SND_JACK_HEADSET; + /* for push button */ + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, RT5670_IL_CMD, 0x40, 0x40); + snd_soc_read(codec, RT5670_IL_CMD); + } else { + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = SND_JACK_HEADPHONE; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + } else { + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x0); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = 0; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + return rt5670->jack_type; +} + +static int rt5670_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5670_IL_CMD); + btn_type = val & 0xff80; + snd_soc_write(codec, RT5670_IL_CMD, val); + if (btn_type != 0) { + msleep(20); + val = snd_soc_read(codec, RT5670_IL_CMD); + snd_soc_write(codec, RT5670_IL_CMD, val); + } + + return btn_type; +} + +static int rt5670_irq_detection(void *data) +{ + struct rt5670_priv *rt5670 = (struct rt5670_priv *)data; + struct snd_soc_jack_gpio *gpio = &rt5670->hp_gpio; + struct snd_soc_jack *jack = rt5670->jack; + int val, btn_type, report = jack->status; + + if (rt5670->pdata.jd_mode == 1) /* 2 port */ + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0070; + else + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0020; + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5670->jack_type == 0) { + report = rt5670_headset_detect(rt5670->codec, 1); + /* for push button and jack out */ + gpio->debounce_time = 25; + break; + } + btn_type = 0; + if (snd_soc_read(rt5670->codec, RT5670_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5670_button_detect(rt5670->codec); + switch (btn_type) { + case 0x2000: /* up */ + report |= SND_JACK_BTN_1; + break; + case 0x0400: /* center */ + report |= SND_JACK_BTN_0; + break; + case 0x0080: /* down */ + report |= SND_JACK_BTN_2; + break; + default: + dev_err(rt5670->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5670->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5670->codec, RT5670_INT_IRQ_ST, 0x1, 0x0); + rt5670_headset_detect(rt5670->codec, 0); + gpio->debounce_time = 150; /* for jack in */ + break; + default: + break; + } + + return report; +} + +int rt5670_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5670->jack = jack; + rt5670->hp_gpio.gpiod_dev = codec->dev; + rt5670->hp_gpio.name = "headphone detect"; + rt5670->hp_gpio.report = SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2; + rt5670->hp_gpio.debounce_time = 150; + rt5670->hp_gpio.wake = true; + rt5670->hp_gpio.data = (struct rt5670_priv *)rt5670; + rt5670->hp_gpio.jack_status_check = rt5670_irq_detection; + + ret = snd_soc_jack_add_gpios(rt5670->jack, 1, + &rt5670->hp_gpio); + if (ret) { + dev_err(codec->dev, "Adding jack GPIO failed\n"); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_set_jack_detect); + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -2506,6 +2671,7 @@ static int rt5670_remove(struct snd_soc_codec *codec) struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5670->regmap, RT5670_RESET, 0); + snd_soc_jack_free_gpios(rt5670->jack, 1, &rt5670->hp_gpio); return 0; } @@ -2665,6 +2831,7 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (dmi_check_system(dmi_platform_intel_braswell)) { rt5670->pdata.dmic_en = true; rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.dev_gpio = true; rt5670->pdata.jd_mode = 1; } @@ -2706,12 +2873,17 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5670->regmap, RT5670_IN2, RT5670_IN_DF2, RT5670_IN_DF2); - if (i2c->irq) { + if (rt5670->pdata.dev_gpio) { + /* for push button */ + regmap_write(rt5670->regmap, RT5670_IL_CMD, 0x0000); + regmap_write(rt5670->regmap, RT5670_IL_CMD2, 0x0010); + regmap_write(rt5670->regmap, RT5670_IL_CMD3, 0x0014); + /* for irq */ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); - + regmap_update_bits(rt5670->regmap, RT5670_DIG_MISC, 0x8, 0x8); } if (rt5670->pdata.jd_mode) { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 0a67adb..0f3255a 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1988,6 +1988,8 @@ struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; struct regmap *regmap; + struct snd_soc_jack *jack; + struct snd_soc_jack_gpio hp_gpio; int sysclk; int sysclk_src; @@ -2004,4 +2006,6 @@ struct rt5670_priv { int jack_type; }; +int rt5670_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); #endif /* __RT5670_H__ */ -- 1.8.1.1.439.g50a6b54

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Re: [alsa-devel] [ALSA/seq] BUG: unable to handle kernel NULL pointer dereference at 00000050
by Takashi Iwai 11 Mar '15

11 Mar '15

At Wed, 11 Mar 2015 18:37:48 +0800, Fengguang Wu wrote: > Greetings, > > 0day kernel testing robot got the below dmesg and the first bad commit is > > git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master > > commit 7c37ae5c625aaa4836466cfaea829a3199dfc571 > Author: Takashi Iwai <tiwai(a)suse.de> > AuthorDate: Thu Feb 12 10:51:59 2015 +0100 > Commit: Takashi Iwai <tiwai(a)suse.de> > CommitDate: Thu Feb 12 11:35:11 2015 +0100 > > ALSA: seq: Rewrite sequencer device binding with standard bus > > We've used the old house-made code for binding the sequencer device > and driver. This can be far better implemented with the standard > bus nowadays. > > This patch refactors the whole sequencer binding code with the bus > /sys/bus/snd_seq. The devices appear as id-card-device on this bus > and are bound with the drivers corresponding to the given id like the > former implementation. The module autoload is also kept like before. > > There is no change in API functions by this patch, and almost all > transitions are kept inside seq_device.c. The proc file output will > change slightly but kept compatible as much as possible. > > Further integration works will follow in later patches. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > +------------------------------------------+------------+------------+-----------------+ > | | 72496edcf8 | 7c37ae5c62 | v4.0-rc3_031010 | > +------------------------------------------+------------+------------+-----------------+ > | boot_successes | 79 | 0 | 0 | > | boot_failures | 1 | 20 | 14 | > | BUG:kernel_boot_crashed | 1 | | | > | BUG:unable_to_handle_kernel | 0 | 20 | 14 | > | Oops | 0 | 20 | 14 | > | EIP_is_at_bus_add_device | 0 | 20 | 14 | > | Kernel_panic-not_syncing:Fatal_exception | 0 | 20 | 14 | > | backtrace:usb_composite_probe | 0 | 20 | 14 | > | backtrace:midi_driver_init | 0 | 20 | 14 | > | backtrace:kernel_init_freeable | 0 | 20 | 14 | > +------------------------------------------+------------+------------+-----------------+ > > [ 1.178531] udc dummy_udc.0: registering UDC driver [MIDI Gadget] > [ 1.179260] MIDI Gadget gadget: adding config #1 'MIDI Gadget'/c1f006c0 > [ 1.184102] MIDI Gadget gadget: adding 'gmidi function'/d1fab310 to config 'MIDI Gadget'/c1f006c0 > [ 1.186606] BUG: unable to handle kernel NULL pointer dereference at 00000050 > [ 1.187008] IP: [<c1590786>] bus_add_device+0xd6/0x160 > [ 1.187008] *pde = 00000000 > [ 1.187008] Oops: 0000 [#1] PREEMPT SMP DEBUG_PAGEALLOC > [ 1.187008] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.19.0-03415-g7c37ae5 #24 > [ 1.187008] Hardware name: Bochs Bochs, BIOS Bochs 01/01/2011 > [ 1.187008] task: d3480000 ti: d346a000 task.ti: d346a000 > [ 1.187008] EIP: 0060:[<c1590786>] EFLAGS: 00010246 CPU: 0 > [ 1.187008] EIP is at bus_add_device+0xd6/0x160 > [ 1.187008] EAX: 00000000 EBX: 00000000 ECX: d1fb7cc0 EDX: d1fa90bc > [ 1.187008] ESI: d1fa90bc EDI: c1f4f800 EBP: d346bb44 ESP: d346bb30 > [ 1.187008] DS: 007b ES: 007b FS: 00d8 GS: 0000 SS: 0068 > [ 1.187008] CR0: 80050033 CR2: 00000050 CR3: 02023000 CR4: 000406d0 > [ 1.187008] Stack: > [ 1.187008] c1291226 d1fa90b4 d1fa90bc d1f78c7c 00000000 d346bb84 c158e80a d1fa90bc > [ 1.187008] d1f78c84 00000000 00000002 00000000 00000000 d1fa90b4 d1fa9048 2e056fb7 > [ 1.187008] 00000002 ffffffff d1fb6dd0 00000000 d238b024 d346bb8c c18c26f0 d346bb98 > [ 1.187008] Call Trace: > [ 1.187008] [<c1291226>] ? acpi_platform_notify+0x19/0x78 > [ 1.187008] [<c158e80a>] device_add+0x33a/0x530 > [ 1.187008] [<c18c26f0>] snd_seq_device_dev_register+0x10/0x20 > [ 1.187008] [<c18a13b6>] snd_device_register+0x46/0x80 > [ 1.187008] [<c18b181d>] snd_rawmidi_dev_register+0x1cd/0x2d0 > [ 1.187008] [<c189e70a>] ? snd_ctl_dev_register+0x2a/0x30 > [ 1.187008] [<c18a1432>] snd_device_register_all+0x42/0x70 > [ 1.187008] [<c189e10d>] snd_card_register+0x3d/0x150 > [ 1.187008] [<c16c284a>] f_midi_bind+0x16a/0x6c0 > [ 1.187008] [<c104de70>] ? irq_exit+0x30/0x60 > [ 1.187008] [<c10881ef>] ? console_unlock+0x38f/0x520 > [ 1.187008] [<c108860a>] ? vprintk_emit+0x28a/0x5f0 > [ 1.187008] [<c158f8df>] ? dev_vprintk_emit+0x14f/0x280 > [ 1.187008] [<c1109c9b>] ? alloc_debug_processing+0xcb/0x150 > [ 1.187008] [<c16c2e32>] ? f_midi_alloc+0x92/0x1a0 > [ 1.187008] [<c16c2e32>] ? f_midi_alloc+0x92/0x1a0 > [ 1.187008] [<c16c2e32>] ? f_midi_alloc+0x92/0x1a0 > [ 1.187008] [<c1faabc0>] ? midi_bind+0xd2/0xd2 > [ 1.187008] [<c158fa2a>] ? dev_printk_emit+0x1a/0x20 > [ 1.187008] [<c158fa76>] ? __dev_printk+0x46/0x90 > [ 1.187008] [<c158fae6>] ? dev_printk+0x26/0x30 > [ 1.187008] [<c16971b9>] usb_add_function+0x79/0x140 > [ 1.187008] [<c1faabc0>] ? midi_bind+0xd2/0xd2 > [ 1.187008] [<c1faabe5>] midi_bind_config+0x25/0x3c > [ 1.187008] [<c169753c>] usb_add_config+0x9c/0x240 > [ 1.187008] [<c1a1ee48>] ? mutex_unlock+0x8/0x10 > [ 1.187008] [<c1faab86>] midi_bind+0x98/0xd2 > [ 1.187008] [<c1697d28>] composite_bind+0x88/0x1a0 > [ 1.187008] [<c169a92f>] udc_bind_to_driver+0x4f/0xf0 > [ 1.187008] [<c169b15d>] usb_gadget_probe_driver+0x6d/0xb0 > [ 1.187008] [<c1f7cb5e>] ? do_one_initcall+0x77/0x151 > [ 1.187008] [<c1697eb8>] usb_composite_probe+0x78/0xa0 > [ 1.187008] [<c1faaadf>] ? midimod_init+0xf/0xf > [ 1.187008] [<c1faaaec>] midi_driver_init+0xd/0xf > [ 1.187008] [<c1f7cbb4>] do_one_initcall+0xcd/0x151 > [ 1.187008] [<c1f7c424>] ? do_early_param+0x73/0x73 > [ 1.187008] [<c1060c00>] ? parse_args+0x1f0/0x430 > [ 1.187008] [<c1f7cd1b>] kernel_init_freeable+0xe3/0x15b > [ 1.187008] [<c1a1411b>] kernel_init+0xb/0xe0 > [ 1.187008] [<c1a22201>] ret_from_kernel_thread+0x21/0x30 > [ 1.187008] [<c1a14110>] ? rest_init+0xb0/0xb0 > [ 1.187008] Code: b6 00 00 00 00 8b 57 14 8b 45 f0 e8 25 d9 ff ff 85 c0 89 c3 75 69 8b 45 f0 8b 48 2c 85 c9 74 6c 8b 45 f0 8d 70 08 8b 47 48 89 f2 <8b> 40 50 83 c0 28 e8 6f 34 bc ff 85 c0 89 c3 74 0d 8b 55 f0 8d > [ 1.187008] EIP: [<c1590786>] bus_add_device+0xd6/0x160 SS:ESP 0068:d346bb30 > [ 1.187008] CR2: 0000000000000050 > [ 1.187008] ---[ end trace 7c40b21b66beeff8 ]--- > [ 1.187008] Kernel panic - not syncing: Fatal exception This is yet another init order problem. Fixed by the patch below. Thanks! Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: seq: Fix init order of snd_seq_device stuff When the sequencer driver is built in kernel, it may panic at boot because of the uninitialized snd_seq_bus_type. Initialize it properly via subsys_initcall() instead of module_init() to assure that the bus is registered beforehand. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Fixes: 7c37ae5c625a ('ALSA: seq: Rewrite sequencer device binding with standard bus') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 355b34269bd1..d99f99d61983 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -311,5 +311,5 @@ static void __exit alsa_seq_device_exit(void) bus_unregister(&snd_seq_bus_type); } -module_init(alsa_seq_device_init) +subsys_initcall(alsa_seq_device_init) module_exit(alsa_seq_device_exit) -- 2.3.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: move the jack creation to Braswell machine driver
by Jin Yao 11 Mar '15

11 Mar '15

The jack creation code was in rt5670 codec driver before due to the jack resources (gpio/irq) were defined under the node of codec device in ACPI on Braswell. We used the snd_soc_jack_new() to create a jack instance. But now snd_soc_jack_new() is removed from upstream and we can't use snd_soc_card_jack_new() in codec driver, so we move the jack creation code to machine driver and pass the jack instance to codec driver for further processing. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Jin Yao <yao.jin(a)linux.intel.com> --- sound/soc/intel/cht_bsw_rt5672.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index c41fae3..4204fc4 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -22,6 +22,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/jack.h> #include "../codecs/rt5670.h" #include "sst-atom-controls.h" @@ -29,6 +30,20 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "rt5670-aif1" +static struct snd_soc_jack cht_bsw_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin cht_bsw_headset_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, +}; + static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) { int i; @@ -178,6 +193,15 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) | RT5670_AD_MONO_L_FILTER | RT5670_AD_MONO_R_FILTER, RT5670_CLK_SEL_I2S1_ASRC); + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2, &cht_bsw_headset, + cht_bsw_headset_pins, ARRAY_SIZE(cht_bsw_headset_pins)); + if (ret) + return ret; + + rt5670_set_jack_detect(codec, &cht_bsw_headset); return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH 24/45] hdspm.h: include stdint.h in userspace
by Mikko Rapeli 11 Mar '15

11 Mar '15

Fixes compilation error: sound/hdspm.h:43:2: error: unknown type name ‘uint32_t’ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/hdspm.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index d956c35..f799828 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,6 +20,12 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef __KERNEL__ +#include <linux/types.h> +#else +#include <stdint.h> +#endif + /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 -- 2.1.4

2 3

[alsa-devel] [PATCH] SoC: rt5670: Add IRQ function
by Bard Liao 11 Mar '15

11 Mar '15

This patch adds the IRQ function support of rt5670. We use a flag named dev_gpio in platform data to inform codec driver if the IRQ function is used or not. Also, we export rt5670_set_jack_detect for machine driver to pass the jack point. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5670.h | 1 + sound/soc/codecs/rt5670.c | 176 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5670.h | 4 ++ 3 files changed, 179 insertions(+), 2 deletions(-) diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h index bd31119..b7d6051 100644 --- a/include/sound/rt5670.h +++ b/include/sound/rt5670.h @@ -14,6 +14,7 @@ struct rt5670_platform_data { int jd_mode; bool in2_diff; + bool dev_gpio; bool dmic_en; unsigned int dmic1_data_pin; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 4b57c85..23ca049 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -403,6 +403,171 @@ static bool rt5670_readable_register(struct device *dev, unsigned int reg) } } +/** + * rt5670_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert) +{ + int val; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_DET_MODE | RT5670_CBJ_MN_JD, + RT5670_CBJ_MN_JD); + snd_soc_write(codec, RT5670_GPIO_CTRL2, 0x0004); + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + snd_soc_update_bits(codec, RT5670_CJ_CTRL1, + RT5670_CBJ_BST1_EN, RT5670_CBJ_BST1_EN); + snd_soc_write(codec, RT5670_JD_CTRL3, 0x00f0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, RT5670_CBJ_MN_JD); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, 0); + msleep(300); + val = snd_soc_read(codec, RT5670_CJ_CTRL3) & 0x7; + if (val == 0x1 || val == 0x2) { + rt5670->jack_type = SND_JACK_HEADSET; + /* for push button */ + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, RT5670_IL_CMD, 0x40, 0x40); + snd_soc_read(codec, RT5670_IL_CMD); + } else { + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = SND_JACK_HEADPHONE; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + } else { + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x0); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = 0; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + return rt5670->jack_type; +} + +static int rt5670_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5670_IL_CMD); + btn_type = val & 0xff80; + snd_soc_write(codec, RT5670_IL_CMD, val); + if (btn_type != 0) { + msleep(20); + val = snd_soc_read(codec, RT5670_IL_CMD); + snd_soc_write(codec, RT5670_IL_CMD, val); + } + + return btn_type; +} + +static int rt5670_irq_detection(void *data) +{ + struct rt5670_priv *rt5670 = (struct rt5670_priv *)data; + struct snd_soc_jack_gpio *gpio = &rt5670->hp_gpio; + struct snd_soc_jack *jack = rt5670->jack; + int val, btn_type, report = jack->status; + + if (rt5670->pdata.jd_mode == 1) /* 2 port */ + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0070; + else + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0020; + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5670->jack_type == 0) { + report = rt5670_headset_detect(rt5670->codec, 1); + /* for push button and jack out */ + gpio->debounce_time = 25; + break; + } + btn_type = 0; + if (snd_soc_read(rt5670->codec, RT5670_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5670_button_detect(rt5670->codec); + switch (btn_type) { + case 0x2000: /* up */ + report |= SND_JACK_BTN_1; + break; + case 0x0400: /* center */ + report |= SND_JACK_BTN_0; + break; + case 0x0080: /* down */ + report |= SND_JACK_BTN_2; + break; + default: + dev_err(rt5670->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5670->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5670->codec, RT5670_INT_IRQ_ST, 0x1, 0x0); + rt5670_headset_detect(rt5670->codec, 0); + gpio->debounce_time = 150; /* for jack in */ + break; + default: + break; + } + + return report; +} + +int rt5670_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5670->jack = jack; + rt5670->hp_gpio.gpiod_dev = codec->dev; + rt5670->hp_gpio.name = "headphone detect"; + rt5670->hp_gpio.report = SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2; + rt5670->hp_gpio.debounce_time = 150; + rt5670->hp_gpio.wake = true; + rt5670->hp_gpio.data = (struct rt5670_priv *)rt5670; + rt5670->hp_gpio.jack_status_check = rt5670_irq_detection; + + ret = snd_soc_jack_add_gpios(rt5670->jack, 1, + &rt5670->hp_gpio); + if (ret) { + dev_err(codec->dev, "Adding jack GPIO failed\n"); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_set_jack_detect); + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -2506,6 +2671,7 @@ static int rt5670_remove(struct snd_soc_codec *codec) struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5670->regmap, RT5670_RESET, 0); + snd_soc_jack_free_gpios(rt5670->jack, 1, &rt5670->hp_gpio); return 0; } @@ -2665,6 +2831,7 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (dmi_check_system(dmi_platform_intel_braswell)) { rt5670->pdata.dmic_en = true; rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.dev_gpio = true; rt5670->pdata.jd_mode = 1; } @@ -2706,12 +2873,17 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5670->regmap, RT5670_IN2, RT5670_IN_DF2, RT5670_IN_DF2); - if (i2c->irq) { + if (rt5670->pdata.dev_gpio) { + /* for push button */ + regmap_write(rt5670->regmap, RT5670_IL_CMD, 0x0000); + regmap_write(rt5670->regmap, RT5670_IL_CMD2, 0x0010); + regmap_write(rt5670->regmap, RT5670_IL_CMD3, 0x0014); + /* for irq */ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); - + regmap_update_bits(rt5670->regmap, RT5670_DIG_MISC, 0x8, 0x8); } if (rt5670->pdata.jd_mode) { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 0a67adb..0f3255a 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1988,6 +1988,8 @@ struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; struct regmap *regmap; + struct snd_soc_jack *jack; + struct snd_soc_jack_gpio hp_gpio; int sysclk; int sysclk_src; @@ -2004,4 +2006,6 @@ struct rt5670_priv { int jack_type; }; +int rt5670_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); #endif /* __RT5670_H__ */ -- 1.8.1.1.439.g50a6b54

1 2

[alsa-devel] [PATCH 1/2] ASoC: rt5670: Add IRQ function
by Bard Liao 11 Mar '15

11 Mar '15

This patch adds the IRQ function support of rt5670. We use a flag named dev_gpio in platform data to inform codec driver if the IRQ function is used or not. Also, we export rt5670_set_jack_detect for machine driver to pass the jack point. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5670.h | 1 + sound/soc/codecs/rt5670.c | 176 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5670.h | 4 ++ 3 files changed, 179 insertions(+), 2 deletions(-) diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h index bd31119..b7d6051 100644 --- a/include/sound/rt5670.h +++ b/include/sound/rt5670.h @@ -14,6 +14,7 @@ struct rt5670_platform_data { int jd_mode; bool in2_diff; + bool dev_gpio; bool dmic_en; unsigned int dmic1_data_pin; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 4b57c85..fdd31e2 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -403,6 +403,171 @@ static bool rt5670_readable_register(struct device *dev, unsigned int reg) } } +/** + * rt5670_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert) +{ + int val; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_DET_MODE | RT5670_CBJ_MN_JD, + RT5670_CBJ_MN_JD); + snd_soc_write(codec, RT5670_GPIO_CTRL2, 0x0004); + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + snd_soc_update_bits(codec, RT5670_CJ_CTRL1, + RT5670_CBJ_BST1_EN, RT5670_CBJ_BST1_EN); + snd_soc_write(codec, RT5670_JD_CTRL3, 0x00f0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, RT5670_CBJ_MN_JD); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, 0); + msleep(300); + val = snd_soc_read(codec, RT5670_CJ_CTRL3) & 0x7; + if (val == 0x1 || val == 0x2) { + rt5670->jack_type = SND_JACK_HEADSET; + /* for push button */ + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, RT5670_IL_CMD, 0x40, 0x40); + snd_soc_read(codec, RT5670_IL_CMD); + } else { + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = SND_JACK_HEADPHONE; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + } else { + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x0); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = 0; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + return rt5670->jack_type; +} + +static int rt5670_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5670_IL_CMD); + btn_type = val & 0xff80; + snd_soc_write(codec, RT5670_IL_CMD, val); + if (btn_type != 0) { + msleep(20); + val = snd_soc_read(codec, RT5670_IL_CMD); + snd_soc_write(codec, RT5670_IL_CMD, val); + } + + return btn_type; +} + +static int rt5670_irq_detection(void *data) +{ + struct rt5670_priv *rt5670 = (struct rt5670_priv *)data; + struct snd_soc_jack_gpio *gpio = &rt5670->hp_gpio; + struct snd_soc_jack *jack = &rt5670->jack; + int val, btn_type, report = jack->status; + + if (rt5670->pdata.jd_mode == 1) /* 2 port */ + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0070; + else + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0020; + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5670->jack_type == 0) { + report = rt5670_headset_detect(rt5670->codec, 1); + /* for push button and jack out */ + gpio->debounce_time = 25; + break; + } + btn_type = 0; + if (snd_soc_read(rt5670->codec, RT5670_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5670_button_detect(rt5670->codec); + switch (btn_type) { + case 0x2000: /* up */ + report |= SND_JACK_BTN_1; + break; + case 0x0400: /* center */ + report |= SND_JACK_BTN_0; + break; + case 0x0080: /* down */ + report |= SND_JACK_BTN_2; + break; + default: + dev_err(rt5670->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5670->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5670->codec, RT5670_INT_IRQ_ST, 0x1, 0x0); + rt5670_headset_detect(rt5670->codec, 0); + gpio->debounce_time = 150; /* for jack in */ + break; + default: + break; + } + + return report; +} + +int rt5670_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5670->jack = jack; + rt5670->hp_gpio.gpiod_dev = codec->dev; + rt5670->hp_gpio.name = "headphone detect"; + rt5670->hp_gpio.report = SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2; + rt5670->hp_gpio.debounce_time = 150; + rt5670->hp_gpio.wake = true; + rt5670->hp_gpio.data = (struct rt5670_priv *)rt5670; + rt5670->hp_gpio.jack_status_check = rt5670_irq_detection; + + ret = snd_soc_jack_add_gpios(rt5670->jack, 1, + &rt5670->hp_gpio); + if (ret) { + dev_err(codec->dev, "Adding jack GPIO failed\n"); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_set_jack_detect); + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -2506,6 +2671,7 @@ static int rt5670_remove(struct snd_soc_codec *codec) struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5670->regmap, RT5670_RESET, 0); + snd_soc_jack_free_gpios(rt5670->jack, 1, &rt5670->hp_gpio); return 0; } @@ -2665,6 +2831,7 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (dmi_check_system(dmi_platform_intel_braswell)) { rt5670->pdata.dmic_en = true; rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.dev_gpio = true; rt5670->pdata.jd_mode = 1; } @@ -2706,12 +2873,17 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5670->regmap, RT5670_IN2, RT5670_IN_DF2, RT5670_IN_DF2); - if (i2c->irq) { + if (rt5670->pdata.dev_gpio) { + /* for push button */ + regmap_write(rt5670->regmap, RT5670_IL_CMD, 0x0000); + regmap_write(rt5670->regmap, RT5670_IL_CMD2, 0x0010); + regmap_write(rt5670->regmap, RT5670_IL_CMD3, 0x0014); + /* for irq */ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); - + regmap_update_bits(rt5670->regmap, RT5670_DIG_MISC, 0x8, 0x8); } if (rt5670->pdata.jd_mode) { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 0a67adb..0f3255a 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1988,6 +1988,8 @@ struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; struct regmap *regmap; + struct snd_soc_jack *jack; + struct snd_soc_jack_gpio hp_gpio; int sysclk; int sysclk_src; @@ -2004,4 +2006,6 @@ struct rt5670_priv { int jack_type; }; +int rt5670_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); #endif /* __RT5670_H__ */ -- 1.8.1.1.439.g50a6b54

1 1

[alsa-devel] [PATCH] ASoC: rsnd: Use %pad to print dma_addr_t in rsnd_dmapp_init()
by Geert Uytterhoeven 11 Mar '15

11 Mar '15

sound/soc/sh/rcar/dma.c: In function 'rsnd_dmapp_init': sound/soc/sh/rcar/dma.c:341:2: warning: format '%x' expects argument of type 'unsigned int', but argument 5 has type 'dma_addr_t' [-Wformat=] dev_dbg(dev, "id/src/dst/chcr = %d/%x/%x/%08x\n", ^ sound/soc/sh/rcar/dma.c:341:2: warning: format '%x' expects argument of type 'unsigned int', but argument 6 has type 'dma_addr_t' [-Wformat=] Fixes: 288f392e729dd4d3 ("ASoC: rsnd: add Audio DMAC peri peri support rework") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Perhaps this can still be folded in the original commit? sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 92fd55044ee6b494..2b73df84a76904dc 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -338,8 +338,8 @@ static int rsnd_dmapp_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, rsnd_dmapp_stop(dma); - dev_dbg(dev, "id/src/dst/chcr = %d/%x/%x/%08x\n", - dmapp->dmapp_id, dma->src_addr, dma->dst_addr, dmapp->chcr); + dev_dbg(dev, "id/src/dst/chcr = %d/%pad/%pad/%08x\n", + dmapp->dmapp_id, &dma->src_addr, &dma->dst_addr, dmapp->chcr); return 0; } -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: Intel: move sysclk source setting to platform_clock_control for balance.
by Jin Yao 10 Mar '15

10 Mar '15

A playback noise happens after suspend/resume on Braswell. The issue is due to the codec PLL and codec ASRC are not enabled correctly due to the incorrect sysclk setting after resume. This patch resets the sysclk source setting in platform clock control widget handler. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Jin Yao <yao.jin(a)linux.intel.com> --- sound/soc/intel/cht_bsw_rt5672.c | 40 ++++++++++++++++++++++++++++------------ 1 file changed, 28 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index bc8dcac..279df4c 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -50,6 +50,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct snd_soc_dai *codec_dai; + int ret; codec_dai = cht_get_codec_dai(card); if (!codec_dai) { @@ -57,17 +58,31 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (!SND_SOC_DAPM_EVENT_OFF(event)) - return 0; - - /* Set codec sysclk source to its internal clock because codec PLL will - * be off when idle and MCLK will also be off by ACPI when codec is - * runtime suspended. Codec needs clock for jack detection and button - * press. - */ - snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, - 0, SND_SOC_CLOCK_IN); - + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, 48000 * 512); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* set codec sysclk source to PLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + 48000 * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + } else { + /* Set codec sysclk source to its internal clock because codec + * PLL will be off when idle and MCLK will also be off by ACPI + * when codec is runtime suspended. Codec needs clock for jack + * detection and button press. + */ + snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, + 48000 * 512, SND_SOC_CLOCK_IN); + } return 0; } @@ -77,7 +92,8 @@ static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_POST_PMD), + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cht_audio_map[] = { -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: rsnd: fixup patches for DMA/debugfs
by Kuninori Morimoto 10 Mar '15

10 Mar '15

Hi Mark These are rsnd fixup patches 1) is for new DMA method 2) is for debugfs these are based on mark/topic/rsnd Kuninori Morimoto (2): 1) ASoC: rsnd: recover PIO mode for new dma interface 2) ASoC: rsnd: add regmap_config::name for debugfs sound/soc/sh/rcar/dma.c | 19 ++++++++++++++++++- sound/soc/sh/rcar/gen.c | 1 + 2 files changed, 19 insertions(+), 1 deletion(-) Best regards --- Kuninori Morimoto

2 3

[alsa-devel] [PATCH 00/14] Fix wrong ctl value references in ASoC codecs
by Takashi Iwai 10 Mar '15

10 Mar '15

Hi, while looking through the ASoC codec codes, I noticed that lots of drivers refer to the wrong union value type for boolean controls. They should be value.integer.value[], but the code accesses to value.enumerated.item[]. They haven't been noticed so far maybe just because both are same on 32bit architectures. But they are incompatible on 64bit architectures, thus must be fixed. Takashi === Takashi Iwai (14): ASoC: adav80x: Fix wrong value references for boolean kctl ASoC: ak4641: Fix wrong value references for boolean kctl ASoC: cs4271: Fix wrong value references for boolean kctl ASoC: es8238: Fix wrong value references for boolean kctl ASoC: pcm1681: Fix wrong value references for boolean kctl ASoC: tas5086: Fix wrong value references for boolean kctl ASoC: wm2000: Fix wrong value references for boolean kctl ASoC: wm8731: Fix wrong value references for boolean kctl ASoC: wm8903: Fix wrong value references for boolean kctl ASoC: wm8904: Fix wrong value references for boolean kctl ASoC: wm8955: Fix wrong value references for boolean kctl ASoC: wm8960: Fix wrong value references for boolean kctl ASoC: wm9712: Fix wrong value references for boolean kctl ASoC: wm9713: Fix wrong value references for boolean kctl sound/soc/codecs/adav80x.c | 4 ++-- sound/soc/codecs/ak4641.c | 4 ++-- sound/soc/codecs/cs4271.c | 4 ++-- sound/soc/codecs/es8328.c | 4 ++-- sound/soc/codecs/pcm1681.c | 4 ++-- sound/soc/codecs/tas5086.c | 4 ++-- sound/soc/codecs/wm2000.c | 8 ++++---- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8955.c | 4 ++-- sound/soc/codecs/wm8960.c | 4 ++-- sound/soc/codecs/wm9712.c | 6 +++--- sound/soc/codecs/wm9713.c | 6 +++--- 14 files changed, 32 insertions(+), 32 deletions(-) -- 2.3.2

5 26

[alsa-devel] [PATCH 0/2] simple-card: Add samplerate and samplewidth constraints
by Jyri Sarha 10 Mar '15

10 Mar '15

These constraints help to sort out the format and samplerate combinations that do not properly work on a specific HW. Jyri Sarha (2): ASoC: simple-card: Remove trailing whitespace ASoC: simple-card: Add support for samplerate and samplewidth constraints .../devicetree/bindings/sound/simple-card.txt | 6 ++ sound/soc/generic/simple-card.c | 89 +++++++++++++++++++++- 2 files changed, 94 insertions(+), 1 deletion(-) -- 1.9.1

4 15

[alsa-devel] [PATCH] ALSA: firewire-lib: leave unit reference counting completely
by Takashi Sakamoto 10 Mar '15

10 Mar '15

With previous commit, this module managed to leave the counting to each drivers, but the isochronous resources functionality still increment/decrement the count. This commit purge such codes to leave the responsibility to each drivers. Fix: c6f224dc20ad ('ALSA: firewire-lib: remove reference counting') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/iso-resources.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c index 5f17b77..f0e4d50 100644 --- a/sound/firewire/iso-resources.c +++ b/sound/firewire/iso-resources.c @@ -26,7 +26,7 @@ int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit) { r->channels_mask = ~0uLL; - r->unit = fw_unit_get(unit); + r->unit = unit; mutex_init(&r->mutex); r->allocated = false; @@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r) { WARN_ON(r->allocated); mutex_destroy(&r->mutex); - fw_unit_put(r->unit); } EXPORT_SYMBOL(fw_iso_resources_destroy); -- 2.1.0

2 1

[alsa-devel] [PATCH 0/8] ALSA: control: refactoring core codes
by Takashi Sakamoto 10 Mar '15

10 Mar '15

This patchset is for refactoring core control codes. This is also a preparation for my further work for ALSA control functionality. Takashi Sakamoto (8): ALSA: control: change type from long due to the definition of sizeof operator ALSA: control: obsolete switch statement with const value table ALSA: control: gathering evaluations to access ALSA: control: add a comment about locking values after creating ALSA: control: rename loop index to i ALSA: control: fix over 80 characters lines ALSA: control: rename to appropriate macro name ALSA: control: arrange snd_ctl_new() as a local function sound/core/control.c | 223 ++++++++++++++++++++++++++------------------------- 1 file changed, 113 insertions(+), 110 deletions(-) -- 2.1.0

4 31

[alsa-devel] [PATCH] ALSA: dice: fix wrong offsets for Dice interface
by Takashi Sakamoto 10 Mar '15

10 Mar '15

For received packet stream, the offset of 'RX_SEQ_START' locates after the offset of 'RX_NUMBER_MIDI', although current macro and proc output includes wrong offsets. Fortunately, this bug doesn't affect streaming functionality because these macro is not used. This commit fixes these wrong macro and outputs. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-interface.h | 18 +++++++++--------- sound/firewire/dice/dice-proc.c | 4 ++-- 2 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index 27b044f..de7602b 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -299,23 +299,23 @@ #define RX_ISOCHRONOUS 0x008 /* - * Index of first quadlet to be interpreted; read/write. If > 0, that many - * quadlets at the beginning of each data block will be ignored, and all the - * audio and MIDI quadlets will follow. - */ -#define RX_SEQ_START 0x00c - -/* * The number of audio channels; read-only. There will be one quadlet per * channel. */ -#define RX_NUMBER_AUDIO 0x010 +#define RX_NUMBER_AUDIO 0x00c /* * The number of MIDI ports, 0-8; read-only. If > 0, there will be one * additional quadlet in each data block, following the audio quadlets. */ -#define RX_NUMBER_MIDI 0x014 +#define RX_NUMBER_MIDI 0x010 + +/* + * Index of first quadlet to be interpreted; read/write. If > 0, that many + * quadlets at the beginning of each data block will be ignored, and all the + * audio and MIDI quadlets will follow. + */ +#define RX_SEQ_START 0x014 /* * Names of all audio channels; read-only. Quadlets are byte-swapped. Names diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index f5c1d1b..ecfe20f 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry, } tx; struct { u32 iso; - u32 seq_start; u32 number_audio; u32 number_midi; + u32 seq_start; char names[RX_NAMES_SIZE]; u32 ac3_caps; u32 ac3_enable; @@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry, break; snd_iprintf(buffer, "rx %u:\n", stream); snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); - snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); snd_iprintf(buffer, " audio channels: %u\n", buf.rx.number_audio); snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); + snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); if (quadlets >= 68) { dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); snd_iprintf(buffer, " names: %s\n", buf.rx.names); -- 2.1.0

2 5

[alsa-devel] [PATCH 1/3] ALSA: seq_oss: Drop superfluous error/debug messages after malloc failures
by Takashi Iwai 10 Mar '15

10 Mar '15

The kernel memory allocators already report the errors when the requested allocation fails, thus we don't need to warn it again in each caller side. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_init.c | 4 +--- sound/core/seq/oss/seq_oss_midi.c | 5 ++--- sound/core/seq/oss/seq_oss_readq.c | 9 ++++----- sound/core/seq/oss/seq_oss_synth.c | 6 ++---- 4 files changed, 9 insertions(+), 15 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b0e32e161dd1..2de3feff70d0 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -188,10 +188,8 @@ snd_seq_oss_open(struct file *file, int level) struct seq_oss_devinfo *dp; dp = kzalloc(sizeof(*dp), GFP_KERNEL); - if (!dp) { - pr_err("ALSA: seq_oss: can't malloc device info\n"); + if (!dp) return -ENOMEM; - } dp->cseq = system_client; dp->port = -1; diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index e79cc44b1394..96e8395ae586 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -173,10 +173,9 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) /* * allocate midi info record */ - if ((mdev = kzalloc(sizeof(*mdev), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc midi info\n"); + mdev = kzalloc(sizeof(*mdev), GFP_KERNEL); + if (!mdev) return -ENOMEM; - } /* copy the port information */ mdev->client = pinfo->addr.client; diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c index 654d17a5023c..c080c73cea04 100644 --- a/sound/core/seq/oss/seq_oss_readq.c +++ b/sound/core/seq/oss/seq_oss_readq.c @@ -47,13 +47,12 @@ snd_seq_oss_readq_new(struct seq_oss_devinfo *dp, int maxlen) { struct seq_oss_readq *q; - if ((q = kzalloc(sizeof(*q), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue\n"); + q = kzalloc(sizeof(*q), GFP_KERNEL); + if (!q) return NULL; - } - if ((q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue buffer\n"); + q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL); + if (!q->q) { kfree(q); return NULL; } diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 835edc80f918..48e4fe1b68ab 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -106,10 +106,9 @@ snd_seq_oss_synth_probe(struct device *_dev) struct snd_seq_oss_reg *reg = SNDRV_SEQ_DEVICE_ARGPTR(dev); unsigned long flags; - if ((rec = kzalloc(sizeof(*rec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc synth info\n"); + rec = kzalloc(sizeof(*rec), GFP_KERNEL); + if (!rec) return -ENOMEM; - } rec->seq_device = -1; rec->synth_type = reg->type; rec->synth_subtype = reg->subtype; @@ -249,7 +248,6 @@ snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp) if (info->nr_voices > 0) { info->ch = kcalloc(info->nr_voices, sizeof(struct seq_oss_chinfo), GFP_KERNEL); if (!info->ch) { - pr_err("ALSA: seq_oss: Cannot malloc voices\n"); rec->oper.close(&info->arg); module_put(rec->oper.owner); snd_use_lock_free(&rec->use_lock); -- 2.3.2

1 2

[alsa-devel] [PATCH] sound/firewire: Fix trivial typos in comments
by Yannick Guerrini 10 Mar '15

10 Mar '15

Change 'propper' to 'proper' Change 'paramters' to 'parameters' Change 'SYT_INTEVAL' to 'SYT_INTERVAL' Change 'aligh'/'alighed' to 'align'/'aligned' Signed-off-by: Yannick Guerrini <yguerrini(a)tomshardware.fr> --- sound/firewire/amdtp.c | 8 ++++---- sound/firewire/fireworks/fireworks_transaction.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 5cc356d..e061355 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -166,10 +166,10 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * One AMDTP packet can include some frames. In blocking mode, the * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's - * preferrable to aligh period/buffer sizes to current SYT_INTERVAL. + * preferrable to align period/buffer sizes to current SYT_INTERVAL. * - * TODO: These constraints can be improved with propper rules. - * Currently apply LCM of SYT_INTEVALs. + * TODO: These constraints can be improved with proper rules. + * Currently apply LCM of SYT_INTERVALs. */ err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); @@ -270,7 +270,7 @@ static void amdtp_read_s32(struct amdtp_stream *s, * @s: the AMDTP stream to configure * @format: the format of the ALSA PCM device * - * The sample format must be set after the other paramters (rate/PCM channels/ + * The sample format must be set after the other parameters (rate/PCM channels/ * MIDI) and before the stream is started, and must not be changed while the * stream is running. */ diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c index 2a85e42..f550808 100644 --- a/sound/firewire/fireworks/fireworks_transaction.c +++ b/sound/firewire/fireworks/fireworks_transaction.c @@ -13,7 +13,7 @@ * * Transaction substance: * At first, 6 data exist. Following to the data, parameters for each command - * exist. All of the parameters are 32 bit alighed to big endian. + * exist. All of the parameters are 32 bit aligned to big endian. * data[0]: Length of transaction substance * data[1]: Transaction version * data[2]: Sequence number. This is incremented by the device -- 1.9.5.msysgit.0

2 1

[alsa-devel] [USB UAC2] How to route pcm stream for UAC2 virtual sound card for lowest latency?
by Xuebing Wang 10 Mar '15

10 Mar '15

Hi Community, I am developing a commercial product based on Freescale iMX6SL + UAC2. I can get UAC2 working for below command. 'hw:1,0' is the virtual sound card. 'hw:0,0' is the physical sound card. export RATE=44100; arecord -D hw:1,0 -c 2 -r $RATE -f S16_LE | aplay -c 2 -r $RATE -f S16_LE & Any suggestions on what is the best way to get PCM stream from UAC2 virtual sound card and feed to the physical sound card, for the lowest latency purpose? Thanks. -- Xuebing

2 1

[alsa-devel] [baohua:sirf-3.18 2573/2782] sound/soc/sirf/atlas7-iacc.c:168:24: sparse: symbol 'atlas7_iacc_dai_ops' was not declared. Should it be static?
by kbuild test robot 10 Mar '15

10 Mar '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/baohua/linux sirf-3.18 head: d61ca44a1d96f134d35d31522ca8efb19a6e1451 commit: 81aba96393135459ac811ce78fe8f91a8b767b82 [2573/2782] ASoC: sirf: atlas7: Add internal audio codec controller driver reproduce: # apt-get install sparse git checkout 81aba96393135459ac811ce78fe8f91a8b767b82 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/sirf/atlas7-iacc.c:168:24: sparse: symbol 'atlas7_iacc_dai_ops' was not declared. Should it be static? sound/soc/sirf/atlas7-iacc.c:482:27: sparse: incorrect type in initializer (incompatible argument 4 (different address spaces)) sound/soc/sirf/atlas7-iacc.c:482:27: expected int ( *copy )( ... ) sound/soc/sirf/atlas7-iacc.c:482:27: got int ( static [toplevel] *<noident> )( ... ) Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

4 4

[alsa-devel] [PATCH 1/8] ASoC: Intel: add function to load firmware image
by han.lu＠intel.com 10 Mar '15

10 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Add a general method to load firmware image, and apply to base firmware image loading. With the method, the driver will support loading multiple different modules in order to support different features. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index b9da030..396d545 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -173,6 +173,16 @@ struct sst_module_runtime_context { }; /* + * Audio DSP Module State + */ +enum sst_module_state { + SST_MODULE_STATE_UNLOADED = 0, /* default state */ + SST_MODULE_STATE_LOADED, + SST_MODULE_STATE_INITIALIZED, /* and inactive */ + SST_MODULE_STATE_ACTIVE, +}; + +/* * Audio DSP Generic Module. * * Each Firmware file can consist of 1..N modules. A module can span multiple @@ -203,6 +213,9 @@ struct sst_module { struct list_head list; /* DSP list of modules */ struct list_head list_fw; /* FW list of modules */ struct list_head runtime_list; /* list of runtime module objects*/ + + /* state */ + enum sst_module_state state; }; /* diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5f71ef6..5e58008 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -498,6 +498,7 @@ struct sst_module *sst_module_new(struct sst_fw *sst_fw, sst_module->scratch_size = template->scratch_size; sst_module->persistent_size = template->persistent_size; sst_module->entry = template->entry; + sst_module->state = SST_MODULE_STATE_UNLOADED; INIT_LIST_HEAD(&sst_module->block_list); INIT_LIST_HEAD(&sst_module->runtime_list); diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 402b728..8ad733b 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -169,6 +169,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, block = (void *)block + sizeof(*block) + block->size; } + mod->state = SST_MODULE_STATE_LOADED; return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 863a9ca..cbd8c00 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1844,6 +1844,8 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) if (ret < 0) dev_err(dev, "error: audio DSP boot failure\n"); + sst_hsw_init_module_state(hsw); + ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, msecs_to_jiffies(IPC_BOOT_MSECS)); if (ret == 0) { @@ -1886,6 +1888,75 @@ struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw) return hsw->dsp; } +void sst_hsw_init_module_state(struct sst_hsw *hsw) +{ + struct sst_module *module; + enum sst_hsw_module_id id; + + /* the base fw contains several modules */ + for (id = SST_HSW_MODULE_BASE_FW; id < SST_HSW_MAX_MODULE_ID; id++) { + module = sst_module_get_from_id(hsw->dsp, id); + if (module) + module->state = SST_MODULE_STATE_ACTIVE; + } +} + +int sst_hsw_module_load(struct sst_hsw *hsw, + u32 module_id, u32 instance_id, char *name) +{ + int ret = 0; + const struct firmware *fw = NULL; + struct sst_fw *hsw_sst_fw; + struct sst_module *module; + struct device *dev = hsw->dev; + struct sst_dsp *dsp = hsw->dsp; + + dev_dbg(dev, "sst_hsw_module_load id=%d, name='%s'", module_id, name); + + module = sst_module_get_from_id(dsp, module_id); + if (module == NULL) { + /* loading for the first time */ + if (module_id == SST_HSW_MODULE_BASE_FW) { + /* for base module: use fw requested in acpi probe */ + fw = dsp->pdata->fw; + if (!fw) { + dev_err(dev, "request Base fw failed\n"); + return -ENODEV; + } + } else { + /* try and load any other optional modules if they are + * available. Use dev_info instead of dev_err in case + * request firmware failed */ + ret = request_firmware(&fw, name, dev); + if (ret) { + dev_info(dev, "fw image %s not available(%d)\n", + name, ret); + return ret; + } + } + hsw_sst_fw = sst_fw_new(dsp, fw, hsw); + if (hsw_sst_fw == NULL) { + dev_err(dev, "error: failed to load firmware\n"); + ret = -ENOMEM; + goto out; + } + module = sst_module_get_from_id(dsp, module_id); + if (module == NULL) { + dev_err(dev, "error: no module %d in firmware %s\n", + module_id, name); + } + dev_info(dev, "firmware %s loaded successfuly\n", name); + } else + dev_info(dev, "module %d (%s) already loaded\n", + module_id, name); +out: + /* release fw, but base fw should be released by acpi driver */ + if (fw && module_id != SST_HSW_MODULE_BASE_FW) + release_firmware(fw); + + return ret; +} + static struct sst_dsp_device hsw_dev = { .thread = hsw_irq_thread, .ops = &haswell_ops, @@ -1947,12 +2018,10 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* keep the DSP in reset state for base FW loading */ sst_dsp_reset(hsw->dsp); - hsw->sst_fw = sst_fw_new(hsw->dsp, pdata->fw, hsw); - if (hsw->sst_fw == NULL) { - ret = -ENODEV; - dev_err(dev, "error: failed to load firmware\n"); + /* load base module and other modules in base firmware image */ + ret = sst_hsw_module_load(hsw, SST_HSW_MODULE_BASE_FW, 0, "Base"); + if (ret < 0) goto fw_err; - } /* allocate scratch mem regions */ ret = sst_block_alloc_scratch(hsw->dsp); @@ -1971,6 +2040,9 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) goto boot_err; } + /* init module state after boot */ + sst_hsw_init_module_state(hsw); + /* get the FW version */ sst_hsw_fw_get_version(hsw, &version); @@ -1986,7 +2058,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) boot_err: sst_dsp_reset(hsw->dsp); - sst_fw_free(hsw->sst_fw); + sst_fw_free_all(hsw->dsp); fw_err: dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, hsw->dx_context, hsw->dx_context_paddr); diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 8580960..e071b3a 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -467,6 +467,12 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata); struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw); +/* fw module function */ +void sst_hsw_init_module_state(struct sst_hsw *hsw); + +int sst_hsw_module_load(struct sst_hsw *hsw, + u32 module_id, u32 instance_id, char *name); + /* runtime module management */ struct sst_module_runtime *sst_hsw_runtime_module_create(struct sst_hsw *hsw, int mod_id, int offset); -- 1.9.1

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[alsa-devel] About iMX6SL sign extension of SSI Receive Configuration Register (SSIx_SRCR)
by Xuebing Wang 10 Mar '15

10 Mar '15

Hi Nicolin and community, iMX6SL reference manual section "48.9.8 SSI Receive Configuration Register (SSIx_SRCR)" says about sign extension. Below is a quote from iMX6SL reference manual: ------------------------------------------------------------- Receive Data Extension. This control bit allows SSI to store the received data word in sign extended form. This bit affects data storage only in case received data is LSB aligned (SRCR[9]=1) 0 OFF — Sign extension turned off. 1 ON — Sign extension turned on. ------------------------------------------------------------- After I receive data (in the format of S16_LE) from SSI/I2S, I got data like 0x7FFF or 0x77d2. Are they supposed to be 0xFFFF or 0xF7d2? After setting bit 10 (RXEXT) of SSIx_SRCR, I still get 0x7FFF or 0x77d2. -- Note: SRCR[9] is 1. Any suggestions? Thanks. -- Xuebing

2 1

[alsa-devel] [DAPM question] DAPM support for single input multiple outputs
by Qiao Zhou 10 Mar '15

10 Mar '15

Hi Mark, I have one question and need your suggestion. Thanks in advance. In our DSP, one stream path is like this: There is one bit controlling whether a signal A goes to A_out or B_out. That's the reverse case of DAPM_MUX(multiple inputs, single output). One way is to use two switches but it seems not good since two switches control the same bit. It's strange. Do you have any opinion about such requirement support? -- Best Regards Qiao

3 4

[alsa-devel] [PATCH 0/24 v2] ASoC: rsnd: move 2nd DMAC in rsnd driver and subnode dma-name support
by Kuninori Morimoto 10 Mar '15

10 Mar '15

Hi Mark, Arnd, Vinod Cc: Laurent, SH-ML These patches add Audio DMAC peri peri implementation in rsnd driver. I will remove current DMAEngine side Audio DMAC peri peri implement if these are accepted. http://thread.gmane.org/gmane.linux.ports.sh.devel/41063/focus=43829 Arnd, Vinod This series adds subnode DT support for 1st DMAC, and it is based on 1) patch. can you please check it ? Mark Actually, 1) patch has no relationship until 20). I guess this patch-set has merge issue. Can you check 2) - 19) as 1st step ? 1) : export of_dma_request_slave_channel() for 20) 2) - 6) : cleanup / tidyup 7) - 12) : Audio DMAC peri peri implementation 13) - 14) : cleanup. these are no longer needed now 15) - 19) : prepare for 1st DMAC dma-names subnode support 20) : 1st DMAC dma-names subnode subnode 21) - 24) : DT bindings text update Kuninori Morimoto (24): 1) dmaengine: export symbol of of_dma_request_slave_channel() 2) ASoC: rsnd: remove SH-DMA-BASE specific implementation 3) ASoC: rsnd: remove un-needed parameter from rsnd_dma_init() 4) ASoC: rsnd: remove unused rsnd_dma_available() 5) ASoC: rsnd: remove un-needed parameter from rsnd_dma_quit() 6) ASoC: rsnd: tidyup rsnd_dma_to_mod() macro declaration position 7) ASoC: rsnd: enable to get resource by name 8) ASoC: rsnd: add rsnd_gen_get_phy_addr() to get physical address 9) ASoC: rsnd: add dma.c for Audio DMAC / Audio DMAC peri peri 10) ASoC: rsnd: move rsnd_gen_dma_addr() from gen.c to dma.c 11) ASoC: rsnd: enable to care 1st / 2nd DMAC on rsnd_dma_xxx() 12) ASoC: rsnd: add Audio DMAC peri peri support rework 13) ASoC: rsnd: remove rsnd_dma::addr 14) ASoC: rsnd: remove rsnd_dma::dir 15) ASoC: rsnd: use union with rsnd_dmaen / rsnd_dmapp 16) ASoC: rsnd: enable rsnd_ssi_use_busif() for DMA 17) ASoC: rsnd: ssi: add rsnd_ssi_of_node() 18) ASoC: rsnd: src: add rsnd_src_of_node() 19) ASoC: rsnd: dvc: add rsnd_dvc_of_node() 20) ASoC: rsnd: 1st DMAC dma-names cares subnode 21) ASoC: rsnd: add sample code of rcar_sound,src irq 22) ASoC: rsnd: add sample code of missing clocks 23) ASoC: rsnd: add sample code of dma entry 24) ASoC: rsnd: add sample code of reg-names .../devicetree/bindings/sound/renesas,rsnd.txt | 125 +++- drivers/dma/of-dma.c | 1 + include/linux/of_dma.h | 4 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 235 +------- sound/soc/sh/rcar/dma.c | 603 ++++++++++++++++++++ sound/soc/sh/rcar/dvc.c | 12 + sound/soc/sh/rcar/gen.c | 145 +---- sound/soc/sh/rcar/rsnd.h | 72 ++- sound/soc/sh/rcar/src.c | 23 +- sound/soc/sh/rcar/ssi.c | 28 +- 11 files changed, 838 insertions(+), 412 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] ASoC: tlv320aic23: remove incorrect __exit markups
by Dmitry Torokhov 09 Mar '15

09 Mar '15

Even if bus is not hot-pluggable, the devices can be unbound from the driver via sysfs, so we should not be using __exit annotations on remove() methods. The only exception is drivers registered with platform_driver_probe() which specifically disables sysfs bind/unbind attributes. Signed-off-by: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> --- sound/soc/codecs/tlv320aic23-i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c index f137019..78a94af 100644 --- a/sound/soc/codecs/tlv320aic23-i2c.c +++ b/sound/soc/codecs/tlv320aic23-i2c.c @@ -31,7 +31,7 @@ static int tlv320aic23_i2c_probe(struct i2c_client *i2c, return tlv320aic23_probe(&i2c->dev, regmap); } -static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +static int tlv320aic23_i2c_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); return 0; @@ -56,7 +56,7 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic23_of_match), }, .probe = tlv320aic23_i2c_probe, - .remove = __exit_p(tlv320aic23_i2c_remove), + .remove = tlv320aic23_i2c_remove, .id_table = tlv320aic23_id, }; -- 2.2.0.rc0.207.ga3a616c -- Dmitry

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[alsa-devel] [PATCH] ASoC: wm2200: match wait_for_completion_timeout return type
by Nicholas Mc Guire 09 Mar '15

09 Mar '15

return type of wait_for_completion_timeout is unsigned long not int. An appropriately named unsigned long is added and the assignment fixed up. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- This was only compile tested for exynos_defconfig + CONFIG_COMPILE_TEST=y, SND_SOC_ALL_CODECS=m (implies CONFIG_SND_SOC_WM2200=m) Patch is against 4.0-rc2 linux-next (localversion-next is -next-20150306) sound/soc/codecs/wm2200.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 1559984..b48694a 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1942,6 +1942,7 @@ static int wm2200_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct wm2200_priv *wm2200 = snd_soc_codec_get_drvdata(codec); struct _fll_div factors; int ret, i, timeout; + unsigned long time_left; if (!Fout) { dev_dbg(codec->dev, "FLL disabled"); @@ -2021,9 +2022,10 @@ static int wm2200_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Poll for the lock; will use the interrupt to exit quickly */ for (i = 0; i < timeout; i++) { if (i2c->irq) { - ret = wait_for_completion_timeout(&wm2200->fll_lock, - msecs_to_jiffies(25)); - if (ret > 0) + time_left = wait_for_completion_timeout( + &wm2200->fll_lock, + msecs_to_jiffies(25)); + if (time_left > 0) break; } else { msleep(1); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: arizona: match wait_for_completion_timeout return type
by Nicholas Mc Guire 09 Mar '15

09 Mar '15

return type of wait_for_completion_timeout is unsigned long not int. An appropriately named unsigned long is added and the assignment fixed up. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Note that the error message "Timed out waiting for lock" might not be that clear in a kernel log message as it could probably be read as relating to a kernel lock "Timed out waiting for FLL lock" might be clearer. This was only compile tested for exynos_defconfig + CONFIG_COMPILE_TEST=y SND_SOC_ALL_CODECS=m, EXTCON_ARIZONA=m, CONFIG_MFD_ARIZONA_SPI=m, CONFIG_MFD_WM5102=y, (implies SND_SOC_ARIZONA=m) Patch is against 4.0-rc2 linux-next (localversion-next is -next-20150306) sound/soc/codecs/arizona.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 95d31d6..57da0ce 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1902,7 +1902,7 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll) static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; - int ret; + unsigned long time_left; bool use_sync = false; int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; @@ -1978,9 +1978,9 @@ static int arizona_enable_fll(struct arizona_fll *fll) regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); - ret = wait_for_completion_timeout(&fll->ok, + time_left = wait_for_completion_timeout(&fll->ok, msecs_to_jiffies(250)); - if (ret == 0) + if (time_left == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); return 0; -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: wm5100: match wait_for_completion_timeout return type
by Nicholas Mc Guire 09 Mar '15

09 Mar '15

return type of wait_for_completion_timeout is unsigned long not int. An appropriately named unsigned long is added and the assignment fixed up. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- This was only compile tested for exynos_defconfig + CONFIG_COMPILE_TEST=y, SND_SOC_ALL_CODECS=m (implies CONFIG_SND_SOC_WM5100=m) Patch is against 4.0-rc2 linux-next (localversion-next is -next-20150306) sound/soc/codecs/wm5100.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index ea09db5..9674037 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1762,6 +1762,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct _fll_div factors; struct wm5100_fll *fll; int ret, base, lock, i, timeout; + unsigned long time_left; switch (fll_id) { case WM5100_FLL1: @@ -1842,9 +1843,9 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Poll for the lock; will use interrupt when we can test */ for (i = 0; i < timeout; i++) { if (i2c->irq) { - ret = wait_for_completion_timeout(&fll->lock, - msecs_to_jiffies(25)); - if (ret > 0) + time_left = wait_for_completion_timeout(&fll->lock, + msecs_to_jiffies(25)); + if (time_left > 0) break; } else { msleep(1); -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: wm8996: match wait_for_completion_timeout return type
by Nicholas Mc Guire 09 Mar '15

09 Mar '15

return type of wait_for_completion_timeout is unsigned long not int. An appropriately named unsigned long is added and the assignment fixed up in case of completion occurring the remaining time is >=1 so ret is set to 1 if no timeout occurred. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- This was only compile tested for exynos_defconfig + CONFIG_COMPILE_TEST=y, SND_SOC_ALL_CODECS=m (implies CONFIG_SND_SOC_WM8996=m) Patch is against 4.0-rc2 linux-next (localversion-next is -next-20150306) sound/soc/codecs/wm8996.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index dc92d5e..24d9705 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2009,7 +2009,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); struct _fll_div fll_div; - unsigned long timeout; + unsigned long timeout, time_left; int ret, reg, retry; /* Any change? */ @@ -2113,10 +2113,11 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, timeout /= 2; for (retry = 0; retry < 10; retry++) { - ret = wait_for_completion_timeout(&wm8996->fll_lock, - timeout); - if (ret != 0) { + time_left = wait_for_completion_timeout(&wm8996->fll_lock, + timeout); + if (time_left != 0) { WARN_ON(!i2c->irq); + ret = 1; break; } -- 1.7.10.4

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